diff --git a/Makefile.am b/Makefile.am
index 125791e444..24ec7f49ea 100644
--- a/Makefile.am
+++ b/Makefile.am
@@ -446,7 +446,9 @@ stream_DATA += \
 	plugins/streams/music.mulaw \
 	plugins/streams/radio.alaw \
 	plugins/streams/test_gstreamer.sh \
-	plugins/streams/test_gstreamer_1.sh
+	plugins/streams/test_gstreamer1.sh \
+	plugins/streams/test_gstreamer_multistream.sh \
+	plugins/streams/test_gstreamer1_multistream.sh
 EXTRA_DIST += \
 	conf/janus.plugin.streaming.jcfg.sample.in \
 	$(stream_DATA)
diff --git a/conf/janus.plugin.streaming.jcfg.sample.in b/conf/janus.plugin.streaming.jcfg.sample.in
index b476bc3138..9e456f7878 100644
--- a/conf/janus.plugin.streaming.jcfg.sample.in
+++ b/conf/janus.plugin.streaming.jcfg.sample.in
@@ -143,6 +143,7 @@ file-ondemand-sample: {
 	#videopt = 126
 	#videortpmap = "H264/90000"
 	#videofmtp = "profile-level-id=42e01f;packetization-mode=1"
+	#secret = "adminpwd"
 #}
 
 #
@@ -162,6 +163,7 @@ file-ondemand-sample: {
 	#video = false
 	#data = true
 	#dataport = 5008
+	#secret = "adminpwd"
 #}
 
 #
@@ -182,6 +184,7 @@ file-ondemand-sample: {
 	#videomcast = "232.3.4.5"
 	#videopt = 100
 	#videortpmap = "VP8/90000"
+	#secret = "adminpwd"
 #}
 
 #
@@ -202,4 +205,54 @@ file-ondemand-sample: {
 	#url = "rtsp://127.0.0.1:8554/unicast"
 	#rtsp_user = "username"
 	#rtsp_pwd = "password"
+	#secret = "adminpwd"
 #}
+
+#
+# Finally, this is an example of how you can configure a live Streaming
+# mountpoint to send multiple streams of the same type at the same time:
+# that is, not simulcasting, but different streams (e.g., two audio
+# streams and two video streams). To do so, you don't set the audio,
+# video and data properties inline, but use an array of properties instead,
+# each identifying a single stream to add, that will then translate to
+# a dedicated m-line in the SDP. For each stream, you specify the type,
+# a unique ID (mid), and can provide a short description (label) so that
+# the client side can know what's what when rendering the streams. Notice
+# how the port/pt/rtpmap/fmtp/etc. stuff is called just like that, without
+# any audio/video/data prefix: in fact, each media stream can be configured
+# the same way, and it's the type that allows us to differentiate them.
+# As such, you can use the same approach for creating regular mountpoints
+# as well (e.g., 1 audio and 1 video) in a much clearer, and cleaner, way.
+#
+multistream-test: {
+	type = "rtp"
+	id = 123
+	description = "Multistream test (1 audio, 2 video)"
+	media = (
+		{
+			type = "audio"
+			mid = "a"
+			label = "Audio stream"
+			port = 5102
+			pt = 111
+			rtpmap = "opus/48000/2"
+		},
+		{
+			type = "video"
+			mid = "v1"
+			label = "Video stream #1"
+			port = 5104
+			pt = 100
+			rtpmap = "VP8/90000"
+		},
+		{
+			type = "video"
+			mid = "v2"
+			label = "Video stream #2"
+			port = 5106
+			pt = 100
+			rtpmap = "VP8/90000"
+		}
+	)
+	secret = "adminpwd"
+}
diff --git a/dtls-bio.c b/dtls-bio.c
index 21eff3507c..1a76945a6a 100644
--- a/dtls-bio.c
+++ b/dtls-bio.c
@@ -3,7 +3,7 @@
  * \copyright GNU General Public License v3
  * \brief    OpenSSL BIO agent writer
  * \details  OpenSSL BIO that writes packets to a libnice agent.
- * 
+ *
  * \ingroup protocols
  * \ref protocols
  */
@@ -110,7 +110,7 @@ static int janus_dtls_bio_agent_free(BIO *bio) {
 #endif
 	return 1;
 }
-	
+
 static int janus_dtls_bio_agent_write(BIO *bio, const char *in, int inl) {
 	JANUS_LOG(LOG_HUGE, "janus_dtls_bio_agent_write: %p, %d\n", in, inl);
 	/* Forward data to the write BIO */
@@ -129,17 +129,12 @@ static int janus_dtls_bio_agent_write(BIO *bio, const char *in, int inl) {
 		JANUS_LOG(LOG_ERR, "No DTLS-SRTP stack, no DTLS bridge...\n");
 		return -1;
 	}
-	janus_ice_component *component = (janus_ice_component *)dtls->component;
-	if(component == NULL) {
-		JANUS_LOG(LOG_ERR, "No component, no DTLS bridge...\n");
-		return -1;
-	}
-	janus_ice_stream *stream = component->stream;
-	if(!stream) {
-		JANUS_LOG(LOG_ERR, "No stream, no DTLS bridge...\n");
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)dtls->pc;
+	if(pc == NULL) {
+		JANUS_LOG(LOG_ERR, "No WebRTC PeerConnection, no DTLS bridge...\n");
 		return -1;
 	}
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle || !handle->agent || !dtls->write_bio) {
 		JANUS_LOG(LOG_ERR, "No handle/agent/bio, no DTLS bridge...\n");
 		return -1;
@@ -149,17 +144,17 @@ static int janus_dtls_bio_agent_write(BIO *bio, const char *in, int inl) {
 		/* FIXME Just a warning for now, this will need to be solved with proper fragmentation */
 		JANUS_LOG(LOG_WARN, "[%"SCNu64"] The DTLS stack is trying to send a packet of %d bytes, this may be larger than the MTU and get dropped!\n", handle->handle_id, inl);
 	}
-	int bytes = nice_agent_send(handle->agent, component->stream_id, component->component_id, inl, in);
+	int bytes = nice_agent_send(handle->agent, pc->stream_id, pc->component_id, inl, in);
 	if(bytes < inl) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error sending DTLS message on component %d of stream %d (%d)\n", handle->handle_id, component->component_id, stream->stream_id, bytes);
+		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error sending DTLS message on component %d of stream %d (%d)\n", handle->handle_id, pc->component_id, pc->stream_id, bytes);
 	} else {
 		JANUS_LOG(LOG_HUGE, "[%"SCNu64"] >> >> ... and sent %d of those bytes on the socket\n", handle->handle_id, bytes);
 	}
 	/* Update stats (TODO Do the same for the last second window as well)
 	 * FIXME: the Data stats includes the bytes used for the handshake */
 	if(bytes > 0) {
-		component->out_stats.data.packets++;
-		component->out_stats.data.bytes += bytes;
+		pc->dtls_out_stats.info[0].packets++;
+		pc->dtls_out_stats.info[0].bytes += bytes;
 	}
 	return bytes;
 }
diff --git a/dtls.c b/dtls.c
index 7b201a763d..df3d9d8dd7 100644
--- a/dtls.c
+++ b/dtls.c
@@ -79,13 +79,10 @@ static void janus_dtls_notify_state_change(janus_dtls_srtp *dtls) {
 		return;
 	if(dtls == NULL)
 		return;
-	janus_ice_component *component = (janus_ice_component *)dtls->component;
-	if(component == NULL)
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)dtls->pc;
+	if(pc == NULL)
 		return;
-	janus_ice_stream *stream = component->stream;
-	if(stream == NULL)
-		return;
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(handle == NULL)
 		return;
 	janus_session *session = (janus_session *)handle->session;
@@ -93,8 +90,8 @@ static void janus_dtls_notify_state_change(janus_dtls_srtp *dtls) {
 		return;
 	json_t *info = json_object();
 	json_object_set_new(info, "dtls", json_string(janus_get_dtls_srtp_state(dtls->dtls_state)));
-	json_object_set_new(info, "stream_id", json_integer(stream->stream_id));
-	json_object_set_new(info, "component_id", json_integer(component->component_id));
+	json_object_set_new(info, "stream_id", json_integer(pc->stream_id));
+	json_object_set_new(info, "component_id", json_integer(pc->component_id));
 	json_object_set_new(info, "retransmissions", json_integer(dtls->retransmissions));
 	janus_events_notify_handlers(JANUS_EVENT_TYPE_WEBRTC, session->session_id, handle->handle_id, handle->opaque_id, info);
 }
@@ -444,7 +441,7 @@ gint janus_dtls_srtp_init(const char *server_pem, const char *server_key, const
 static void janus_dtls_srtp_free(const janus_refcount *dtls_ref) {
 	janus_dtls_srtp *dtls = janus_refcount_containerof(dtls_ref, janus_dtls_srtp, ref);
 	/* This stack can be destroyed, free all the resources */
-	dtls->component = NULL;
+	dtls->pc = NULL;
 	if(dtls->ssl != NULL) {
 		SSL_free(dtls->ssl);
 		dtls->ssl = NULL;
@@ -486,18 +483,13 @@ void janus_dtls_srtp_cleanup(void) {
 }
 
 
-janus_dtls_srtp *janus_dtls_srtp_create(void *ice_component, janus_dtls_role role) {
-	janus_ice_component *component = (janus_ice_component *)ice_component;
-	if(component == NULL) {
-		JANUS_LOG(LOG_ERR, "No component, no DTLS...\n");
-		return NULL;
-	}
-	janus_ice_stream *stream = component->stream;
-	if(!stream) {
-		JANUS_LOG(LOG_ERR, "No stream, no DTLS...\n");
+janus_dtls_srtp *janus_dtls_srtp_create(void *webrtc, janus_dtls_role role) {
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)webrtc;
+	if(pc == NULL) {
+		JANUS_LOG(LOG_ERR, "No WebRTC PeerConnection, no DTLS...\n");
 		return NULL;
 	}
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle || !handle->agent) {
 		JANUS_LOG(LOG_ERR, "No handle/agent, no DTLS...\n");
 		return NULL;
@@ -561,7 +553,7 @@ janus_dtls_srtp *janus_dtls_srtp_create(void *ice_component, janus_dtls_role rol
 #endif
 	/* Done */
 	dtls->dtls_connected = 0;
-	dtls->component = component;
+	dtls->pc = pc;
 	return dtls;
 }
 
@@ -588,21 +580,18 @@ int janus_dtls_srtp_create_sctp(janus_dtls_srtp *dtls) {
 #ifdef HAVE_SCTP
 	if(dtls == NULL)
 		return -1;
-	janus_ice_component *component = (janus_ice_component *)dtls->component;
-	if(component == NULL)
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)dtls->pc;
+	if(pc == NULL)
 		return -2;
-	janus_ice_stream *stream = component->stream;
-	if(!stream)
-		return -3;
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle || !handle->agent)
+		return -3;
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT))
 		return -4;
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT))
-		return -5;
 	dtls->sctp = janus_sctp_association_create(dtls, handle, 5000);
 	if(dtls->sctp == NULL) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error creating SCTP association...\n", handle->handle_id);
-		return -6;
+		return -5;
 	}
 	return 0;
 #else
@@ -616,27 +605,22 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 		JANUS_LOG(LOG_ERR, "No DTLS-SRTP stack, no incoming message...\n");
 		return;
 	}
-	janus_ice_component *component = (janus_ice_component *)dtls->component;
-	if(component == NULL) {
-		JANUS_LOG(LOG_ERR, "No component, no DTLS...\n");
-		return;
-	}
-	janus_ice_stream *stream = component->stream;
-	if(!stream) {
-		JANUS_LOG(LOG_ERR, "No stream, no DTLS...\n");
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)dtls->pc;
+	if(pc == NULL) {
+		JANUS_LOG(LOG_ERR, "No WebRTC PeerConnection, no DTLS...\n");
 		return;
 	}
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle || !handle->agent) {
 		JANUS_LOG(LOG_ERR, "No handle/agent, no DTLS...\n");
 		return;
 	}
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT)) {
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT)) {
 		JANUS_LOG(LOG_WARN, "[%"SCNu64"] Alert already triggered, clearing up...\n", handle->handle_id);
 		return;
 	}
 	if(!dtls->ssl || !dtls->read_bio) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"] No DTLS stuff for component %d in stream %d??\n", handle->handle_id, component->component_id, stream->stream_id);
+		JANUS_LOG(LOG_ERR, "[%"SCNu64"] No DTLS stuff for component %d in stream %d??\n", handle->handle_id, pc->component_id, pc->stream_id);
 		return;
 	}
 	if(dtls->dtls_started == 0) {
@@ -664,7 +648,7 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 			return;
 		}
 	}
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP) || janus_is_stopping()) {
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP) || janus_is_stopping()) {
 		/* DTLS alert triggered, we should end it here */
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Forced to stop it here...\n", handle->handle_id);
 		return;
@@ -679,6 +663,14 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 		JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Any data available?\n", handle->handle_id);
 #ifdef HAVE_SCTP
 		if(dtls->sctp != NULL && read > 0) {
+			/* Update stats for the data medium */
+			janus_handle_webrtc_medium *medium = g_hash_table_lookup(pc->media_bytype,
+					GINT_TO_POINTER(JANUS_MEDIA_DATA));
+			if(medium != NULL) {
+				medium->in_stats.info[0].packets++;
+				medium->in_stats.info[0].bytes += read;
+			}
+			/* Pass the data to the SCTP stack */
 			JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Sending data (%d bytes) to the SCTP stack...\n", handle->handle_id, read);
 			janus_sctp_data_from_dtls(dtls->sctp, data, read);
 		}
@@ -699,7 +691,7 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 			unsigned char rfingerprint[EVP_MAX_MD_SIZE];
 			char remote_fingerprint[160];
 			char *rfp = (char *)&remote_fingerprint;
-			if(stream->remote_hashing && !strcasecmp(stream->remote_hashing, "sha-1")) {
+			if(pc->remote_hashing && !strcasecmp(pc->remote_hashing, "sha-1")) {
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Computing sha-1 fingerprint of remote certificate...\n", handle->handle_id);
 				X509_digest(rcert, EVP_sha1(), (unsigned char *)rfingerprint, &rsize);
 			} else {
@@ -715,8 +707,8 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 			}
 			*(rfp-1) = 0;
 			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Remote fingerprint (%s) of the client is %s\n",
-				handle->handle_id, stream->remote_hashing ? stream->remote_hashing : "sha-256", remote_fingerprint);
-			if(!strcasecmp(remote_fingerprint, stream->remote_fingerprint ? stream->remote_fingerprint : "(none)")) {
+				handle->handle_id, pc->remote_hashing ? pc->remote_hashing : "sha-256", remote_fingerprint);
+			if(!strcasecmp(remote_fingerprint, pc->remote_fingerprint ? pc->remote_fingerprint : "(none)")) {
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"]  Fingerprint is a match!\n", handle->handle_id);
 				dtls->dtls_state = JANUS_DTLS_STATE_CONNECTED;
 				dtls->dtls_connected = janus_get_monotonic_time();
@@ -724,7 +716,7 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 				janus_dtls_notify_state_change(dtls);
 			} else {
 				/* FIXME NOT a match! MITM? */
-				JANUS_LOG(LOG_ERR, "[%"SCNu64"]  Fingerprint is NOT a match! got %s, expected %s\n", handle->handle_id, remote_fingerprint, stream->remote_fingerprint);
+				JANUS_LOG(LOG_ERR, "[%"SCNu64"]  Fingerprint is NOT a match! got %s, expected %s\n", handle->handle_id, remote_fingerprint, pc->remote_fingerprint);
 				dtls->dtls_state = JANUS_DTLS_STATE_FAILED;
 				/* Notify event handlers */
 				janus_dtls_notify_state_change(dtls);
@@ -776,7 +768,7 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 				if(!SSL_export_keying_material(dtls->ssl, material, master_length*2, "EXTRACTOR-dtls_srtp", 19, NULL, 0, 0)) {
 					/* Oops... */
 					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Oops, couldn't extract SRTP keying material for component %d in stream %d?? (%s)\n",
-						handle->handle_id, component->component_id, stream->stream_id, ERR_reason_error_string(ERR_get_error()));
+						handle->handle_id, pc->component_id, pc->stream_id, ERR_reason_error_string(ERR_get_error()));
 					goto done;
 				}
 				/* Key derivation (http://tools.ietf.org/html/rfc5764#section-4.2) */
@@ -866,23 +858,23 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 				srtp_err_status_t res = srtp_create(&(dtls->srtp_in), &(dtls->remote_policy));
 				if(res != srtp_err_status_ok) {
 					/* Something went wrong... */
-					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Oops, error creating inbound SRTP session for component %d in stream %d??\n", handle->handle_id, component->component_id, stream->stream_id);
+					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Oops, error creating inbound SRTP session for component %d in stream %d??\n", handle->handle_id, pc->component_id, pc->stream_id);
 					JANUS_LOG(LOG_ERR, "[%"SCNu64"]  -- %d (%s)\n", handle->handle_id, res, janus_srtp_error_str(res));
 					goto done;
 				}
-				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Created inbound SRTP session for component %d in stream %d\n", handle->handle_id, component->component_id, stream->stream_id);
+				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Created inbound SRTP session for component %d in stream %d\n", handle->handle_id, pc->component_id, pc->stream_id);
 				res = srtp_create(&(dtls->srtp_out), &(dtls->local_policy));
 				if(res != srtp_err_status_ok) {
 					/* Something went wrong... */
-					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Oops, error creating outbound SRTP session for component %d in stream %d??\n", handle->handle_id, component->component_id, stream->stream_id);
+					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Oops, error creating outbound SRTP session for component %d in stream %d??\n", handle->handle_id, pc->component_id, pc->stream_id);
 					JANUS_LOG(LOG_ERR, "[%"SCNu64"]  -- %d (%s)\n", handle->handle_id, res, janus_srtp_error_str(res));
 					goto done;
 				}
 				dtls->srtp_profile = srtp_profile->id;
 				dtls->srtp_valid = 1;
-				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Created outbound SRTP session for component %d in stream %d\n", handle->handle_id, component->component_id, stream->stream_id);
+				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Created outbound SRTP session for component %d in stream %d\n", handle->handle_id, pc->component_id, pc->stream_id);
 #ifdef HAVE_SCTP
-				if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS)) {
+				if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_DATA_CHANNELS)) {
 					/* Create SCTP association as well */
 					janus_dtls_srtp_create_sctp(dtls);
 				}
@@ -890,13 +882,13 @@ void janus_dtls_srtp_incoming_msg(janus_dtls_srtp *dtls, char *buf, uint16_t len
 				dtls->ready = 1;
 			}
 done:
-			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT) && dtls->srtp_valid) {
+			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT) && dtls->srtp_valid) {
 				/* Handshake successfully completed */
-				janus_ice_dtls_handshake_done(handle, component);
+				janus_handle_dtls_handshake_done(handle);
 			} else {
 				/* Something went wrong in either DTLS or SRTP... tell the plugin about it */
 				janus_dtls_callback(dtls->ssl, SSL_CB_ALERT, 0);
-				janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING);
+				janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING);
 			}
 		}
 	}
@@ -939,23 +931,18 @@ void janus_dtls_callback(const SSL *ssl, int where, int ret) {
 		JANUS_LOG(LOG_ERR, "No DTLS session related to this alert...\n");
 		return;
 	}
-	janus_ice_component *component = dtls->component;
-	if(component == NULL) {
-		JANUS_LOG(LOG_ERR, "No ICE component related to this alert...\n");
-		return;
-	}
-	janus_ice_stream *stream = component->stream;
-	if(!stream) {
-		JANUS_LOG(LOG_ERR, "No ICE stream related to this alert...\n");
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)dtls->pc;
+	if(pc == NULL) {
+		JANUS_LOG(LOG_ERR, "No WebRTC PeerConnection related to this alert...\n");
 		return;
 	}
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle) {
 		JANUS_LOG(LOG_ERR, "No ICE handle related to this alert...\n");
 		return;
 	}
-	JANUS_LOG(LOG_VERB, "[%"SCNu64"] DTLS alert triggered on stream %u (component %u), closing...\n", handle->handle_id, stream->stream_id, component->component_id);
-	janus_ice_webrtc_hangup(handle, "DTLS alert");
+	JANUS_LOG(LOG_VERB, "[%"SCNu64"] DTLS alert triggered on stream %u (component %u), closing...\n", handle->handle_id, pc->stream_id, pc->component_id);
+	janus_handle_webrtc_hangup(handle, "DTLS alert");
 }
 
 /* DTLS certificate verification callback */
@@ -974,10 +961,23 @@ void janus_dtls_wrap_sctp_data(janus_dtls_srtp *dtls, char *label, char *buf, in
 int janus_dtls_send_sctp_data(janus_dtls_srtp *dtls, char *buf, int len) {
 	if(dtls == NULL || !dtls->ready || buf == NULL || len < 1)
 		return -1;
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)dtls->pc;
+	if(pc == NULL) {
+		JANUS_LOG(LOG_ERR, "No WebRTC PeerConnection related to this data...\n");
+		return -2;
+	}
 	int res = SSL_write(dtls->ssl, buf, len);
 	if(res <= 0) {
 		unsigned long err = SSL_get_error(dtls->ssl, res);
 		JANUS_LOG(LOG_ERR, "Error sending data: %s\n", ERR_reason_error_string(err));
+	} else {
+		/* Update stats for the data medium */
+		janus_handle_webrtc_medium *medium = g_hash_table_lookup(pc->media_bytype,
+				GINT_TO_POINTER(JANUS_MEDIA_DATA));
+		if(medium != NULL) {
+			medium->out_stats.info[0].packets++;
+			medium->out_stats.info[0].bytes += res;
+		}
 	}
 	return res;
 }
@@ -985,17 +985,12 @@ int janus_dtls_send_sctp_data(janus_dtls_srtp *dtls, char *buf, int len) {
 void janus_dtls_notify_data(janus_dtls_srtp *dtls, char *label, char *buf, int len) {
 	if(dtls == NULL || buf == NULL || len < 1)
 		return;
-	janus_ice_component *component = (janus_ice_component *)dtls->component;
-	if(component == NULL) {
-		JANUS_LOG(LOG_ERR, "No component...\n");
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)dtls->pc;
+	if(pc == NULL) {
+		JANUS_LOG(LOG_ERR, "No WebRTC PeerConnection...\n");
 		return;
 	}
-	janus_ice_stream *stream = component->stream;
-	if(!stream) {
-		JANUS_LOG(LOG_ERR, "No stream...\n");
-		return;
-	}
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle || !handle->agent || !dtls->write_bio) {
 		JANUS_LOG(LOG_ERR, "No handle...\n");
 		return;
@@ -1008,16 +1003,13 @@ gboolean janus_dtls_retry(gpointer stack) {
 	janus_dtls_srtp *dtls = (janus_dtls_srtp *)stack;
 	if(dtls == NULL)
 		return FALSE;
-	janus_ice_component *component = (janus_ice_component *)dtls->component;
-	if(component == NULL)
-		return FALSE;
-	janus_ice_stream *stream = component->stream;
-	if(!stream)
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)dtls->pc;
+	if(pc == NULL)
 		goto stoptimer;
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle)
 		goto stoptimer;
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP))
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP))
 		goto stoptimer;
 	if(dtls->dtls_state == JANUS_DTLS_STATE_CONNECTED) {
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] DTLS already set up, disabling retransmission timer!\n", handle->handle_id);
@@ -1026,8 +1018,8 @@ gboolean janus_dtls_retry(gpointer stack) {
 	if(janus_get_monotonic_time() - dtls->dtls_started >= 20*G_USEC_PER_SEC) {
 		/* FIXME Should we really give up after 20 seconds waiting for DTLS? */
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] DTLS taking too much time for component %d in stream %d...\n",
-			handle->handle_id, component->component_id, stream->stream_id);
-		janus_ice_webrtc_hangup(handle, "DTLS timeout");
+			handle->handle_id, pc->component_id, pc->stream_id);
+		janus_handle_webrtc_hangup(handle, "DTLS timeout");
 		goto stoptimer;
 	}
 	struct timeval timeout = {0};
@@ -1039,7 +1031,7 @@ gboolean janus_dtls_retry(gpointer stack) {
 	JANUS_LOG(LOG_HUGE, "[%"SCNu64"] DTLSv1_get_timeout: %"SCNu64"\n", handle->handle_id, timeout_value);
 	if(timeout_value == 0) {
 		dtls->retransmissions++;
-		JANUS_LOG(LOG_VERB, "[%"SCNu64"] DTLS timeout on component %d of stream %d, retransmitting\n", handle->handle_id, component->component_id, stream->stream_id);
+		JANUS_LOG(LOG_VERB, "[%"SCNu64"] DTLS timeout on component %d of stream %d, retransmitting\n", handle->handle_id, pc->component_id, pc->stream_id);
 		/* Notify event handlers */
 		janus_dtls_notify_state_change(dtls);
 		/* Retransmit the packet */
@@ -1048,10 +1040,10 @@ gboolean janus_dtls_retry(gpointer stack) {
 	return TRUE;
 
 stoptimer:
-	if(component->dtlsrt_source != NULL) {
-		g_source_destroy(component->dtlsrt_source);
-		g_source_unref(component->dtlsrt_source);
-		component->dtlsrt_source = NULL;
+	if(pc->dtlsrt_source != NULL) {
+		g_source_destroy(pc->dtlsrt_source);
+		g_source_unref(pc->dtlsrt_source);
+		pc->dtlsrt_source = NULL;
 	}
 	return FALSE;
 }
diff --git a/dtls.h b/dtls.h
index d095b1a2cf..6c25924924 100644
--- a/dtls.h
+++ b/dtls.h
@@ -58,8 +58,8 @@ typedef enum janus_dtls_state {
 
 /*! \brief Janus DTLS-SRTP handle */
 typedef struct janus_dtls_srtp {
-	/*! \brief Opaque pointer to the component this DTLS-SRTP context belongs to */
-	void *component;
+	/*! \brief Opaque pointer to the WebRTC PeerConnection this DTLS-SRTP context belongs to */
+	void *pc;
 	/*! \brief DTLS role of the server for this stream: 1=client, 0=server */
 	janus_dtls_role dtls_role;
 	/*! \brief DTLS state of this component: -1=failed, 0=nothing, 1=trying, 2=connected */
@@ -102,10 +102,10 @@ typedef struct janus_dtls_srtp {
 
 
 /*! \brief Create a janus_dtls_srtp instance
- * @param[in] component Opaque pointer to the component owning that will use the stack
+ * @param[in] webrtc Opaque pointer to the WebRTC PeerConnection owning that will use the stack
  * @param[in] role The role of the DTLS stack (client/server)
  * @returns A new janus_dtls_srtp instance if successful, NULL otherwise */
-janus_dtls_srtp *janus_dtls_srtp_create(void *component, janus_dtls_role role);
+janus_dtls_srtp *janus_dtls_srtp_create(void *webrtc, janus_dtls_role role);
 /*! \brief Start a DTLS handshake
  * @param[in] dtls The janus_dtls_srtp instance to start the handshake on */
 void janus_dtls_srtp_handshake(janus_dtls_srtp *dtls);
diff --git a/events.c b/events.c
index b22907fbbc..c8f9d298e7 100644
--- a/events.c
+++ b/events.c
@@ -58,6 +58,7 @@ int janus_events_init(gboolean enabled, char *server_name, GHashTable *handlers)
 			g_free(server);
 			g_async_queue_unref(events);
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Events handler thread...\n", error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			return -1;
 		}
 	}
diff --git a/events/janus_mqttevh.c b/events/janus_mqttevh.c
index 0772191f31..d5352abdee 100644
--- a/events/janus_mqttevh.c
+++ b/events/janus_mqttevh.c
@@ -820,6 +820,7 @@ static int janus_mqttevh_init(const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_FATAL, "Got error %d (%s) trying to launch the MQTT EventHandler handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		goto error;
 	}
 
diff --git a/events/janus_rabbitmqevh.c b/events/janus_rabbitmqevh.c
index 0cae51db91..758e0f1a7a 100644
--- a/events/janus_rabbitmqevh.c
+++ b/events/janus_rabbitmqevh.c
@@ -349,6 +349,7 @@ int janus_rabbitmqevh_init(const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_FATAL, "Got error %d (%s) trying to launch the RabbitMQEventHandler handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		goto error;
 	}
 
diff --git a/events/janus_sampleevh.c b/events/janus_sampleevh.c
index c6b09d6a6b..23fd633f2a 100644
--- a/events/janus_sampleevh.c
+++ b/events/janus_sampleevh.c
@@ -244,6 +244,7 @@ int janus_sampleevh_init(const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the SampleEventHandler handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_SAMPLEEVH_NAME);
diff --git a/html/audiobridgetest.js b/html/audiobridgetest.js
index e091def17f..1bd96906e5 100644
--- a/html/audiobridgetest.js
+++ b/html/audiobridgetest.js
@@ -52,6 +52,7 @@ var janus = null;
 var mixertest = null;
 var opaqueId = "audiobridgetest-"+Janus.randomString(12);
 
+var remoteStream = null;
 var spinner = null;
 
 var myroom = 1234;	// Demo room
@@ -120,6 +121,19 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
+								},
+								webrtcState: function(on) {
+									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
+								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message :::");
 									Janus.debug(msg);
@@ -273,24 +287,42 @@ $(document).ready(function() {
 										mixertest.handleRemoteJsep({jsep: jsep});
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									Janus.debug(stream);
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
 									// We're not going to attach the local audio stream
 									$('#audiojoin').hide();
 									$('#room').removeClass('hide').show();
 									$('#participant').removeClass('hide').html(myusername).show();
 								},
-								onremotestream: function(stream) {
+								onremotetrack: function(track, mid, on) {
+									Janus.debug(" ::: Got a remote track event :::");
+									Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									if(remoteStream || track.kind !== "audio")
+										return;
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										if(remoteStream) {
+											try {
+												var tracks = remoteStream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										remoteStream = null;
+										$('#roomaudio').remove();
+										return;
+									}
+									remoteStream = new MediaStream();
+									remoteStream.addTrack(track.clone());
 									$('#room').removeClass('hide').show();
-									var addButtons = false;
 									if($('#roomaudio').length === 0) {
-										addButtons = true;
 										$('#mixedaudio').append('<audio class="rounded centered" id="roomaudio" width="100%" height="100%" autoplay/>');
 									}
-									Janus.attachMediaStream($('#roomaudio').get(0), stream);
-									if(!addButtons)
-										return;
+									Janus.attachMediaStream($('#roomaudio').get(0), remoteStream);
 									// Mute button
 									audioenabled = true;
 									$('#toggleaudio').click(
@@ -311,6 +343,7 @@ $(document).ready(function() {
 									$('#list').empty();
 									$('#mixedaudio').empty();
 									$('#room').hide();
+									remoteStream = null;
 								}
 							});
 					},
diff --git a/html/demos.html b/html/demos.html
index 89a56e71ff..ac2d4650ae 100644
--- a/html/demos.html
+++ b/html/demos.html
@@ -20,7 +20,7 @@
 <link rel="stylesheet" href="css/demo.css" type="text/css"/>
 </head>
 <body>
-	
+
 <a href="https://github.com/meetecho/janus-gateway"><img style="position: absolute; top: 0; left: 0; border: 0; z-index: 1001;" src="https://s3.amazonaws.com/github/ribbons/forkme_left_darkblue_121621.png" alt="Fork me on GitHub"></a>
 
 <nav class="navbar navbar-default navbar-static-top">
@@ -56,6 +56,10 @@ <h1>Janus WebRTC Server: Demo Tests</h1>
 					<td><a href="videoroomtest.html">Video Room</a></td>
 					<td>A videoconferencing demo, allowing you to join a video room with up to six users.</td>
 				</tr>
+				<tr>
+					<td><a href="mvideoroomtest.html">Video Room (multistream)</a></td>
+					<td>The same videoconferencing demo, but using one PeerConnection to receive multiple streams.</td>
+				</tr>
 				<tr>
 					<td><a href="audiobridgetest.html">Audio Room</a></td>
 					<td>An audio mixing/bridge demo, allowing you join an Audio Room room.</td>
diff --git a/html/devicetest.js b/html/devicetest.js
index 42c9b3dade..eb5857bb35 100644
--- a/html/devicetest.js
+++ b/html/devicetest.js
@@ -53,6 +53,8 @@ var echotest = null;
 var opaqueId = "devicetest-"+Janus.randomString(12);
 
 var firstTime = true;
+var localTracks = {}, localVideos = 0,
+	remoteTracks = {}, remoteVideos = 0;
 var bitrateTimer = null;
 var spinner = null;
 
@@ -245,6 +247,20 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
+								},
+								webrtcState: function(on) {
+									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
+									$("#videoleft").parent().unblock();
+								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message :::");
 									Janus.debug(msg);
@@ -260,9 +276,8 @@ $(document).ready(function() {
 											if(spinner !== null && spinner !== undefined)
 												spinner.stop();
 											spinner = null;
-											$('#myvideo').remove();
+											$('video').remove();
 											$('#waitingvideo').remove();
-											$('#peervideo').remove();
 											audioenabled = true;
 											$('#toggleaudio').attr('disabled', true).html("Disable audio").removeClass("btn-success").addClass("btn-danger");
 											videoenabled = true;
@@ -298,98 +313,181 @@ $(document).ready(function() {
 										updateSimulcastButtons(substream, temporal);
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									Janus.debug(stream);
-									if($('#myvideo').length === 0) {
-										$('#videos').removeClass('hide').show();
-										$('#videoleft').append('<video class="rounded centered" id="myvideo" width=320 height=240 autoplay playsinline muted="muted"/>');
-									}
-									Janus.attachMediaStream($('#myvideo').get(0), stream);
-									$("#myvideo").get(0).muted = "muted";
-									if(echotest.webrtcStuff.pc.iceConnectionState !== "completed" &&
-											echotest.webrtcStuff.pc.iceConnectionState !== "connected") {
-										// No remote video yet
-										$('#videoright').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
-										if(spinner == null) {
-											var target = document.getElementById('videoright');
-											spinner = new Spinner({top:100}).spin(target);
-										} else {
-											spinner.spin();
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									// Reset devices controls
+									$('#audio-device, #video-device').removeAttr('disabled');
+									$('#change-devices').removeAttr('disabled');
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
 										}
-										var videoTracks = stream.getVideoTracks();
-										if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-											// No webcam
-											$('#myvideo').hide();
-											$('#videoleft').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoleft .no-video-container').length === 0) {
+													$('#videoleft').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
 										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No webcam
-										$('#myvideo').hide();
-										if($('#videoleft .no-video-container').length === 0) {
-											$('#videoleft').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
+									if($('#videoleft video').length === 0) {
+										$('#videos').removeClass('hide').show();
+									}
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoleft .no-video-container').length === 0) {
+												$('#videoleft').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
 										}
 									} else {
+										// New video track: create a stream out of it
+										localVideos++;
 										$('#videoleft .no-video-container').remove();
-										$('#myvideo').removeClass('hide').show();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										$('#videoleft').append('<video class="rounded centered" id="myvideo' + trackId + '" width=320 height=240 autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
+									if(echotest.webrtcStuff.pc.iceConnectionState !== "completed" &&
+											echotest.webrtcStuff.pc.iceConnectionState !== "connected") {
+										$("#videoleft").parent().block({
+											message: '<b>Publishing...</b>',
+											css: {
+												border: 'none',
+												backgroundColor: 'transparent',
+												color: 'white'
+											}
+										});
 									}
-									// Reset devices controls
-									$('#audio-device, #video-device').removeAttr('disabled');
-									$('#change-devices').removeAttr('disabled');
 								},
-								onremotestream: function(stream) {
-									Janus.debug(" ::: Got a remote stream :::");
-									Janus.debug(stream);
+								onremotetrack: function(track, mid, on) {
+									Janus.debug(" ::: Got a remote track event :::");
+									Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = remoteTracks[mid];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										$('#peervideo' + mid).remove();
+										if(track.kind === "video") {
+											remoteVideos--;
+											if(remoteVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoright .no-video-container').length === 0) {
+													$('#videoright').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No remote video available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete remoteTracks[mid];
+										return;
+									}
+									// If we're here, a new track was added
 									var addButtons = false;
-									if($('#peervideo').length === 0) {
+									if($('#videoright audio').length === 0 && $('#videoright video').length === 0) {
 										addButtons = true;
 										$('#videos').removeClass('hide').show();
-										$('#videoright').append('<video class="rounded centered hide" id="peervideo" width=320 height=240 autoplay playsinline/>');
-										// Show the video, hide the spinner and show the resolution when we get a playing event
-										$("#peervideo").bind("playing", function () {
-											$('#waitingvideo').remove();
-											if(this.videoWidth)
-												$('#peervideo').removeClass('hide').show();
-											if(spinner !== null && spinner !== undefined)
-												spinner.stop();
-											spinner = null;
-											var width = this.videoWidth;
-											var height = this.videoHeight;
-											$('#curres').removeClass('hide').text(width+'x'+height).show();
-										});
 									}
-									Janus.attachMediaStream($('#peervideo').get(0), stream);
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#peervideo').hide();
-										if($('#videoright .no-video-container').length === 0) {
-											$('#videoright').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
+									if(track.kind === "audio") {
+										// New audio track: create a stream out of it, and use a hidden <audio> element
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote audio stream:", stream);
+										if($('#peervideo'+mid).length === 0)
+											$('#videoright').append('<audio class="hide" id="peervideo' + mid + '" autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										if(remoteVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoright .no-video-container').length === 0) {
+												$('#videoright').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
 										}
 									} else {
+										// New video track: create a stream out of it
+										remoteVideos++;
 										$('#videoright .no-video-container').remove();
-										$('#peervideo').removeClass('hide').show();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote video stream:", stream);
+										if($('#peervideo'+mid).length === 0)
+											$('#videoright').append('<video class="rounded centered" id="peervideo' + mid + '" width=320 height=240 autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										// Note: we'll need this for additional videos too
+										if(!bitrateTimer) {
+											$('#curbitrate').removeClass('hide').show();
+											bitrateTimer = setInterval(function() {
+												if(!$("#peervideo" + mid + ' video').get(0))
+													return;
+												// Display updated bitrate, if supported
+												var bitrate = echotest.getBitrate();
+												//~ Janus.debug("Current bitrate is " + echotest.getBitrate());
+												$('#curbitrate').text(bitrate);
+												// Check if the resolution changed too
+												var width = $("#peervideo" + mid).get(0).videoWidth;
+												var height = $("#peervideo" + mid).get(0).videoHeight;
+												if(width > 0 && height > 0)
+													$('#curres').removeClass('hide').text(width+'x'+height).show();
+											}, 1000);
+										}
 									}
 									if(!addButtons)
 										return;
 									// Enable audio/video buttons and bitrate limiter
 									audioenabled = true;
 									videoenabled = true;
-									$('#toggleaudio').removeAttr('disabled').click(
+									$('#toggleaudio').click(
 										function() {
 											audioenabled = !audioenabled;
 											if(audioenabled)
@@ -398,7 +496,7 @@ $(document).ready(function() {
 												$('#toggleaudio').html("Enable audio").removeClass("btn-danger").addClass("btn-success");
 											echotest.send({"message": { "audio": audioenabled }});
 										});
-									$('#togglevideo').removeAttr('disabled').click(
+									$('#togglevideo').click(
 										function() {
 											videoenabled = !videoenabled;
 											if(videoenabled)
@@ -408,7 +506,7 @@ $(document).ready(function() {
 											echotest.send({"message": { "video": videoenabled }});
 										});
 									$('#toggleaudio').parent().removeClass('hide').show();
-									$('#bitrate a').removeAttr('disabled').click(function() {
+									$('#bitrate a').click(function() {
 										var id = $(this).attr("id");
 										var bitrate = parseInt(id)*1000;
 										if(bitrate === 0) {
@@ -420,21 +518,6 @@ $(document).ready(function() {
 										echotest.send({"message": { "bitrate": bitrate }});
 										return false;
 									});
-									if(Janus.webRTCAdapter.browserDetails.browser === "chrome" || Janus.webRTCAdapter.browserDetails.browser === "firefox" ||
-											Janus.webRTCAdapter.browserDetails.browser === "safari") {
-										$('#curbitrate').removeClass('hide').show();
-										bitrateTimer = setInterval(function() {
-											// Display updated bitrate, if supported
-											var bitrate = echotest.getBitrate();
-											//~ Janus.debug("Current bitrate is " + echotest.getBitrate());
-											$('#curbitrate').text(bitrate);
-											// Check if the resolution changed too
-											var width = $("#peervideo").get(0).videoWidth;
-											var height = $("#peervideo").get(0).videoHeight;
-											if(width > 0 && height > 0)
-												$('#curres').removeClass('hide').text(width+'x'+height).show();
-										}, 1000);
-									}
 								},
 								ondataopen: function(data) {
 									Janus.log("The DataChannel is available!");
@@ -447,13 +530,11 @@ $(document).ready(function() {
 								},
 								oncleanup: function() {
 									Janus.log(" ::: Got a cleanup notification :::");
+									$("#videoleft").empty().parent().unblock();
+									$('#videoright').empty();
 									if(spinner !== null && spinner !== undefined)
 										spinner.stop();
 									spinner = null;
-									$('#myvideo').remove();
-									$('#waitingvideo').remove();
-									$('#peervideo').remove();
-									$('.no-video-container').remove();
 									audioenabled = true;
 									$('#toggleaudio').attr('disabled', true).html("Disable audio").removeClass("btn-success").addClass("btn-danger");
 									videoenabled = true;
@@ -470,6 +551,10 @@ $(document).ready(function() {
 									$('#outputdeviceset').html('Output device<span class="caret"></span>');
 									simulcastStarted = false;
 									$('#simulcast').remove();
+									localTracks = {};
+									localVideos = 0;
+									remoteTracks = {};
+									remoteVideos = 0;
 								}
 							});
 					},
diff --git a/html/echotest.js b/html/echotest.js
index deb879647a..549dd226f6 100644
--- a/html/echotest.js
+++ b/html/echotest.js
@@ -52,6 +52,8 @@ var janus = null;
 var echotest = null;
 var opaqueId = "echotest-"+Janus.randomString(12);
 
+var localTracks = {}, localVideos = 0,
+	remoteTracks = {}, remoteVideos = 0;
 var bitrateTimer = null;
 var spinner = null;
 
@@ -163,16 +165,16 @@ $(document).ready(function() {
 								iceState: function(state) {
 									Janus.log("ICE state changed to " + state);
 								},
-								mediaState: function(medium, on) {
-									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium);
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
 								},
 								webrtcState: function(on) {
 									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
 									$("#videoleft").parent().unblock();
 								},
-								slowLink: function(uplink, lost) {
+								slowLink: function(uplink, lost, mid) {
 									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
-										" packets on this PeerConnection (" + lost + " lost packets)");
+										" packets on mid " + mid + " (" + lost + " lost packets)");
 								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message :::");
@@ -190,9 +192,8 @@ $(document).ready(function() {
 											if(spinner !== null && spinner !== undefined)
 												spinner.stop();
 											spinner = null;
-											$('#myvideo').remove();
+											$('video').remove();
 											$('#waitingvideo').remove();
-											$('#peervideo').remove();
 											$('#toggleaudio').attr('disabled', true);
 											$('#togglevideo').attr('disabled', true);
 											$('#bitrate').attr('disabled', true);
@@ -220,15 +221,73 @@ $(document).ready(function() {
 										updateSimulcastButtons(substream, temporal);
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									Janus.debug(stream);
-									if($('#myvideo').length === 0) {
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoleft .no-video-container').length === 0) {
+													$('#videoleft').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
+									if($('#videoleft video').length === 0) {
 										$('#videos').removeClass('hide').show();
-										$('#videoleft').append('<video class="rounded centered" id="myvideo" width=320 height=240 autoplay playsinline muted="muted"/>');
 									}
-									Janus.attachMediaStream($('#myvideo').get(0), stream);
-									$("#myvideo").get(0).muted = "muted";
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoleft .no-video-container').length === 0) {
+												$('#videoleft').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videoleft .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										$('#videoleft').append('<video class="rounded centered" id="myvideo' + trackId + '" width=320 height=240 autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
 									if(echotest.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											echotest.webrtcStuff.pc.iceConnectionState !== "connected") {
 										$("#videoleft").parent().block({
@@ -239,67 +298,94 @@ $(document).ready(function() {
 												color: 'white'
 											}
 										});
-										// No remote video yet
-										$('#videoright').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
-										if(spinner == null) {
-											var target = document.getElementById('videoright');
-											spinner = new Spinner({top:100}).spin(target);
-										} else {
-											spinner.spin();
-										}
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No webcam
-										$('#myvideo').hide();
-										if($('#videoleft .no-video-container').length === 0) {
-											$('#videoleft').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
+								},
+								onremotetrack: function(track, mid, on) {
+									Janus.debug(" ::: Got a remote track event :::");
+									Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = remoteTracks[mid];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
 										}
-									} else {
-										$('#videoleft .no-video-container').remove();
-										$('#myvideo').removeClass('hide').show();
+										$('#peervideo' + mid).remove();
+										if(track.kind === "video") {
+											remoteVideos--;
+											if(remoteVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoright .no-video-container').length === 0) {
+													$('#videoright').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No remote video available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete remoteTracks[mid];
+										return;
 									}
-								},
-								onremotestream: function(stream) {
-									Janus.debug(" ::: Got a remote stream :::");
-									Janus.debug(stream);
+									// If we're here, a new track was added
 									var addButtons = false;
-									if($('#peervideo').length === 0) {
+									if($('#videoright audio').length === 0 && $('#videoright video').length === 0) {
 										addButtons = true;
 										$('#videos').removeClass('hide').show();
-										$('#videoright').append('<video class="rounded centered hide" id="peervideo" width=320 height=240 autoplay playsinline/>');
-										// Show the video, hide the spinner and show the resolution when we get a playing event
-										$("#peervideo").bind("playing", function () {
-											$('#waitingvideo').remove();
-											if(this.videoWidth)
-												$('#peervideo').removeClass('hide').show();
-											if(spinner !== null && spinner !== undefined)
-												spinner.stop();
-											spinner = null;
-											var width = this.videoWidth;
-											var height = this.videoHeight;
-											$('#curres').removeClass('hide').text(width+'x'+height).show();
-										});
 									}
-									Janus.attachMediaStream($('#peervideo').get(0), stream);
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#peervideo').hide();
-										if($('#videoright .no-video-container').length === 0) {
-											$('#videoright').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
+									if(track.kind === "audio") {
+										// New audio track: create a stream out of it, and use a hidden <audio> element
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote audio stream:", stream);
+										if($('#peervideo'+mid).length === 0)
+											$('#videoright').append('<audio class="hide" id="peervideo' + mid + '" autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										if(remoteVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoright .no-video-container').length === 0) {
+												$('#videoright').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
 										}
 									} else {
+										// New video track: create a stream out of it
+										remoteVideos++;
 										$('#videoright .no-video-container').remove();
-										$('#peervideo').removeClass('hide').show();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote video stream:", stream);
+										if($('#peervideo'+mid).length === 0)
+											$('#videoright').append('<video class="rounded centered" id="peervideo' + mid + '" width=320 height=240 autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										// Note: we'll need this for additional videos too
+										if(!bitrateTimer) {
+											$('#curbitrate').removeClass('hide').show();
+											bitrateTimer = setInterval(function() {
+												if(!$("#peervideo" + mid + ' video').get(0))
+													return;
+												// Display updated bitrate, if supported
+												var bitrate = echotest.getBitrate();
+												//~ Janus.debug("Current bitrate is " + echotest.getBitrate());
+												$('#curbitrate').text(bitrate);
+												// Check if the resolution changed too
+												var width = $("#peervideo" + mid).get(0).videoWidth;
+												var height = $("#peervideo" + mid).get(0).videoHeight;
+												if(width > 0 && height > 0)
+													$('#curres').removeClass('hide').text(width+'x'+height).show();
+											}, 1000);
+										}
 									}
 									if(!addButtons)
 										return;
@@ -337,21 +423,6 @@ $(document).ready(function() {
 										echotest.send({"message": { "bitrate": bitrate }});
 										return false;
 									});
-									if(Janus.webRTCAdapter.browserDetails.browser === "chrome" || Janus.webRTCAdapter.browserDetails.browser === "firefox" ||
-											Janus.webRTCAdapter.browserDetails.browser === "safari") {
-										$('#curbitrate').removeClass('hide').show();
-										bitrateTimer = setInterval(function() {
-											// Display updated bitrate, if supported
-											var bitrate = echotest.getBitrate();
-											//~ Janus.debug("Current bitrate is " + echotest.getBitrate());
-											$('#curbitrate').text(bitrate);
-											// Check if the resolution changed too
-											var width = $("#peervideo").get(0).videoWidth;
-											var height = $("#peervideo").get(0).videoHeight;
-											if(width > 0 && height > 0)
-												$('#curres').removeClass('hide').text(width+'x'+height).show();
-										}, 1000);
-									}
 								},
 								ondataopen: function(data) {
 									Janus.log("The DataChannel is available!");
@@ -370,10 +441,8 @@ $(document).ready(function() {
 									if(bitrateTimer)
 										clearInterval(bitrateTimer);
 									bitrateTimer = null;
-									$('#myvideo').remove();
-									$('#waitingvideo').remove();
-									$("#videoleft").parent().unblock();
-									$('#peervideo').remove();
+									$("#videoleft").empty().parent().unblock();
+									$('#videoright').empty();
 									$('#toggleaudio').attr('disabled', true);
 									$('#togglevideo').attr('disabled', true);
 									$('#bitrate').attr('disabled', true);
@@ -382,6 +451,10 @@ $(document).ready(function() {
 									$('#datasend').attr('disabled', true);
 									simulcastStarted = false;
 									$('#simulcast').remove();
+									localTracks = {};
+									localVideos = 0;
+									remoteTracks = {};
+									remoteVideos = 0;
 								}
 							});
 					},
diff --git a/html/janus.js b/html/janus.js
index 6bcc53a3a9..c2987c252e 100644
--- a/html/janus.js
+++ b/html/janus.js
@@ -721,7 +721,7 @@ function Janus(gatewayCallbacks) {
 				Janus.debug("This handle is not attached to this session");
 				return;
 			}
-			pluginHandle.mediaState(json["type"], json["receiving"]);
+			pluginHandle.mediaState(json["type"], json["receiving"], json["mid"]);
 		} else if(json["janus"] === "slowlink") {
 			Janus.debug("Got a slowlink event on session " + sessionId);
 			Janus.debug(json);
@@ -736,7 +736,7 @@ function Janus(gatewayCallbacks) {
 				Janus.debug("This handle is not attached to this session");
 				return;
 			}
-			pluginHandle.slowLink(json["uplink"], json["lost"]);
+			pluginHandle.slowLink(json["uplink"], json["lost"], json["mid"]);
 		} else if(json["janus"] === "error") {
 			// Oops, something wrong happened
 			Janus.error("Ooops: " + json["error"].code + " " + json["error"].reason);	// FIXME
@@ -1069,8 +1069,8 @@ function Janus(gatewayCallbacks) {
 		callbacks.webrtcState = (typeof callbacks.webrtcState == "function") ? callbacks.webrtcState : Janus.noop;
 		callbacks.slowLink = (typeof callbacks.slowLink == "function") ? callbacks.slowLink : Janus.noop;
 		callbacks.onmessage = (typeof callbacks.onmessage == "function") ? callbacks.onmessage : Janus.noop;
-		callbacks.onlocalstream = (typeof callbacks.onlocalstream == "function") ? callbacks.onlocalstream : Janus.noop;
-		callbacks.onremotestream = (typeof callbacks.onremotestream == "function") ? callbacks.onremotestream : Janus.noop;
+		callbacks.onlocaltrack = (typeof callbacks.onlocaltrack == "function") ? callbacks.onlocaltrack : Janus.noop;
+		callbacks.onremotetrack = (typeof callbacks.onremotetrack == "function") ? callbacks.onremotetrack : Janus.noop;
 		callbacks.ondata = (typeof callbacks.ondata == "function") ? callbacks.ondata : Janus.noop;
 		callbacks.ondataopen = (typeof callbacks.ondataopen == "function") ? callbacks.ondataopen : Janus.noop;
 		callbacks.oncleanup = (typeof callbacks.oncleanup == "function") ? callbacks.oncleanup : Janus.noop;
@@ -1123,31 +1123,20 @@ function Janus(gatewayCallbacks) {
 							dtmfSender : null,
 							trickle : true,
 							iceDone : false,
-							volume : {
-								value : null,
-								timer : null
-							},
-							bitrate : {
-								value : null,
-								bsnow : null,
-								bsbefore : null,
-								tsnow : null,
-								tsbefore : null,
-								timer : null
-							}
+							bitrate : {}
 						},
 						getId : function() { return handleId; },
 						getPlugin : function() { return plugin; },
-						getVolume : function() { return getVolume(handleId, true); },
-						getRemoteVolume : function() { return getVolume(handleId, true); },
-						getLocalVolume : function() { return getVolume(handleId, false); },
-						isAudioMuted : function() { return isMuted(handleId, false); },
-						muteAudio : function() { return mute(handleId, false, true); },
-						unmuteAudio : function() { return mute(handleId, false, false); },
-						isVideoMuted : function() { return isMuted(handleId, true); },
-						muteVideo : function() { return mute(handleId, true, true); },
-						unmuteVideo : function() { return mute(handleId, true, false); },
-						getBitrate : function() { return getBitrate(handleId); },
+						getVolume : function(mid, result) { return getVolume(handleId, mid, true, result); },
+						getRemoteVolume : function(mid, result) { return getVolume(handleId, mid, true, result); },
+						getLocalVolume : function(mid, result) { return getVolume(handleId, mid, false, result); },
+						isAudioMuted : function(mid) { return isMuted(handleId, mid, false); },
+						muteAudio : function(mid) { return mute(handleId, mid, false, true); },
+						unmuteAudio : function(mid) { return mute(handleId, mid, false, false); },
+						isVideoMuted : function(mid) { return isMuted(handleId, mid, true); },
+						muteVideo : function(mid) { return mute(handleId, mid, true, true); },
+						unmuteVideo : function(mid) { return mute(handleId, mid, true, false); },
+						getBitrate : function(mid) { return getBitrate(handleId, mid); },
 						send : function(callbacks) { sendMessage(handleId, callbacks); },
 						data : function(callbacks) { sendData(handleId, callbacks); },
 						dtmf : function(callbacks) { sendDtmf(handleId, callbacks); },
@@ -1160,8 +1149,8 @@ function Janus(gatewayCallbacks) {
 						createOffer : function(callbacks) { prepareWebrtc(handleId, true, callbacks); },
 						createAnswer : function(callbacks) { prepareWebrtc(handleId, false, callbacks); },
 						handleRemoteJsep : function(callbacks) { prepareWebrtcPeer(handleId, callbacks); },
-						onlocalstream : callbacks.onlocalstream,
-						onremotestream : callbacks.onremotestream,
+						onlocaltrack : callbacks.onlocaltrack,
+						onremotetrack : callbacks.onremotetrack,
 						ondata : callbacks.ondata,
 						ondataopen : callbacks.ondataopen,
 						oncleanup : callbacks.oncleanup,
@@ -1208,31 +1197,20 @@ function Janus(gatewayCallbacks) {
 							dtmfSender : null,
 							trickle : true,
 							iceDone : false,
-							volume : {
-								value : null,
-								timer : null
-							},
-							bitrate : {
-								value : null,
-								bsnow : null,
-								bsbefore : null,
-								tsnow : null,
-								tsbefore : null,
-								timer : null
-							}
+							bitrate : {}
 						},
 						getId : function() { return handleId; },
 						getPlugin : function() { return plugin; },
-						getVolume : function() { return getVolume(handleId, true); },
-						getRemoteVolume : function() { return getVolume(handleId, true); },
-						getLocalVolume : function() { return getVolume(handleId, false); },
-						isAudioMuted : function() { return isMuted(handleId, false); },
-						muteAudio : function() { return mute(handleId, false, true); },
-						unmuteAudio : function() { return mute(handleId, false, false); },
-						isVideoMuted : function() { return isMuted(handleId, true); },
-						muteVideo : function() { return mute(handleId, true, true); },
-						unmuteVideo : function() { return mute(handleId, true, false); },
-						getBitrate : function() { return getBitrate(handleId); },
+						getVolume : function(mid, result) { return getVolume(handleId, mid, true, result); },
+						getRemoteVolume : function(mid, result) { return getVolume(handleId, mid, true, result); },
+						getLocalVolume : function(mid, result) { return getVolume(handleId, mid, false, result); },
+						isAudioMuted : function(mid) { return isMuted(handleId, mid, false); },
+						muteAudio : function(mid) { return mute(handleId, mid, false, true); },
+						unmuteAudio : function(mid) { return mute(handleId, mid, false, false); },
+						isVideoMuted : function(mid) { return isMuted(handleId, mid, true); },
+						muteVideo : function(mid) { return mute(handleId, mid, true, true); },
+						unmuteVideo : function(mid) { return mute(handleId, mid, true, false); },
+						getBitrate : function(mid) { return getBitrate(handleId, mid); },
 						send : function(callbacks) { sendMessage(handleId, callbacks); },
 						data : function(callbacks) { sendData(handleId, callbacks); },
 						dtmf : function(callbacks) { sendDtmf(handleId, callbacks); },
@@ -1245,8 +1223,8 @@ function Janus(gatewayCallbacks) {
 						createOffer : function(callbacks) { prepareWebrtc(handleId, true, callbacks); },
 						createAnswer : function(callbacks) { prepareWebrtc(handleId, false, callbacks); },
 						handleRemoteJsep : function(callbacks) { prepareWebrtcPeer(handleId, callbacks); },
-						onlocalstream : callbacks.onlocalstream,
-						onremotestream : callbacks.onremotestream,
+						onlocaltrack : callbacks.onlocaltrack,
+						onremotetrack : callbacks.onremotetrack,
 						ondata : callbacks.ondata,
 						ondataopen : callbacks.ondataopen,
 						oncleanup : callbacks.oncleanup,
@@ -1669,13 +1647,10 @@ function Janus(gatewayCallbacks) {
 							}
 						}
 					}
-					if(audioTransceiver && audioTransceiver.sender) {
-						audioTransceiver.sender.replaceTrack(stream.getAudioTracks()[0]);
-					} else {
-						config.pc.addTrack(stream.getAudioTracks()[0], stream);
-					}
+				}
+				if(audioTransceiver && audioTransceiver.sender) {
+					audioTransceiver.sender.replaceTrack(stream.getAudioTracks()[0]);
 				} else {
-					Janus.log((media.replaceAudio ? "Replacing" : "Adding") + " audio track:", stream.getAudioTracks()[0]);
 					config.pc.addTrack(stream.getAudioTracks()[0], stream);
 				}
 			}
@@ -1697,24 +1672,22 @@ function Janus(gatewayCallbacks) {
 							}
 						}
 					}
-					if(videoTransceiver && videoTransceiver.sender) {
-						videoTransceiver.sender.replaceTrack(stream.getVideoTracks()[0]);
-					} else {
-						config.pc.addTrack(stream.getVideoTracks()[0], stream);
-					}
+				}
+				if(videoTransceiver && videoTransceiver.sender) {
+					videoTransceiver.sender.replaceTrack(stream.getVideoTracks()[0]);
 				} else {
-					Janus.log((media.replaceVideo ? "Replacing" : "Adding") + " video track:", stream.getVideoTracks()[0]);
 					config.pc.addTrack(stream.getVideoTracks()[0], stream);
 				}
 			}
 		}
 		// If we still need to create a PeerConnection, let's do that
 		if(!config.pc) {
-			var pc_config = {"iceServers": iceServers, "iceTransportPolicy": iceTransportPolicy, "bundlePolicy": bundlePolicy};
-			if(Janus.webRTCAdapter.browserDetails.browser === "chrome") {
-				// For Chrome versions before 72, we force a plan-b semantic, and unified-plan otherwise
-				pc_config["sdpSemantics"] = (Janus.webRTCAdapter.browserDetails.version < 72) ? "plan-b" : "unified-plan";
-			}
+			var pc_config = {
+				iceServers: iceServers,
+				iceTransportPolicy: iceTransportPolicy,
+				bundlePolicy: bundlePolicy
+			};
+			pc_config.sdpSemantics = Janus.unifiedPlan ? "unified-plan" : "plan-b";
 			var pc_constraints = {
 				"optional": [{"DtlsSrtpKeyAgreement": true}]
 			};
@@ -1736,10 +1709,6 @@ function Janus(gatewayCallbacks) {
 			Janus.debug(pc_constraints);
 			config.pc = new RTCPeerConnection(pc_config, pc_constraints);
 			Janus.debug(config.pc);
-			if(config.pc.getStats) {	// FIXME
-				config.volume = {};
-				config.bitrate.value = "0 kbits/sec";
-			}
 			Janus.log("Preparing local SDP and gathering candidates (trickle=" + config.trickle + ")");
 			config.pc.oniceconnectionstatechange = function(e) {
 				if(config.pc)
@@ -1777,27 +1746,44 @@ function Janus(gatewayCallbacks) {
 				if(!event.streams)
 					return;
 				config.remoteStream = event.streams[0];
-				pluginHandle.onremotestream(config.remoteStream);
-				if(event.track.onended)
-					return;
-				Janus.log("Adding onended callback to track:", event.track);
-				event.track.onended = function(ev) {
-					Janus.log("Remote track muted/removed:", ev);
-					if(config.remoteStream) {
-						config.remoteStream.removeTrack(ev.target);
-						pluginHandle.onremotestream(config.remoteStream);
-					}
-				};
-				event.track.onmute = event.track.onended;
-				event.track.onunmute = function(ev) {
-					Janus.log("Remote track flowing again:", ev);
+				if(event.track) {
+					// Notify about the new track
+					var mid = event.transceiver ? event.transceiver.mid : event.track.id;
 					try {
-						config.remoteStream.addTrack(ev.target);
-						pluginHandle.onremotestream(config.remoteStream);
+						pluginHandle.onremotetrack(event.track, mid, true);
 					} catch(e) {
 						Janus.error(e);
 					};
-				};
+					if(event.track.onended)
+						return;
+					Janus.log("Adding onended callback to track:", event.track);
+					event.track.onended = function(ev) {
+						Janus.log("Remote track removed:", ev);
+						if(config.remoteStream) {
+							config.remoteStream.removeTrack(ev.target);
+							var mid = ev.target.id;
+							if(Janus.unifiedPlan) {
+								var transceiver = config.pc.getTransceivers().find(
+									t => t.receiver.track === ev.target);
+								mid = transceiver.mid;
+							}
+							try {
+								pluginHandle.onremotetrack(ev.target, mid, false);
+							} catch(e) {
+								Janus.error(e);
+							};
+						}
+					};
+					event.track.onmute = event.track.onended;
+					event.track.onunmute = function(ev) {
+						Janus.log("Remote track flowing again:", ev);
+						try {
+							pluginHandle.onremotetrack(ev.target, mid, true);
+						} catch(e) {
+							Janus.error(e);
+						};
+					};
+				}
 			};
 		}
 		if(addTracks && stream !== null && stream !== undefined) {
@@ -1836,8 +1822,26 @@ function Janus(gatewayCallbacks) {
 			};
 		}
 		// If there's a new local stream, let's notify the application
-		if(config.myStream)
-			pluginHandle.onlocalstream(config.myStream);
+		if(config.myStream) {
+			var tracks = config.myStream.getTracks();
+			for(var i in tracks) {
+				var track = tracks[i];
+				track.onended = function(ev) {
+					// FIXME What does this event contain? Is there a reference to the track?
+					Janus.log("Local track removed:", ev);
+					try {
+						pluginHandle.onlocaltrack(ev.track, false);
+					} catch(e) {
+						Janus.error(e);
+					};
+				}
+				try {
+					pluginHandle.onlocaltrack(track, true);
+				} catch(e) {
+					Janus.error(e);
+				};
+			}
+		}
 		// Create offer/answer now
 		if(jsep === null || jsep === undefined) {
 			createOffer(handleId, media, callbacks);
@@ -2042,6 +2046,11 @@ function Janus(gatewayCallbacks) {
 					var s = config.myStream.getAudioTracks()[0];
 					Janus.log("Removing audio track:", s);
 					config.myStream.removeTrack(s);
+					try {
+						pluginHandle.onlocaltrack(s, false);
+					} catch(e) {
+						Janus.error(e);
+					};
 					try {
 						s.stop();
 					} catch(e) {};
@@ -2068,6 +2077,11 @@ function Janus(gatewayCallbacks) {
 					var s = config.myStream.getVideoTracks()[0];
 					Janus.log("Removing video track:", s);
 					config.myStream.removeTrack(s);
+					try {
+						pluginHandle.onlocaltrack(s, false);
+					} catch(e) {
+						Janus.error(e);
+					};
 					try {
 						s.stop();
 					} catch(e) {};
@@ -2917,55 +2931,85 @@ function Janus(gatewayCallbacks) {
 		callbacks.success(config.mySdp);
 	}
 
-	function getVolume(handleId, remote) {
+	function getVolume(handleId, mid, remote, result) {
+		result = (typeof result == "function") ? result : Janus.noop;
 		var pluginHandle = pluginHandles[handleId];
 		if(pluginHandle === null || pluginHandle === undefined ||
 				pluginHandle.webrtcStuff === null || pluginHandle.webrtcStuff === undefined) {
 			Janus.warn("Invalid handle");
-			return 0;
+			result(0);
+			return;
 		}
 		var stream = remote ? "remote" : "local";
 		var config = pluginHandle.webrtcStuff;
-		if(!config.volume[stream])
-			config.volume[stream] = { value: 0 };
-		// Start getting the volume, if getStats is supported
-		if(config.pc.getStats && Janus.webRTCAdapter.browserDetails.browser === "chrome") {
+		// Note well: this relies on the audioLevel property as returned
+		// by getStats, so it won't work if the browser doesn't do them.
+		// At the time of writing, only Chrome (and maybe Safari?) support
+		// audioLevel, while Firefox doesn't return it. Besides, it looks
+		// like audioLevel is set only for remote tracks, while it's always
+		// 0 for local tracks: as such, getLocalVolume is useless right now.
+		if(config.pc.getStats) {
 			if(remote && (config.remoteStream === null || config.remoteStream === undefined)) {
 				Janus.warn("Remote stream unavailable");
-				return 0;
+				result(0);
+				return;
 			} else if(!remote && (config.myStream === null || config.myStream === undefined)) {
 				Janus.warn("Local stream unavailable");
-				return 0;
-			}
-			if(config.volume[stream].timer === null || config.volume[stream].timer === undefined) {
-				Janus.log("Starting " + stream + " volume monitor");
-				config.volume[stream].timer = setInterval(function() {
-					config.pc.getStats()
-						.then(function(stats) {
-							var results = stats.result();
-							for(var i=0; i<results.length; i++) {
-								var res = results[i];
-								if(res.type == 'ssrc') {
-									if(remote && res.stat('audioOutputLevel'))
-										config.volume[stream].value = parseInt(res.stat('audioOutputLevel'));
-									else if(!remote && res.stat('audioInputLevel'))
-										config.volume[stream].value = parseInt(res.stat('audioInputLevel'));
-								}
-							}
-						});
-				}, 200);
-				return 0;	// We don't have a volume to return yet
+				result(0);
+				return;
 			}
-			return config.volume[stream].value;
+			// Are we interested in a mid in particular? (only if transceivers are in use)
+			var query = config.pc;
+			if(mid && Janus.unifiedPlan) {
+				var transceiver = config.pc.getTransceivers()
+					.find(t => (t.mid === mid && t.receiver.track.kind === "audio"));
+				if(!transceiver) {
+					Janus.warn("No audio transceiver with mid " + mid);
+					result(0);
+					return;
+				}
+				if(remote && !transceiver.receiver) {
+					Janus.warn("Remote transceiver track unavailable");
+					result(0);
+					return;
+				} else if(!remote && !transceiver.sender) {
+					Janus.warn("Local transceiver track unavailable");
+					result(0);
+					return;
+				}
+				query = remote ? transceiver.receiver : transceiver.sender;
+			}
+			var currentVolume = -1;
+			query.getStats()
+				.then(function(stats) {
+					var size = stats.size;
+					stats.forEach(function (res) {
+						size--;
+						if(!res || !res.id || res.type !== "track") {
+							if(size === 0 && currentVolume === -1)
+								result(0);
+							return;
+						}
+						console.log(res);
+						if((remote && res.id.indexOf("receiver") === -1) ||
+								(!remote && res.id.indexOf("sender") === -1)) {
+							if(size === 0 && currentVolume === -1)
+								result(0);
+							return;
+						}
+						if(currentVolume !== -1)
+							return;
+						currentVolume = res.audioLevel;
+						result(currentVolume);
+					});
+				});
 		} else {
-			// audioInputLevel and audioOutputLevel seem only available in Chrome? audioLevel
-			// seems to be available on Chrome and Firefox, but they don't seem to work
 			Janus.warn("Getting the " + stream + " volume unsupported by browser");
-			return 0;
+			result(0);
 		}
 	}
 
-	function isMuted(handleId, video) {
+	function isMuted(handleId, mid, video) {
 		var pluginHandle = pluginHandles[handleId];
 		if(pluginHandle === null || pluginHandle === undefined ||
 				pluginHandle.webrtcStuff === null || pluginHandle.webrtcStuff === undefined) {
@@ -2989,7 +3033,21 @@ function Janus(gatewayCallbacks) {
 				Janus.warn("No video track");
 				return true;
 			}
-			return !config.myStream.getVideoTracks()[0].enabled;
+			if(mid && Janus.unifiedPlan) {
+				var transceiver = config.pc.getTransceivers()
+					.find(t => (t.mid === mid && t.receiver.track.kind === "video"));
+				if(!transceiver) {
+					Janus.warn("No video transceiver with mid " + mid);
+					return true;
+				}
+				if(!transceiver.sender || !transceiver.sender.track) {
+					Janus.warn("No video sender with mid " + mid);
+					return true;
+				}
+				return !transceiver.sender.track.enabled;
+			} else {
+				return !config.myStream.getVideoTracks()[0].enabled;
+			}
 		} else {
 			// Check audio track
 			if(config.myStream.getAudioTracks() === null
@@ -2998,11 +3056,25 @@ function Janus(gatewayCallbacks) {
 				Janus.warn("No audio track");
 				return true;
 			}
-			return !config.myStream.getAudioTracks()[0].enabled;
+			if(mid && Janus.unifiedPlan) {
+				var transceiver = config.pc.getTransceivers()
+					.find(t => (t.mid === mid && t.receiver.track.kind === "audio"));
+				if(!transceiver) {
+					Janus.warn("No audio transceiver with mid " + mid);
+					return true;
+				}
+				if(!transceiver.sender || !transceiver.sender.track) {
+					Janus.warn("No audio sender with mid " + mid);
+					return true;
+				}
+				return !transceiver.sender.track.enabled;
+			} else {
+				return !config.myStream.getAudioTracks()[0].enabled;
+			}
 		}
 	}
 
-	function mute(handleId, video, mute) {
+	function mute(handleId, mid, video, mute) {
 		var pluginHandle = pluginHandles[handleId];
 		if(pluginHandle === null || pluginHandle === undefined ||
 				pluginHandle.webrtcStuff === null || pluginHandle.webrtcStuff === undefined) {
@@ -3026,8 +3098,21 @@ function Janus(gatewayCallbacks) {
 				Janus.warn("No video track");
 				return false;
 			}
-			config.myStream.getVideoTracks()[0].enabled = mute ? false : true;
-			return true;
+			if(mid && Janus.unifiedPlan) {
+				var transceiver = config.pc.getTransceivers()
+					.find(t => (t.mid === mid && t.receiver.track.kind === "video"));
+				if(!transceiver) {
+					Janus.warn("No video transceiver with mid " + mid);
+					return false;
+				}
+				if(!transceiver.sender || !transceiver.sender.track) {
+					Janus.warn("No video sender with mid " + mid);
+					return false;
+				}
+				transceiver.sender.track.enabled = mute ? false : true;
+			} else {
+				config.myStream.getVideoTracks()[0].enabled = mute ? false : true;
+			}
 		} else {
 			// Mute/unmute audio track
 			if(config.myStream.getAudioTracks() === null
@@ -3036,12 +3121,26 @@ function Janus(gatewayCallbacks) {
 				Janus.warn("No audio track");
 				return false;
 			}
-			config.myStream.getAudioTracks()[0].enabled = mute ? false : true;
-			return true;
+			if(mid && Janus.unifiedPlan) {
+				var transceiver = config.pc.getTransceivers()
+					.find(t => (t.mid === mid && t.receiver.track.kind === "audio"));
+				if(!transceiver) {
+					Janus.warn("No audio transceiver with mid " + mid);
+					return false;
+				}
+				if(!transceiver.sender || !transceiver.sender.track) {
+					Janus.warn("No audio sender with mid " + mid);
+					return false;
+				}
+				transceiver.sender.track.enabled = mute ? false : true;
+			} else {
+				config.myStream.getAudioTracks()[0].enabled = mute ? false : true;
+			}
 		}
+		return true;
 	}
 
-	function getBitrate(handleId) {
+	function getBitrate(handleId, mid) {
 		var pluginHandle = pluginHandles[handleId];
 		if(pluginHandle === null || pluginHandle === undefined ||
 				pluginHandle.webrtcStuff === null || pluginHandle.webrtcStuff === undefined) {
@@ -3049,14 +3148,41 @@ function Janus(gatewayCallbacks) {
 			return "Invalid handle";
 		}
 		var config = pluginHandle.webrtcStuff;
-		if(config.pc === null || config.pc === undefined)
+		if(config.pc === null || config.pc === undefined) {
+			Janus.warn("Invalid PeerConnection");
 			return "Invalid PeerConnection";
+		}
 		// Start getting the bitrate, if getStats is supported
 		if(config.pc.getStats) {
-			if(config.bitrate.timer === null || config.bitrate.timer === undefined) {
-				Janus.log("Starting bitrate timer (via getStats)");
-				config.bitrate.timer = setInterval(function() {
-					config.pc.getStats()
+			var query = config.pc;
+			var target = mid ? mid : "default";
+			if(mid && Janus.unifiedPlan) {
+				var transceiver = config.pc.getTransceivers()
+					.find(t => (t.mid === mid && t.receiver.track.kind === "video"));
+				if(!transceiver) {
+					Janus.warn("No video transceiver with mid " + mid);
+					return ("No video transceiver with mid " + mid);
+				}
+				if(!transceiver.receiver) {
+					Janus.warn("No video receiver with mid " + mid);
+					return ("No video receiver with mid " + mid);
+				}
+				query = transceiver.receiver;
+			}
+			if(!config.bitrate[target]) {
+				config.bitrate[target] = {
+					timer: null,
+					bsnow: null,
+					bsbefore: null,
+					tsnow: null,
+					tsbefore: null,
+					value: "0 kbits/sec"
+				};
+			}
+			if(!config.bitrate[target].timer) {
+				Janus.log("Starting bitrate timer" + (mid ? (" for mid " + mid) : "") + " (via getStats)");
+				config.bitrate[target].timer = setInterval(function() {
+					query.getStats()
 						.then(function(stats) {
 							stats.forEach(function (res) {
 								if(!res)
@@ -3074,32 +3200,31 @@ function Janus(gatewayCallbacks) {
 								}
 								// Parse stats now
 								if(inStats) {
-									config.bitrate.bsnow = res.bytesReceived;
-									config.bitrate.tsnow = res.timestamp;
-									if(config.bitrate.bsbefore === null || config.bitrate.tsbefore === null) {
+									config.bitrate[target].bsnow = res.bytesReceived;
+									config.bitrate[target].tsnow = res.timestamp;
+									if(config.bitrate[target].bsbefore === null || config.bitrate[target].tsbefore === null) {
 										// Skip this round
-										config.bitrate.bsbefore = config.bitrate.bsnow;
-										config.bitrate.tsbefore = config.bitrate.tsnow;
+										config.bitrate[target].bsbefore = config.bitrate[target].bsnow;
+										config.bitrate[target].tsbefore = config.bitrate[target].tsnow;
 									} else {
 										// Calculate bitrate
-										var timePassed = config.bitrate.tsnow - config.bitrate.tsbefore;
+										var timePassed = config.bitrate[target].tsnow - config.bitrate[target].tsbefore;
 										if(Janus.webRTCAdapter.browserDetails.browser == "safari")
 											timePassed = timePassed/1000;	// Apparently the timestamp is in microseconds, in Safari
-										var bitRate = Math.round((config.bitrate.bsnow - config.bitrate.bsbefore) * 8 / timePassed);
+										var bitRate = Math.round((config.bitrate[target].bsnow - config.bitrate[target].bsbefore) * 8 / timePassed);
 										if(Janus.webRTCAdapter.browserDetails.browser === 'safari')
 											bitRate = parseInt(bitRate/1000);
-										config.bitrate.value = bitRate + ' kbits/sec';
-										//~ Janus.log("Estimated bitrate is " + config.bitrate.value);
-										config.bitrate.bsbefore = config.bitrate.bsnow;
-										config.bitrate.tsbefore = config.bitrate.tsnow;
+										config.bitrate[target].value = bitRate + ' kbits/sec';
+										//~ Janus.log("Estimated bitrate is " + config.bitrate[target].value);
+										config.bitrate[target].bsbefore = config.bitrate[target].bsnow;
+										config.bitrate[target].tsbefore = config.bitrate[target].tsnow;
 									}
 								}
 							});
 						});
 				}, 1000);
-				return "0 kbits/sec";	// We don't have a bitrate value yet
 			}
-			return config.bitrate.value;
+			return config.bitrate[target].value;
 		} else {
 			Janus.warn("Getting the video bitrate unsupported by browser");
 			return "Feature unsupported by browser";
@@ -3142,21 +3267,11 @@ function Janus(gatewayCallbacks) {
 			}
 			// Cleanup stack
 			config.remoteStream = null;
-			if(config.volume) {
-				if(config.volume["local"] && config.volume["local"].timer)
-					clearInterval(config.volume["local"].timer);
-				if(config.volume["remote"] && config.volume["remote"].timer)
-					clearInterval(config.volume["remote"].timer);
-			}
-			config.volume = {};
-			if(config.bitrate.timer)
-				clearInterval(config.bitrate.timer);
-			config.bitrate.timer = null;
-			config.bitrate.bsnow = null;
-			config.bitrate.bsbefore = null;
-			config.bitrate.tsnow = null;
-			config.bitrate.tsbefore = null;
-			config.bitrate.value = null;
+			for(var i in config.bitrate) {
+				if(config.bitrate[i].timer)
+					clearInterval(config.bitrate[i].timer);
+			}
+			config.bitrate = {};
 			try {
 				// Try a MediaStreamTrack.stop() for each track
 				if(!config.streamExternal && config.myStream !== null && config.myStream !== undefined) {
diff --git a/html/mvideoroomtest.html b/html/mvideoroomtest.html
new file mode 100644
index 0000000000..0246c05a35
--- /dev/null
+++ b/html/mvideoroomtest.html
@@ -0,0 +1,159 @@
+<!DOCTYPE html>
+<html xmlns="http://www.w3.org/1999/xhtml">
+<head>
+<meta charset="utf-8">
+<meta name="viewport" content="width=device-width, initial-scale=1.0"/>
+<meta http-equiv="Content-Type" content="text/html; charset=UTF-8" />
+<title>Janus WebRTC Server: Video Room Demo (multistream)</title>
+<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/webrtc-adapter/6.0.3/adapter.min.js" ></script>
+<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/jquery/1.7.2/jquery.min.js" ></script>
+<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/jquery.blockUI/2.70/jquery.blockUI.min.js" ></script>
+<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/twitter-bootstrap/3.0.2/js/bootstrap.min.js"></script>
+<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/bootbox.js/4.1.0/bootbox.min.js"></script>
+<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/spin.js/2.3.2/spin.min.js"></script>
+<script type="text/javascript" src="https://cdnjs.cloudflare.com/ajax/libs/toastr.js/2.1.3/toastr.min.js"></script>
+<script type="text/javascript" src="janus.js" ></script>
+<script type="text/javascript" src="mvideoroomtest.js"></script>
+<script>
+	$(function() {
+		$(".navbar-static-top").load("navbar.html", function() {
+			$(".navbar-static-top li.dropdown").addClass("active");
+			$(".navbar-static-top a[href='mvideoroomtest.html']").parent().addClass("active");
+		});
+		$(".footer").load("footer.html");
+	});
+</script>
+<link rel="stylesheet" href="https://cdnjs.cloudflare.com/ajax/libs/bootswatch/3.3.7/cerulean/bootstrap.min.css" type="text/css"/>
+<link rel="stylesheet" href="css/demo.css" type="text/css"/>
+<link rel="stylesheet" href="https://cdnjs.cloudflare.com/ajax/libs/font-awesome/4.6.2/css/font-awesome.min.css"/>
+<link rel="stylesheet" href="https://cdnjs.cloudflare.com/ajax/libs/toastr.js/2.1.3/toastr.css"/>
+</head>
+<body>
+
+<a href="https://github.com/meetecho/janus-gateway"><img style="position: absolute; top: 0; left: 0; border: 0; z-index: 1001;" src="https://s3.amazonaws.com/github/ribbons/forkme_left_darkblue_121621.png" alt="Fork me on GitHub"></a>
+
+<nav class="navbar navbar-default navbar-static-top">
+</nav>
+
+<div class="container">
+	<div class="row">
+		<div class="col-md-12">
+			<div class="page-header">
+				<h1>Plugin Demo: Video Room (multistream)
+					<button class="btn btn-default" autocomplete="off" id="start">Start</button>
+				</h1>
+			</div>
+			<div class="container" id="details">
+				<div class="row">
+					<div class="col-md-12">
+						<h3>Demo details</h3>
+						<p>This is a variant of the <a href="videoroomtest.html">original VideoRoom demo</a>,
+						but taking advantage of the recently added multistream support to use a
+						single PeerConnection to receive all the contributions, rather than
+						creating a different one for each subscription. Apart from this, the
+						same considerations apply.</p>
+						<p>To use the demo, just insert a username to join the default room that
+						is configured. This will add you to the list of participants, and allow
+						you to automatically send your audio/video frames and receive the other
+						participants' feeds. The other participants will appear in separate
+						panels, whose title will be the names they chose when registering at
+						the demo.</p>
+						<p>Press the <code>Start</code> button above to launch the demo.</p>
+					</div>
+				</div>
+			</div>
+			<div class="container hide" id="videojoin">
+				<div class="row">
+					<span class="label label-info" id="you"></span>
+					<div class="col-md-12" id="controls">
+						<div class="input-group margin-bottom-md hide" id="registernow">
+							<span class="input-group-addon">@</span>
+							<input autocomplete="off" class="form-control" autocomplete="off" type="text" placeholder="Choose a display name" id="username" onkeypress="return checkEnter(this, event);"></input>
+							<span class="input-group-btn">
+								<button class="btn btn-success" autocomplete="off" id="register">Join the room</button>
+							</span>
+						</div>
+					</div>
+				</div>
+			</div>
+			<div class="container hide" id="videos">
+				<div class="row">
+					<div class="col-md-4">
+						<div class="panel panel-default">
+							<div class="panel-heading">
+								<h3 class="panel-title">Local Video <span class="label label-primary hide" id="publisher"></span>
+									<div class="btn-group btn-group-xs pull-right hide">
+										<div class="btn-group btn-group-xs">
+											<button id="bitrateset" autocomplete="off" class="btn btn-primary dropdown-toggle" data-toggle="dropdown">
+												Bandwidth<span class="caret"></span>
+											</button>
+											<ul id="bitrate" class="dropdown-menu" role="menu">
+												<li><a href="#" id="0">No limit</a></li>
+												<li><a href="#" id="128">Cap to 128kbit</a></li>
+												<li><a href="#" id="256">Cap to 256kbit</a></li>
+												<li><a href="#" id="512">Cap to 512kbit</a></li>
+												<li><a href="#" id="1024">Cap to 1mbit</a></li>
+												<li><a href="#" id="1500">Cap to 1.5mbit</a></li>
+												<li><a href="#" id="2000">Cap to 2mbit</a></li>
+											</ul>
+										</div>
+									</div>
+								</h3>
+							</div>
+							<div class="panel-body" id="videolocal"></div>
+						</div>
+					</div>
+					<div class="col-md-4">
+						<div class="panel panel-default">
+							<div class="panel-heading">
+								<h3 class="panel-title">Remote Video #1 <span class="label label-info hide" id="remote1"></span></h3>
+							</div>
+							<div class="panel-body relative" id="videoremote1"></div>
+						</div>
+					</div>
+					<div class="col-md-4">
+						<div class="panel panel-default">
+							<div class="panel-heading">
+								<h3 class="panel-title">Remote Video #2 <span class="label label-info hide" id="remote2"></span></h3>
+							</div>
+							<div class="panel-body relative" id="videoremote2"></div>
+						</div>
+					</div>
+				</div>
+				<div class="row">
+					<div class="col-md-4">
+						<div class="panel panel-default">
+							<div class="panel-heading">
+								<h3 class="panel-title">Remote Video #3 <span class="label label-info hide" id="remote3"></span></h3>
+							</div>
+							<div class="panel-body relative" id="videoremote3"></div>
+						</div>
+					</div>
+					<div class="col-md-4">
+						<div class="panel panel-default">
+							<div class="panel-heading">
+								<h3 class="panel-title">Remote Video #4 <span class="label label-info hide" id="remote4"></span></h3>
+							</div>
+							<div class="panel-body relative" id="videoremote4"></div>
+						</div>
+					</div>
+					<div class="col-md-4">
+						<div class="panel panel-default">
+							<div class="panel-heading">
+								<h3 class="panel-title">Remote Video #5 <span class="label label-info hide" id="remote5"></span></h3>
+							</div>
+							<div class="panel-body relative" id="videoremote5"></div>
+						</div>
+					</div>
+				</div>
+			</div>
+		</div>
+	</div>
+
+	<hr>
+	<div class="footer">
+	</div>
+</div>
+
+</body>
+</html>
diff --git a/html/mvideoroomtest.js b/html/mvideoroomtest.js
new file mode 100644
index 0000000000..1e50c0efc4
--- /dev/null
+++ b/html/mvideoroomtest.js
@@ -0,0 +1,993 @@
+// We make use of this 'server' variable to provide the address of the
+// REST Janus API. By default, in this example we assume that Janus is
+// co-located with the web server hosting the HTML pages but listening
+// on a different port (8088, the default for HTTP in Janus), which is
+// why we make use of the 'window.location.hostname' base address. Since
+// Janus can also do HTTPS, and considering we don't really want to make
+// use of HTTP for Janus if your demos are served on HTTPS, we also rely
+// on the 'window.location.protocol' prefix to build the variable, in
+// particular to also change the port used to contact Janus (8088 for
+// HTTP and 8089 for HTTPS, if enabled).
+// In case you place Janus behind an Apache frontend (as we did on the
+// online demos at http://janus.conf.meetecho.com) you can just use a
+// relative path for the variable, e.g.:
+//
+// 		var server = "/janus";
+//
+// which will take care of this on its own.
+//
+//
+// If you want to use the WebSockets frontend to Janus, instead, you'll
+// have to pass a different kind of address, e.g.:
+//
+// 		var server = "ws://" + window.location.hostname + ":8188";
+//
+// Of course this assumes that support for WebSockets has been built in
+// when compiling the server. WebSockets support has not been tested
+// as much as the REST API, so handle with care!
+//
+//
+// If you have multiple options available, and want to let the library
+// autodetect the best way to contact your server (or pool of servers),
+// you can also pass an array of servers, e.g., to provide alternative
+// means of access (e.g., try WebSockets first and, if that fails, fall
+// back to plain HTTP) or just have failover servers:
+//
+//		var server = [
+//			"ws://" + window.location.hostname + ":8188",
+//			"/janus"
+//		];
+//
+// This will tell the library to try connecting to each of the servers
+// in the presented order. The first working server will be used for
+// the whole session.
+//
+var server = null;
+if(window.location.protocol === 'http:')
+	server = "http://" + window.location.hostname + ":8088/janus";
+else
+	server = "https://" + window.location.hostname + ":8089/janus";
+
+var janus = null;
+var sfutest = null;
+var opaqueId = "videoroomtest-"+Janus.randomString(12);
+
+var myroom = 1234;	// Demo room
+var myusername = null;
+var myid = null;
+var mystream = null;
+// We use this other ID just to map our subscriptions to us
+var mypvtid = null;
+
+var remoteFeed = null;
+var feeds = {}, feedStreams = {}, subStreams = {}, slots = {}, mids = {};
+var localTracks = {}, localVideos = 0, remoteTracks = {};
+var bitrateTimer = [], simulcastStarted = {};
+
+var doSimulcast = (getQueryStringValue("simulcast") === "yes" || getQueryStringValue("simulcast") === "true");
+
+$(document).ready(function() {
+	// Initialize the library (all console debuggers enabled)
+	Janus.init({debug: "all", callback: function() {
+		// Use a button to start the demo
+		$('#start').one('click', function() {
+			$(this).attr('disabled', true).unbind('click');
+			// Make sure the browser supports WebRTC
+			if(!Janus.isWebrtcSupported()) {
+				bootbox.alert("No WebRTC support... ");
+				return;
+			}
+			// Create session
+			janus = new Janus(
+				{
+					server: server,
+					success: function() {
+						// Attach to video room test plugin
+						janus.attach(
+							{
+								plugin: "janus.plugin.videoroom",
+								opaqueId: opaqueId,
+								success: function(pluginHandle) {
+									$('#details').remove();
+									sfutest = pluginHandle;
+									Janus.log("Plugin attached! (" + sfutest.getPlugin() + ", id=" + sfutest.getId() + ")");
+									Janus.log("  -- This is a publisher/manager");
+									// Prepare the username registration
+									$('#videojoin').removeClass('hide').show();
+									$('#registernow').removeClass('hide').show();
+									$('#register').click(registerUsername);
+									$('#username').focus();
+									$('#start').removeAttr('disabled').html("Stop")
+										.click(function() {
+											$(this).attr('disabled', true);
+											janus.destroy();
+										});
+								},
+								error: function(error) {
+									Janus.error("  -- Error attaching plugin...", error);
+									bootbox.alert("Error attaching plugin... " + error);
+								},
+								consentDialog: function(on) {
+									Janus.debug("Consent dialog should be " + (on ? "on" : "off") + " now");
+									if(on) {
+										// Darken screen and show hint
+										$.blockUI({
+											message: '<div><img src="up_arrow.png"/></div>',
+											css: {
+												border: 'none',
+												padding: '15px',
+												backgroundColor: 'transparent',
+												color: '#aaa',
+												top: '10px',
+												left: (navigator.mozGetUserMedia ? '-100px' : '300px')
+											} });
+									} else {
+										// Restore screen
+										$.unblockUI();
+									}
+								},
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
+								},
+								webrtcState: function(on) {
+									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
+									$("#videolocal").parent().parent().unblock();
+									if(!on)
+										return;
+									$('#publish').remove();
+									// This controls allows us to override the global room bitrate cap
+									$('#bitrate').parent().parent().removeClass('hide').show();
+									$('#bitrate a').click(function() {
+										var id = $(this).attr("id");
+										var bitrate = parseInt(id)*1000;
+										if(bitrate === 0) {
+											Janus.log("Not limiting bandwidth via REMB");
+										} else {
+											Janus.log("Capping bandwidth to " + bitrate + " via REMB");
+										}
+										$('#bitrateset').html($(this).html() + '<span class="caret"></span>').parent().removeClass('open');
+										sfutest.send({"message": { "request": "configure", "bitrate": bitrate }});
+										return false;
+									});
+								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
+								onmessage: function(msg, jsep) {
+									Janus.debug(" ::: Got a message (publisher) :::");
+									Janus.debug(msg);
+									var event = msg["videoroom"];
+									Janus.debug("Event: " + event);
+									if(event != undefined && event != null) {
+										if(event === "joined") {
+											// Publisher/manager created, negotiate WebRTC and attach to existing feeds, if any
+											myid = msg["id"];
+											mypvtid = msg["private_id"];
+											Janus.log("Successfully joined room " + msg["room"] + " with ID " + myid);
+											publishOwnFeed(true);
+											// Any new feed to attach to?
+											if(msg["publishers"] !== undefined && msg["publishers"] !== null) {
+												var list = msg["publishers"];
+												Janus.debug("Got a list of available publishers/feeds:");
+												Janus.debug(list);
+												var sources = null;
+												for(var f in list) {
+													var id = list[f]["id"];
+													var display = list[f]["display"];
+													var streams = list[f]["streams"];
+													for(var i in streams) {
+														var stream = streams[i];
+														stream["id"] = id;
+														stream["display"] = display;
+													}
+													feedStreams[id] = {
+														id: id,
+														display: display,
+														streams: streams
+													}
+													Janus.debug("  >> [" + id + "] " + display + ":", streams);
+													if(sources === null)
+														sources = [];
+													sources.push(streams);
+												}
+												if(sources)
+													subscribeTo(sources);
+											}
+										} else if(event === "destroyed") {
+											// The room has been destroyed
+											Janus.warn("The room has been destroyed!");
+											bootbox.alert("The room has been destroyed", function() {
+												window.location.reload();
+											});
+										} else if(event === "event") {
+											// Any info on our streams or a new feed to attach to?
+											if(msg["streams"] !== undefined && msg["streams"] !== null) {
+												var streams = msg["streams"];
+												for(var i in streams) {
+													var stream = streams[i];
+													stream["id"] = myid;
+													stream["display"] = myusername;
+												}
+												feedStreams[myid] = {
+													id: myid,
+													display: myusername,
+													streams: streams
+												}
+											} else if(msg["publishers"] !== undefined && msg["publishers"] !== null) {
+												var list = msg["publishers"];
+												Janus.debug("Got a list of available publishers/feeds:");
+												Janus.debug(list);
+												var sources = null;
+												for(var f in list) {
+													var id = list[f]["id"];
+													var display = list[f]["display"];
+													var streams = list[f]["streams"];
+													for(var i in streams) {
+														var stream = streams[i];
+														stream["id"] = id;
+														stream["display"] = display;
+													}
+													feedStreams[id] = {
+														id: id,
+														display: display,
+														streams: streams
+													}
+													Janus.debug("  >> [" + id + "] " + display + ":", streams);
+													if(sources === null)
+														sources = [];
+													sources.push(streams);
+												}
+												if(sources)
+													subscribeTo(sources);
+											} else if(msg["leaving"] !== undefined && msg["leaving"] !== null) {
+												// One of the publishers has gone away?
+												var leaving = msg["leaving"];
+												Janus.log("Publisher left: " + leaving);
+												unsubscribeFrom(leaving);
+											} else if(msg["unpublished"] !== undefined && msg["unpublished"] !== null) {
+												// One of the publishers has unpublished?
+												var unpublished = msg["unpublished"];
+												Janus.log("Publisher left: " + unpublished);
+												if(unpublished === 'ok') {
+													// That's us
+													sfutest.hangup();
+													return;
+												}
+												unsubscribeFrom(unpublished);
+											} else if(msg["error"] !== undefined && msg["error"] !== null) {
+												if(msg["error_code"] === 426) {
+													// This is a "no such room" error: give a more meaningful description
+													bootbox.alert(
+														"<p>Apparently room <code>" + myroom + "</code> (the one this demo uses as a test room) " +
+														"does not exist...</p><p>Do you have an updated <code>janus.plugin.videoroom.cfg</code> " +
+														"configuration file? If not, make sure you copy the details of room <code>" + myroom + "</code> " +
+														"from that sample in your current configuration file, then restart Janus and try again."
+													);
+												} else {
+													bootbox.alert(msg["error"]);
+												}
+											}
+										}
+									}
+									if(jsep !== undefined && jsep !== null) {
+										Janus.debug("Handling SDP as well...");
+										Janus.debug(jsep);
+										sfutest.handleRemoteJsep({jsep: jsep});
+										// Check if any of the media we wanted to publish has
+										// been rejected (e.g., wrong or unsupported codec)
+										var audio = msg["audio_codec"];
+										if(mystream && mystream.getAudioTracks() && mystream.getAudioTracks().length > 0 && !audio) {
+											// Audio has been rejected
+											toastr.warning("Our audio stream has been rejected, viewers won't hear us");
+										}
+										var video = msg["video_codec"];
+										if(mystream && mystream.getVideoTracks() && mystream.getVideoTracks().length > 0 && !video) {
+											// Video has been rejected
+											toastr.warning("Our video stream has been rejected, viewers won't see us");
+											// Hide the webcam video
+											$('#myvideo').hide();
+											$('#videolocal').append(
+												'<div class="no-video-container">' +
+													'<i class="fa fa-video-camera fa-5 no-video-icon" style="height: 100%;"></i>' +
+													'<span class="no-video-text" style="font-size: 16px;">Video rejected, no webcam</span>' +
+												'</div>');
+										}
+									}
+								},
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videolocal .no-video-container').length === 0) {
+													$('#videolocal').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
+									$('#videos').removeClass('hide').show();
+									if($('#mute').length === 0) {
+										// Add a 'mute' button
+										$('#videolocal').append('<button class="btn btn-warning btn-xs" id="mute" style="position: absolute; bottom: 0px; left: 0px; margin: 15px;">Mute</button>');
+										$('#mute').click(toggleMute);
+										// Add an 'unpublish' button
+										$('#videolocal').append('<button class="btn btn-warning btn-xs" id="unpublish" style="position: absolute; bottom: 0px; right: 0px; margin: 15px;">Unpublish</button>');
+										$('#unpublish').click(unpublishOwnFeed);
+									}
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videolocal .no-video-container').length === 0) {
+												$('#videolocal').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videolocal .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										Janus.log(stream.getTracks());
+										Janus.log(stream.getVideoTracks());
+										$('#videolocal').append('<video class="rounded centered" id="myvideo' + trackId + '" width=100% autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
+									if(sfutest.webrtcStuff.pc.iceConnectionState !== "completed" &&
+											sfutest.webrtcStuff.pc.iceConnectionState !== "connected") {
+										$("#videolocal").parent().parent().block({
+											message: '<b>Publishing...</b>',
+											css: {
+												border: 'none',
+												backgroundColor: 'transparent',
+												color: 'white'
+											}
+										});
+									}
+								},
+								onremotetrack: function(track, mid, on) {
+									// The publisher stream is sendonly, we don't expect anything here
+								},
+								oncleanup: function() {
+									Janus.log(" ::: Got a cleanup notification: we are unpublished now :::");
+									mystream = null;
+									delete feedStreams[myid];
+									$('#videolocal').html('<button id="publish" class="btn btn-primary">Publish</button>');
+									$('#publish').click(function() { publishOwnFeed(true); });
+									$("#videolocal").parent().parent().unblock();
+									$('#bitrate').parent().parent().addClass('hide');
+									$('#bitrate a').unbind('click');
+									localTracks = {};
+									localVideos = 0;
+								}
+							});
+					},
+					error: function(error) {
+						Janus.error(error);
+						bootbox.alert(error, function() {
+							window.location.reload();
+						});
+					},
+					destroyed: function() {
+						window.location.reload();
+					}
+				});
+		});
+	}});
+});
+
+function checkEnter(field, event) {
+	var theCode = event.keyCode ? event.keyCode : event.which ? event.which : event.charCode;
+	if(theCode == 13) {
+		registerUsername();
+		return false;
+	} else {
+		return true;
+	}
+}
+
+function registerUsername() {
+	if($('#username').length === 0) {
+		// Create fields to register
+		$('#register').click(registerUsername);
+		$('#username').focus();
+	} else {
+		// Try a registration
+		$('#username').attr('disabled', true);
+		$('#register').attr('disabled', true).unbind('click');
+		var username = $('#username').val();
+		if(username === "") {
+			$('#you')
+				.removeClass().addClass('label label-warning')
+				.html("Insert your display name (e.g., pippo)");
+			$('#username').removeAttr('disabled');
+			$('#register').removeAttr('disabled').click(registerUsername);
+			return;
+		}
+		if(/[^a-zA-Z0-9]/.test(username)) {
+			$('#you')
+				.removeClass().addClass('label label-warning')
+				.html('Input is not alphanumeric');
+			$('#username').removeAttr('disabled').val("");
+			$('#register').removeAttr('disabled').click(registerUsername);
+			return;
+		}
+		var register = { "request": "join", "room": myroom, "ptype": "publisher", "display": username };
+		myusername = username;
+		sfutest.send({"message": register});
+	}
+}
+
+function publishOwnFeed(useAudio) {
+	// Publish our stream
+	$('#publish').attr('disabled', true).unbind('click');
+	sfutest.createOffer(
+		{
+			// Add data:true here if you want to publish datachannels as well
+			media: { audioRecv: false, videoRecv: false, audioSend: useAudio, videoSend: true },	// Publishers are sendonly
+			// If you want to test simulcasting (Chrome and Firefox only), then
+			// pass a ?simulcast=true when opening this demo page: it will turn
+			// the following 'simulcast' property to pass to janus.js to true
+			simulcast: doSimulcast,
+			success: function(jsep) {
+				Janus.debug("Got publisher SDP!");
+				Janus.debug(jsep);
+				var publish = { "request": "configure", "audio": useAudio, "video": true };
+				// You can force a specific codec to use when publishing by using the
+				// audiocodec and videocodec properties, for instance:
+				// 		publish["audiocodec"] = "opus"
+				// to force Opus as the audio codec to use, or:
+				// 		publish["videocodec"] = "vp9"
+				// to force VP9 as the videocodec to use. In both case, though, forcing
+				// a codec will only work if: (1) the codec is actually in the SDP (and
+				// so the browser supports it), and (2) the codec is in the list of
+				// allowed codecs in a room. With respect to the point (2) above,
+				// refer to the text in janus.plugin.videoroom.cfg for more details
+				sfutest.send({"message": publish, "jsep": jsep});
+			},
+			error: function(error) {
+				Janus.error("WebRTC error:", error);
+				if (useAudio) {
+					 publishOwnFeed(false);
+				} else {
+					bootbox.alert("WebRTC error... " + JSON.stringify(error));
+					$('#publish').removeAttr('disabled').click(function() { publishOwnFeed(true); });
+				}
+			}
+		});
+}
+
+function toggleMute() {
+	var muted = sfutest.isAudioMuted();
+	Janus.log((muted ? "Unmuting" : "Muting") + " local stream...");
+	if(muted)
+		sfutest.unmuteAudio();
+	else
+		sfutest.muteAudio();
+	muted = sfutest.isAudioMuted();
+	$('#mute').html(muted ? "Unmute" : "Mute");
+}
+
+function unpublishOwnFeed() {
+	// Unpublish our stream
+	$('#unpublish').attr('disabled', true).unbind('click');
+	var unpublish = { "request": "unpublish" };
+	sfutest.send({"message": unpublish});
+}
+
+var creatingFeed = false;
+function subscribeTo(sources) {
+	// New feeds are available, do we need create a new plugin handle first?
+	if(remoteFeed !== null) {
+		// Prepare the streams to subscribe to, as an array: we have the list of
+		// streams the feeds are publishing, so we can choose what to pick or skip
+		var subscription = [];
+		for(var s in sources) {
+			var streams = sources[s];
+			for(var i in streams) {
+				var stream = streams[i];
+				// If the publisher is VP8/VP9 and this is an older Safari, let's avoid video
+				if(stream.type === "video" && Janus.webRTCAdapter.browserDetails.browser === "safari" &&
+						(stream.codec === "vp9" || (stream.codec === "vp8" && !Janus.safariVp8))) {
+					toastr.warning("Publisher is using " + stream.codec.toUpperCase +
+						", but Safari doesn't support it: disabling video stream #" + stream.mindex);
+					continue;
+				}
+				// Find an empty slot in the UI for each new source
+				if(!feedStreams[stream.id].slot) {
+					var slot;
+					for(var i=1;i<6;i++) {
+						if(feeds[i] === undefined || feeds[i] === null) {
+							slot = i;
+							feeds[slot] = stream.id;
+							feedStreams[stream.id].slot = slot;
+							feedStreams[stream.id].remoteVideos = 0;
+							$('#remote' + slot).removeClass('hide').html(stream.display).show();
+							break;
+						}
+					}
+				}
+				subscription.push({
+					feed: stream.id,	// This is mandatory
+					mid: stream.mid		// This is optional (all streams, if missing)
+				});
+			}
+		}
+		remoteFeed.send({ message: {
+			request: "subscribe",
+			streams: subscription
+		}});
+		return;
+	}
+	// We don't have a handle yet, but we may be creating one already
+	if(creatingFeed) {
+		// Still working on the handle
+		setTimeout(function() {
+			subscribeTo(sources);
+		}, 500);
+		return;
+	}
+	creatingFeed = true;
+	janus.attach(
+		{
+			plugin: "janus.plugin.videoroom",
+			opaqueId: opaqueId,
+			success: function(pluginHandle) {
+				remoteFeed = pluginHandle;
+				remoteTracks = {};
+				Janus.log("Plugin attached! (" + remoteFeed.getPlugin() + ", id=" + remoteFeed.getId() + ")");
+				Janus.log("  -- This is a multistream subscriber");
+				// Prepare the streams to subscribe to, as an array: we have the list of
+				// streams the feed is publishing, so we can choose what to pick or skip
+				var subscription = [];
+				for(var s in sources) {
+					var streams = sources[s];
+					for(var i in streams) {
+						var stream = streams[i];
+						// If the publisher is VP8/VP9 and this is an older Safari, let's avoid video
+						if(stream.type === "video" && Janus.webRTCAdapter.browserDetails.browser === "safari" &&
+								(stream.codec === "vp9" || (stream.codec === "vp8" && !Janus.safariVp8))) {
+							toastr.warning("Publisher is using " + stream.codec.toUpperCase +
+								", but Safari doesn't support it: disabling video stream #" + stream.mindex);
+							continue;
+						}
+						// Find an empty slot in the UI for each new source
+						if(!feedStreams[stream.id].slot) {
+							var slot;
+							for(var i=1;i<6;i++) {
+								if(feeds[i] === undefined || feeds[i] === null) {
+									slot = i;
+									feeds[slot] = stream.id;
+									feedStreams[stream.id].slot = slot;
+									feedStreams[stream.id].remoteVideos = 0;
+									$('#remote' + slot).removeClass('hide').html(stream.display).show();
+									break;
+								}
+							}
+						}
+						subscription.push({
+							feed: stream.id,	// This is mandatory
+							mid: stream.mid		// This is optional (all streams, if missing)
+						});
+					}
+				}
+				// We wait for the plugin to send us an offer
+				var subscribe = {
+					request: "join",
+					room: myroom,
+					ptype: "subscriber",
+					streams: subscription,
+					private_id: mypvtid
+				};
+				remoteFeed.send({ message: subscribe });
+			},
+			error: function(error) {
+				Janus.error("  -- Error attaching plugin...", error);
+				bootbox.alert("Error attaching plugin... " + error);
+			},
+			iceState: function(state) {
+				Janus.log("ICE state (remote feed) changed to " + state);
+			},
+			webrtcState: function(on) {
+				Janus.log("Janus says this WebRTC PeerConnection (remote feed) is " + (on ? "up" : "down") + " now");
+			},
+			slowLink: function(uplink, lost, mid) {
+				Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+					" packets on mid " + mid + " (" + lost + " lost packets)");
+			},
+			onmessage: function(msg, jsep) {
+				Janus.debug(" ::: Got a message (subscriber) :::");
+				Janus.debug(msg);
+				var event = msg["videoroom"];
+				Janus.debug("Event: " + event);
+				if(msg["error"] !== undefined && msg["error"] !== null) {
+					bootbox.alert(msg["error"]);
+				} else if(event != undefined && event != null) {
+					if(event === "attached") {
+						creatingFeed = false;
+						Janus.log("Successfully attached to feed in room " + msg["room"]);
+					} else if(event === "event") {
+						// Check if we got an event on a simulcast-related event from this publisher
+						var mid = msg["mid"];
+						var substream = msg["substream"];
+						var temporal = msg["temporal"];
+						if((substream !== null && substream !== undefined) || (temporal !== null && temporal !== undefined)) {
+							// Check which this feed this refers to
+							var sub = subStreams[mid];
+							var feed = feedStreams[sub.feed_id];
+							var slot = slots[mid];
+							if(!simulcastStarted[slot]) {
+								simulcastStarted[slot] = true;
+								// Add some new buttons
+								addSimulcastButtons(slot, true);
+							}
+							// We just received notice that there's been a switch, update the buttons
+							updateSimulcastButtons(slot, substream, temporal);
+						}
+					} else {
+						// What has just happened?
+					}
+				}
+				if(msg["streams"]) {
+					// Update map of subscriptions by mid
+					for(var i in msg["streams"]) {
+						var mid = msg["streams"][i]["mid"];
+						subStreams[mid] = msg["streams"][i];
+						var feed = feedStreams[msg["streams"][i]["feed_id"]];
+						if(feed && feed.slot) {
+							slots[mid] = feed.slot;
+							mids[feed.slot] = mid;
+						}
+					}
+				}
+				if(jsep !== undefined && jsep !== null) {
+					Janus.debug("Handling SDP as well...");
+					Janus.debug(jsep);
+					// Answer and attach
+					remoteFeed.createAnswer(
+						{
+							jsep: jsep,
+							// Add data:true here if you want to subscribe to datachannels as well
+							// (obviously only works if the publisher offered them in the first place)
+							media: { audioSend: false, videoSend: false },	// We want recvonly audio/video
+							success: function(jsep) {
+								Janus.debug("Got SDP!");
+								Janus.debug(jsep);
+								var body = { request: "start", room: myroom };
+								remoteFeed.send({ message: body, jsep: jsep });
+							},
+							error: function(error) {
+								Janus.error("WebRTC error:", error);
+								bootbox.alert("WebRTC error... " + JSON.stringify(error));
+							}
+						});
+				}
+			},
+			onlocaltrack: function(track, on) {
+				// The subscriber stream is recvonly, we don't expect anything here
+			},
+			onremotetrack: function(track, mid, on) {
+				Janus.debug(" ::: Got a remote track event ::: (remote feed)");
+				Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+				// Which publisher are we getting on this mid?
+				var sub = subStreams[mid];
+				var feed = feedStreams[sub.feed_id];
+				Janus.debug(" >> This track is coming from feed " + sub.feed_id + ":", feed);
+				var slot = slots[mid];
+				if(feed && !slot) {
+					slot = feed.slot;
+					slots[mid] = feed.slot;
+					mids[feed.slot] = mid;
+				}
+				Janus.debug(" >> mid " + mid + " is in slot " + slot);
+				if(!on) {
+					// Track removed, get rid of the stream and the rendering
+					var stream = remoteTracks[mid];
+					if(stream) {
+						try {
+							var tracks = stream.getTracks();
+							for(var i in tracks) {
+								var mst = tracks[i];
+								if(mst !== null && mst !== undefined)
+									mst.stop();
+							}
+						} catch(e) {}
+					}
+					$('#remotevideo' + slot + '-' + mid).remove();
+					if(track.kind === "video" && feed) {
+						feed.remoteVideos--;
+						if(feed.remoteVideos === 0) {
+							// No video, at least for now: show a placeholder
+							if($('#videoremote' + slot + ' .no-video-container').length === 0) {
+								$('#videoremote' + slot).append(
+									'<div class="no-video-container">' +
+										'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+										'<span class="no-video-text">No remote video available</span>' +
+									'</div>');
+							}
+						}
+					}
+					delete remoteTracks[mid];
+					delete slots[mid];
+					delete mids[slot];
+					return;
+				}
+				// If we're here, a new track was added
+				if(feed.spinner !== undefined && feed.spinner !== null) {
+					feed.spinner.stop();
+					feed.spinner = null;
+				}
+				if($('#remotevideo' + slot + '-' + mid).length > 0)
+					return;
+				if(track.kind === "audio") {
+					// New audio track: create a stream out of it, and use a hidden <audio> element
+					stream = new MediaStream();
+					stream.addTrack(track.clone());
+					remoteTracks[mid] = stream;
+					Janus.log("Created remote audio stream:", stream);
+					$('#videoremote' + slot).append('<audio class="hide" id="remotevideo' + slot + '-' + mid + '" autoplay playsinline/>');
+					Janus.attachMediaStream($('#remotevideo' + slot + '-' + mid).get(0), stream);
+					if(feed.remoteVideos === 0) {
+						// No video, at least for now: show a placeholder
+						if($('#videoremote' + slot + ' .no-video-container').length === 0) {
+							$('#videoremote' + slot).append(
+								'<div class="no-video-container">' +
+									'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+									'<span class="no-video-text">No remote video available</span>' +
+								'</div>');
+						}
+					}
+				} else {
+					// New video track: create a stream out of it
+					feed.remoteVideos++;
+					$('#videoremote' + slot + ' .no-video-container').remove();
+					stream = new MediaStream();
+					stream.addTrack(track.clone());
+					remoteTracks[mid] = stream;
+					Janus.log("Created remote video stream:", stream);
+					$('#videoremote' + slot).append('<video class="rounded centered" id="remotevideo' + slot + '-' + mid + '" width=100% autoplay playsinline/>');
+					$('#videoremote' + slot).append(
+						'<span class="label label-primary hide" id="curres'+slot+'" style="position: absolute; bottom: 0px; left: 0px; margin: 15px;"></span>' +
+						'<span class="label label-info hide" id="curbitrate'+slot+'" style="position: absolute; bottom: 0px; right: 0px; margin: 15px;"></span>');
+					Janus.attachMediaStream($('#remotevideo' + slot + '-' + mid).get(0), stream);
+					// Note: we'll need this for additional videos too
+					if(!bitrateTimer[slot]) {
+						$('#curbitrate' + slot).removeClass('hide').show();
+						bitrateTimer[slot] = setInterval(function() {
+							if(!$("#videoremote" + slot + ' video').get(0))
+								return;
+							// Display updated bitrate, if supported
+							var bitrate = remoteFeed.getBitrate(mid);
+							$('#curbitrate' + slot).text(bitrate);
+							// Check if the resolution changed too
+							var width = $("#videoremote" + slot + ' video').get(0).videoWidth;
+							var height = $("#videoremote" + slot + ' video').get(0).videoHeight;
+							if(width > 0 && height > 0)
+								$('#curres' + slot).removeClass('hide').text(width+'x'+height).show();
+						}, 1000);
+					}
+				}
+			},
+			oncleanup: function() {
+				Janus.log(" ::: Got a cleanup notification (remote feed) :::");
+				for(var i=1;i<6;i++) {
+					$('#remotevideo'+i).remove();
+					$('#waitingvideo'+i).remove();
+					$('#novideo'+i).remove();
+					$('#curbitrate'+i).remove();
+					$('#curres'+i).remove();
+					if(bitrateTimer[i] !== null && bitrateTimer[i] !== null)
+						clearInterval(bitrateTimer[i]);
+					bitrateTimer[i] = null;
+					feedStreams[i].simulcastStarted = false;
+					feedStreams[i].remoteVideos = 0;
+					$('#simulcast'+i).remove();
+				}
+				remoteTracks = {};
+			}
+		});
+}
+
+function unsubscribeFrom(id) {
+	// Unsubscribe from this publisher
+	var feed = feedStreams[id];
+	if(!feed)
+		return;
+	Janus.debug("Feed " + id + " (" + feed.display + ") has left the room, detaching");
+	if(bitrateTimer[feed.slot] !== null && bitrateTimer[feed.slot] !== null)
+		clearInterval(bitrateTimer[feed.slot]);
+	bitrateTimer[feed.slot] = null;
+	$('#remote' + feed.slot).empty().hide();
+	$('#videoremote' + feed.slot).empty();
+	delete simulcastStarted[feed.slot];
+	$('#simulcast' + feed.slot).remove();
+	delete feeds[feed.slot];
+	feeds.slot = 0;
+	delete feedStreams[id];
+	// Send an unsubscribe request
+	var unsubscribe = {
+		request: "unsubscribe",
+		streams: [{ feed: id }]
+	};
+	if(remoteFeed != null)
+		remoteFeed.send({ message: unsubscribe });
+}
+
+// Helper to parse query string
+function getQueryStringValue(name) {
+	name = name.replace(/[\[]/, "\\[").replace(/[\]]/, "\\]");
+	var regex = new RegExp("[\\?&]" + name + "=([^&#]*)"),
+		results = regex.exec(location.search);
+	return results === null ? "" : decodeURIComponent(results[1].replace(/\+/g, " "));
+}
+
+// Helpers to create Simulcast-related UI, if enabled
+function addSimulcastButtons(feed, temporal) {
+	var index = feed;
+	$('#remote'+index).parent().append(
+		'<div id="simulcast'+index+'" class="btn-group-vertical btn-group-vertical-xs pull-right">' +
+		'	<div class"row">' +
+		'		<div class="btn-group btn-group-xs" style="width: 100%">' +
+		'			<button id="sl'+index+'-2" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to higher quality" style="width: 33%">SL 2</button>' +
+		'			<button id="sl'+index+'-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to normal quality" style="width: 33%">SL 1</button>' +
+		'			<button id="sl'+index+'-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to lower quality" style="width: 34%">SL 0</button>' +
+		'		</div>' +
+		'	</div>' +
+		'	<div class"row">' +
+		'		<div class="btn-group btn-group-xs hide" style="width: 100%">' +
+		'			<button id="tl'+index+'-2" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 2" style="width: 34%">TL 2</button>' +
+		'			<button id="tl'+index+'-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 1" style="width: 33%">TL 1</button>' +
+		'			<button id="tl'+index+'-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 0" style="width: 33%">TL 0</button>' +
+		'		</div>' +
+		'	</div>' +
+		'</div>'
+	);
+	// Enable the simulcast selection buttons
+	$('#sl' + index + '-0').removeClass('btn-primary btn-success').addClass('btn-primary')
+		.unbind('click').click(function() {
+			var index = $(this).attr('id').split('sl')[1].split('-')[0];
+			toastr.info("Switching simulcast substream (mid=" + mids[index] + "), wait for it... (lower quality)", null, {timeOut: 2000});
+			if(!$('#sl' + index + '-2').hasClass('btn-success'))
+				$('#sl' + index + '-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#sl' + index + '-1').hasClass('btn-success'))
+				$('#sl' + index + '-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#sl' + index + '-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			remoteFeed.send({message: { request: "configure", mid: mids[index], substream: 0 }});
+		});
+	$('#sl' + index + '-1').removeClass('btn-primary btn-success').addClass('btn-primary')
+		.unbind('click').click(function() {
+			var index = $(this).attr('id').split('sl')[1].split('-')[0];
+			toastr.info("Switching simulcast substream (mid=" + mids[index] + "), wait for it... (normal quality)", null, {timeOut: 2000});
+			if(!$('#sl' + index + '-2').hasClass('btn-success'))
+				$('#sl' + index + '-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#sl' + index + '-1').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			if(!$('#sl' + index + '-0').hasClass('btn-success'))
+				$('#sl' + index + '-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			remoteFeed.send({message: { request: "configure", mid: mids[index], substream: 1 }});
+		});
+	$('#sl' + index + '-2').removeClass('btn-primary btn-success').addClass('btn-primary')
+		.unbind('click').click(function() {
+			var index = $(this).attr('id').split('sl')[1].split('-')[0];
+			toastr.info("Switching simulcast substream (mid=" + mids[index] + "), wait for it... (higher quality)", null, {timeOut: 2000});
+			$('#sl' + index + '-2').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			if(!$('#sl' + index + '-1').hasClass('btn-success'))
+				$('#sl' + index + '-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#sl' + index + '-0').hasClass('btn-success'))
+				$('#sl' + index + '-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			remoteFeed.send({message: { request: "configure", mid: mids[index], substream: 2 }});
+		});
+	if(!temporal)	// No temporal layer support
+		return;
+	$('#tl' + index + '-0').parent().removeClass('hide');
+	$('#tl' + index + '-0').removeClass('btn-primary btn-success').addClass('btn-primary')
+		.unbind('click').click(function() {
+			var index = $(this).attr('id').split('tl')[1].split('-')[0];
+			toastr.info("Capping simulcast temporal layer (mid=" + mids[index] + "), wait for it... (lowest FPS)", null, {timeOut: 2000});
+			if(!$('#tl' + index + '-2').hasClass('btn-success'))
+				$('#tl' + index + '-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#tl' + index + '-1').hasClass('btn-success'))
+				$('#tl' + index + '-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#tl' + index + '-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			remoteFeed.send({message: { request: "configure", mid: mids[index], temporal: 0 }});
+		});
+	$('#tl' + index + '-1').removeClass('btn-primary btn-success').addClass('btn-primary')
+		.unbind('click').click(function() {
+			var index = $(this).attr('id').split('tl')[1].split('-')[0];
+			toastr.info("Capping simulcast temporal layer (mid=" + mids[index] + "), wait for it... (medium FPS)", null, {timeOut: 2000});
+			if(!$('#tl' + index + '-2').hasClass('btn-success'))
+				$('#tl' + index + '-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#tl' + index + '-1').removeClass('btn-primary btn-info').addClass('btn-info');
+			if(!$('#tl' + index + '-0').hasClass('btn-success'))
+				$('#tl' + index + '-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			remoteFeed.send({message: { request: "configure", mid: mids[index], temporal: 1 }});
+		});
+	$('#tl' + index + '-2').removeClass('btn-primary btn-success').addClass('btn-primary')
+		.unbind('click').click(function() {
+			var index = $(this).attr('id').split('tl')[1].split('-')[0];
+			toastr.info("Capping simulcast temporal layer (mid=" + mids[index] + "), wait for it... (highest FPS)", null, {timeOut: 2000});
+			$('#tl' + index + '-2').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			if(!$('#tl' + index + '-1').hasClass('btn-success'))
+				$('#tl' + index + '-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#tl' + index + '-0').hasClass('btn-success'))
+				$('#tl' + index + '-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			remoteFeed.send({message: { request: "configure", mid: mids[index], temporal: 2 }});
+		});
+}
+
+function updateSimulcastButtons(feed, substream, temporal) {
+	// Check the substream
+	var index = feed;
+	if(substream === 0) {
+		toastr.success("Switched simulcast substream! (lower quality)", null, {timeOut: 2000});
+		$('#sl' + index + '-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#sl' + index + '-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#sl' + index + '-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+	} else if(substream === 1) {
+		toastr.success("Switched simulcast substream! (normal quality)", null, {timeOut: 2000});
+		$('#sl' + index + '-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#sl' + index + '-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#sl' + index + '-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+	} else if(substream === 2) {
+		toastr.success("Switched simulcast substream! (higher quality)", null, {timeOut: 2000});
+		$('#sl' + index + '-2').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#sl' + index + '-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#sl' + index + '-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+	}
+	// Check the temporal layer
+	if(temporal === 0) {
+		toastr.success("Capped simulcast temporal layer! (lowest FPS)", null, {timeOut: 2000});
+		$('#tl' + index + '-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#tl' + index + '-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#tl' + index + '-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+	} else if(temporal === 1) {
+		toastr.success("Capped simulcast temporal layer! (medium FPS)", null, {timeOut: 2000});
+		$('#tl' + index + '-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#tl' + index + '-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#tl' + index + '-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+	} else if(temporal === 2) {
+		toastr.success("Capped simulcast temporal layer! (highest FPS)", null, {timeOut: 2000});
+		$('#tl' + index + '-2').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#tl' + index + '-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#tl' + index + '-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+	}
+}
diff --git a/html/navbar.html b/html/navbar.html
index e32b1faa54..94a00f9839 100644
--- a/html/navbar.html
+++ b/html/navbar.html
@@ -19,6 +19,7 @@
 					<li><a href="videocalltest.html">Video Call</a></li>
 					<li><a href="siptest.html">SIP Gateway</a></li>
 					<li><a href="videoroomtest.html">Video Room</a></li>
+					<li><a href="mvideoroomtest.html">Video Room (multistream)</a></li>
 					<li><a href="audiobridgetest.html">Audio Bridge</a></li>
 					<li><a href="textroomtest.html">Text Room</a></li>
 					<li><a href="voicemailtest.html">Voice Mail</a></li>
diff --git a/html/nosiptest.js b/html/nosiptest.js
index 279e8aedd2..42e7b4f941 100644
--- a/html/nosiptest.js
+++ b/html/nosiptest.js
@@ -53,6 +53,9 @@ var janus = null;
 // We'll need two handles for this demo: a caller and a callee
 var caller = null, callee = null;
 var opaqueId = Janus.randomString(12);
+// The local and remote tracks only refer to the caller, though (we ignore the callee)
+var localTracks = {}, localVideos = 0,
+	remoteTracks = {}, remoteVideos = 0;
 
 var spinner = null;
 
@@ -139,16 +142,16 @@ $(document).ready(function() {
 								iceState: function(state) {
 									Janus.log("[caller] ICE state changed to " + state);
 								},
-								mediaState: function(medium, on) {
-									Janus.log("[caller] Janus " + (on ? "started" : "stopped") + " receiving our " + medium);
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
 								},
 								webrtcState: function(on) {
 									Janus.log("[caller] Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
 									$("#videoleft").parent().unblock();
 								},
-								slowLink: function(uplink, lost) {
+								slowLink: function(uplink, lost, mid) {
 									Janus.warn("[caller] Janus reports problems " + (uplink ? "sending" : "receiving") +
-										" packets on this PeerConnection (" + lost + " lost packets)");
+										" packets on mid " + mid + " (" + lost + " lost packets)");
 								},
 								onmessage: function(msg, jsep) {
 									Janus.debug("[caller]  ::: Got a message :::");
@@ -189,14 +192,73 @@ $(document).ready(function() {
 										}
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug("[caller]  ::: Got a local stream :::");
-									Janus.debug(stream);
-									$('#videos').removeClass('hide').show();
-									if($('#myvideo').length === 0)
-										$('#videoleft').append('<video class="rounded centered" id="myvideo" width=320 height=240 autoplay playsinline muted="muted"/>');
-									Janus.attachMediaStream($('#myvideo').get(0), stream);
-									$("#myvideo").get(0).muted = "muted";
+								onlocaltrack: function(track, on) {
+									Janus.debug("[caller]  ::: Got a local track event :::");
+									Janus.debug("[caller] Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoleft .no-video-container').length === 0) {
+													$('#videoleft').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
+									if($('#videoleft video').length === 0) {
+										$('#videos').removeClass('hide').show();
+									}
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoleft .no-video-container').length === 0) {
+												$('#videoleft').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videoleft .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("[caller] Created local stream:", stream);
+										$('#videoleft').append('<video class="rounded centered" id="myvideo' + trackId + '" width=320 height=240 autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
 									if(caller.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											caller.webrtcStuff.pc.iceConnectionState !== "connected") {
 										$("#videoleft").parent().block({
@@ -207,39 +269,46 @@ $(document).ready(function() {
 												color: 'white'
 											}
 										});
-										// No remote video yet
-										$('#videoright').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
-										if(spinner == null) {
-											var target = document.getElementById('videoright');
-											spinner = new Spinner({top:100}).spin(target);
-										} else {
-											spinner.spin();
-										}
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No webcam
-										$('#myvideo').hide();
-										if($('#videoleft .no-video-container').length === 0) {
-											$('#videoleft').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
+								},
+								onremotetrack: function(track, mid, on) {
+									Janus.debug("[caller]  ::: Got a remote track event :::");
+									Janus.debug("[caller] Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = remoteTracks[mid];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
 										}
-									} else {
-										$('#videoleft .no-video-container').remove();
-										$('#myvideo').removeClass('hide').show();
+										$('#peervideo' + mid).remove();
+										if(track.kind === "video") {
+											remoteVideos--;
+											if(remoteVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoright .no-video-container').length === 0) {
+													$('#videoright').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No remote video available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete remoteTracks[mid];
+										return;
 									}
-								},
-								onremotestream: function(stream) {
-									Janus.debug("[caller]  ::: Got a remote stream :::");
-									Janus.debug(stream);
-									if($('#peervideo').length === 0) {
+									// If we're here, a new track was added
+									if($('#videoright audio').length === 0 && $('#videoright video').length === 0) {
+										$('#videos').removeClass('hide').show();
 										$('#videoright').parent().find('h3').html(
 											'Send DTMF: <span id="dtmf" class="btn-group btn-group-xs"></span>');
-										$('#videoright').append(
-											'<video class="rounded centered hide" id="peervideo" width=320 height=240 autoplay playsinline/>');
 										for(var i=0; i<12; i++) {
 											if(i<10)
 												$('#dtmf').append('<button class="btn btn-info dtmf">' + i + '</button>');
@@ -251,34 +320,36 @@ $(document).ready(function() {
 										$('.dtmf').click(function() {
 											// Send DTMF tone (inband)
 											caller.dtmf({dtmf: { tones: $(this).text()}});
-											// Notice you can also send DTMF tones using SIP INFO
-											// 		caller.send({message: {request: "dtmf_info", digit: $(this).text()}});
-										});
-										// Show the peer and hide the spinner when we get a playing event
-										$("#peervideo").bind("playing", function () {
-											$('#waitingvideo').remove();
-											if(this.videoWidth)
-												$('#peervideo').removeClass('hide').show();
-											if(spinner !== null && spinner !== undefined)
-												spinner.stop();
-											spinner = null;
 										});
 									}
-									Janus.attachMediaStream($('#peervideo').get(0), stream);
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#peervideo').hide();
-										if($('#videoright .no-video-container').length === 0) {
-											$('#videoright').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
+									if(track.kind === "audio") {
+										// New audio track: create a stream out of it, and use a hidden <audio> element
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("[caller] Created remote audio stream:", stream);
+										$('#videoright').append('<audio class="hide" id="peervideo' + mid + '" autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										if(remoteVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoright .no-video-container').length === 0) {
+												$('#videoright').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No remote video available</span>' +
+													'</div>');
+											}
 										}
 									} else {
+										// New video track: create a stream out of it
+										remoteVideos++;
 										$('#videoright .no-video-container').remove();
-										$('#peervideo').removeClass('hide').show();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("[caller] Created remote video stream:", stream);
+										$('#videoright').append('<video class="rounded centered" id="peervideo' + mid + '" width=320 height=240 autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
 									}
 								},
 								oncleanup: function() {
@@ -286,10 +357,13 @@ $(document).ready(function() {
 									if(spinner !== null && spinner !== undefined)
 										spinner.stop();
 									spinner = null;
-									$('#myvideo').remove();
-									$('#waitingvideo').remove();
-									$("#videoleft").parent().unblock();
-									$('#peervideo').remove();
+									$("#videoleft").empty().parent().unblock();
+									$('#videoright').empty();
+									$('#dtmf').parent().html("Remote UA");
+									localTracks = {};
+									localVideos = 0;
+									remoteTracks = {};
+									remoteVideos = 0;
 								}
 							});
 						// Attach to nosip plugin as a callee
@@ -334,9 +408,9 @@ $(document).ready(function() {
 									Janus.log("[callee] Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
 									$("#videoleft").parent().unblock();
 								},
-								slowLink: function(uplink, lost) {
+								slowLink: function(uplink, lost, mid) {
 									Janus.warn("[callee] Janus reports problems " + (uplink ? "sending" : "receiving") +
-										" packets on this PeerConnection (" + lost + " lost packets)");
+										" packets on mid " + mid + " (" + lost + " lost packets)");
 								},
 								onmessage: function(msg, jsep) {
 									Janus.debug("[callee]  ::: Got a message :::");
@@ -400,10 +474,10 @@ $(document).ready(function() {
 										}
 									}
 								},
-								onlocalstream: function(stream) {
+								onlocaltrack: function(track, on) {
 									// The callee is our fake peer, we don't display anything
 								},
-								onremotestream: function(stream) {
+								onremotetrack: function(track, mid, on) {
 									// The callee is our fake peer, we don't display anything
 								},
 								oncleanup: function() {
diff --git a/html/recordplaytest.js b/html/recordplaytest.js
index 8ac66c3d95..981637abe7 100644
--- a/html/recordplaytest.js
+++ b/html/recordplaytest.js
@@ -52,6 +52,8 @@ var janus = null;
 var recordplay = null;
 var opaqueId = "recordplaytest-"+Janus.randomString(12);
 
+var localTracks = {}, localVideos = 0,
+	remoteTracks = {}, remoteVideos = 0;
 var spinner = null;
 var bandwidth = 1024 * 1024;
 
@@ -123,10 +125,20 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
+								},
 								webrtcState: function(on) {
 									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
 									$("#videobox").parent().unblock();
 								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message :::");
 									Janus.debug(msg);
@@ -226,18 +238,78 @@ $(document).ready(function() {
 										updateRecsList();
 									}
 								},
-								onlocalstream: function(stream) {
+								onlocaltrack: function(track, on) {
 									if(playing === true)
 										return;
-									Janus.debug(" ::: Got a local stream :::");
-									Janus.debug(stream);
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#thevideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videobox .no-video-container').length === 0) {
+													$('#videobox').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
 									$('#videotitle').html("Recording...");
 									$('#stop').unbind('click').click(stop);
 									$('#video').removeClass('hide').show();
-									if($('#thevideo').length === 0)
-										$('#videobox').append('<video class="rounded centered" id="thevideo" width=320 height=240 autoplay playsinline muted="muted"/>');
-									Janus.attachMediaStream($('#thevideo').get(0), stream);
-									$("#thevideo").get(0).muted = "muted";
+									if($('#videobox video').length === 0) {
+										$('#videos').removeClass('hide').show();
+									}
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videobox .no-video-container').length === 0) {
+												$('#videobox').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videobox .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										$('#videobox').append('<video class="rounded centered" id="thevideo' + trackId + '" width=320 height=240 autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#thevideo' + trackId).get(0), stream);
+									}
 									if(recordplay.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											recordplay.webrtcStuff.pc.iceConnectionState !== "connected") {
 										$("#videobox").parent().block({
@@ -249,70 +321,81 @@ $(document).ready(function() {
 											}
 										});
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#thevideo').hide();
-										if($('#videobox .no-video-container').length === 0) {
-											$('#videobox').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
-										}
-									} else {
-										$('#videobox .no-video-container').remove();
-										$('#thevideo').removeClass('hide').show();
-									}
 								},
-								onremotestream: function(stream) {
+								onremotetrack: function(track, mid, on) {
 									if(playing === false)
 										return;
-									Janus.debug(" ::: Got a remote stream :::");
-									Janus.debug(stream);
-									if($('#thevideo').length === 0) {
+									Janus.debug(" ::: Got a remote track event :::");
+									Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = remoteTracks[mid];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										$('#thevideo' + mid).remove();
+										if(track.kind === "video") {
+											remoteVideos--;
+											if(remoteVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videobox .no-video-container').length === 0) {
+													$('#videobox').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No remote video available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete remoteTracks[mid];
+										return;
+									}
+									// If we're here, a new track was added
+									if($('#videobox audio').length === 0 && $('#videobox video').length === 0) {
 										$('#videotitle').html(selectedRecordingInfo);
 										$('#stop').unbind('click').click(stop);
 										$('#video').removeClass('hide').show();
-										$('#videobox').append('<video class="rounded centered hide" id="thevideo" width=320 height=240 autoplay playsinline/>');
-										// No remote video yet
-										$('#videobox').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
-										if(spinner == null) {
-											var target = document.getElementById('videobox');
-											spinner = new Spinner({top:100}).spin(target);
-										} else {
-											spinner.spin();
-										}
-										// Show the video, hide the spinner and show the resolution when we get a playing event
-										$("#thevideo").bind("playing", function () {
-											$('#waitingvideo').remove();
-											$('#thevideo').removeClass('hide');
-											if(spinner !== null && spinner !== undefined)
-												spinner.stop();
-											spinner = null;
-										});
 									}
-									Janus.attachMediaStream($('#thevideo').get(0), stream);
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#thevideo').hide();
-										if($('#videobox .no-video-container').length === 0) {
-											$('#videobox').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
+									if(track.kind === "audio") {
+										// New audio track: create a stream out of it, and use a hidden <audio> element
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote audio stream:", stream);
+										$('#videobox').append('<audio class="hide" id="thevideo' + mid + '" autoplay playsinline/>');
+										Janus.attachMediaStream($('#thevideo' + mid).get(0), stream);
+										if(remoteVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videobox .no-video-container').length === 0) {
+												$('#videobox').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No remote video available</span>' +
+													'</div>');
+											}
 										}
 									} else {
+										// New video track: create a stream out of it
+										remoteVideos++;
 										$('#videobox .no-video-container').remove();
-										$('#thevideo').removeClass('hide').show();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote video stream:", stream);
+										$('#videobox').append('<video class="rounded centered" id="thevideo' + mid + '" width=320 height=240 autoplay playsinline/>');
+										Janus.attachMediaStream($('#thevideo' + mid).get(0), stream);
 									}
 								},
 								oncleanup: function() {
 									Janus.log(" ::: Got a cleanup notification :::");
 									// FIXME Reset status
-									$('#waitingvideo').remove();
 									if(spinner !== null && spinner !== undefined)
 										spinner.stop();
 									spinner = null;
@@ -327,6 +410,10 @@ $(document).ready(function() {
 									$('#recset').removeAttr('disabled');
 									$('#recslist').removeAttr('disabled');
 									updateRecsList();
+									localTracks = {};
+									localVideos = 0;
+									remoteTracks = {};
+									remoteVideos = 0;
 								}
 							});
 					},
diff --git a/html/screensharingtest.js b/html/screensharingtest.js
index fc1b646a7f..11538bfbfb 100644
--- a/html/screensharingtest.js
+++ b/html/screensharingtest.js
@@ -61,6 +61,8 @@ var role = null;
 var room = null;
 var source = null;
 
+var localTracks = {}, localVideos = 0,
+	remoteTracks = {}, remoteVideos = 0;
 var spinner = null;
 
 
@@ -135,6 +137,12 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium);
+								},
 								webrtcState: function(on) {
 									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
 									$("#screencapture").parent().unblock();
@@ -147,6 +155,10 @@ $(document).ready(function() {
 										});
 									}
 								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message (publisher) :::");
 									Janus.debug(msg);
@@ -221,18 +233,74 @@ $(document).ready(function() {
 										screentest.handleRemoteJsep({jsep: jsep});
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									Janus.debug(stream);
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#screenvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#screencapture .no-video-container').length === 0) {
+													$('#screencapture').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
 									$('#screenmenu').hide();
-									$('#room').removeClass('hide').show();
-									if($('#screenvideo').length === 0) {
-										$('#screencapture').append('<video class="rounded centered" id="screenvideo" width="100%" height="100%" autoplay playsinline muted="muted"/>');
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#screencapture .no-video-container').length === 0) {
+												$('#screencapture').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#screencapture .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										$('#screencapture').append('<video class="rounded centered" id="screenvideo' + trackId + '" width=100% autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#screenvideo' + trackId).get(0), stream);
 									}
-									Janus.attachMediaStream($('#screenvideo').get(0), stream);
 									if(screentest.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											screentest.webrtcStuff.pc.iceConnectionState !== "connected") {
-										$("#screencapture").parent().block({
+										$("#screencapture").parent().parent().block({
 											message: '<b>Publishing...</b>',
 											css: {
 												border: 'none',
@@ -242,7 +310,7 @@ $(document).ready(function() {
 										});
 									}
 								},
-								onremotestream: function(stream) {
+								onremotetrack: function(track, mid, on) {
 									// The publisher stream is sendonly, we don't expect anything here
 								},
 								oncleanup: function() {
@@ -250,6 +318,8 @@ $(document).ready(function() {
 									$('#screencapture').empty();
 									$("#screencapture").parent().unblock();
 									$('#room').hide();
+									localTracks = {};
+									localVideos = 0;
 								}
 							});
 					},
@@ -394,6 +464,8 @@ function newRemoteFeed(id, display) {
 			opaqueId: opaqueId,
 			success: function(pluginHandle) {
 				remoteFeed = pluginHandle;
+				remoteFeed.remoteTracks = {};
+				remoteFeed.remoteVideos = 0;
 				Janus.log("Plugin attached! (" + remoteFeed.getPlugin() + ", id=" + remoteFeed.getId() + ")");
 				Janus.log("  -- This is a subscriber");
 				// We wait for the plugin to send us an offer
@@ -404,6 +476,16 @@ function newRemoteFeed(id, display) {
 				Janus.error("  -- Error attaching plugin...", error);
 				bootbox.alert("Error attaching plugin... " + error);
 			},
+			iceState: function(state) {
+				Janus.log("ICE state (feed #" + remoteFeed.rfindex + ") changed to " + state);
+			},
+			webrtcState: function(on) {
+				Janus.log("Janus says this WebRTC PeerConnection (feed #" + remoteFeed.rfindex + ") is " + (on ? "up" : "down") + " now");
+			},
+			slowLink: function(uplink, lost, mid) {
+				Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+					" packets on mid " + mid + " (" + lost + " lost packets)");
+			},
 			onmessage: function(msg, jsep) {
 				Janus.debug(" ::: Got a message (listener) :::");
 				Janus.debug(msg);
@@ -446,24 +528,76 @@ function newRemoteFeed(id, display) {
 						});
 				}
 			},
-			onlocalstream: function(stream) {
+			onlocaltrack: function(track, on) {
 				// The subscriber stream is recvonly, we don't expect anything here
 			},
-			onremotestream: function(stream) {
-				if($('#screenvideo').length === 0) {
-					// No remote video yet
-					$('#screencapture').append('<video class="rounded centered" id="waitingvideo" width="100%" height="100%" />');
-					$('#screencapture').append('<video class="rounded centered hide" id="screenvideo" width="100%" height="100%" autoplay playsinline/>');
-					// Show the video, hide the spinner and show the resolution when we get a playing event
-					$("#screenvideo").bind("playing", function () {
-						$('#waitingvideo').remove();
-						$('#screenvideo').removeClass('hide');
-						if(spinner !== null && spinner !== undefined)
-							spinner.stop();
-						spinner = null;
-					});
+			onremotetrack: function(track, mid, on) {
+				Janus.debug(" ::: Got a remote track event :::");
+				Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+				if(!on) {
+					// Track removed, get rid of the stream and the rendering
+					var stream = remoteTracks[mid];
+					if(stream) {
+						try {
+							var tracks = stream.getTracks();
+							for(var i in tracks) {
+								var mst = tracks[i];
+								if(mst !== null && mst !== undefined)
+									mst.stop();
+							}
+						} catch(e) {}
+					}
+					$('#screenvideo' + mid).remove();
+					if(track.kind === "video") {
+						remoteVideos--;
+						if(remoteVideos === 0) {
+							// No video, at least for now: show a placeholder
+							if($('#screencapture .no-video-container').length === 0) {
+								$('#screencapture').append(
+									'<div class="no-video-container">' +
+										'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+										'<span class="no-video-text">No remote video available</span>' +
+									'</div>');
+							}
+						}
+					}
+					delete remoteTracks[mid];
+					return;
+				}
+				// If we're here, a new track was added
+				if(spinner !== undefined && spinner !== null) {
+					spinner.stop();
+					spinner = null;
+				}
+				if(track.kind === "audio") {
+					// New audio track: create a stream out of it, and use a hidden <audio> element
+					stream = new MediaStream();
+					stream.addTrack(track.clone());
+					remoteTracks[mid] = stream;
+					Janus.log("Created remote audio stream:", stream);
+					$('#screencapture').append('<audio class="hide" id="screenvideo' + mid + '" autoplay playsinline/>');
+					Janus.attachMediaStream($('#screenvideo' + mid).get(0), stream);
+					if(remoteVideos === 0) {
+						// No video, at least for now: show a placeholder
+						if($('#screencapture .no-video-container').length === 0) {
+							$('#screencapture').append(
+								'<div class="no-video-container">' +
+									'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+									'<span class="no-video-text">No remote video available</span>' +
+								'</div>');
+						}
+					}
+				} else {
+					// New video track: create a stream out of it
+					remoteVideos++;
+					$('#screencapture .no-video-container').remove();
+					stream = new MediaStream();
+					stream.addTrack(track.clone());
+					remoteFeed.remoteTracks[mid] = stream;
+					Janus.log("Created remote video stream:", stream);
+					$('#screencapture').append('<video class="rounded centered" id="screenvideo' + mid + '" width=100% autoplay playsinline/>');
+					Janus.attachMediaStream($('#screenvideo' + mid).get(0), stream);
 				}
-				Janus.attachMediaStream($('#screenvideo').get(0), stream);
 			},
 			oncleanup: function() {
 				Janus.log(" ::: Got a cleanup notification (remote feed " + id + ") :::");
@@ -471,6 +605,8 @@ function newRemoteFeed(id, display) {
 				if(spinner !== null && spinner !== undefined)
 					spinner.stop();
 				spinner = null;
+				remoteFeed.remoteTracks = {};
+				remoteFeed.remoteVideos = 0;
 			}
 		});
 }
diff --git a/html/sipretest.js b/html/sipretest.js
index 2107d4bd0a..304c28525d 100644
--- a/html/sipretest.js
+++ b/html/sipretest.js
@@ -52,6 +52,8 @@ var janus = null;
 var sipcall = null;
 var opaqueId = "sipretest-"+Janus.randomString(12);
 
+var localTracks = {}, localVideos = 0,
+	remoteTracks = {}, remoteVideos = 0;
 var spinner = null;
 
 var selectedApproach = null;
@@ -124,7 +126,7 @@ $(document).ready(function() {
 									Janus.debug("Consent dialog should be " + (on ? "on" : "off") + " now");
 									if(on) {
 										// Darken screen and show hint
-										$.blockUI({ 
+										$.blockUI({
 											message: '<div><img src="up_arrow.png"/></div>',
 											css: {
 												border: 'none',
@@ -139,13 +141,20 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
-								mediaState: function(medium, on) {
-									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium);
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
 								},
 								webrtcState: function(on) {
 									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
 									$("#videoleft").parent().unblock();
 								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message :::");
 									Janus.debug(msg);
@@ -290,7 +299,7 @@ $(document).ready(function() {
 														}
 													}
 												}
-											});											
+											});
 										} else if(event === 'accepting') {
 											// Response to an offerless INVITE, let's wait for an 'accepted'
 										} else if(event === 'progress') {
@@ -326,14 +335,73 @@ $(document).ready(function() {
 										}
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									Janus.debug(stream);
-									$('#videos').removeClass('hide').show();
-									if($('#myvideo').length === 0)
-										$('#videoleft').append('<video class="rounded centered" id="myvideo" width=320 height=240 autoplay playsinline muted="muted"/>');
-									Janus.attachMediaStream($('#myvideo').get(0), stream);
-									$("#myvideo").get(0).muted = "muted";
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoleft .no-video-container').length === 0) {
+													$('#videoleft').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
+									if($('#videoleft video').length === 0) {
+										$('#videos').removeClass('hide').show();
+									}
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoleft .no-video-container').length === 0) {
+												$('#videoleft').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videoleft .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										$('#videoleft').append('<video class="rounded centered" id="myvideo' + trackId + '" width=320 height=240 autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
 									if(sipcall.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											sipcall.webrtcStuff.pc.iceConnectionState !== "connected") {
 										$("#videoleft").parent().block({
@@ -344,39 +412,46 @@ $(document).ready(function() {
 												color: 'white'
 											}
 										});
-										// No remote video yet
-										$('#videoright').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
-										if(spinner == null) {
-											var target = document.getElementById('videoright');
-											spinner = new Spinner({top:100}).spin(target);
-										} else {
-											spinner.spin();
-										}
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No webcam
-										$('#myvideo').hide();
-										if($('#videoleft .no-video-container').length === 0) {
-											$('#videoleft').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
+								},
+								onremotetrack: function(track, mid, on) {
+									Janus.debug(" ::: Got a remote track event :::");
+									Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = remoteTracks[mid];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
 										}
-									} else {
-										$('#videoleft .no-video-container').remove();
-										$('#myvideo').removeClass('hide').show();
+										$('#peervideo' + mid).remove();
+										if(track.kind === "video") {
+											remoteVideos--;
+											if(remoteVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoright .no-video-container').length === 0) {
+													$('#videoright').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No remote video available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete remoteTracks[mid];
+										return;
 									}
-								},
-								onremotestream: function(stream) {
-									Janus.debug(" ::: Got a remote stream :::");
-									Janus.debug(stream);
-									if($('#remotevideo').length === 0) {
+									// If we're here, a new track was added
+									if($('#videoright audio').length === 0 && $('#videoright video').length === 0) {
+										$('#videos').removeClass('hide').show();
 										$('#videoright').parent().find('h3').html(
 											'Send DTMF: <span id="dtmf" class="btn-group btn-group-xs"></span>');
-										$('#videoright').append(
-											'<video class="rounded centered hide" id="remotevideo" width=320 height=240 autoplay playsinline/>');
 										for(var i=0; i<12; i++) {
 											if(i<10)
 												$('#dtmf').append('<button class="btn btn-info dtmf">' + i + '</button>');
@@ -391,41 +466,47 @@ $(document).ready(function() {
 											// Notice you can also send DTMF tones using SIP INFO
 											// 		sipcall.send({message: {request: "dtmf_info", digit: $(this).text()}});
 										});
-										// Show the peer and hide the spinner when we get a playing event
-										$("#remotevideo").bind("playing", function () {
-											$('#waitingvideo').remove();
-											if(this.videoWidth)
-												$('#remotevideo').removeClass('hide').show();
-											if(spinner !== null && spinner !== undefined)
-												spinner.stop();
-											spinner = null;
-										});
 									}
-									Janus.attachMediaStream($('#remotevideo').get(0), stream);
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#remotevideo').hide();
-										if($('#videoright .no-video-container').length === 0) {
-											$('#videoright').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
+									if(track.kind === "audio") {
+										// New audio track: create a stream out of it, and use a hidden <audio> element
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote audio stream:", stream);
+										$('#videoright').append('<audio class="hide" id="peervideo' + mid + '" autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										if(remoteVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoright .no-video-container').length === 0) {
+												$('#videoright').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No remote video available</span>' +
+													'</div>');
+											}
 										}
 									} else {
+										// New video track: create a stream out of it
+										remoteVideos++;
 										$('#videoright .no-video-container').remove();
-										$('#remotevideo').removeClass('hide').show();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote video stream:", stream);
+										$('#videoright').append('<video class="rounded centered" id="peervideo' + mid + '" width=320 height=240 autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
 									}
 								},
 								oncleanup: function() {
 									Janus.log(" ::: Got a cleanup notification :::");
-									$('#myvideo').remove();
-									$('#waitingvideo').remove();
-									$('#remotevideo').remove();
-									$('.no-video-container').remove();
+									$("#videoleft").empty().parent().unblock();
+									$('#videoright').empty();
 									$('#videos').hide();
 									$('#dtmf').parent().html("Remote UA");
+									localTracks = {};
+									localVideos = 0;
+									remoteTracks = {};
+									remoteVideos = 0;
 								}
 							});
 					},
@@ -442,7 +523,7 @@ $(document).ready(function() {
 		});
 	}});
 });
-	
+
 function checkEnter(field, event) {
 	var theCode = event.keyCode ? event.keyCode : event.which ? event.which : event.charCode;
 	if(theCode == 13) {
@@ -525,7 +606,7 @@ function registerUsername() {
 						$('#register').removeAttr('disabled').click(registerUsername);
 						$('#registerset').removeAttr('disabled');
 					}
-				}); 
+				});
 		} else {
 			sipcall.send({"message": register});
 		}
@@ -586,7 +667,7 @@ function registerUsername() {
 					$('#register').removeAttr('disabled').click(registerUsername);
 					$('#registerset').removeAttr('disabled');
 				}
-			}); 
+			});
 	} else {
 		register["proxy"] = sipserver;
 		// Uncomment this if you want to see an outbound proxy too
@@ -617,7 +698,7 @@ function doCall() {
 	}
 	// Call this URI
 	doVideo = $('#dovideo').is(':checked');
-	Janus.log("This is a SIP " + (doVideo ? "video" : "audio") + " call (dovideo=" + doVideo + ")"); 
+	Janus.log("This is a SIP " + (doVideo ? "video" : "audio") + " call (dovideo=" + doVideo + ")");
 	sipcall.createOffer(
 		{
 			media: {
diff --git a/html/siptest.js b/html/siptest.js
index 394faf146a..289f2bb534 100644
--- a/html/siptest.js
+++ b/html/siptest.js
@@ -52,6 +52,8 @@ var janus = null;
 var sipcall = null;
 var opaqueId = "siptest-"+Janus.randomString(12);
 
+var localTracks = {}, localVideos = 0,
+	remoteTracks = {}, remoteVideos = 0;
 var spinner = null;
 
 var selectedApproach = null;
@@ -143,13 +145,20 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
-								mediaState: function(medium, on) {
-									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium);
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
 								},
 								webrtcState: function(on) {
 									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
 									$("#videoleft").parent().unblock();
 								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message :::");
 									Janus.debug(msg);
@@ -409,14 +418,73 @@ $(document).ready(function() {
 										}
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									Janus.debug(stream);
-									$('#videos').removeClass('hide').show();
-									if($('#myvideo').length === 0)
-										$('#videoleft').append('<video class="rounded centered" id="myvideo" width=320 height=240 autoplay playsinline muted="muted"/>');
-									Janus.attachMediaStream($('#myvideo').get(0), stream);
-									$("#myvideo").get(0).muted = "muted";
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoleft .no-video-container').length === 0) {
+													$('#videoleft').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
+									if($('#videoleft video').length === 0) {
+										$('#videos').removeClass('hide').show();
+									}
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoleft .no-video-container').length === 0) {
+												$('#videoleft').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videoleft .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										$('#videoleft').append('<video class="rounded centered" id="myvideo' + trackId + '" width=320 height=240 autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
 									if(sipcall.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											sipcall.webrtcStuff.pc.iceConnectionState !== "connected") {
 										$("#videoleft").parent().block({
@@ -427,39 +495,46 @@ $(document).ready(function() {
 												color: 'white'
 											}
 										});
-										// No remote video yet
-										$('#videoright').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
-										if(spinner == null) {
-											var target = document.getElementById('videoright');
-											spinner = new Spinner({top:100}).spin(target);
-										} else {
-											spinner.spin();
-										}
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No webcam
-										$('#myvideo').hide();
-										if($('#videoleft .no-video-container').length === 0) {
-											$('#videoleft').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
+								},
+								onremotetrack: function(track, mid, on) {
+									Janus.debug(" ::: Got a remote track event :::");
+									Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = remoteTracks[mid];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
 										}
-									} else {
-										$('#videoleft .no-video-container').remove();
-										$('#myvideo').removeClass('hide').show();
+										$('#peervideo' + mid).remove();
+										if(track.kind === "video") {
+											remoteVideos--;
+											if(remoteVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoright .no-video-container').length === 0) {
+													$('#videoright').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No remote video available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete remoteTracks[mid];
+										return;
 									}
-								},
-								onremotestream: function(stream) {
-									Janus.debug(" ::: Got a remote stream :::");
-									Janus.debug(stream);
-									if($('#remotevideo').length === 0) {
+									// If we're here, a new track was added
+									if($('#videoright audio').length === 0 && $('#videoright video').length === 0) {
+										$('#videos').removeClass('hide').show();
 										$('#videoright').parent().find('h3').html(
 											'Send DTMF: <span id="dtmf" class="btn-group btn-group-xs"></span>');
-										$('#videoright').append(
-											'<video class="rounded centered hide" id="remotevideo" width=320 height=240 autoplay playsinline/>');
 										for(var i=0; i<12; i++) {
 											if(i<10)
 												$('#dtmf').append('<button class="btn btn-info dtmf">' + i + '</button>');
@@ -474,41 +549,47 @@ $(document).ready(function() {
 											// Notice you can also send DTMF tones using SIP INFO
 											// 		sipcall.send({message: {request: "dtmf_info", digit: $(this).text()}});
 										});
-										// Show the peer and hide the spinner when we get a playing event
-										$("#remotevideo").bind("playing", function () {
-											$('#waitingvideo').remove();
-											if(this.videoWidth)
-												$('#remotevideo').removeClass('hide').show();
-											if(spinner !== null && spinner !== undefined)
-												spinner.stop();
-											spinner = null;
-										});
 									}
-									Janus.attachMediaStream($('#remotevideo').get(0), stream);
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#remotevideo').hide();
-										if($('#videoright .no-video-container').length === 0) {
-											$('#videoright').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
+									if(track.kind === "audio") {
+										// New audio track: create a stream out of it, and use a hidden <audio> element
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote audio stream:", stream);
+										$('#videoright').append('<audio class="hide" id="peervideo' + mid + '" autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										if(remoteVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoright .no-video-container').length === 0) {
+												$('#videoright').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No remote video available</span>' +
+													'</div>');
+											}
 										}
 									} else {
+										// New video track: create a stream out of it
+										remoteVideos++;
 										$('#videoright .no-video-container').remove();
-										$('#remotevideo').removeClass('hide').show();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote video stream:", stream);
+										$('#videoright').append('<video class="rounded centered" id="peervideo' + mid + '" width=320 height=240 autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
 									}
 								},
 								oncleanup: function() {
 									Janus.log(" ::: Got a cleanup notification :::");
-									$('#myvideo').remove();
-									$('#waitingvideo').remove();
-									$('#remotevideo').remove();
-									$('#videos .no-video-container').remove();
+									$("#videoleft").empty().parent().unblock();
+									$('#videoright').empty();
 									$('#videos').hide();
 									$('#dtmf').parent().html("Remote UA");
+									localTracks = {};
+									localVideos = 0;
+									remoteTracks = {};
+									remoteVideos = 0;
 								}
 							});
 					},
@@ -1240,10 +1321,8 @@ function addHelper(helperCreated) {
 			},
 			oncleanup: function() {
 				Janus.log("[Helper #" + helperId + "]  ::: Got a cleanup notification :::");
-				$('#myvideo' + helperId).remove();
-				$('#waitingvideo' + helperId).remove();
-				$('#remotevideo' + helperId).remove();
-				$('#videos' + helperId + ' .no-video-container').remove();
+				$('#videoleft' + helperId).empty().parent().unblock();
+				$('#videoright' + helperId).empty();
 				$('#videos' + helperId).hide();
 				$('#dtmf' + helperId).parent().html("Remote UA");
 			}
diff --git a/html/streamingtest.html b/html/streamingtest.html
index c54c951dd1..4622b3ab59 100644
--- a/html/streamingtest.html
+++ b/html/streamingtest.html
@@ -86,17 +86,7 @@ <h3 class="panel-title">Streams <i id="update-streams" class="fa fa-refresh" tit
 							</div>
 						</div>
 					</div>
-					<div class="col-md-6">
-						<div class="panel panel-default">
-							<div class="panel-heading">
-								<h3 class="panel-title">Stream
-									<span class="label label-info hide" id="status"></span>
-									<span class="label label-primary hide" id="curres"></span>
-									<span class="label label-info hide" id="curbitrate"></span>
-								</h3>
-							</div>
-							<div class="panel-body" id="stream"></div>
-						</div>
+					<div class="col-md-6" id="videos">
 					</div>
 				</div>
 			</div>
diff --git a/html/streamingtest.js b/html/streamingtest.js
index e0e612fc06..86743f4459 100644
--- a/html/streamingtest.js
+++ b/html/streamingtest.js
@@ -52,11 +52,13 @@ var janus = null;
 var streaming = null;
 var opaqueId = "streamingtest-"+Janus.randomString(12);
 
-var bitrateTimer = null;
-var spinner = null;
+var remoteTracks = {}, remoteVideos = 0, dataMid = null;
+var bitrateTimer = {};
+var spinner = {};
 
-var simulcastStarted = false, svcStarted = false;
+var simulcastStarted = {}, svcStarted = {};
 
+var streamsList = {};
 var selectedStream = null;
 
 
@@ -91,7 +93,9 @@ $(document).ready(function() {
 									$('#start').removeAttr('disabled').html("Stop")
 										.click(function() {
 											$(this).attr('disabled', true);
-											clearInterval(bitrateTimer);
+											for(var i in bitrateTimer)
+												clearInterval(bitrateTimer[i]);
+											bitrateTimer = {};
 											janus.destroy();
 											$('#streamslist').attr('disabled', true);
 											$('#watch').attr('disabled', true).unbind('click');
@@ -102,6 +106,16 @@ $(document).ready(function() {
 									Janus.error("  -- Error attaching plugin... ", error);
 									bootbox.alert("Error attaching plugin... " + error);
 								},
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								webrtcState: function(on) {
+									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
+								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message :::");
 									Janus.debug(msg);
@@ -116,27 +130,29 @@ $(document).ready(function() {
 											else if(status === 'stopped')
 												stopStream();
 										} else if(msg["streaming"] === "event") {
+											// Does this event refer to a mid in particular?
+											var mid = result["mid"] ? result["mid"] : "0";
 											// Is simulcast in place?
 											var substream = result["substream"];
 											var temporal = result["temporal"];
 											if((substream !== null && substream !== undefined) || (temporal !== null && temporal !== undefined)) {
-												if(!simulcastStarted) {
-													simulcastStarted = true;
-													addSimulcastButtons(temporal !== null && temporal !== undefined);
+												if(!simulcastStarted[mid]) {
+													simulcastStarted[mid] = true;
+													addSimulcastButtons(mid, temporal !== null && temporal !== undefined);
 												}
 												// We just received notice that there's been a switch, update the buttons
-												updateSimulcastButtons(substream, temporal);
+												updateSimulcastButtons(mid, substream, temporal);
 											}
 											// Is VP9/SVC in place?
 											var spatial = result["spatial_layer"];
 											temporal = result["temporal_layer"];
 											if((spatial !== null && spatial !== undefined) || (temporal !== null && temporal !== undefined)) {
-												if(!svcStarted) {
-													svcStarted = true;
-													addSvcButtons();
+												if(!svcStarted[mid]) {
+													svcStarted[mid] = true;
+													addSvcButtons(mid);
 												}
 												// We just received notice that there's been a switch, update the buttons
-												updateSvcButtons(spatial, temporal);
+												updateSvcButtons(mid, spatial, temporal);
 											}
 										}
 									} else if(msg["error"] !== undefined && msg["error"] !== null) {
@@ -158,7 +174,7 @@ $(document).ready(function() {
 													Janus.debug(jsep);
 													var body = { "request": "start" };
 													streaming.send({"message": body, "jsep": jsep});
-													$('#watch').html("Stop").removeAttr('disabled').click(stopStream);
+													$('#watch').html("Stop").removeAttr('disabled').unbind('click').click(stopStream);
 												},
 												error: function(error) {
 													Janus.error("WebRTC error:", error);
@@ -167,82 +183,127 @@ $(document).ready(function() {
 											});
 									}
 								},
-								onremotestream: function(stream) {
-									Janus.debug(" ::: Got a remote stream :::");
-									Janus.debug(stream);
+								onremotetrack: function(track, mid, on) {
+									Janus.debug(" ::: Got a remote track event :::");
+									Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									var mstreamId = "mstream"+mid;
+									if(streamsList[selectedStream] && streamsList[selectedStream].legacy)
+										mstreamId = "mstream0";
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = remoteTracks[mid];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										$('#remotevideo' + mid).remove();
+										if(track.kind === "video") {
+											remoteVideos--;
+											if(remoteVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#'+mstreamId+' .no-video-container').length === 0) {
+												$('#'+mstreamId).append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No remote video available</span>' +
+													'</div>');
+												}
+											}
+										}
+										delete remoteTracks[mid];
+										return;
+									}
+									// If we're here, a new track was added
 									var addButtons = false;
-									if($('#remotevideo').length === 0) {
+									if($('#'+mstreamId+' audio').length === 0 && $('#'+mstreamId+' video').length === 0) {
 										addButtons = true;
-										$('#stream').append('<video class="rounded centered hide" id="remotevideo" width=320 height=240 autoplay playsinline/>');
-										// Show the stream and hide the spinner when we get a playing event
-										$("#remotevideo").bind("playing", function () {
-											$('#waitingvideo').remove();
-											if(this.videoWidth)
-												$('#remotevideo').removeClass('hide').show();
-											if(spinner !== null && spinner !== undefined)
-												spinner.stop();
-											spinner = null;
-											var videoTracks = stream.getVideoTracks();
-											if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0)
-												return;
-											var width = this.videoWidth;
-											var height = this.videoHeight;
-											$('#curres').removeClass('hide').text(width+'x'+height).show();
-											if(Janus.webRTCAdapter.browserDetails.browser === "firefox") {
-												// Firefox Stable has a bug: width and height are not immediately available after a playing
-												setTimeout(function() {
-													var width = $("#remotevideo").get(0).videoWidth;
-													var height = $("#remotevideo").get(0).videoHeight;
-													$('#curres').removeClass('hide').text(width+'x'+height).show();
-												}, 2000);
-											}
-										});
 									}
-									Janus.attachMediaStream($('#remotevideo').get(0), stream);
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#remotevideo').hide();
-										if($('#stream .no-video-container').length === 0) {
-											$('#stream').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
+									if(track.kind === "audio") {
+										// New audio track: create a stream out of it, and use a hidden <audio> element
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote audio stream:", stream);
+										$('#'+mstreamId).append('<audio class="hide" id="remotevideo' + mid + '" autoplay playsinline/>');
+										if(remoteVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#'+mstreamId+' .no-video-container').length === 0) {
+												$('#'+mstreamId).append(
+													'<div class="no-video-container audioonly">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
 										}
 									} else {
-										$('#stream .no-video-container').remove();
-										$('#remotevideo').removeClass('hide').show();
+										// New video track: create a stream out of it
+										remoteVideos++;
+										$('.no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote video stream:", stream);
+										$('#'+mstreamId).append('<video class="rounded centered hide" id="remotevideo' + mid + '" width=320 height=240 autoplay playsinline/>');
+										// Note: we'll need this for additional videos too
+										if(!bitrateTimer[mid]) {
+											$('#curbitrate'+mid).removeClass('hide').show();
+											bitrateTimer[mid] = setInterval(function() {
+												if(!$("#remotevideo" + mid).get(0))
+													return;
+												// Display updated bitrate, if supported
+												var bitrate = streaming.getBitrate(mid);
+												//~ Janus.debug("Current bitrate is " + streaming.getBitrate());
+												$('#curbitrate'+mid).text(bitrate);
+												// Check if the resolution changed too
+												var width = $("#remotevideo" + mid).get(0).videoWidth;
+												var height = $("#remotevideo" + mid).get(0).videoHeight;
+												if(width > 0 && height > 0)
+													$('#curres'+mid).removeClass('hide').text(width+'x'+height).show();
+											}, 1000);
+										}
 									}
+									// Play the stream and hide the spinner when we get a playing event
+									$("#remotevideo" + mid).bind("playing", function (ev) {
+										$('.waitingvideo').remove();
+										if(spinner[mid] !== null && spinner[mid] !== undefined)
+											spinner[mid].stop();
+										spinner[mid] = null;
+										if(!this.videoWidth)
+											return;
+										$('#'+ev.target.id).removeClass('hide').show();
+										var width = this.videoWidth;
+										var height = this.videoHeight;
+										$('#curres'+mid).removeClass('hide').text(width+'x'+height).show();
+										if(Janus.webRTCAdapter.browserDetails.browser === "firefox") {
+											// Firefox Stable has a bug: width and height are not immediately available after a playing
+											setTimeout(function() {
+												var width = $('#'+ev.target.id).get(0).videoWidth;
+												var height = $('#'+ev.target.id).get(0).videoHeight;
+												$('#curres'+mid).removeClass('hide').text(width+'x'+height).show();
+											}, 2000);
+										}
+									});
+									Janus.attachMediaStream($('#remotevideo' + mid).get(0), stream);
 									if(!addButtons)
 										return;
-									if(videoTracks && videoTracks.length &&
-											(Janus.webRTCAdapter.browserDetails.browser === "chrome" ||
-												Janus.webRTCAdapter.browserDetails.browser === "firefox" ||
-												Janus.webRTCAdapter.browserDetails.browser === "safari")) {
-										$('#curbitrate').removeClass('hide').show();
-										bitrateTimer = setInterval(function() {
-											// Display updated bitrate, if supported
-											var bitrate = streaming.getBitrate();
-											//~ Janus.debug("Current bitrate is " + streaming.getBitrate());
-											$('#curbitrate').text(bitrate);
-											// Check if the resolution changed too
-											var width = $("#remotevideo").get(0).videoWidth;
-											var height = $("#remotevideo").get(0).videoHeight;
-											if(width > 0 && height > 0)
-												$('#curres').removeClass('hide').text(width+'x'+height).show();
-										}, 1000);
-									}
 								},
 								ondataopen: function(data) {
 									Janus.log("The DataChannel is available!");
-									$('#waitingvideo').remove();
-									$('#stream').append(
+									$('.waitingvideo').remove();
+									$('#mstream'+dataMid).append(
 										'<input class="form-control" type="text" id="datarecv" disabled></input>'
 									);
-									if(spinner !== null && spinner !== undefined)
-										spinner.stop();
-									spinner = null;
+									for(var i in spinner) {
+										if(spinner[i])
+											spinner[i].stop();
+									}
+									spinner = {};
 								},
 								ondata: function(data) {
 									Janus.debug("We got data from the DataChannel! " + data);
@@ -250,19 +311,23 @@ $(document).ready(function() {
 								},
 								oncleanup: function() {
 									Janus.log(" ::: Got a cleanup notification :::");
-									$('#waitingvideo').remove();
-									$('#remotevideo').remove();
-									$('#datarecv').remove();
-									$('.no-video-container').remove();
-									$('#bitrate').attr('disabled', true);
-									$('#bitrateset').html('Bandwidth<span class="caret"></span>');
-									$('#curbitrate').hide();
-									if(bitrateTimer !== null && bitrateTimer !== undefined)
-										clearInterval(bitrateTimer);
-									bitrateTimer = null;
-									$('#curres').hide();
-									$('#simulcast').remove();
+									$('#videos').empty();
+									for(var i in bitrateTimer)
+										clearInterval(bitrateTimer[i]);
+									bitrateTimer = {};
+									for(var i in spinner) {
+										if(spinner[i])
+											spinner[i].stop();
+									}
+									spinner = {};
 									simulcastStarted = false;
+									remoteTracks = {};
+									remoteVideos = 0;
+									dataMid = null;
+									$('#streamset').removeAttr('disabled');
+									$('#streamslist').removeAttr('disabled');
+									$('#watch').html("Watch or Listen").removeAttr('disabled')
+										.unbind('click').click(startStream);
 								}
 							});
 					},
@@ -286,7 +351,7 @@ function updateStreamsList() {
 	Janus.debug("Sending message (" + JSON.stringify(body) + ")");
 	streaming.send({"message": body, success: function(result) {
 		setTimeout(function() {
-			$('#update-streams').removeClass('fa-spin').click(updateStreamsList);
+			$('#update-streams').removeClass('fa-spin').unbind('click').click(updateStreamsList);
 		}, 500);
 		if(result === null || result === undefined) {
 			bootbox.alert("Got no response to our query for available streams");
@@ -299,9 +364,29 @@ function updateStreamsList() {
 			var list = result["list"];
 			Janus.log("Got a list of available streams");
 			Janus.debug(list);
+			streamsList = {};
 			for(var mp in list) {
 				Janus.debug("  >> [" + list[mp]["id"] + "] " + list[mp]["description"] + " (" + list[mp]["type"] + ")");
 				$('#streamslist').append("<li><a href='#' id='" + list[mp]["id"] + "'>" + list[mp]["description"] + " (" + list[mp]["type"] + ")" + "</a></li>");
+				// Check the nature of the available streams, and if there are some multistream ones
+				list[mp].legacy = true;
+				if(list[mp].media) {
+					var audios = 0, videos = 0;
+					for(var mi in list[mp].media) {
+						if(!list[mp].media[mi])
+							continue;
+						if(list[mp].media[mi].type === "audio")
+							audios++;
+						else if(list[mp].media[mi].type === "video")
+							videos++;
+						if(audios > 1 || videos > 1) {
+							list[mp].legacy = false;
+							break;
+						}
+					}
+				}
+				// Keep track of all the available streams
+				streamsList[list[mp]["id"]] = list[mp];
 			}
 			$('#streamslist a').unbind('click').click(function() {
 				selectedStream = $(this).attr("id");
@@ -316,23 +401,73 @@ function updateStreamsList() {
 
 function startStream() {
 	Janus.log("Selected video id #" + selectedStream);
-	if(selectedStream === undefined || selectedStream === null) {
+	if(selectedStream === null || selectedStream === undefined || !streamsList[selectedStream]) {
 		bootbox.alert("Select a stream from the list");
 		return;
 	}
 	$('#streamset').attr('disabled', true);
 	$('#streamslist').attr('disabled', true);
 	$('#watch').attr('disabled', true).unbind('click');
-	var body = { "request": "watch", id: parseInt(selectedStream) };
-	streaming.send({"message": body});
-	// No remote video yet
-	$('#stream').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
-	if(spinner == null) {
-		var target = document.getElementById('stream');
-		spinner = new Spinner({top:100}).spin(target);
+	// Add some panels to host the remote streams
+	if(streamsList[selectedStream].legacy) {
+		// At max 1-audio/1-video, so use a single panel
+		var mid = null;
+		for(mi in streamsList[selectedStream].media) {
+			// Add a new panel
+			var type = streamsList[selectedStream].media[mi].type;
+			if(type === "video") {
+				mid = streamsList[selectedStream].media[mi].mid;
+				break;
+			}
+		}
+		if($('#mstream0').length === 0) {
+			addPanel("0", mid);
+			// No remote video yet
+			$('#mstream0').append('<video class="rounded centered waitingvideo" id="waitingvideo0" width=320 height=240 />');
+		}
+		if(mid) {
+			if(spinner[mid] == null) {
+				var target = document.getElementById('mstream0');
+				spinner[mid] = new Spinner({top:100}).spin(target);
+			} else {
+				spinner[mid].spin();
+			}
+		}
+		dataMid = "0";
 	} else {
-		spinner.spin();
+		// Multistream mountpoint, create a panel for each stream
+		for(mi in streamsList[selectedStream].media) {
+			// Add a new panel
+			var type = streamsList[selectedStream].media[mi].type;
+			var mid = streamsList[selectedStream].media[mi].mid;
+			var label = streamsList[selectedStream].media[mi].label;
+			if($('#mstream'+mid).length === 0) {
+				addPanel(mid, mid, label);
+				// No remote media yet
+				$('#mstream'+mid).append('<video class="rounded centered waitingvideo" id="waitingvideo'+mid+'" width=320 height=240 />');
+			}
+			if(spinner[mid] == null) {
+				var target = document.getElementById('mstream'+mid);
+				spinner[mid] = new Spinner({top:100}).spin(target);
+			} else {
+				spinner[mid].spin();
+			}
+			if(type === 'data')
+				dataMid = mid;
+		}
 	}
+	// Prepare the request to start streaming and send it
+	var body = { "request": "watch", id: parseInt(selectedStream) };
+	// Notice that, for RTP mountpoints, you can subscribe to a subset
+	// of the mountpoint media, rather than them all, by adding a "stream"
+	// array containing the list of stream mids you're interested in, e.g.:
+	//
+	//		body.streams = [ "0", "2" ];
+	//
+	// to only subscribe to the first and third stream, and skip the second
+	// (assuming those are the mids you got from a "list" or "info" request).
+	// By default, you always subscribe to all the streams in a mountpoint
+	streaming.send({"message": body});
 }
 
 function stopStream() {
@@ -340,239 +475,255 @@ function stopStream() {
 	var body = { "request": "stop" };
 	streaming.send({"message": body});
 	streaming.hangup();
-	$('#streamset').removeAttr('disabled');
-	$('#streamslist').removeAttr('disabled');
-	$('#watch').html("Watch or Listen").removeAttr('disabled').unbind('click').click(startStream);
-	$('#status').empty().hide();
-	$('#bitrate').attr('disabled', true);
-	$('#bitrateset').html('Bandwidth<span class="caret"></span>');
-	$('#curbitrate').hide();
-	if(bitrateTimer !== null && bitrateTimer !== undefined)
-		clearInterval(bitrateTimer);
-	bitrateTimer = null;
-	$('#curres').empty().hide();
-	$('#simulcast').remove();
-	simulcastStarted = false;
+}
+
+// Helper to add a new panel to the 'videos' div
+function addPanel(panelId, mid, desc) {
+	$('#videos').append(
+		'<div class="row" id="panel' + panelId + '">' +
+		'	<div class="panel panel-default">' +
+		'		<div class="panel-heading">' +
+		'			<h3 class="panel-title">' + (desc ? desc : "Stream") +
+		'				<span class="label label-info hide" id="status' + mid + '"></span>' +
+		'				<span class="label label-primary hide" id="curres' + mid + '"></span>' +
+		'				<span class="label label-info hide" id="curbitrate' + mid + '"></span>' +
+		'			</h3>' +
+		'		</div>' +
+		'		<div class="panel-body" id="mstream' + panelId + '"></div>' +
+		'	</div>' +
+		'</div>'
+	);
 }
 
 // Helpers to create Simulcast-related UI, if enabled
-function addSimulcastButtons(temporal) {
-	$('#curres').parent().append(
-		'<div id="simulcast" class="btn-group-vertical btn-group-vertical-xs pull-right">' +
+function addSimulcastButtons(mid, temporal) {
+	$('#curres'+mid).parent().append(
+		'<div id="simulcast'+mid+'" class="btn-group-vertical btn-group-vertical-xs pull-right">' +
 		'	<div class"row">' +
 		'		<div class="btn-group btn-group-xs" style="width: 100%">' +
-		'			<button id="sl-2" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to higher quality" style="width: 33%">SL 2</button>' +
-		'			<button id="sl-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to normal quality" style="width: 33%">SL 1</button>' +
-		'			<button id="sl-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to lower quality" style="width: 34%">SL 0</button>' +
+		'			<button id="m-'+mid+'-sl-2" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to higher quality" style="width: 33%">SL 2</button>' +
+		'			<button id="m-'+mid+'-sl-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to normal quality" style="width: 33%">SL 1</button>' +
+		'			<button id="m-'+mid+'-sl-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to lower quality" style="width: 34%">SL 0</button>' +
 		'		</div>' +
 		'	</div>' +
 		'	<div class"row">' +
 		'		<div class="btn-group btn-group-xs hide" style="width: 100%">' +
-		'			<button id="tl-2" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 2" style="width: 34%">TL 2</button>' +
-		'			<button id="tl-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 1" style="width: 33%">TL 1</button>' +
-		'			<button id="tl-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 0" style="width: 33%">TL 0</button>' +
+		'			<button id="m-'+mid+'-tl-2" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 2" style="width: 34%">TL 2</button>' +
+		'			<button id="m-'+mid+'-tl-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 1" style="width: 33%">TL 1</button>' +
+		'			<button id="m-'+mid+'-tl-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 0" style="width: 33%">TL 0</button>' +
 		'		</div>' +
 		'	</div>' +
 		'</div>');
 	// Enable the simulcast selection buttons
-	$('#sl-0').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-sl')[0];
 			toastr.info("Switching simulcast substream, wait for it... (lower quality)", null, {timeOut: 2000});
-			if(!$('#sl-2').hasClass('btn-success'))
-				$('#sl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
-			if(!$('#sl-1').hasClass('btn-success'))
-				$('#sl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
-			$('#sl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			streaming.send({message: { request: "configure", substream: 0 }});
+			if(!$('#m-'+mid+'-sl-2').hasClass('btn-success'))
+				$('#m-'+mid+'-sl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#m-'+mid+'-sl-1').hasClass('btn-success'))
+				$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			streaming.send({message: { request: "configure", mid: mid, substream: 0 }});
 		});
-	$('#sl-1').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-sl')[0];
 			toastr.info("Switching simulcast substream, wait for it... (normal quality)", null, {timeOut: 2000});
-			if(!$('#sl-2').hasClass('btn-success'))
-				$('#sl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
-			$('#sl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			if(!$('#sl-0').hasClass('btn-success'))
-				$('#sl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
-			streaming.send({message: { request: "configure", substream: 1 }});
+			if(!$('#m-'+mid+'-sl-2').hasClass('btn-success'))
+				$('#m-'+mid+'-sl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			if(!$('#m-'+mid+'-sl-0').hasClass('btn-success'))
+				$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			streaming.send({message: { request: "configure", mid: mid, substream: 1 }});
 		});
-	$('#sl-2').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-sl-2').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-sl')[0];
 			toastr.info("Switching simulcast substream, wait for it... (higher quality)", null, {timeOut: 2000});
-			$('#sl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			if(!$('#sl-1').hasClass('btn-success'))
-				$('#sl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
-			if(!$('#sl-0').hasClass('btn-success'))
-				$('#sl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
-			streaming.send({message: { request: "configure", substream: 2 }});
+			$('#m-'+mid+'-sl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			if(!$('#m-'+mid+'-sl-1').hasClass('btn-success'))
+				$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#m-'+mid+'-sl-0').hasClass('btn-success'))
+				$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			streaming.send({message: { request: "configure", mid: mid, substream: 2 }});
 		});
 	if(!temporal)	// No temporal layer support
 		return;
-	$('#tl-0').parent().removeClass('hide');
-	$('#tl-0').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-tl-0').parent().removeClass('hide');
+	$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-tl')[0];
 			toastr.info("Capping simulcast temporal layer, wait for it... (lowest FPS)", null, {timeOut: 2000});
-			if(!$('#tl-2').hasClass('btn-success'))
-				$('#tl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
-			if(!$('#tl-1').hasClass('btn-success'))
-				$('#tl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
-			$('#tl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			streaming.send({message: { request: "configure", temporal: 0 }});
+			if(!$('#m-'+mid+'-tl-2').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#m-'+mid+'-tl-1').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			streaming.send({message: { request: "configure", mid: mid, temporal: 0 }});
 		});
-	$('#tl-1').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-tl')[0];
 			toastr.info("Capping simulcast temporal layer, wait for it... (medium FPS)", null, {timeOut: 2000});
-			if(!$('#tl-2').hasClass('btn-success'))
-				$('#tl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
-			$('#tl-1').removeClass('btn-primary btn-info').addClass('btn-info');
-			if(!$('#tl-0').hasClass('btn-success'))
-				$('#tl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
-			streaming.send({message: { request: "configure", temporal: 1 }});
+			if(!$('#m-'+mid+'-tl-2').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-info').addClass('btn-info');
+			if(!$('#m-'+mid+'-tl-0').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			streaming.send({message: { request: "configure", mid: mid, temporal: 1 }});
 		});
-	$('#tl-2').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-tl')[0];
 			toastr.info("Capping simulcast temporal layer, wait for it... (highest FPS)", null, {timeOut: 2000});
-			$('#tl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			if(!$('#tl-1').hasClass('btn-success'))
-				$('#tl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
-			if(!$('#tl-0').hasClass('btn-success'))
-				$('#tl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
-			streaming.send({message: { request: "configure", temporal: 2 }});
+			$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			if(!$('#m-'+mid+'-tl-1').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#m-'+mid+'-tl-0').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			streaming.send({message: { request: "configure", mid: mid, temporal: 2 }});
 		});
 }
 
-function updateSimulcastButtons(substream, temporal) {
+function updateSimulcastButtons(mid, substream, temporal) {
 	// Check the substream
 	if(substream === 0) {
 		toastr.success("Switched simulcast substream! (lower quality)", null, {timeOut: 2000});
-		$('#sl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#sl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#sl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-sl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
 	} else if(substream === 1) {
 		toastr.success("Switched simulcast substream! (normal quality)", null, {timeOut: 2000});
-		$('#sl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#sl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
-		$('#sl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-sl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
 	} else if(substream === 2) {
 		toastr.success("Switched simulcast substream! (higher quality)", null, {timeOut: 2000});
-		$('#sl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
-		$('#sl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#sl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-sl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
 	}
 	// Check the temporal layer
 	if(temporal === 0) {
 		toastr.success("Capped simulcast temporal layer! (lowest FPS)", null, {timeOut: 2000});
-		$('#tl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#tl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#tl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
 	} else if(temporal === 1) {
 		toastr.success("Capped simulcast temporal layer! (medium FPS)", null, {timeOut: 2000});
-		$('#tl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#tl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
-		$('#tl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
 	} else if(temporal === 2) {
 		toastr.success("Capped simulcast temporal layer! (highest FPS)", null, {timeOut: 2000});
-		$('#tl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
-		$('#tl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#tl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
 	}
 }
 
 // Helpers to create SVC-related UI for a new viewer
-function addSvcButtons() {
+function addSvcButtons(mid) {
 	if($('#svc').length > 0)
 		return;
-	$('#curres').parent().append(
-		'<div id="svc" class="btn-group-vertical btn-group-vertical-xs pull-right">' +
+	$('#curres'+mid).parent().append(
+		'<div id="svc'+mid+'" class="btn-group-vertical btn-group-vertical-xs pull-right">' +
 		'	<div class"row">' +
 		'		<div class="btn-group btn-group-xs" style="width: 100%">' +
-		'			<button id="sl-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to normal resolution" style="width: 50%">SL 1</button>' +
-		'			<button id="sl-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to low resolution" style="width: 50%">SL 0</button>' +
+		'			<button id="m-'+mid+'-sl-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to normal resolution" style="width: 50%">SL 1</button>' +
+		'			<button id="m-'+mid+'-sl-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Switch to low resolution" style="width: 50%">SL 0</button>' +
 		'		</div>' +
 		'	</div>' +
 		'	<div class"row">' +
 		'		<div class="btn-group btn-group-xs" style="width: 100%">' +
-		'			<button id="tl-2" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 2 (high FPS)" style="width: 34%">TL 2</button>' +
-		'			<button id="tl-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 1 (medium FPS)" style="width: 33%">TL 1</button>' +
-		'			<button id="tl-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 0 (low FPS)" style="width: 33%">TL 0</button>' +
+		'			<button id="m-'+mid+'-tl-2" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 2 (high FPS)" style="width: 34%">TL 2</button>' +
+		'			<button id="m-'+mid+'-tl-1" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 1 (medium FPS)" style="width: 33%">TL 1</button>' +
+		'			<button id="m-'+mid+'-tl-0" type="button" class="btn btn-primary" data-toggle="tooltip" title="Cap to temporal layer 0 (low FPS)" style="width: 33%">TL 0</button>' +
 		'		</div>' +
 		'	</div>' +
 		'</div>'
 	);
-	// Enable the VP8 simulcast selection buttons
-	$('#sl-0').removeClass('btn-primary btn-success').addClass('btn-primary')
+	// Enable the SVC selection buttons
+	$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-sl')[0];
 			toastr.info("Switching SVC spatial layer, wait for it... (low resolution)", null, {timeOut: 2000});
-			if(!$('#sl-1').hasClass('btn-success'))
-				$('#sl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
-			$('#sl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			streaming.send({message: { request: "configure", spatial_layer: 0 }});
+			if(!$('#m-'+mid+'-sl-1').hasClass('btn-success'))
+				$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			streaming.send({message: { request: "configure", mid: mid, spatial_layer: 0 }});
 		});
-	$('#sl-1').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-sl')[0];
 			toastr.info("Switching SVC spatial layer, wait for it... (normal resolution)", null, {timeOut: 2000});
-			$('#sl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			if(!$('#sl-0').hasClass('btn-success'))
-				$('#sl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
-			streaming.send({message: { request: "configure", spatial_layer: 1 }});
+			$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			if(!$('#m-'+mid+'-sl-0').hasClass('btn-success'))
+				$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			streaming.send({message: { request: "configure", mid: mid, spatial_layer: 1 }});
 		});
-	$('#tl-0').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-tl')[0];
 			toastr.info("Capping SVC temporal layer, wait for it... (lowest FPS)", null, {timeOut: 2000});
-			if(!$('#tl-2').hasClass('btn-success'))
-				$('#tl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
-			if(!$('#tl-1').hasClass('btn-success'))
-				$('#tl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
-			$('#tl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			streaming.send({message: { request: "configure", temporal_layer: 0 }});
+			if(!$('#m-'+mid+'-tl-2').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#m-'+mid+'-tl-1').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			streaming.send({message: { request: "configure", mid: mid, temporal_layer: 0 }});
 		});
-	$('#tl-1').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-tl')[0];
 			toastr.info("Capping SVC temporal layer, wait for it... (medium FPS)", null, {timeOut: 2000});
-			if(!$('#tl-2').hasClass('btn-success'))
-				$('#tl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
-			$('#tl-1').removeClass('btn-primary btn-info').addClass('btn-info');
-			if(!$('#tl-0').hasClass('btn-success'))
-				$('#tl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
-			streaming.send({message: { request: "configure", temporal_layer: 1 }});
+			if(!$('#m-'+mid+'-tl-2').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-info').addClass('btn-primary');
+			$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-info').addClass('btn-info');
+			if(!$('#m-'+mid+'-tl-0').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			streaming.send({message: { request: "configure", mid: mid, temporal_layer: 1 }});
 		});
-	$('#tl-2').removeClass('btn-primary btn-success').addClass('btn-primary')
+	$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-success').addClass('btn-primary')
 		.unbind('click').click(function() {
+			var mid = $(this).attr('id').split('m-')[1].split('-tl')[0];
 			toastr.info("Capping SVC temporal layer, wait for it... (highest FPS)", null, {timeOut: 2000});
-			$('#tl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
-			if(!$('#tl-1').hasClass('btn-success'))
-				$('#tl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
-			if(!$('#tl-0').hasClass('btn-success'))
-				$('#tl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
-			streaming.send({message: { request: "configure", temporal_layer: 2 }});
+			$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-info');
+			if(!$('#m-'+mid+'-tl-1').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-info').addClass('btn-primary');
+			if(!$('#m-'+mid+'-tl-0').hasClass('btn-success'))
+				$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-info').addClass('btn-primary');
+			streaming.send({message: { request: "configure", mid: mid, temporal_layer: 2 }});
 		});
 }
 
-function updateSvcButtons(spatial, temporal) {
+function updateSvcButtons(mid, spatial, temporal) {
 	// Check the spatial layer
 	if(spatial === 0) {
 		toastr.success("Switched SVC spatial layer! (lower resolution)", null, {timeOut: 2000});
-		$('#sl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#sl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
 	} else if(spatial === 1) {
 		toastr.success("Switched SVC spatial layer! (normal resolution)", null, {timeOut: 2000});
-		$('#sl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
-		$('#sl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-sl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-sl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
 	}
 	// Check the temporal layer
 	if(temporal === 0) {
 		toastr.success("Capped SVC temporal layer! (lowest FPS)", null, {timeOut: 2000});
-		$('#tl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#tl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#tl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
 	} else if(temporal === 1) {
 		toastr.success("Capped SVC temporal layer! (medium FPS)", null, {timeOut: 2000});
-		$('#tl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#tl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
-		$('#tl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
 	} else if(temporal === 2) {
 		toastr.success("Capped SVC temporal layer! (highest FPS)", null, {timeOut: 2000});
-		$('#tl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
-		$('#tl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
-		$('#tl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-2').removeClass('btn-primary btn-info btn-success').addClass('btn-success');
+		$('#m-'+mid+'-tl-1').removeClass('btn-primary btn-success').addClass('btn-primary');
+		$('#m-'+mid+'-tl-0').removeClass('btn-primary btn-success').addClass('btn-primary');
 	}
 }
diff --git a/html/videocalltest.js b/html/videocalltest.js
index 3d3f797bab..e4b5441511 100644
--- a/html/videocalltest.js
+++ b/html/videocalltest.js
@@ -52,6 +52,8 @@ var janus = null;
 var videocall = null;
 var opaqueId = "videocalltest-"+Janus.randomString(12);
 
+var localTracks = {}, localVideos = 0,
+	remoteTracks = {}, remoteVideos = 0;
 var bitrateTimer = null;
 var spinner = null;
 
@@ -125,13 +127,20 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
-								mediaState: function(medium, on) {
-									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium);
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
 								},
 								webrtcState: function(on) {
 									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
 									$("#videoleft").parent().unblock();
 								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message :::");
 									Janus.debug(msg);
@@ -310,14 +319,73 @@ $(document).ready(function() {
 										bitrateTimer = null;
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									Janus.debug(stream);
-									$('#videos').removeClass('hide').show();
-									if($('#myvideo').length === 0)
-										$('#videoleft').append('<video class="rounded centered" id="myvideo" width=320 height=240 autoplay playsinline muted="muted"/>');
-									Janus.attachMediaStream($('#myvideo').get(0), stream);
-									$("#myvideo").get(0).muted = "muted";
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoleft .no-video-container').length === 0) {
+													$('#videoleft').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
+									if($('#videoleft video').length === 0) {
+										$('#videos').removeClass('hide').show();
+									}
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoleft .no-video-container').length === 0) {
+												$('#videoleft').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videoleft .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										$('#videoleft').append('<video class="rounded centered" id="myvideo' + trackId + '" width=320 height=240 autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
 									if(videocall.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											videocall.webrtcStuff.pc.iceConnectionState !== "connected") {
 										$("#videoleft").parent().block({
@@ -328,75 +396,99 @@ $(document).ready(function() {
 												color: 'white'
 											}
 										});
-										// No remote video yet
-										$('#videoright').append('<video class="rounded centered" id="waitingvideo" width=320 height=240 />');
-										if(spinner == null) {
-											var target = document.getElementById('videoright');
-											spinner = new Spinner({top:100}).spin(target);
-										} else {
-											spinner.spin();
-										}
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No webcam
-										$('#myvideo').hide();
-										if($('#videoleft .no-video-container').length === 0) {
-											$('#videoleft').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
+								},
+								onremotetrack: function(track, mid, on) {
+									Janus.debug(" ::: Got a remote track event :::");
+									Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = remoteTracks[mid];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
 										}
-									} else {
-										$('#videoleft .no-video-container').remove();
-										$('#myvideo').removeClass('hide').show();
+										$('#peervideo' + mid).remove();
+										if(track.kind === "video") {
+											remoteVideos--;
+											if(remoteVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videoright .no-video-container').length === 0) {
+													$('#videoright').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No remote video available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete remoteTracks[mid];
+										return;
 									}
-								},
-								onremotestream: function(stream) {
-									Janus.debug(" ::: Got a remote stream :::");
-									Janus.debug(stream);
+									// If we're here, a new track was added
 									var addButtons = false;
-									if($('#remotevideo').length === 0) {
+									if($('#videoright audio').length === 0 && $('#videoright video').length === 0) {
 										addButtons = true;
-										$('#videoright').append('<video class="rounded centered hide" id="remotevideo" width=320 height=240 autoplay playsinline/>');
-										// Show the video, hide the spinner and show the resolution when we get a playing event
-										$("#remotevideo").bind("playing", function () {
-											$('#waitingvideo').remove();
-											if(this.videoWidth)
-												$('#remotevideo').removeClass('hide').show();
-											if(spinner !== null && spinner !== undefined)
-												spinner.stop();
-											spinner = null;
-											var width = this.videoWidth;
-											var height = this.videoHeight;
-											$('#curres').removeClass('hide').text(width+'x'+height).show();
-										});
-										$('#callee').removeClass('hide').html(yourusername).show();
+										$('#videos').removeClass('hide').show();
 									}
-									Janus.attachMediaStream($('#remotevideo').get(0), stream);
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No remote video
-										$('#remotevideo').hide();
-										if($('#videoright .no-video-container').length === 0) {
-											$('#videoright').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No remote video available</span>' +
-												'</div>');
+									if(track.kind === "audio") {
+										// New audio track: create a stream out of it, and use a hidden <audio> element
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote audio stream:", stream);
+										$('#videoright').append('<audio class="hide" id="peervideo' + mid + '" autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										if(remoteVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videoright .no-video-container').length === 0) {
+												$('#videoright').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
 										}
 									} else {
+										// New video track: create a stream out of it
+										remoteVideos++;
 										$('#videoright .no-video-container').remove();
-										$('#remotevideo').removeClass('hide').show();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										remoteTracks[mid] = stream;
+										Janus.log("Created remote video stream:", stream);
+										$('#videoright').append('<video class="rounded centered" id="peervideo' + mid + '" width=320 height=240 autoplay playsinline/>');
+										Janus.attachMediaStream($('#peervideo' + mid).get(0), stream);
+										// Note: we'll need this for additional videos too
+										if(!bitrateTimer) {
+											$('#curbitrate').removeClass('hide').show();
+											bitrateTimer = setInterval(function() {
+												if(!$("#peervideo" + mid).get(0))
+													return;
+												// Display updated bitrate, if supported
+												var bitrate = videocall.getBitrate();
+												//~ Janus.debug("Current bitrate is " + videocall.getBitrate());
+												$('#curbitrate').text(bitrate);
+												// Check if the resolution changed too
+												var width = $("#peervideo" + mid).get(0).videoWidth;
+												var height = $("#peervideo" + mid).get(0).videoHeight;
+												if(width > 0 && height > 0)
+													$('#curres').removeClass('hide').text(width+'x'+height).show();
+											}, 1000);
+										}
 									}
 									if(!addButtons)
 										return;
 									// Enable audio/video buttons and bitrate limiter
 									audioenabled = true;
 									videoenabled = true;
-									$('#toggleaudio').html("Disable audio").removeClass("btn-success").addClass("btn-danger")
-											.unbind('click').removeAttr('disabled').click(
+									$('#toggleaudio').removeAttr('disabled').click(
 										function() {
 											audioenabled = !audioenabled;
 											if(audioenabled)
@@ -405,19 +497,17 @@ $(document).ready(function() {
 												$('#toggleaudio').html("Enable audio").removeClass("btn-danger").addClass("btn-success");
 											videocall.send({"message": { "request": "set", "audio": audioenabled }});
 										});
-									$('#togglevideo').html("Disable video").removeClass("btn-success").addClass("btn-danger")
-											.unbind('click').removeAttr('disabled').click(
+									$('#togglevideo').removeAttr('disabled').click(
 										function() {
 											videoenabled = !videoenabled;
 											if(videoenabled)
 												$('#togglevideo').html("Disable video").removeClass("btn-success").addClass("btn-danger");
 											else
 												$('#togglevideo').html("Enable video").removeClass("btn-danger").addClass("btn-success");
-											videocall.send({"message": { "request": "set", "video": videoenabled }});
+											videocall.send({"message": { "request": "set", "video": audioenabled }});
 										});
 									$('#toggleaudio').parent().removeClass('hide').show();
-									$('#bitrateset').html("Bandwidth");
-									$('#bitrate a').unbind('click').removeAttr('disabled').click(function() {
+									$('#bitrate a').removeAttr('disabled').click(function() {
 										var id = $(this).attr("id");
 										var bitrate = parseInt(id)*1000;
 										if(bitrate === 0) {
@@ -425,24 +515,10 @@ $(document).ready(function() {
 										} else {
 											Janus.log("Capping bandwidth to " + bitrate + " via REMB");
 										}
-										$('#bitrateset').html($(this).html()).parent().removeClass('open');
+										$('#bitrateset').html($(this).html() + '<span class="caret"></span>').parent().removeClass('open');
 										videocall.send({"message": { "request": "set", "bitrate": bitrate }});
 										return false;
 									});
-									if(Janus.webRTCAdapter.browserDetails.browser === "chrome" || Janus.webRTCAdapter.browserDetails.browser === "firefox" ||
-											Janus.webRTCAdapter.browserDetails.browser === "safari") {
-										$('#curbitrate').removeClass('hide').show();
-										bitrateTimer = setInterval(function() {
-											// Display updated bitrate, if supported
-											var bitrate = videocall.getBitrate();
-											$('#curbitrate').text(bitrate);
-											// Check if the resolution changed too
-											var width = $("#remotevideo").get(0).videoWidth;
-											var height = $("#remotevideo").get(0).videoHeight;
-											if(width > 0 && height > 0)
-												$('#curres').removeClass('hide').text(width+'x'+height).show();
-										}, 1000);
-									}
 								},
 								ondataopen: function(data) {
 									Janus.log("The DataChannel is available!");
@@ -455,10 +531,8 @@ $(document).ready(function() {
 								},
 								oncleanup: function() {
 									Janus.log(" ::: Got a cleanup notification :::");
-									$('#myvideo').remove();
-									$('#remotevideo').remove();
-									$("#videoleft").parent().unblock();
-									$('.no-video-container').remove();
+									$("#videoleft").empty().parent().unblock();
+									$('#videoright').empty();
 									$('#callee').empty().hide();
 									yourusername = null;
 									$('#curbitrate').hide();
@@ -472,7 +546,6 @@ $(document).ready(function() {
 									if(bitrateTimer !== null && bitrateTimer !== null)
 										clearInterval(bitrateTimer);
 									bitrateTimer = null;
-									$('#waitingvideo').remove();
 									$('#videos').hide();
 									simulcastStarted = false;
 									$('#simulcast').remove();
@@ -480,6 +553,10 @@ $(document).ready(function() {
 									$('#call').removeAttr('disabled').html('Call')
 										.removeClass("btn-danger").addClass("btn-success")
 										.unbind('click').click(doCall);
+									localTracks = {};
+									localVideos = 0;
+									remoteTracks = {};
+									remoteVideos = 0;
 								}
 							});
 					},
diff --git a/html/videoroomtest.html b/html/videoroomtest.html
index e88f34b9ae..c785a6f94e 100644
--- a/html/videoroomtest.html
+++ b/html/videoroomtest.html
@@ -54,6 +54,12 @@ <h3>Demo details</h3>
 						No mixing is involved: all media are just relayed in a publisher/subscriber
 						approach. This means that the plugin acts as a SFU (Selective Forwarding Unit)
 						rather than an MCU (Multipoint Control Unit).</p>
+						<div class="alert alert-info">Notice that this is the <b>original</b> VideoRoom
+						demo, and uses a different PeerConnections per each subscription: if
+						you want to test the new multistream support, instead, try the
+						<a href="mvideoroomtest.html">multistream VideoRoom demo</a>
+						instead. The two demos are interoperable, if you want to see how
+						different subscription mechanisms are used on the same sources.</div>
 						<p>If you're interested in testing how simulcasting can be used within
 						the context of a videoconferencing application, just pass the
 						<code>?simulcast=true</code> query string to the url of this page and
diff --git a/html/videoroomtest.js b/html/videoroomtest.js
index d761048725..d22e7617d3 100644
--- a/html/videoroomtest.js
+++ b/html/videoroomtest.js
@@ -59,7 +59,8 @@ var mystream = null;
 // We use this other ID just to map our subscriptions to us
 var mypvtid = null;
 
-var feeds = [];
+var localTracks = {}, localVideos = 0;
+var feeds = [], feedStreams = {};
 var bitrateTimer = [];
 
 var doSimulcast = (getQueryStringValue("simulcast") === "yes" || getQueryStringValue("simulcast") === "true");
@@ -125,8 +126,11 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
-								mediaState: function(medium, on) {
-									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium);
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
 								},
 								webrtcState: function(on) {
 									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
@@ -149,6 +153,10 @@ $(document).ready(function() {
 										return false;
 									});
 								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message (publisher) :::");
 									Janus.debug(msg);
@@ -169,10 +177,15 @@ $(document).ready(function() {
 												for(var f in list) {
 													var id = list[f]["id"];
 													var display = list[f]["display"];
-													var audio = list[f]["audio_codec"];
-													var video = list[f]["video_codec"];
-													Janus.debug("  >> [" + id + "] " + display + " (audio: " + audio + ", video: " + video + ")");
-													newRemoteFeed(id, display, audio, video);
+													var streams = list[f]["streams"];
+													for(var i in streams) {
+														var stream = streams[i];
+														stream["id"] = id;
+														stream["display"] = display;
+													}
+													feedStreams[id] = streams;
+													Janus.debug("  >> [" + id + "] " + display + ":", streams);
+													newRemoteFeed(id, display, streams);
 												}
 											}
 										} else if(event === "destroyed") {
@@ -182,18 +195,31 @@ $(document).ready(function() {
 												window.location.reload();
 											});
 										} else if(event === "event") {
-											// Any new feed to attach to?
-											if(msg["publishers"] !== undefined && msg["publishers"] !== null) {
+											// Any info on our streams or a new feed to attach to?
+											if(msg["streams"] !== undefined && msg["streams"] !== null) {
+												var streams = msg["streams"];
+												for(var i in streams) {
+													var stream = streams[i];
+													stream["id"] = myid;
+													stream["display"] = myusername;
+												}
+												feedStreams[myid] = streams;
+											} else if(msg["publishers"] !== undefined && msg["publishers"] !== null) {
 												var list = msg["publishers"];
 												Janus.debug("Got a list of available publishers/feeds:");
 												Janus.debug(list);
 												for(var f in list) {
 													var id = list[f]["id"];
 													var display = list[f]["display"];
-													var audio = list[f]["audio_codec"];
-													var video = list[f]["video_codec"];
-													Janus.debug("  >> [" + id + "] " + display + " (audio: " + audio + ", video: " + video + ")");
-													newRemoteFeed(id, display, audio, video);
+													var streams = list[f]["streams"];
+													for(var i in streams) {
+														var stream = streams[i];
+														stream["id"] = id;
+														stream["display"] = display;
+													}
+													feedStreams[id] = streams;
+													Janus.debug("  >> [" + id + "] " + display + ":", streams);
+													newRemoteFeed(id, display, streams);
 												}
 											} else if(msg["leaving"] !== undefined && msg["leaving"] !== null) {
 												// One of the publishers has gone away?
@@ -213,6 +239,7 @@ $(document).ready(function() {
 													feeds[remoteFeed.rfindex] = null;
 													remoteFeed.detach();
 												}
+												delete feedStreams[leaving];
 											} else if(msg["unpublished"] !== undefined && msg["unpublished"] !== null) {
 												// One of the publishers has unpublished?
 												var unpublished = msg["unpublished"];
@@ -276,14 +303,49 @@ $(document).ready(function() {
 										}
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									mystream = stream;
-									Janus.debug(stream);
-									$('#videojoin').hide();
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videolocal .no-video-container').length === 0) {
+													$('#videolocal').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
 									$('#videos').removeClass('hide').show();
-									if($('#myvideo').length === 0) {
-										$('#videolocal').append('<video class="rounded centered" id="myvideo" width="100%" height="100%" autoplay playsinline muted="muted"/>');
+									if($('#mute').length === 0) {
 										// Add a 'mute' button
 										$('#videolocal').append('<button class="btn btn-warning btn-xs" id="mute" style="position: absolute; bottom: 0px; left: 0px; margin: 15px;">Mute</button>');
 										$('#mute').click(toggleMute);
@@ -291,9 +353,31 @@ $(document).ready(function() {
 										$('#videolocal').append('<button class="btn btn-warning btn-xs" id="unpublish" style="position: absolute; bottom: 0px; right: 0px; margin: 15px;">Unpublish</button>');
 										$('#unpublish').click(unpublishOwnFeed);
 									}
-									$('#publisher').removeClass('hide').html(myusername).show();
-									Janus.attachMediaStream($('#myvideo').get(0), stream);
-									$("#myvideo").get(0).muted = "muted";
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videolocal .no-video-container').length === 0) {
+												$('#videolocal').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videolocal .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										Janus.log(stream.getTracks());
+										Janus.log(stream.getVideoTracks());
+										$('#videolocal').append('<video class="rounded centered" id="myvideo' + trackId + '" width=100% autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
 									if(sfutest.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											sfutest.webrtcStuff.pc.iceConnectionState !== "connected") {
 										$("#videolocal").parent().parent().block({
@@ -305,33 +389,21 @@ $(document).ready(function() {
 											}
 										});
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No webcam
-										$('#myvideo').hide();
-										if($('#videolocal .no-video-container').length === 0) {
-											$('#videolocal').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
-										}
-									} else {
-										$('#videolocal .no-video-container').remove();
-										$('#myvideo').removeClass('hide').show();
-									}
 								},
-								onremotestream: function(stream) {
+								onremotetrack: function(track, mid, on) {
 									// The publisher stream is sendonly, we don't expect anything here
 								},
 								oncleanup: function() {
 									Janus.log(" ::: Got a cleanup notification: we are unpublished now :::");
 									mystream = null;
+									delete feedStreams[myid];
 									$('#videolocal').html('<button id="publish" class="btn btn-primary">Publish</button>');
 									$('#publish').click(function() { publishOwnFeed(true); });
 									$("#videolocal").parent().parent().unblock();
 									$('#bitrate').parent().parent().addClass('hide');
 									$('#bitrate a').unbind('click');
+									localTracks = {};
+									localVideos = 0;
 								}
 							});
 					},
@@ -449,39 +521,66 @@ function unpublishOwnFeed() {
 	sfutest.send({"message": unpublish});
 }
 
-function newRemoteFeed(id, display, audio, video) {
+function newRemoteFeed(id, display, streams) {
 	// A new feed has been published, create a new plugin handle and attach to it as a subscriber
 	var remoteFeed = null;
+	if(!streams)
+		streams = feedStreams[id];
 	janus.attach(
 		{
 			plugin: "janus.plugin.videoroom",
 			opaqueId: opaqueId,
 			success: function(pluginHandle) {
 				remoteFeed = pluginHandle;
+				remoteFeed.remoteTracks = {};
+				remoteFeed.remoteVideos = 0;
 				remoteFeed.simulcastStarted = false;
 				Janus.log("Plugin attached! (" + remoteFeed.getPlugin() + ", id=" + remoteFeed.getId() + ")");
 				Janus.log("  -- This is a subscriber");
-				// We wait for the plugin to send us an offer
-				var subscribe = { "request": "join", "room": myroom, "ptype": "subscriber", "feed": id, "private_id": mypvtid };
-				// In case you don't want to receive audio, video or data, even if the
-				// publisher is sending them, set the 'offer_audio', 'offer_video' or
-				// 'offer_data' properties to false (they're true by default), e.g.:
-				// 		subscribe["offer_video"] = false;
-				// For example, if the publisher is VP8 and this is Safari, let's avoid video
-				if(Janus.webRTCAdapter.browserDetails.browser === "safari" &&
-						(video === "vp9" || (video === "vp8" && !Janus.safariVp8))) {
-					if(video)
-						video = video.toUpperCase()
-					toastr.warning("Publisher is using " + video + ", but Safari doesn't support it: disabling video");
-					subscribe["offer_video"] = false;
+				// Prepare the streams to subscribe to, as an array: we have the list of
+				// streams the feed is publishing, so we can choose what to pick or skip
+				var subscription = [];
+				for(var i in streams) {
+					var stream = streams[i];
+					// If the publisher is VP8/VP9 and this is an older Safari, let's avoid video
+					if(stream.type === "video" && Janus.webRTCAdapter.browserDetails.browser === "safari" &&
+							(stream.codec === "vp9" || (stream.codec === "vp8" && !Janus.safariVp8))) {
+						toastr.warning("Publisher is using " + stream.codec.toUpperCase +
+							", but Safari doesn't support it: disabling video stream #" + stream.mindex);
+						continue;
+					}
+					subscription.push({
+						feed: stream.id,	// This is mandatory
+						mid: stream.mid		// This is optional (all streams, if missing)
+					});
+					// FIXME Right now, this is always the same feed: in the future, it won't
+					remoteFeed.rfid = stream.id;
+					remoteFeed.rfdisplay = stream.display;
 				}
-				remoteFeed.videoCodec = video;
+				// We wait for the plugin to send us an offer
+				var subscribe = {
+					request: "join",
+					room: myroom,
+					ptype: "subscriber",
+					streams: subscription,
+					private_id: mypvtid
+				};
 				remoteFeed.send({"message": subscribe});
 			},
 			error: function(error) {
 				Janus.error("  -- Error attaching plugin...", error);
 				bootbox.alert("Error attaching plugin... " + error);
 			},
+			iceState: function(state) {
+				Janus.log("ICE state (feed #" + remoteFeed.rfindex + ") changed to " + state);
+			},
+			webrtcState: function(on) {
+				Janus.log("Janus says this WebRTC PeerConnection (feed #" + remoteFeed.rfindex + ") is " + (on ? "up" : "down") + " now");
+			},
+			slowLink: function(uplink, lost, mid) {
+				Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+					" packets on mid " + mid + " (" + lost + " lost packets)");
+			},
 			onmessage: function(msg, jsep) {
 				Janus.debug(" ::: Got a message (subscriber) :::");
 				Janus.debug(msg);
@@ -499,15 +598,13 @@ function newRemoteFeed(id, display, audio, video) {
 								break;
 							}
 						}
-						remoteFeed.rfid = msg["id"];
-						remoteFeed.rfdisplay = msg["display"];
 						if(remoteFeed.spinner === undefined || remoteFeed.spinner === null) {
 							var target = document.getElementById('videoremote'+remoteFeed.rfindex);
 							remoteFeed.spinner = new Spinner({top:100}).spin(target);
 						} else {
 							remoteFeed.spinner.spin();
 						}
-						Janus.log("Successfully attached to feed " + remoteFeed.rfid + " (" + remoteFeed.rfdisplay + ") in room " + msg["room"]);
+						Janus.log("Successfully attached to feed in room " + msg["room"]);
 						$('#remote'+remoteFeed.rfindex).removeClass('hide').html(remoteFeed.rfdisplay).show();
 					} else if(event === "event") {
 						// Check if we got an event on a simulcast-related event from this publisher
@@ -549,75 +646,96 @@ function newRemoteFeed(id, display, audio, video) {
 						});
 				}
 			},
-			webrtcState: function(on) {
-				Janus.log("Janus says this WebRTC PeerConnection (feed #" + remoteFeed.rfindex + ") is " + (on ? "up" : "down") + " now");
-			},
-			onlocalstream: function(stream) {
+			onlocaltrack: function(track, on) {
 				// The subscriber stream is recvonly, we don't expect anything here
 			},
-			onremotestream: function(stream) {
-				Janus.debug("Remote feed #" + remoteFeed.rfindex);
-				var addButtons = false;
-				if($('#remotevideo'+remoteFeed.rfindex).length === 0) {
-					addButtons = true;
-					// No remote video yet
-					$('#videoremote'+remoteFeed.rfindex).append('<video class="rounded centered" id="waitingvideo' + remoteFeed.rfindex + '" width=320 height=240 />');
-					$('#videoremote'+remoteFeed.rfindex).append('<video class="rounded centered relative hide" id="remotevideo' + remoteFeed.rfindex + '" width="100%" height="100%" autoplay playsinline/>');
+			onremotetrack: function(track, mid, on) {
+				Janus.debug(" ::: Got a remote track event ::: (Remote feed #" + remoteFeed.rfindex + ")");
+				Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+				if(!on) {
+					// Track removed, get rid of the stream and the rendering
+					var stream = remoteFeed.remoteTracks[mid];
+					if(stream) {
+						try {
+							var tracks = stream.getTracks();
+							for(var i in tracks) {
+								var mst = tracks[i];
+								if(mst !== null && mst !== undefined)
+									mst.stop();
+							}
+						} catch(e) {}
+					}
+					$('#remotevideo'+remoteFeed.rfindex + '-' + mid).remove();
+					if(track.kind === "video") {
+						remoteFeed.remoteVideos--;
+						if(remoteFeed.remoteVideos === 0) {
+							// No video, at least for now: show a placeholder
+							if($('#videoremote'+remoteFeed.rfindex + ' .no-video-container').length === 0) {
+								$('#videoremote'+remoteFeed.rfindex).append(
+									'<div class="no-video-container">' +
+										'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+										'<span class="no-video-text">No remote video available</span>' +
+									'</div>');
+							}
+						}
+					}
+					delete remoteFeed.remoteTracks[mid];
+					return;
+				}
+				// If we're here, a new track was added
+				if(remoteFeed.spinner !== undefined && remoteFeed.spinner !== null) {
+					remoteFeed.spinner.stop();
+					remoteFeed.spinner = null;
+				}
+				if($('#remotevideo' + remoteFeed.rfindex + '-' + mid).length > 0)
+					return;
+				if(track.kind === "audio") {
+					// New audio track: create a stream out of it, and use a hidden <audio> element
+					stream = new MediaStream();
+					stream.addTrack(track.clone());
+					remoteFeed.remoteTracks[mid] = stream;
+					Janus.log("Created remote audio stream:", stream);
+					$('#videoremote'+remoteFeed.rfindex).append('<audio class="hide" id="remotevideo' + remoteFeed.rfindex + '-' + mid + '" autoplay playsinline/>');
+					Janus.attachMediaStream($('#remotevideo' + remoteFeed.rfindex + '-' + mid).get(0), stream);
+					if(remoteFeed.remoteVideos === 0) {
+						// No video, at least for now: show a placeholder
+						if($('#videoremote'+remoteFeed.rfindex + ' .no-video-container').length === 0) {
+							$('#videoremote'+remoteFeed.rfindex).append(
+								'<div class="no-video-container">' +
+									'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+									'<span class="no-video-text">No remote video available</span>' +
+								'</div>');
+						}
+					}
+				} else {
+					// New video track: create a stream out of it
+					remoteFeed.remoteVideos++;
+					$('#videoremote'+remoteFeed.rfindex + ' .no-video-container').remove();
+					stream = new MediaStream();
+					stream.addTrack(track.clone());
+					remoteFeed.remoteTracks[mid] = stream;
+					Janus.log("Created remote video stream:", stream);
+					$('#videoremote'+remoteFeed.rfindex).append('<video class="rounded centered" id="remotevideo' + remoteFeed.rfindex + '-' + mid + '" width=100% autoplay playsinline/>');
 					$('#videoremote'+remoteFeed.rfindex).append(
 						'<span class="label label-primary hide" id="curres'+remoteFeed.rfindex+'" style="position: absolute; bottom: 0px; left: 0px; margin: 15px;"></span>' +
 						'<span class="label label-info hide" id="curbitrate'+remoteFeed.rfindex+'" style="position: absolute; bottom: 0px; right: 0px; margin: 15px;"></span>');
-					// Show the video, hide the spinner and show the resolution when we get a playing event
-					$("#remotevideo"+remoteFeed.rfindex).bind("playing", function () {
-						if(remoteFeed.spinner !== undefined && remoteFeed.spinner !== null)
-							remoteFeed.spinner.stop();
-						remoteFeed.spinner = null;
-						$('#waitingvideo'+remoteFeed.rfindex).remove();
-						if(this.videoWidth)
-							$('#remotevideo'+remoteFeed.rfindex).removeClass('hide').show();
-						var width = this.videoWidth;
-						var height = this.videoHeight;
-						$('#curres'+remoteFeed.rfindex).removeClass('hide').text(width+'x'+height).show();
-						if(Janus.webRTCAdapter.browserDetails.browser === "firefox") {
-							// Firefox Stable has a bug: width and height are not immediately available after a playing
-							setTimeout(function() {
-								var width = $("#remotevideo"+remoteFeed.rfindex).get(0).videoWidth;
-								var height = $("#remotevideo"+remoteFeed.rfindex).get(0).videoHeight;
+					Janus.attachMediaStream($('#remotevideo' + remoteFeed.rfindex + '-' + mid).get(0), stream);
+					// Note: we'll need this for additional videos too
+					if(!bitrateTimer[remoteFeed.rfindex]) {
+						$('#curbitrate'+remoteFeed.rfindex).removeClass('hide').show();
+						bitrateTimer[remoteFeed.rfindex] = setInterval(function() {
+							if(!$("#videoremote" + remoteFeed.rfindex + ' video').get(0))
+								return;
+							// Display updated bitrate, if supported
+							var bitrate = remoteFeed.getBitrate();
+							$('#curbitrate'+remoteFeed.rfindex).text(bitrate);
+							// Check if the resolution changed too
+							var width = $("#videoremote" + remoteFeed.rfindex + ' video').get(0).videoWidth;
+							var height = $("#videoremote" + remoteFeed.rfindex + ' video').get(0).videoHeight;
+							if(width > 0 && height > 0)
 								$('#curres'+remoteFeed.rfindex).removeClass('hide').text(width+'x'+height).show();
-							}, 2000);
-						}
-					});
-				}
-				Janus.attachMediaStream($('#remotevideo'+remoteFeed.rfindex).get(0), stream);
-				var videoTracks = stream.getVideoTracks();
-				if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-					// No remote video
-					$('#remotevideo'+remoteFeed.rfindex).hide();
-					if($('#videoremote'+remoteFeed.rfindex + ' .no-video-container').length === 0) {
-						$('#videoremote'+remoteFeed.rfindex).append(
-							'<div class="no-video-container">' +
-								'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-								'<span class="no-video-text">No remote video available</span>' +
-							'</div>');
+						}, 1000);
 					}
-				} else {
-					$('#videoremote'+remoteFeed.rfindex+ ' .no-video-container').remove();
-					$('#remotevideo'+remoteFeed.rfindex).removeClass('hide').show();
-				}
-				if(!addButtons)
-					return;
-				if(Janus.webRTCAdapter.browserDetails.browser === "chrome" || Janus.webRTCAdapter.browserDetails.browser === "firefox" ||
-						Janus.webRTCAdapter.browserDetails.browser === "safari") {
-					$('#curbitrate'+remoteFeed.rfindex).removeClass('hide').show();
-					bitrateTimer[remoteFeed.rfindex] = setInterval(function() {
-						// Display updated bitrate, if supported
-						var bitrate = remoteFeed.getBitrate();
-						$('#curbitrate'+remoteFeed.rfindex).text(bitrate);
-						// Check if the resolution changed too
-						var width = $("#remotevideo"+remoteFeed.rfindex).get(0).videoWidth;
-						var height = $("#remotevideo"+remoteFeed.rfindex).get(0).videoHeight;
-						if(width > 0 && height > 0)
-							$('#curres'+remoteFeed.rfindex).removeClass('hide').text(width+'x'+height).show();
-					}, 1000);
 				}
 			},
 			oncleanup: function() {
@@ -635,6 +753,8 @@ function newRemoteFeed(id, display, audio, video) {
 				bitrateTimer[remoteFeed.rfindex] = null;
 				remoteFeed.simulcastStarted = false;
 				$('#simulcast'+remoteFeed.rfindex).remove();
+				remoteFeed.remoteTracks = {};
+				remoteFeed.remoteVideos = 0;
 			}
 		});
 }
diff --git a/html/voicemailtest.js b/html/voicemailtest.js
index 153b5bf528..ad47514114 100644
--- a/html/voicemailtest.js
+++ b/html/voicemailtest.js
@@ -104,7 +104,7 @@ $(document).ready(function() {
 									Janus.debug("Consent dialog should be " + (on ? "on" : "off") + " now");
 									if(on) {
 										// Darken screen and show hint
-										$.blockUI({ 
+										$.blockUI({
 											message: '<div><img src="up_arrow.png"/></div>',
 											css: {
 												border: 'none',
@@ -171,10 +171,10 @@ $(document).ready(function() {
 										vmailtest.handleRemoteJsep({jsep: jsep});
 									}
 								},
-								onlocalstream: function(stream) {
+								onlocaltrack: function(track, on) {
 									// We're not going to attach the local audio stream
 								},
-								onremotestream: function(stream) {
+								onremotetrack: function(track, mid, on) {
 									// We're not going to receive anything from the plugin
 								},
 								oncleanup: function() {
diff --git a/html/vp9svctest.js b/html/vp9svctest.js
index c115d9669d..c07d256a14 100644
--- a/html/vp9svctest.js
+++ b/html/vp9svctest.js
@@ -59,6 +59,7 @@ var mystream = null;
 // We use this other ID just to map our subscriptions to us
 var mypvtid = null;
 
+var localTracks = {}, localVideos = 0;
 var feeds = [];
 var bitrateTimer = [];
 
@@ -123,8 +124,11 @@ $(document).ready(function() {
 										$.unblockUI();
 									}
 								},
-								mediaState: function(medium, on) {
-									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium);
+								iceState: function(state) {
+									Janus.log("ICE state changed to " + state);
+								},
+								mediaState: function(medium, on, mid) {
+									Janus.log("Janus " + (on ? "started" : "stopped") + " receiving our " + medium + " (mid=" + mid + ")");
 								},
 								webrtcState: function(on) {
 									Janus.log("Janus says our WebRTC PeerConnection is " + (on ? "up" : "down") + " now");
@@ -147,6 +151,10 @@ $(document).ready(function() {
 										return false;
 									});
 								},
+								slowLink: function(uplink, lost, mid) {
+									Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+										" packets on mid " + mid + " (" + lost + " lost packets)");
+								},
 								onmessage: function(msg, jsep) {
 									Janus.debug(" ::: Got a message (publisher) :::");
 									Janus.debug(msg);
@@ -251,14 +259,49 @@ $(document).ready(function() {
 										sfutest.handleRemoteJsep({jsep: jsep});
 									}
 								},
-								onlocalstream: function(stream) {
-									Janus.debug(" ::: Got a local stream :::");
-									mystream = stream;
-									Janus.debug(stream);
-									$('#videojoin').hide();
+								onlocaltrack: function(track, on) {
+									Janus.debug(" ::: Got a local track event :::");
+									Janus.debug("Local track " + (on ? "added" : "removed") + ":", track);
+									// We use the track ID as name of the element, but it may contain invalid characters
+									var trackId = track.id.replace(/[{}]/g, "");
+									if(!on) {
+										// Track removed, get rid of the stream and the rendering
+										var stream = localTracks[trackId];
+										if(stream) {
+											try {
+												var tracks = stream.getTracks();
+												for(var i in tracks) {
+													var mst = tracks[i];
+													if(mst !== null && mst !== undefined)
+														mst.stop();
+												}
+											} catch(e) {}
+										}
+										if(track.kind === "video") {
+											$('#myvideo' + trackId).remove();
+											localVideos--;
+											if(localVideos === 0) {
+												// No video, at least for now: show a placeholder
+												if($('#videolocal .no-video-container').length === 0) {
+													$('#videolocal').append(
+														'<div class="no-video-container">' +
+															'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+															'<span class="no-video-text">No webcam available</span>' +
+														'</div>');
+												}
+											}
+										}
+										delete localTracks[trackId];
+										return;
+									}
+									// If we're here, a new track was added
+									var stream = localTracks[trackId];
+									if(stream) {
+										// We've been here already
+										return;
+									}
 									$('#videos').removeClass('hide').show();
-									if($('#myvideo').length === 0) {
-										$('#videolocal').append('<video class="rounded centered" id="myvideo" width="100%" height="100%" autoplay playsinline muted="muted"/>');
+									if($('#mute').length === 0) {
 										// Add a 'mute' button
 										$('#videolocal').append('<button class="btn btn-warning btn-xs" id="mute" style="position: absolute; bottom: 0px; left: 0px; margin: 15px;">Mute</button>');
 										$('#mute').click(toggleMute);
@@ -266,9 +309,29 @@ $(document).ready(function() {
 										$('#videolocal').append('<button class="btn btn-warning btn-xs" id="unpublish" style="position: absolute; bottom: 0px; right: 0px; margin: 15px;">Unpublish</button>');
 										$('#unpublish').click(unpublishOwnFeed);
 									}
-									$('#publisher').removeClass('hide').html(myusername).show();
-									Janus.attachMediaStream($('#myvideo').get(0), stream);
-									$("#myvideo").get(0).muted = "muted";
+									if(track.kind === "audio") {
+										// We ignore local audio tracks, they'd generate echo anyway
+										if(localVideos === 0) {
+											// No video, at least for now: show a placeholder
+											if($('#videolocal .no-video-container').length === 0) {
+												$('#videolocal').append(
+													'<div class="no-video-container">' +
+														'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+														'<span class="no-video-text">No webcam available</span>' +
+													'</div>');
+											}
+										}
+									} else {
+										// New video track: create a stream out of it
+										localVideos++;
+										$('#videolocal .no-video-container').remove();
+										stream = new MediaStream();
+										stream.addTrack(track.clone());
+										localTracks[trackId] = stream;
+										Janus.log("Created local stream:", stream);
+										$('#videolocal').append('<video class="rounded centered" id="myvideo' + trackId + '" width=100% autoplay playsinline muted="muted"/>');
+										Janus.attachMediaStream($('#myvideo' + trackId).get(0), stream);
+									}
 									if(sfutest.webrtcStuff.pc.iceConnectionState !== "completed" &&
 											sfutest.webrtcStuff.pc.iceConnectionState !== "connected") {
 										$("#videolocal").parent().parent().block({
@@ -280,23 +343,8 @@ $(document).ready(function() {
 											}
 										});
 									}
-									var videoTracks = stream.getVideoTracks();
-									if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-										// No webcam
-										$('#myvideo').hide();
-										if($('#videolocal .no-video-container').length === 0) {
-											$('#videolocal').append(
-												'<div class="no-video-container">' +
-													'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-													'<span class="no-video-text">No webcam available</span>' +
-												'</div>');
-										}
-									} else {
-										$('#videolocal .no-video-container').remove();
-										$('#myvideo').removeClass('hide').show();
-									}
 								},
-								onremotestream: function(stream) {
+								onremotetrack: function(track, mid, on) {
 									// The publisher stream is sendonly, we don't expect anything here
 								},
 								oncleanup: function() {
@@ -307,6 +355,8 @@ $(document).ready(function() {
 									$("#videolocal").parent().parent().unblock();
 									$('#bitrate').parent().parent().addClass('hide');
 									$('#bitrate a').unbind('click');
+									localTracks = {};
+									localVideos = 0;
 								}
 							});
 					},
@@ -418,6 +468,8 @@ function newRemoteFeed(id, display) {
 			opaqueId: opaqueId,
 			success: function(pluginHandle) {
 				remoteFeed = pluginHandle;
+				remoteFeed.remoteTracks = {};
+				remoteFeed.remoteVideos = 0;
 				Janus.log("Plugin attached! (" + remoteFeed.getPlugin() + ", id=" + remoteFeed.getId() + ")");
 				Janus.log("  -- This is a subscriber");
 				// We wait for the plugin to send us an offer
@@ -435,6 +487,16 @@ function newRemoteFeed(id, display) {
 				Janus.error("  -- Error attaching plugin...", error);
 				bootbox.alert("Error attaching plugin... " + error);
 			},
+			iceState: function(state) {
+				Janus.log("ICE state (feed #" + remoteFeed.rfindex + ") changed to " + state);
+			},
+			webrtcState: function(on) {
+				Janus.log("Janus says this WebRTC PeerConnection (feed #" + remoteFeed.rfindex + ") is " + (on ? "up" : "down") + " now");
+			},
+			slowLink: function(uplink, lost, mid) {
+				Janus.warn("Janus reports problems " + (uplink ? "sending" : "receiving") +
+					" packets on mid " + mid + " (" + lost + " lost packets)");
+			},
 			onmessage: function(msg, jsep) {
 				Janus.debug(" ::: Got a message (listener) :::");
 				Janus.debug(msg);
@@ -502,75 +564,91 @@ function newRemoteFeed(id, display) {
 						});
 				}
 			},
-			webrtcState: function(on) {
-				Janus.log("Janus says this WebRTC PeerConnection (feed #" + remoteFeed.rfindex + ") is " + (on ? "up" : "down") + " now");
-			},
-			onlocalstream: function(stream) {
+			onlocaltrack: function(track, on) {
 				// The subscriber stream is recvonly, we don't expect anything here
 			},
-			onremotestream: function(stream) {
-				Janus.debug("Remote feed #" + remoteFeed.rfindex);
-				var addButtons = false;
-				if($('#remotevideo'+remoteFeed.rfindex).length === 0) {
-					addButtons = true;
-					// No remote video yet
-					$('#videoremote'+remoteFeed.rfindex).append('<video class="rounded centered" id="waitingvideo' + remoteFeed.rfindex + '" width=320 height=240 />');
-					$('#videoremote'+remoteFeed.rfindex).append('<video class="rounded centered relative hide" id="remotevideo' + remoteFeed.rfindex + '" width="100%" height="100%" autoplay playsinline/>');
-					$('#videoremote'+remoteFeed.rfindex).append(
-						'<span class="label label-primary hide" id="curres'+remoteFeed.rfindex+'" style="position: absolute; bottom: 0px; left: 0px; margin: 15px;"></span>' +
-						'<span class="label label-info hide" id="curbitrate'+remoteFeed.rfindex+'" style="position: absolute; bottom: 0px; right: 0px; margin: 15px;"></span>');
-					// Show the video, hide the spinner and show the resolution when we get a playing event
-					$("#remotevideo"+remoteFeed.rfindex).bind("playing", function () {
-						if(remoteFeed.spinner !== undefined && remoteFeed.spinner !== null)
-							remoteFeed.spinner.stop();
-						remoteFeed.spinner = null;
-						$('#waitingvideo'+remoteFeed.rfindex).remove();
-						if(this.videoWidth)
-							$('#remotevideo'+remoteFeed.rfindex).removeClass('hide').show();
-						var width = this.videoWidth;
-						var height = this.videoHeight;
-						$('#curres'+remoteFeed.rfindex).removeClass('hide').text(width+'x'+height).show();
-						if(Janus.webRTCAdapter.browserDetails.browser === "firefox") {
-							// Firefox Stable has a bug: width and height are not immediately available after a playing
-							setTimeout(function() {
-								var width = $("#remotevideo"+remoteFeed.rfindex).get(0).videoWidth;
-								var height = $("#remotevideo"+remoteFeed.rfindex).get(0).videoHeight;
-								$('#curres'+remoteFeed.rfindex).removeClass('hide').text(width+'x'+height).show();
-							}, 2000);
+			onremotetrack: function(track, mid, on) {
+				Janus.debug(" ::: Got a remote track event ::: (Remote feed #" + remoteFeed.rfindex + ")");
+				Janus.debug("Remote track (mid=" + mid + ") " + (on ? "added" : "removed") + ":", track);
+				if(!on) {
+					// Track removed, get rid of the stream and the rendering
+					var stream = remoteFeed.remoteTracks[mid];
+					if(stream) {
+						try {
+							var tracks = stream.getTracks();
+							for(var i in tracks) {
+								var mst = tracks[i];
+								if(mst !== null && mst !== undefined)
+									mst.stop();
+							}
+						} catch(e) {}
+					}
+					$('#remotevideo'+remoteFeed.rfindex + '-' + mid).remove();
+					if(track.kind === "video") {
+						remoteFeed.remoteVideos--;
+						if(remoteFeed.remoteVideos === 0) {
+							// No video, at least for now: show a placeholder
+							if($('#videoremote'+remoteFeed.rfindex + ' .no-video-container').length === 0) {
+								$('#videoremote'+remoteFeed.rfindex).append(
+									'<div class="no-video-container">' +
+										'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+										'<span class="no-video-text">No remote video available</span>' +
+									'</div>');
+							}
 						}
-					});
+					}
+					delete remoteFeed.remoteTracks[mid];
+					return;
+				}
+				// If we're here, a new track was added
+				if(remoteFeed.spinner !== undefined && remoteFeed.spinner !== null) {
+					remoteFeed.spinner.stop();
+					remoteFeed.spinner = null;
 				}
-				Janus.attachMediaStream($('#remotevideo'+remoteFeed.rfindex).get(0), stream);
-				var videoTracks = stream.getVideoTracks();
-				if(videoTracks === null || videoTracks === undefined || videoTracks.length === 0) {
-					// No remote video
-					$('#remotevideo'+remoteFeed.rfindex).hide();
-					if($('#remotevideo'+remoteFeed.rfindex + ' .no-video-container').length === 0) {
-						$('#remotevideo'+remoteFeed.rfindex).append(
-							'<div class="no-video-container">' +
-								'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
-								'<span class="no-video-text">No remote video available</span>' +
-							'</div>');
+				if(track.kind === "audio") {
+					// New audio track: create a stream out of it, and use a hidden <audio> element
+					stream = new MediaStream();
+					stream.addTrack(track.clone());
+					remoteFeed.remoteTracks[mid] = stream;
+					Janus.log("Created remote audio stream:", stream);
+					$('#videoremote'+remoteFeed.rfindex).append('<audio class="hide" id="remotevideo' + remoteFeed.rfindex + '-' + mid + '" autoplay playsinline/>');
+					Janus.attachMediaStream($('#remotevideo' + remoteFeed.rfindex + '-' + mid).get(0), stream);
+					if(remoteFeed.remoteVideos === 0) {
+						// No video, at least for now: show a placeholder
+						if($('#videoremote'+remoteFeed.rfindex + ' .no-video-container').length === 0) {
+							$('#videoremote'+remoteFeed.rfindex).append(
+								'<div class="no-video-container">' +
+									'<i class="fa fa-video-camera fa-5 no-video-icon"></i>' +
+									'<span class="no-video-text">No remote video available</span>' +
+								'</div>');
+						}
 					}
 				} else {
-					$('#remotevideo'+remoteFeed.rfindex+ ' .no-video-container').remove();
-					$('#remotevideo'+remoteFeed.rfindex).removeClass('hide').show();
-				}
-				if(!addButtons)
-					return;
-				if(Janus.webRTCAdapter.browserDetails.browser === "chrome" || Janus.webRTCAdapter.browserDetails.browser === "firefox" ||
-						Janus.webRTCAdapter.browserDetails.browser === "safari") {
-					$('#curbitrate'+remoteFeed.rfindex).removeClass('hide').show();
-					bitrateTimer[remoteFeed.rfindex] = setInterval(function() {
-						// Display updated bitrate, if supported
-						var bitrate = remoteFeed.getBitrate();
-						$('#curbitrate'+remoteFeed.rfindex).text(bitrate);
-						// Check if the resolution changed too
-						var width = $("#remotevideo"+remoteFeed.rfindex).get(0).videoWidth;
-						var height = $("#remotevideo"+remoteFeed.rfindex).get(0).videoHeight;
-						if(width > 0 && height > 0)
-							$('#curres'+remoteFeed.rfindex).removeClass('hide').text(width+'x'+height).show();
-					}, 1000);
+					// New video track: create a stream out of it
+					remoteFeed.remoteVideos++;
+					$('#videoremote'+remoteFeed.rfindex + ' .no-video-container').remove();
+					stream = new MediaStream();
+					stream.addTrack(track.clone());
+					remoteFeed.remoteTracks[mid] = stream;
+					Janus.log("Created remote video stream:", stream);
+					$('#videoremote'+remoteFeed.rfindex).append('<video class="rounded centered" id="remotevideo' + remoteFeed.rfindex + '-' + mid + '" width=100% autoplay playsinline/>');
+					Janus.attachMediaStream($('#remotevideo' + remoteFeed.rfindex + '-' + mid).get(0), stream);
+					// Note: we'll need this for additional videos too
+					if(!bitrateTimer) {
+						$('#curbitrate'+remoteFeed.rfindex).removeClass('hide').show();
+						bitrateTimer[remoteFeed.rfindex] = setInterval(function() {
+							if(!$("#remotevideo"+remoteFeed.rfindex).get(0))
+								return;
+							// Display updated bitrate, if supported
+							var bitrate = remoteFeed.getBitrate();
+							$('#curbitrate'+remoteFeed.rfindex).text(bitrate);
+							// Check if the resolution changed too
+							var width = $("#remotevideo"+remoteFeed.rfindex).get(0).videoWidth;
+							var height = $("#remotevideo"+remoteFeed.rfindex).get(0).videoHeight;
+							if(width > 0 && height > 0)
+								$('#curres'+remoteFeed.rfindex).removeClass('hide').text(width+'x'+height).show();
+						}, 1000);
+					}
 				}
 			},
 			oncleanup: function() {
@@ -592,6 +670,8 @@ function newRemoteFeed(id, display) {
 				$('#tl'+remoteFeed.rfindex+'-1').unbind('click');
 				$('#tl'+remoteFeed.rfindex+'-0').unbind('click');
 				$('#layers'+remoteFeed.rfindex).addClass('hide');
+				remoteFeed.remoteTracks = {};
+				remoteFeed.remoteVideos = 0;
 			}
 		});
 }
diff --git a/ice.c b/ice.c
index 4071ced19e..8275440460 100644
--- a/ice.c
+++ b/ice.c
@@ -1,9 +1,13 @@
 /*! \file    ice.c
  * \author   Lorenzo Miniero <lorenzo@meetecho.com>
  * \copyright GNU General Public License v3
- * \brief    ICE/STUN/TURN processing
- * \details  Implementation (based on libnice) of the ICE process. The
- * code handles the whole ICE process, from the gathering of candidates
+ * \brief    Janus handles and ICE/STUN/TURN processing
+ * \details  A Janus handle represents an abstraction of the communication
+ * between a user and a specific plugin, within a Janus session. This is
+ * particularly important in terms of media connectivity, as each handle
+ * can be associated with a single WebRTC PeerConnection. This code also
+ * contains the implementation (based on libnice) of a WebRTC PeerConnection.
+ * The code handles the whole ICE process, from the gathering of candidates
  * to the final setup of a virtual channel RTP and RTCP can be transported
  * on. Incoming RTP and RTCP packets from peers are relayed to the associated
  * plugins by means of the incoming_rtp and incoming_rtcp callbacks. Packets
@@ -160,6 +164,7 @@ void janus_ice_set_static_event_loops(int loops) {
 			g_free(loop);
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch a new event loop thread...\n",
 				error->code, error->message ? error->message : "??");
+			g_error_free(error);
 		} else {
 			event_loops = g_slist_append(event_loops, loop);
 			static_event_loops++;
@@ -298,6 +303,7 @@ uint16_t rtp_range_max = 0;
 #define JANUS_ICE_PACKET_SCTP	3
 /* Janus enqueued (S)RTP/(S)RTCP packet to send */
 typedef struct janus_ice_queued_packet {
+	gint mindex;
 	char *data;
 	char *label;
 	gint length;
@@ -313,39 +319,39 @@ static janus_ice_queued_packet janus_ice_dtls_handshake,
 	janus_ice_hangup_peerconnection, janus_ice_detach_handle;
 
 /* Janus NACKed packet we're tracking (to avoid duplicates) */
-typedef struct janus_ice_nacked_packet {
-	janus_ice_handle *handle;
+typedef struct janus_nacked_packet {
+	janus_handle_webrtc_medium *medium;
 	int vindex;
 	guint16 seq_number;
-} janus_ice_nacked_packet;
-static gboolean janus_ice_nacked_packet_cleanup(gpointer user_data) {
-	janus_ice_nacked_packet *pkt = (janus_ice_nacked_packet *)user_data;
+} janus_nacked_packet;
+static gboolean janus_nacked_packet_cleanup(gpointer user_data) {
+	janus_nacked_packet *pkt = (janus_nacked_packet *)user_data;
 
-	if(pkt->handle->stream){
+	if(pkt->medium && pkt->medium->pc && pkt->medium->pc->handle) {
 		JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Cleaning up NACKed packet %"SCNu16" (SSRC %"SCNu32", vindex %d)...\n",
-			pkt->handle->handle_id, pkt->seq_number, pkt->handle->stream->video_ssrc_peer[pkt->vindex], pkt->vindex);
-		g_hash_table_remove(pkt->handle->stream->rtx_nacked[pkt->vindex], GUINT_TO_POINTER(pkt->seq_number));
+			pkt->medium->pc->handle->handle_id, pkt->seq_number, pkt->medium->ssrc_peer[pkt->vindex], pkt->vindex);
+		g_hash_table_remove(pkt->medium->rtx_nacked[pkt->vindex], GUINT_TO_POINTER(pkt->seq_number));
 	}
 
 	return G_SOURCE_REMOVE;
 }
 
 /* Deallocation helpers for handles and related structs */
-static void janus_ice_handle_free(const janus_refcount *handle_ref);
-static void janus_ice_webrtc_free(janus_ice_handle *handle);
+static void janus_handle_free(const janus_refcount *handle_ref);
+static void janus_ice_webrtc_free(janus_handle *handle);
 static void janus_ice_plugin_session_free(const janus_refcount *app_handle_ref);
-static void janus_ice_stream_free(const janus_refcount *handle_ref);
-static void janus_ice_component_free(const janus_refcount *handle_ref);
+static void janus_handle_webrtc_free(const janus_refcount *handle_ref);
+static void janus_handle_webrtc_medium_free(const janus_refcount *handle_ref);
 
 /* Custom GSource for outgoing traffic */
 typedef struct janus_ice_outgoing_traffic {
 	GSource parent;
-	janus_ice_handle *handle;
+	janus_handle *handle;
 	GDestroyNotify destroy;
 } janus_ice_outgoing_traffic;
 static gboolean janus_ice_outgoing_rtcp_handle(gpointer user_data);
 static gboolean janus_ice_outgoing_stats_handle(gpointer user_data);
-static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janus_ice_queued_packet *pkt);
+static gboolean janus_ice_outgoing_traffic_handle(janus_handle *handle, janus_ice_queued_packet *pkt);
 static gboolean janus_ice_outgoing_traffic_prepare(GSource *source, gint *timeout) {
 	janus_ice_outgoing_traffic *t = (janus_ice_outgoing_traffic *)source;
 	return (g_async_queue_length(t->handle->queued_packets) > 0);
@@ -380,7 +386,7 @@ static GSourceFuncs janus_ice_outgoing_traffic_funcs = {
 	janus_ice_outgoing_traffic_finalize,
 	NULL, NULL
 };
-static GSource *janus_ice_outgoing_traffic_create(janus_ice_handle *handle, GDestroyNotify destroy) {
+static GSource *janus_ice_outgoing_traffic_create(janus_handle *handle, GDestroyNotify destroy) {
 	GSource *source = g_source_new(&janus_ice_outgoing_traffic_funcs, sizeof(janus_ice_outgoing_traffic));
 	janus_ice_outgoing_traffic *t = (janus_ice_outgoing_traffic *)source;
 	char name[255];
@@ -449,7 +455,7 @@ gboolean janus_is_rfc4588_enabled(void) {
 	return rfc4588_enabled;
 }
 
-static inline void janus_ice_free_rtp_packet(janus_rtp_packet *pkt) {
+static inline void janus_rtp_packet_free(janus_rtp_packet *pkt) {
 	if(pkt == NULL) {
 		return;
 	}
@@ -485,35 +491,30 @@ uint janus_get_max_nack_queue(void) {
 	return max_nack_queue;
 }
 /* Helper to clean old NACK packets in the buffer when they exceed the queue time limit */
-static void janus_cleanup_nack_buffer(gint64 now, janus_ice_stream *stream, gboolean audio, gboolean video) {
-	if(stream && stream->component) {
-		janus_ice_component *component = stream->component;
-		if(audio && component->audio_retransmit_buffer) {
-			janus_rtp_packet *p = (janus_rtp_packet *)g_queue_peek_head(component->audio_retransmit_buffer);
-			while(p && (!now || (now - p->created >= (gint64)max_nack_queue*1000))) {
-				/* Packet is too old, get rid of it */
-				g_queue_pop_head(component->audio_retransmit_buffer);
-				/* Remove from hashtable too */
-				janus_rtp_header *header = (janus_rtp_header *)p->data;
-				guint16 seq = ntohs(header->seq_number);
-				g_hash_table_remove(component->audio_retransmit_seqs, GUINT_TO_POINTER(seq));
-				/* Free the packet */
-				janus_ice_free_rtp_packet(p);
-				p = (janus_rtp_packet *)g_queue_peek_head(component->audio_retransmit_buffer);
-			}
-		}
-		if(video && component->video_retransmit_buffer) {
-			janus_rtp_packet *p = (janus_rtp_packet *)g_queue_peek_head(component->video_retransmit_buffer);
+static void janus_cleanup_nack_buffer(gint64 now, janus_handle_webrtc *pc, gboolean audio, gboolean video) {
+	if(!pc)
+		return;
+	/* Iterate on all media */
+	janus_handle_webrtc_medium *medium = NULL;
+	uint mi=0;
+	for(mi=0; mi<g_hash_table_size(pc->media); mi++) {
+		medium = g_hash_table_lookup(pc->media, GUINT_TO_POINTER(mi));
+		if(!medium)
+			continue;
+		if((medium->type == JANUS_MEDIA_AUDIO && !audio) || (medium->type == JANUS_MEDIA_VIDEO && !video))
+			continue;
+		if(medium->retransmit_buffer) {
+			janus_rtp_packet *p = (janus_rtp_packet *)g_queue_peek_head(medium->retransmit_buffer);
 			while(p && (!now || (now - p->created >= (gint64)max_nack_queue*1000))) {
 				/* Packet is too old, get rid of it */
-				g_queue_pop_head(component->video_retransmit_buffer);
+				g_queue_pop_head(medium->retransmit_buffer);
 				/* Remove from hashtable too */
 				janus_rtp_header *header = (janus_rtp_header *)p->data;
 				guint16 seq = ntohs(header->seq_number);
-				g_hash_table_remove(component->video_retransmit_seqs, GUINT_TO_POINTER(seq));
+				g_hash_table_remove(medium->retransmit_seqs, GUINT_TO_POINTER(seq));
 				/* Free the packet */
-				janus_ice_free_rtp_packet(p);
-				p = (janus_rtp_packet *)g_queue_peek_head(component->video_retransmit_buffer);
+				janus_rtp_packet_free(p);
+				p = (janus_rtp_packet *)g_queue_peek_head(medium->retransmit_buffer);
 			}
 		}
 	}
@@ -572,7 +573,7 @@ static int janus_seq_in_range(guint16 seqn, guint16 start, guint16 len) {
 
 
 /* Internal method for relaying RTCP messages, optionally filtering them in case they come from plugins */
-void janus_ice_relay_rtcp_internal(janus_ice_handle *handle, int video, char *buf, int len, gboolean filter_rtcp);
+static void janus_ice_relay_rtcp_internal(janus_handle *handle, int mindex, gboolean video, char *buf, int len, gboolean filter_rtcp);
 
 
 /* Map of active plugin sessions */
@@ -597,7 +598,7 @@ static void janus_plugin_session_dereference(janus_plugin_session *plugin_sessio
 }
 
 
-static void janus_ice_clear_queued_packets(janus_ice_handle *handle) {
+static void janus_ice_clear_queued_packets(janus_handle *handle) {
 	if(handle == NULL || handle->queued_packets == NULL) {
 		return;
 	}
@@ -609,7 +610,7 @@ static void janus_ice_clear_queued_packets(janus_ice_handle *handle) {
 }
 
 
-static void janus_ice_notify_trickle(janus_ice_handle *handle, char *buffer) {
+static void janus_ice_notify_trickle(janus_handle *handle, char *buffer) {
 	if(handle == NULL)
 		return;
 	char cbuffer[200];
@@ -627,7 +628,7 @@ static void janus_ice_notify_trickle(janus_ice_handle *handle, char *buffer) {
 		json_object_set_new(event, "opaque_id", json_string(handle->opaque_id));
 	json_t *candidate = json_object();
 	if(buffer != NULL) {
-		json_object_set_new(candidate, "sdpMid", json_string(handle->stream_mid));
+		json_object_set_new(candidate, "sdpMid", json_string(handle->pc_mid));
 		json_object_set_new(candidate, "sdpMLineIndex", json_integer(0));
 		json_object_set_new(candidate, "candidate", json_string(cbuffer));
 	} else {
@@ -640,12 +641,12 @@ static void janus_ice_notify_trickle(janus_ice_handle *handle, char *buffer) {
 	janus_session_notify_event(session, event);
 }
 
-static void janus_ice_notify_media(janus_ice_handle *handle, gboolean video, gboolean up) {
+static void janus_ice_notify_media(janus_handle *handle, char *mid, gboolean video, gboolean up) {
 	if(handle == NULL)
 		return;
 	/* Prepare JSON event to notify user/application */
-	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Notifying that we %s receiving %s\n",
-		handle->handle_id, up ? "are" : "are NOT", video ? "video" : "audio");
+	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Notifying that we %s receiving %s on mid %s\n",
+		handle->handle_id, up ? "are" : "are NOT", video ? "video" : "audio", mid);
 	janus_session *session = (janus_session *)handle->session;
 	if(session == NULL)
 		return;
@@ -655,6 +656,7 @@ static void janus_ice_notify_media(janus_ice_handle *handle, gboolean video, gbo
 	json_object_set_new(event, "sender", json_integer(handle->handle_id));
 	if(opaqueid_in_api && handle->opaque_id != NULL)
 		json_object_set_new(event, "opaque_id", json_string(handle->opaque_id));
+	json_object_set_new(event, "mid", json_string(mid));
 	json_object_set_new(event, "type", json_string(video ? "video" : "audio"));
 	json_object_set_new(event, "receiving", up ? json_true() : json_false());
 	if(!up && no_media_timer > 1)
@@ -666,6 +668,7 @@ static void janus_ice_notify_media(janus_ice_handle *handle, gboolean video, gbo
 	if(janus_events_is_enabled()) {
 		json_t *info = json_object();
 		json_object_set_new(info, "media", json_string(video ? "video" : "audio"));
+		json_object_set_new(info, "mid", json_string(mid));
 		json_object_set_new(info, "receiving", up ? json_true() : json_false());
 		if(!up && no_media_timer > 1)
 			json_object_set_new(info, "seconds", json_integer(no_media_timer));
@@ -673,7 +676,7 @@ static void janus_ice_notify_media(janus_ice_handle *handle, gboolean video, gbo
 	}
 }
 
-void janus_ice_notify_hangup(janus_ice_handle *handle, const char *reason) {
+void janus_ice_notify_hangup(janus_handle *handle, const char *reason) {
 	if(handle == NULL)
 		return;
 	/* Prepare JSON event to notify user/application */
@@ -704,10 +707,10 @@ void janus_ice_notify_hangup(janus_ice_handle *handle, const char *reason) {
 
 
 /* Trickle helpers */
-janus_ice_trickle *janus_ice_trickle_new(const char *transaction, json_t *candidate) {
+janus_trickle *janus_trickle_new(const char *transaction, json_t *candidate) {
 	if(transaction == NULL || candidate == NULL)
 		return NULL;
-	janus_ice_trickle *trickle = g_malloc(sizeof(janus_ice_trickle));
+	janus_trickle *trickle = g_malloc(sizeof(janus_trickle));
 	trickle->handle = NULL;
 	trickle->received = janus_get_monotonic_time();
 	trickle->transaction = g_strdup(transaction);
@@ -715,7 +718,7 @@ janus_ice_trickle *janus_ice_trickle_new(const char *transaction, json_t *candid
 	return trickle;
 }
 
-gint janus_ice_trickle_parse(janus_ice_handle *handle, json_t *candidate, const char **error) {
+gint janus_trickle_parse(janus_handle *handle, json_t *candidate, const char **error) {
 	const char *ignore_error = NULL;
 	if(error == NULL) {
 		error = &ignore_error;
@@ -727,7 +730,7 @@ gint janus_ice_trickle_parse(janus_ice_handle *handle, json_t *candidate, const
 	/* Parse trickle candidate */
 	if(!json_is_object(candidate) || json_object_get(candidate, "completed") != NULL) {
 		JANUS_LOG(LOG_VERB, "No more remote candidates for handle %"SCNu64"!\n", handle->handle_id);
-		janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALL_TRICKLES);
+		janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALL_TRICKLES);
 	} else {
 		/* Handle remote candidate */
 		json_t *mid = json_object_get(candidate, "sdpMid");
@@ -757,18 +760,18 @@ gint janus_ice_trickle_parse(janus_ice_handle *handle, json_t *candidate, const
 		/* Parse it */
 		int sdpMLineIndex = mline ? json_integer_value(mline) : -1;
 		const char *sdpMid = json_string_value(mid);
-		if(sdpMLineIndex > 0 || (handle->stream_mid && sdpMid && strcmp(handle->stream_mid, sdpMid))) {
+		if(sdpMLineIndex > 0 || (handle->pc_mid && sdpMid && strcmp(handle->pc_mid, sdpMid))) {
 			/* FIXME We bundle everything, so we ignore candidates for anything beyond the first m-line */
 			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Got a mid='%s' candidate (index %d) but we're bundling, ignoring...\n",
 				handle->handle_id, json_string_value(mid), sdpMLineIndex);
 			return 0;
 		}
-		janus_ice_stream *stream = handle->stream;
-		if(stream == NULL) {
-			*error = "Trickle error: invalid element type (no such stream)";
+		janus_handle_webrtc *pc = handle->pc;
+		if(pc == NULL) {
+			*error = "Trickle error: invalid element type (no WebRTC PeerConnection)";
 			return JANUS_ERROR_TRICKE_INVALID_STREAM;
 		}
-		int res = janus_sdp_parse_candidate(stream, json_string_value(rc), 1);
+		int res = janus_sdp_parse_candidate(pc, json_string_value(rc), 1);
 		if(res != 0) {
 			JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to parse candidate... (%d)\n", handle->handle_id, res);
 			/* FIXME Should we return an error? */
@@ -777,7 +780,7 @@ gint janus_ice_trickle_parse(janus_ice_handle *handle, json_t *candidate, const
 	return 0;
 }
 
-void janus_ice_trickle_destroy(janus_ice_trickle *trickle) {
+void janus_trickle_destroy(janus_trickle *trickle) {
 	if(trickle == NULL)
 		return;
 	g_free(trickle->transaction);
@@ -1107,8 +1110,8 @@ const gchar *janus_get_ice_state_name(gint state) {
 
 
 /* Thread to take care of the handle loop */
-static void *janus_ice_handle_thread(void *data) {
-	janus_ice_handle *handle = data;
+static void *janus_handle_thread(void *data) {
+	janus_handle *handle = data;
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Handle thread started; %p\n", handle->handle_id, handle);
 	if(handle->mainloop == NULL) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Invalid loop...\n", handle->handle_id);
@@ -1127,11 +1130,11 @@ static void *janus_ice_handle_thread(void *data) {
 	return NULL;
 }
 
-janus_ice_handle *janus_ice_handle_create(void *core_session, const char *opaque_id) {
+janus_handle *janus_handle_create(void *core_session, const char *opaque_id) {
 	if(core_session == NULL)
 		return NULL;
 	janus_session *session = (janus_session *)core_session;
-	janus_ice_handle *handle = NULL;
+	janus_handle *handle = NULL;
 	guint64 handle_id = 0;
 	while(handle_id == 0) {
 		handle_id = janus_random_uint64();
@@ -1142,9 +1145,9 @@ janus_ice_handle *janus_ice_handle_create(void *core_session, const char *opaque
 			handle_id = 0;
 		}
 	}
-	handle = (janus_ice_handle *)g_malloc0(sizeof(janus_ice_handle));
+	handle = (janus_handle *)g_malloc0(sizeof(janus_handle));
 	JANUS_LOG(LOG_INFO, "Creating new handle in session %"SCNu64": %"SCNu64"; %p %p\n", session->session_id, handle_id, core_session, handle);
-	janus_refcount_init(&handle->ref, janus_ice_handle_free);
+	janus_refcount_init(&handle->ref, janus_handle_free);
 	janus_refcount_increase(&session->ref);
 	handle->session = core_session;
 	if(opaque_id)
@@ -1159,7 +1162,7 @@ janus_ice_handle *janus_ice_handle_create(void *core_session, const char *opaque
 	return handle;
 }
 
-gint janus_ice_handle_attach_plugin(void *core_session, janus_ice_handle *handle, janus_plugin *plugin) {
+gint janus_handle_attach_plugin(void *core_session, janus_handle *handle, janus_plugin *plugin) {
 	if(core_session == NULL)
 		return JANUS_ERROR_SESSION_NOT_FOUND;
 	janus_session *session = (janus_session *)core_session;
@@ -1217,13 +1220,14 @@ gint janus_ice_handle_attach_plugin(void *core_session, janus_ice_handle *handle
 		char tname[16];
 		g_snprintf(tname, sizeof(tname), "hloop %"SCNu64, handle->handle_id);
 		janus_refcount_increase(&handle->ref);
-		handle->thread = g_thread_try_new(tname, &janus_ice_handle_thread, handle, &terror);
+		handle->thread = g_thread_try_new(tname, &janus_handle_thread, handle, &terror);
 		if(terror != NULL) {
 			/* FIXME We should clear some resources... */
 			JANUS_LOG(LOG_ERR, "[%"SCNu64"] Got error %d (%s) trying to launch the handle thread...\n",
 				handle->handle_id, terror->code, terror->message ? terror->message : "??");
+			g_error_free(terror);
 			janus_refcount_decrease(&handle->ref);	/* This is for the thread reference we just added */
-			janus_ice_handle_destroy(session, handle);
+			janus_handle_destroy(session, handle);
 			return -1;
 		}
 	}
@@ -1234,7 +1238,7 @@ gint janus_ice_handle_attach_plugin(void *core_session, janus_ice_handle *handle
 	return 0;
 }
 
-gint janus_ice_handle_destroy(void *core_session, janus_ice_handle *handle) {
+gint janus_handle_destroy(void *core_session, janus_handle *handle) {
 	/* session->mutex has to be locked when calling this function */
 	janus_session *session = (janus_session *)core_session;
 	if(session == NULL)
@@ -1244,8 +1248,8 @@ gint janus_ice_handle_destroy(void *core_session, janus_ice_handle *handle) {
 	if(!g_atomic_int_compare_and_exchange(&handle->destroyed, 0, 1))
 		return 0;
 	/* First of all, hangup the PeerConnection, if any */
-	janus_ice_webrtc_hangup(handle, "Detach");
-	janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP);
+	janus_handle_webrtc_hangup(handle, "Detach");
+	janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP);
 	/* Remove the session from active sessions map */
 	janus_mutex_lock(&plugin_sessions_mutex);
 	gboolean found = g_hash_table_remove(plugin_sessions, handle->app_handle);
@@ -1257,8 +1261,8 @@ gint janus_ice_handle_destroy(void *core_session, janus_ice_handle *handle) {
 	janus_plugin *plugin_t = (janus_plugin *)handle->app;
 	if(plugin_t == NULL) {
 		/* There was no plugin attached, probably something went wrong there */
-		janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT);
-		janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP);
+		janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT);
+		janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP);
 		if(handle->mainloop != NULL) {
 			if(handle->stream_id > 0) {
 				nice_agent_attach_recv(handle->agent, handle->stream_id, 1, g_main_loop_get_context(handle->mainloop), NULL, NULL);
@@ -1290,8 +1294,8 @@ gint janus_ice_handle_destroy(void *core_session, janus_ice_handle *handle) {
 	return 0;
 }
 
-static void janus_ice_handle_free(const janus_refcount *handle_ref) {
-	janus_ice_handle *handle = janus_refcount_containerof(handle_ref, janus_ice_handle, ref);
+static void janus_handle_free(const janus_refcount *handle_ref) {
+	janus_handle *handle = janus_refcount_containerof(handle_ref, janus_handle, ref);
 	/* This stack can be destroyed, free all the resources */
 	janus_mutex_lock(&handle->mutex);
 	if(handle->queued_packets != NULL) {
@@ -1322,26 +1326,24 @@ static void janus_ice_plugin_session_free(const janus_refcount *app_handle_ref)
 	janus_plugin_session *app_handle = janus_refcount_containerof(app_handle_ref, janus_plugin_session, ref);
 	/* This app handle can be destroyed, free all the resources */
 	if(app_handle->gateway_handle != NULL) {
-		janus_ice_handle *handle = (janus_ice_handle *)app_handle->gateway_handle;
+		janus_handle *handle = (janus_handle *)app_handle->gateway_handle;
 		app_handle->gateway_handle = NULL;
 		janus_refcount_decrease(&handle->ref);
 	}
 	g_free(app_handle);
 }
 
-void janus_ice_webrtc_hangup(janus_ice_handle *handle, const char *reason) {
+void janus_handle_webrtc_hangup(janus_handle *handle, const char *reason) {
 	if(handle == NULL)
 		return;
 	g_atomic_int_set(&handle->closepc, 0);
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT))
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT))
 		return;
-	janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT);
-	janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEGOTIATED);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO);
+	janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT);
+	janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_OFFER);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEGOTIATED);
 	/* User will be notified only after the actual hangup */
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Hanging up PeerConnection because of a %s\n",
 		handle->handle_id, reason);
@@ -1361,22 +1363,22 @@ void janus_ice_webrtc_hangup(janus_ice_handle *handle, const char *reason) {
 	}
 }
 
-static void janus_ice_webrtc_free(janus_ice_handle *handle) {
+static void janus_ice_webrtc_free(janus_handle *handle) {
 	if(handle == NULL)
 		return;
 	janus_mutex_lock(&handle->mutex);
 	if(!handle->agent_created) {
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY);
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING);
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AGENT);
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY);
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING);
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_HAS_AGENT);
 		janus_mutex_unlock(&handle->mutex);
 		return;
 	}
 	handle->agent_created = 0;
-	if(handle->stream != NULL) {
-		janus_ice_stream_destroy(handle->stream);
-		handle->stream = NULL;
+	if(handle->pc != NULL) {
+		janus_handle_webrtc_destroy(handle->pc);
+		handle->pc = NULL;
 	}
 	if(handle->agent != NULL) {
 		if(G_IS_OBJECT(handle->agent))
@@ -1387,9 +1389,9 @@ static void janus_ice_webrtc_free(janus_ice_handle *handle) {
 		while(handle->pending_trickles) {
 			GList *temp = g_list_first(handle->pending_trickles);
 			handle->pending_trickles = g_list_remove_link(handle->pending_trickles, temp);
-			janus_ice_trickle *trickle = (janus_ice_trickle *)temp->data;
+			janus_trickle *trickle = (janus_trickle *)temp->data;
 			g_list_free(temp);
-			janus_ice_trickle_destroy(trickle);
+			janus_trickle_destroy(trickle);
 		}
 	}
 	handle->pending_trickles = NULL;
@@ -1399,19 +1401,14 @@ static void janus_ice_webrtc_free(janus_ice_handle *handle) {
 	handle->local_sdp = NULL;
 	g_free(handle->remote_sdp);
 	handle->remote_sdp = NULL;
-	handle->stream_mid = NULL;
-	g_free(handle->audio_mid);
-	handle->audio_mid = NULL;
-	g_free(handle->video_mid);
-	handle->video_mid = NULL;
-	g_free(handle->data_mid);
-	handle->data_mid = NULL;
+	g_free(handle->pc_mid);
+	handle->pc_mid = NULL;
 	handle->thread = NULL;
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AGENT);
-	if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP) && handle->hangup_reason) {
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_HAS_AGENT);
+	if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP) && handle->hangup_reason) {
 		janus_ice_notify_hangup(handle, handle->hangup_reason);
 	}
 	handle->hangup_reason = NULL;
@@ -1419,140 +1416,58 @@ static void janus_ice_webrtc_free(janus_ice_handle *handle) {
 	JANUS_LOG(LOG_INFO, "[%"SCNu64"] WebRTC resources freed; %p %p\n", handle->handle_id, handle, handle->session);
 }
 
-void janus_ice_stream_destroy(janus_ice_stream *stream) {
-	if(stream == NULL)
+void janus_handle_webrtc_destroy(janus_handle_webrtc *pc) {
+	if(pc == NULL)
 		return;
-	if(stream->component != NULL) {
-		janus_ice_component_destroy(stream->component);
-		stream->component = NULL;
-	}
-	janus_ice_handle *handle = stream->handle;
+	/* Remove all media instances */
+	g_hash_table_remove_all(pc->media);
+	g_hash_table_remove_all(pc->media_byssrc);
+	g_hash_table_remove_all(pc->media_bymid);
+	g_hash_table_remove_all(pc->media_bytype);
+	/* Get rid of the DTLS stack */
+	if(pc->dtlsrt_source != NULL) {
+		g_source_destroy(pc->dtlsrt_source);
+		g_source_unref(pc->dtlsrt_source);
+		pc->dtlsrt_source = NULL;
+	}
+	if(pc->dtls != NULL) {
+		janus_dtls_srtp_destroy(pc->dtls);
+		janus_refcount_decrease(&pc->dtls->ref);
+		pc->dtls = NULL;
+	}
+	janus_handle *handle = pc->handle;
 	if(handle != NULL) {
 		janus_refcount_decrease(&handle->ref);
-		stream->handle = NULL;
+		pc->handle = NULL;
 	}
-	janus_refcount_decrease(&stream->ref);
+	janus_refcount_decrease(&pc->ref);
 }
 
-static void janus_ice_stream_free(const janus_refcount *stream_ref) {
-	janus_ice_stream *stream = janus_refcount_containerof(stream_ref, janus_ice_stream, ref);
-	/* This stream can be destroyed, free all the resources */
-	stream->handle = NULL;
-	g_free(stream->remote_hashing);
-	stream->remote_hashing = NULL;
-	g_free(stream->remote_fingerprint);
-	stream->remote_fingerprint = NULL;
-	g_free(stream->ruser);
-	stream->ruser = NULL;
-	g_free(stream->rpass);
-	stream->rpass = NULL;
-	g_free(stream->rid[0]);
-	stream->rid[0] = NULL;
-	g_free(stream->rid[1]);
-	stream->rid[1] = NULL;
-	g_free(stream->rid[2]);
-	stream->rid[2] = NULL;
-	g_list_free(stream->audio_payload_types);
-	stream->audio_payload_types = NULL;
-	g_list_free(stream->video_payload_types);
-	stream->video_payload_types = NULL;
-	if(stream->rtx_payload_types != NULL)
-		g_hash_table_destroy(stream->rtx_payload_types);
-	stream->rtx_payload_types = NULL;
-	g_free(stream->audio_codec);
-	stream->audio_codec = NULL;
-	g_free(stream->video_codec);
-	stream->video_codec = NULL;
-	g_free(stream->audio_rtcp_ctx);
-	stream->audio_rtcp_ctx = NULL;
-	g_free(stream->video_rtcp_ctx[0]);
-	stream->video_rtcp_ctx[0] = NULL;
-	g_free(stream->video_rtcp_ctx[1]);
-	stream->video_rtcp_ctx[1] = NULL;
-	g_free(stream->video_rtcp_ctx[2]);
-	stream->video_rtcp_ctx[2] = NULL;
-	if(stream->rtx_nacked[0])
-		g_hash_table_destroy(stream->rtx_nacked[0]);
-	stream->rtx_nacked[0] = NULL;
-	if(stream->rtx_nacked[1])
-		g_hash_table_destroy(stream->rtx_nacked[1]);
-	stream->rtx_nacked[1] = NULL;
-	if(stream->rtx_nacked[2])
-		g_hash_table_destroy(stream->rtx_nacked[2]);
-	stream->rtx_nacked[2] = NULL;
-	g_slist_free_full(stream->transport_wide_received_seq_nums, (GDestroyNotify)g_free);
-	stream->transport_wide_received_seq_nums = NULL;
-	stream->audio_first_ntp_ts = 0;
-	stream->audio_first_rtp_ts = 0;
-	stream->video_first_ntp_ts[0] = 0;
-	stream->video_first_ntp_ts[1] = 0;
-	stream->video_first_ntp_ts[2] = 0;
-	stream->video_first_rtp_ts[0] = 0;
-	stream->video_first_rtp_ts[1] = 0;
-	stream->video_first_rtp_ts[2] = 0;
-	stream->audio_last_ts = 0;
-	stream->video_last_ts = 0;
-	g_free(stream);
-	stream = NULL;
-}
-
-void janus_ice_component_destroy(janus_ice_component *component) {
-	if(component == NULL)
-		return;
-	janus_ice_stream *stream = component->stream;
-	if(stream != NULL) {
-		janus_refcount_decrease(&stream->ref);
-		component->stream = NULL;
-	}
-	janus_dtls_srtp_destroy(component->dtls);
-	janus_refcount_decrease(&component->ref);
-}
-
-static void janus_ice_component_free(const janus_refcount *component_ref) {
-	janus_ice_component *component = janus_refcount_containerof(component_ref, janus_ice_component, ref);
-	if(component->icestate_source != NULL) {
-		g_source_destroy(component->icestate_source);
-		g_source_unref(component->icestate_source);
-		component->icestate_source = NULL;
-	}
-	if(component->dtlsrt_source != NULL) {
-		g_source_destroy(component->dtlsrt_source);
-		g_source_unref(component->dtlsrt_source);
-		component->dtlsrt_source = NULL;
-	}
-	if(component->dtls != NULL) {
-		janus_dtls_srtp_destroy(component->dtls);
-		janus_refcount_decrease(&component->dtls->ref);
-		component->dtls = NULL;
-	}
-	if(component->audio_retransmit_buffer != NULL) {
-		janus_rtp_packet *p = NULL;
-		while((p = (janus_rtp_packet *)g_queue_pop_head(component->audio_retransmit_buffer)) != NULL) {
-			/* Remove from hashtable too */
-			janus_rtp_header *header = (janus_rtp_header *)p->data;
-			guint16 seq = ntohs(header->seq_number);
-			g_hash_table_remove(component->audio_retransmit_seqs, GUINT_TO_POINTER(seq));
-			/* Free the packet */
-			janus_ice_free_rtp_packet(p);
-		}
-		g_queue_free(component->audio_retransmit_buffer);
-		g_hash_table_destroy(component->audio_retransmit_seqs);
-	}
-	if(component->video_retransmit_buffer != NULL) {
-		janus_rtp_packet *p = NULL;
-		while((p = (janus_rtp_packet *)g_queue_pop_head(component->video_retransmit_buffer)) != NULL) {
-			/* Remove from hashtable too */
-			janus_rtp_header *header = (janus_rtp_header *)p->data;
-			guint16 seq = ntohs(header->seq_number);
-			g_hash_table_remove(component->video_retransmit_seqs, GUINT_TO_POINTER(seq));
-			/* Free the packet */
-			janus_ice_free_rtp_packet(p);
-		}
-		g_queue_free(component->video_retransmit_buffer);
-		g_hash_table_destroy(component->video_retransmit_seqs);
-	}
-	if(component->candidates != NULL) {
-		GSList *i = NULL, *candidates = component->candidates;
+static void janus_handle_webrtc_free(const janus_refcount *pc_ref) {
+	janus_handle_webrtc *pc = janus_refcount_containerof(pc_ref, janus_handle_webrtc, ref);
+	/* This PeerConnection can be destroyed, free all the resources */
+	pc->handle = NULL;
+	g_hash_table_destroy(pc->media);
+	g_hash_table_destroy(pc->media_byssrc);
+	g_hash_table_destroy(pc->media_bymid);
+	g_hash_table_destroy(pc->media_bytype);
+	if(pc->icestate_source != NULL) {
+		g_source_destroy(pc->icestate_source);
+		g_source_unref(pc->icestate_source);
+		pc->icestate_source = NULL;
+	}
+	g_free(pc->remote_hashing);
+	pc->remote_hashing = NULL;
+	g_free(pc->remote_fingerprint);
+	pc->remote_fingerprint = NULL;
+	g_free(pc->ruser);
+	pc->ruser = NULL;
+	g_free(pc->rpass);
+	pc->rpass = NULL;
+	g_slist_free_full(pc->transport_wide_received_seq_nums, (GDestroyNotify)g_free);
+	pc->transport_wide_received_seq_nums = NULL;
+	if(pc->candidates != NULL) {
+		GSList *i = NULL, *candidates = pc->candidates;
 		for(i = candidates; i; i = i->next) {
 			NiceCandidate *c = (NiceCandidate *) i->data;
 			if(c != NULL) {
@@ -1563,9 +1478,9 @@ static void janus_ice_component_free(const janus_refcount *component_ref) {
 		g_slist_free(candidates);
 		candidates = NULL;
 	}
-	component->candidates = NULL;
-	if(component->local_candidates != NULL) {
-		GSList *i = NULL, *candidates = component->local_candidates;
+	pc->candidates = NULL;
+	if(pc->local_candidates != NULL) {
+		GSList *i = NULL, *candidates = pc->local_candidates;
 		for(i = candidates; i; i = i->next) {
 			gchar *c = (gchar *) i->data;
 			g_free(c);
@@ -1573,9 +1488,9 @@ static void janus_ice_component_free(const janus_refcount *component_ref) {
 		g_slist_free(candidates);
 		candidates = NULL;
 	}
-	component->local_candidates = NULL;
-	if(component->remote_candidates != NULL) {
-		GSList *i = NULL, *candidates = component->remote_candidates;
+	pc->local_candidates = NULL;
+	if(pc->remote_candidates != NULL) {
+		GSList *i = NULL, *candidates = pc->remote_candidates;
 		for(i = candidates; i; i = i->next) {
 			gchar *c = (gchar *) i->data;
 			g_free(c);
@@ -1583,36 +1498,136 @@ static void janus_ice_component_free(const janus_refcount *component_ref) {
 		g_slist_free(candidates);
 		candidates = NULL;
 	}
-	component->remote_candidates = NULL;
-	g_free(component->selected_pair);
-	component->selected_pair = NULL;
-	if(component->last_seqs_audio)
-		janus_seq_list_free(&component->last_seqs_audio);
-	if(component->last_seqs_video[0])
-		janus_seq_list_free(&component->last_seqs_video[0]);
-	if(component->last_seqs_video[1])
-		janus_seq_list_free(&component->last_seqs_video[1]);
-	if(component->last_seqs_video[2])
-		janus_seq_list_free(&component->last_seqs_video[2]);
-	g_free(component);
+	pc->remote_candidates = NULL;
+	g_free(pc->selected_pair);
+	pc->selected_pair = NULL;
+	g_free(pc);
+	pc = NULL;
+}
+
+janus_handle_webrtc_medium *janus_handle_webrtc_medium_create(janus_handle *handle, janus_media_type type) {
+	if(handle == NULL || handle->pc == NULL || type == JANUS_MEDIA_UNKNOWN)
+		return NULL;
+	janus_handle_webrtc *pc = handle->pc;
+	janus_handle_webrtc_medium *medium = g_malloc0(sizeof(janus_handle_webrtc_medium));
+	medium->pc = pc;
+	medium->type = type;
+	medium->mindex = g_hash_table_size(pc->media);
+	janus_mutex_init(&medium->mutex);
+	janus_refcount_init(&medium->ref, janus_handle_webrtc_medium_free);
+	janus_refcount_increase(&pc->ref);
+	g_hash_table_insert(pc->media, GINT_TO_POINTER(medium->mindex), medium);
+	/* If this is audio or video, fill in some other fields too */
+	if(type != JANUS_MEDIA_DATA) {
+		medium->payload_type = -1;
+		medium->rtx_payload_type = -1;
+		medium->ssrc = janus_random_uint32();	/* FIXME Should we look for conflicts? */
+		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX)) {
+			/* Create an SSRC for RFC4588 as well */
+			medium->ssrc_rtx = janus_random_uint32();	/* FIXME Should we look for conflicts? */
+		}
+		medium->rtcp_ctx[0] = g_malloc0(sizeof(janus_rtcp_context));
+		medium->rtcp_ctx[0]->tb = (type == JANUS_MEDIA_VIDEO ? 90000 : 48000);	/* May change later */
+		/* We can address media by SSRC */
+		g_hash_table_insert(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc), medium);
+		janus_refcount_increase(&medium->ref);
+		if(medium->ssrc_rtx > 0) {
+			g_hash_table_insert(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_rtx), medium);
+			janus_refcount_increase(&medium->ref);
+		}
+		g_hash_table_insert(pc->media_bytype, GINT_TO_POINTER(type), medium);
+		janus_refcount_increase(&medium->ref);
+	}
+	/* For backwards compatibility, we address media by type too (e.g., first video stream) */
+	g_hash_table_insert(pc->media_bytype, GINT_TO_POINTER(type), medium);
+	janus_refcount_increase(&medium->ref);
+	return medium;
+}
+
+static void janus_handle_webrtc_medium_destroy(janus_handle_webrtc_medium *medium) {
+	if(medium == NULL)
+		return;
+	janus_handle_webrtc *pc = medium->pc;
+	if(pc != NULL) {
+		janus_refcount_decrease(&pc->ref);
+		medium->pc = NULL;
+	}
+	janus_refcount_decrease(&medium->ref);
+}
+
+static void janus_handle_webrtc_medium_dereference(janus_handle_webrtc_medium *medium) {
+	if(medium == NULL)
+		return;
+	janus_refcount_decrease(&medium->ref);
+}
+
+static void janus_handle_webrtc_medium_free(const janus_refcount *medium_ref) {
+	janus_handle_webrtc_medium *medium = janus_refcount_containerof(medium_ref, janus_handle_webrtc_medium, ref);
+	g_free(medium->mid);
+	g_free(medium->rid[0]);
+	medium->rid[0] = NULL;
+	g_free(medium->rid[1]);
+	medium->rid[1] = NULL;
+	g_free(medium->rid[2]);
+	medium->rid[2] = NULL;
+	g_list_free(medium->payload_types);
+	medium->payload_types = NULL;
+	if(medium->rtx_payload_types != NULL)
+		g_hash_table_destroy(medium->rtx_payload_types);
+	medium->rtx_payload_types = NULL;
+	g_free(medium->codec);
+	medium->codec = NULL;
+	g_free(medium->rtcp_ctx[0]);
+	medium->rtcp_ctx[0] = NULL;
+	g_free(medium->rtcp_ctx[1]);
+	medium->rtcp_ctx[1] = NULL;
+	g_free(medium->rtcp_ctx[2]);
+	medium->rtcp_ctx[2] = NULL;
+	if(medium->rtx_nacked[0])
+		g_hash_table_destroy(medium->rtx_nacked[0]);
+	medium->rtx_nacked[0] = NULL;
+	if(medium->rtx_nacked[1])
+		g_hash_table_destroy(medium->rtx_nacked[1]);
+	medium->rtx_nacked[1] = NULL;
+	if(medium->rtx_nacked[2])
+		g_hash_table_destroy(medium->rtx_nacked[2]);
+	medium->rtx_nacked[2] = NULL;
+	if(medium->retransmit_buffer != NULL) {
+		janus_rtp_packet *p = NULL;
+		while((p = (janus_rtp_packet *)g_queue_pop_head(medium->retransmit_buffer)) != NULL) {
+			/* Remove from hashtable too */
+			janus_rtp_header *header = (janus_rtp_header *)p->data;
+			guint16 seq = ntohs(header->seq_number);
+			g_hash_table_remove(medium->retransmit_seqs, GUINT_TO_POINTER(seq));
+			/* Free the packet */
+			janus_rtp_packet_free(p);
+		}
+		g_queue_free(medium->retransmit_buffer);
+		g_hash_table_destroy(medium->retransmit_seqs);
+	}
+	if(medium->last_seqs[0])
+		janus_seq_list_free(&medium->last_seqs[0]);
+	if(medium->last_seqs[1])
+		janus_seq_list_free(&medium->last_seqs[1]);
+	if(medium->last_seqs[2])
+		janus_seq_list_free(&medium->last_seqs[2]);
+	g_free(medium);
 	//~ janus_mutex_unlock(&handle->mutex);
 }
 
 /* Call plugin slow_link callback if a minimum of lost packets are detected within a second */
 static void
-janus_slow_link_update(janus_ice_component *component, janus_ice_handle *handle,
-		gboolean video, gboolean uplink, guint lost) {
+janus_slow_link_update(janus_handle_webrtc_medium *medium, janus_handle *handle, gboolean uplink, guint lost) {
 	/* We keep the counters in different janus_ice_stats objects, depending on the direction */
-	guint sl_lost_last_count = uplink ?
-		(video ? component->in_stats.sl_lost_count_video : component->in_stats.sl_lost_count_audio) :
-		(video ? component->out_stats.sl_lost_count_video : component->out_stats.sl_lost_count_audio);
+	gboolean video = (medium->type == JANUS_MEDIA_VIDEO);
+	guint sl_lost_last_count = uplink ? medium->in_stats.sl_lost_count : medium->out_stats.sl_lost_count;
 	guint sl_lost_recently = (lost >= sl_lost_last_count) ? (lost - sl_lost_last_count) : 0;
 	if(slowlink_threshold > 0 && sl_lost_recently >= slowlink_threshold) {
 		/* Tell the plugin */
 		janus_plugin *plugin = (janus_plugin *)handle->app;
 		if(plugin && plugin->slow_link && janus_plugin_session_is_alive(handle->app_handle) &&
 				!g_atomic_int_get(&handle->destroyed))
-			plugin->slow_link(handle->app_handle, uplink, video);
+			plugin->slow_link(handle->app_handle, medium->mindex, video, uplink);
 		/* Notify the user/application too */
 		janus_session *session = (janus_session *)handle->session;
 		if(session != NULL) {
@@ -1622,6 +1637,7 @@ janus_slow_link_update(janus_ice_component *component, janus_ice_handle *handle,
 			json_object_set_new(event, "sender", json_integer(handle->handle_id));
 			if(opaqueid_in_api && handle->opaque_id != NULL)
 				json_object_set_new(event, "opaque_id", json_string(handle->opaque_id));
+			json_object_set_new(event, "mid", json_string(medium->mid));
 			json_object_set_new(event, "media", json_string(video ? "video" : "audio"));
 			json_object_set_new(event, "uplink", uplink ? json_true() : json_false());
 			json_object_set_new(event, "lost", json_integer(sl_lost_recently));
@@ -1631,6 +1647,7 @@ janus_slow_link_update(janus_ice_component *component, janus_ice_handle *handle,
 			/* Finally, notify event handlers */
 			if(janus_events_is_enabled()) {
 				json_t *info = json_object();
+				json_object_set_new(info, "mid", json_string(medium->mid));
 				json_object_set_new(info, "media", json_string(video ? "video" : "audio"));
 				json_object_set_new(info, "slow_link", json_string(uplink ? "uplink" : "downlink"));
 				json_object_set_new(info, "lost_lastsec", json_integer(sl_lost_recently));
@@ -1640,89 +1657,80 @@ janus_slow_link_update(janus_ice_component *component, janus_ice_handle *handle,
 	}
 	/* Update the counter */
 	if(uplink) {
-		if(video)
-			component->in_stats.sl_lost_count_video = lost;
-		else
-			component->in_stats.sl_lost_count_audio = lost;
+		medium->in_stats.sl_lost_count = lost;
 	} else {
-		if(video)
-			component->out_stats.sl_lost_count_video = lost;
-		else
-			component->out_stats.sl_lost_count_audio = lost;
+		medium->out_stats.sl_lost_count = lost;
 	}
 }
 
 
 /* ICE state check timer (needed to check if a failed really is definitive or if things can still improve) */
 static gboolean janus_ice_check_failed(gpointer data) {
-	janus_ice_component *component = (janus_ice_component *)data;
-	if(component == NULL)
-		return FALSE;
-	janus_ice_stream *stream = component->stream;
-	if(!stream)
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)data;
+	if(!pc)
 		goto stoptimer;
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle)
 		goto stoptimer;
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP) ||
-			janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT))
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP) ||
+			janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT))
 		goto stoptimer;
-	if(component->state == NICE_COMPONENT_STATE_CONNECTED || component->state == NICE_COMPONENT_STATE_READY) {
+	if(pc->state == NICE_COMPONENT_STATE_CONNECTED || pc->state == NICE_COMPONENT_STATE_READY) {
 		/* ICE succeeded in the meanwhile, get rid of this timer */
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] ICE succeeded, disabling ICE state check timer!\n", handle->handle_id);
 		goto stoptimer;
 	}
 	/* Still in the failed state, how much time passed since we first detected it? */
-	if(janus_get_monotonic_time() - component->icefailed_detected < 5*G_USEC_PER_SEC) {
+	if(janus_get_monotonic_time() - pc->icefailed_detected < 5*G_USEC_PER_SEC) {
 		/* Let's wait a little longer */
 		return TRUE;
 	}
 	/* If we got here it means the timer expired, and we should check if this is a failure */
-	gboolean trickle_recv = (!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE) || janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALL_TRICKLES));
-	gboolean answer_recv = janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER);
-	gboolean alert_set = janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT);
+	gboolean trickle_recv = (!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE) || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALL_TRICKLES));
+	gboolean answer_recv = janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER);
+	gboolean alert_set = janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT);
 	/* We may still be waiting for something... but we don't wait forever */
 	gboolean do_wait = TRUE;
-	if(janus_get_monotonic_time() - component->icefailed_detected >= 15*G_USEC_PER_SEC) {
+	if(janus_get_monotonic_time() - pc->icefailed_detected >= 15*G_USEC_PER_SEC) {
 		do_wait = FALSE;
 	}
 	if(!do_wait || (handle && trickle_recv && answer_recv && !alert_set)) {
 		/* FIXME Should we really give up for what may be a failure in only one of the media? */
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] ICE failed for component %d in stream %d...\n",
-			handle->handle_id, component->component_id, stream->stream_id);
-		janus_ice_webrtc_hangup(handle, "ICE failed");
+			handle->handle_id, pc->component_id, pc->stream_id);
+		janus_handle_webrtc_hangup(handle, "ICE failed");
 		goto stoptimer;
 	}
 	/* Let's wait a little longer */
 	JANUS_LOG(LOG_WARN, "[%"SCNu64"] ICE failed for component %d in stream %d, but we're still waiting for some info so we don't care... (trickle %s, answer %s, alert %s)\n",
-		handle->handle_id, component->component_id, stream->stream_id,
+		handle->handle_id, pc->component_id, pc->stream_id,
 		trickle_recv ? "received" : "pending",
 		answer_recv ? "received" : "pending",
 		alert_set ? "set" : "not set");
 	return TRUE;
 
 stoptimer:
-	if(component->icestate_source != NULL) {
-		g_source_destroy(component->icestate_source);
-		g_source_unref(component->icestate_source);
-		component->icestate_source = NULL;
+	if(pc->icestate_source != NULL) {
+		g_source_destroy(pc->icestate_source);
+		g_source_unref(pc->icestate_source);
+		pc->icestate_source = NULL;
 	}
 	return FALSE;
 }
 
 /* Callbacks */
 static void janus_ice_cb_candidate_gathering_done(NiceAgent *agent, guint stream_id, gpointer user_data) {
-	janus_ice_handle *handle = (janus_ice_handle *)user_data;
+	janus_handle *handle = (janus_handle *)user_data;
 	if(!handle)
 		return;
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Gathering done for stream %d\n", handle->handle_id, stream_id);
 	handle->cdone++;
-	janus_ice_stream *stream = handle->stream;
-	if(!stream || stream->stream_id != stream_id) {
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc || pc->stream_id != stream_id) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"]  No stream %d??\n", handle->handle_id, stream_id);
 		return;
 	}
-	stream->cdone = 1;
+	pc->cdone = 1;
 	/* If we're doing full-trickle, send an event to the user too */
 	if(janus_full_trickle_enabled) {
 		/* Send a "trickle" event with completed:true to the browser */
@@ -1731,7 +1739,7 @@ static void janus_ice_cb_candidate_gathering_done(NiceAgent *agent, guint stream
 }
 
 static void janus_ice_cb_component_state_changed(NiceAgent *agent, guint stream_id, guint component_id, guint state, gpointer ice) {
-	janus_ice_handle *handle = (janus_ice_handle *)ice;
+	janus_handle *handle = (janus_handle *)ice;
 	if(!handle)
 		return;
 	if(component_id > 1) {
@@ -1740,17 +1748,12 @@ static void janus_ice_cb_component_state_changed(NiceAgent *agent, guint stream_
 	}
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Component state changed for component %d in stream %d: %d (%s)\n",
 		handle->handle_id, component_id, stream_id, state, janus_get_ice_state_name(state));
-	janus_ice_stream *stream = handle->stream;
-	if(!stream || stream->stream_id != stream_id) {
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc || pc->stream_id != stream_id) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No stream %d??\n", handle->handle_id, stream_id);
 		return;
 	}
-	janus_ice_component *component = stream->component;
-	if(!component || component->component_id != component_id) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No component %d in stream %d??\n", handle->handle_id, component_id, stream_id);
-		return;
-	}
-	component->state = state;
+	pc->state = state;
 	/* Notify event handlers */
 	if(janus_events_is_enabled()) {
 		janus_session *session = (janus_session *)handle->session;
@@ -1763,22 +1766,22 @@ static void janus_ice_cb_component_state_changed(NiceAgent *agent, guint stream_
 	/* FIXME Even in case the state is 'connected', we wait for the 'new-selected-pair' callback to do anything */
 	if(state == NICE_COMPONENT_STATE_FAILED) {
 		/* Failed doesn't mean necessarily we need to give up: we may be trickling */
-		gboolean alert_set = janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT);
+		gboolean alert_set = janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT);
 		if(alert_set)
 			return;
-		gboolean trickle_recv = (!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE) || janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALL_TRICKLES));
-		gboolean answer_recv = janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER);
+		gboolean trickle_recv = (!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE) || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALL_TRICKLES));
+		gboolean answer_recv = janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER);
 		JANUS_LOG(LOG_WARN, "[%"SCNu64"] ICE failed for component %d in stream %d, but let's give it some time... (trickle %s, answer %s, alert %s)\n",
 			handle->handle_id, component_id, stream_id,
 			trickle_recv ? "received" : "pending",
 			answer_recv ? "received" : "pending",
 			alert_set ? "set" : "not set");
 		/* In case we haven't started a timer yet, let's do it now */
-		if(component->icestate_source == NULL && component->icefailed_detected == 0) {
-			component->icefailed_detected = janus_get_monotonic_time();
-			component->icestate_source = g_timeout_source_new(500);
-			g_source_set_callback(component->icestate_source, janus_ice_check_failed, component, NULL);
-			guint id = g_source_attach(component->icestate_source, handle->mainctx);
+		if(pc->icestate_source == NULL && pc->icefailed_detected == 0) {
+			pc->icefailed_detected = janus_get_monotonic_time();
+			pc->icestate_source = g_timeout_source_new(500);
+			g_source_set_callback(pc->icestate_source, janus_ice_check_failed, pc, NULL);
+			guint id = g_source_attach(pc->icestate_source, handle->mainctx);
 			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Creating ICE state check timer with ID %u\n", handle->handle_id, id);
 		}
 	}
@@ -1789,7 +1792,7 @@ static void janus_ice_cb_new_selected_pair (NiceAgent *agent, guint stream_id, g
 #else
 static void janus_ice_cb_new_selected_pair (NiceAgent *agent, guint stream_id, guint component_id, NiceCandidate *local, NiceCandidate *remote, gpointer ice) {
 #endif
-	janus_ice_handle *handle = (janus_ice_handle *)ice;
+	janus_handle *handle = (janus_handle *)ice;
 	if(!handle)
 		return;
 	if(component_id > 1) {
@@ -1801,16 +1804,11 @@ static void janus_ice_cb_new_selected_pair (NiceAgent *agent, guint stream_id, g
 #else
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] New selected pair for component %d in stream %d: %s <-> %s\n", handle ? handle->handle_id : 0, component_id, stream_id, local->foundation, remote->foundation);
 #endif
-	janus_ice_stream *stream = handle->stream;
-	if(!stream || stream->stream_id != stream_id) {
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc || pc->stream_id != stream_id) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No stream %d??\n", handle->handle_id, stream_id);
 		return;
 	}
-	janus_ice_component *component = stream->component;
-	if(!component || component->component_id != component_id) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No component %d in stream %d??\n", handle->handle_id, component_id, stream_id);
-		return;
-	}
 	char sp[200];
 #ifndef HAVE_LIBNICE_TCP
 	g_snprintf(sp, 200, "%s <-> %s", local, remote);
@@ -1859,10 +1857,10 @@ static void janus_ice_cb_new_selected_pair (NiceAgent *agent, guint stream_id, g
 		raddress, rport, rtype, remote->transport == NICE_CANDIDATE_TRANSPORT_UDP ? "udp" : "tcp");
 #endif
 	gboolean newpair = FALSE;
-	if(component->selected_pair == NULL || strcmp(sp, component->selected_pair)) {
+	if(pc->selected_pair == NULL || strcmp(sp, pc->selected_pair)) {
 		newpair = TRUE;
-		gchar *prev_selected_pair = component->selected_pair;
-		component->selected_pair = g_strdup(sp);
+		gchar *prev_selected_pair = pc->selected_pair;
+		pc->selected_pair = g_strdup(sp);
 		g_clear_pointer(&prev_selected_pair, g_free);
 	}
 	/* Notify event handlers */
@@ -1893,13 +1891,13 @@ static void janus_ice_cb_new_selected_pair (NiceAgent *agent, guint stream_id, g
 		janus_events_notify_handlers(JANUS_EVENT_TYPE_WEBRTC, session->session_id, handle->handle_id, handle->opaque_id, info);
 	}
 	/* Have we been here before? (might happen, when trickling) */
-	if(component->component_connected > 0)
+	if(pc->ice_connected > 0)
 		return;
 	/* FIXME Clear the queue */
 	janus_ice_clear_queued_packets(handle);
 	/* Now we can start the DTLS handshake (FIXME This was on the 'connected' state notification, before) */
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"]   Component is ready enough, starting DTLS handshake...\n", handle->handle_id);
-	component->component_connected = janus_get_monotonic_time();
+	pc->ice_connected = janus_get_monotonic_time();
 	/* Start the DTLS handshake, at last */
 #if GLIB_CHECK_VERSION(2, 46, 0)
 	g_async_queue_push_front(handle->queued_packets, &janus_ice_dtls_handshake);
@@ -1910,7 +1908,7 @@ static void janus_ice_cb_new_selected_pair (NiceAgent *agent, guint stream_id, g
 }
 
 /* Candidates management */
-static int janus_ice_candidate_to_string(janus_ice_handle *handle, NiceCandidate *c, char *buffer, int buflen, gboolean log_candidate);
+static int janus_ice_candidate_to_string(janus_handle *handle, NiceCandidate *c, char *buffer, int buflen, gboolean log_candidate);
 #ifndef HAVE_LIBNICE_TCP
 static void janus_ice_cb_new_local_candidate (NiceAgent *agent, guint stream_id, guint component_id, gchar *foundation, gpointer ice) {
 #else
@@ -1918,10 +1916,10 @@ static void janus_ice_cb_new_local_candidate (NiceAgent *agent, NiceCandidate *c
 #endif
 	if(!janus_full_trickle_enabled) {
 		/* Ignore if we're not full-trickling: for half-trickle
-		 * janus_ice_candidates_to_sdp() is used instead */
+		 * janus_handle_candidates_to_sdp() is used instead */
 		return;
 	}
-	janus_ice_handle *handle = (janus_ice_handle *)ice;
+	janus_handle *handle = (janus_handle *)ice;
 	if(!handle)
 		return;
 #ifndef HAVE_LIBNICE_TCP
@@ -1952,16 +1950,11 @@ static void janus_ice_cb_new_local_candidate (NiceAgent *agent, NiceCandidate *c
 		/* New remote candidate for a component we don't need anymore (rtcp-mux) */
 		return;
 	}
-	janus_ice_stream *stream = handle->stream;
-	if(!stream || stream->stream_id != stream_id) {
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc || pc->stream_id != stream_id) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No stream %d??\n", handle->handle_id, stream_id);
 		return;
 	}
-	janus_ice_component *component = stream->component;
-	if(!component || component->component_id != component_id) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No component %d in stream %d??\n", handle->handle_id, component_id, stream_id);
-		return;
-	}
 #ifndef HAVE_LIBNICE_TCP
 	/* Get local candidates and look for the related foundation */
 	NiceCandidate *candidate = NULL;
@@ -2003,7 +1996,7 @@ static void janus_ice_cb_new_remote_candidate (NiceAgent *agent, guint stream_id
 #else
 static void janus_ice_cb_new_remote_candidate (NiceAgent *agent, NiceCandidate *candidate, gpointer ice) {
 #endif
-	janus_ice_handle *handle = (janus_ice_handle *)ice;
+	janus_handle *handle = (janus_handle *)ice;
 	if(!handle)
 		return;
 #ifndef HAVE_LIBNICE_TCP
@@ -2034,16 +2027,11 @@ static void janus_ice_cb_new_remote_candidate (NiceAgent *agent, NiceCandidate *
 		/* New remote candidate for a component we don't need anymore (rtcp-mux) */
 		return;
 	}
-	janus_ice_stream *stream = handle->stream;
-	if(!stream || stream->stream_id != stream_id) {
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc || pc->stream_id != stream_id) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No stream %d??\n", handle->handle_id, stream_id);
 		return;
 	}
-	janus_ice_component *component = stream->component;
-	if(!component || component->component_id != component_id) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No component %d in stream %d??\n", handle->handle_id, component_id, stream_id);
-		return;
-	}
 #ifndef HAVE_LIBNICE_TCP
 	/* Get remote candidates and look for the related foundation */
 	NiceCandidate *candidate = NULL;
@@ -2141,7 +2129,7 @@ static void janus_ice_cb_new_remote_candidate (NiceAgent *agent, NiceCandidate *
 	}
 
 	/* Now parse the candidate as if we received it from the Janus API */
-	int res = janus_sdp_parse_candidate(stream, buffer, 1);
+	int res = janus_sdp_parse_candidate(pc, buffer, 1);
 	if(res != 0) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to parse prflx candidate... (%d)\n", handle->handle_id, res);
 	}
@@ -2154,23 +2142,18 @@ static void janus_ice_cb_new_remote_candidate (NiceAgent *agent, NiceCandidate *
 }
 
 static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint component_id, guint len, gchar *buf, gpointer ice) {
-	janus_ice_component *component = (janus_ice_component *)ice;
-	if(!component) {
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)ice;
+	if(!pc) {
 		JANUS_LOG(LOG_ERR, "No component %d in stream %d??\n", component_id, stream_id);
 		return;
 	}
-	janus_ice_stream *stream = component->stream;
-	if(!stream) {
-		JANUS_LOG(LOG_ERR, "No stream %d??\n", stream_id);
-		return;
-	}
-	janus_ice_handle *handle = stream->handle;
+	janus_handle *handle = pc->handle;
 	if(!handle) {
 		JANUS_LOG(LOG_ERR, "No handle for stream %d??\n", stream_id);
 		return;
 	}
 	janus_session *session = (janus_session *)handle->session;
-	if(!component->dtls) {	/* Still waiting for the DTLS stack */
+	if(!pc->dtls) {	/* Still waiting for the DTLS stack */
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Still waiting for the DTLS stack for component %d in stream %d...\n", handle->handle_id, component_id, stream_id);
 		return;
 	}
@@ -2178,130 +2161,119 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 	if(janus_is_dtls(buf) || (!janus_is_rtp(buf, len) && !janus_is_rtcp(buf, len))) {
 		/* This is DTLS: either handshake stuff, or data coming from SCTP DataChannels */
 		JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Looks like DTLS!\n", handle->handle_id);
-		janus_dtls_srtp_incoming_msg(component->dtls, buf, len);
-		/* Update stats (TODO Do the same for the last second window as well) */
-		component->in_stats.data.packets++;
-		component->in_stats.data.bytes += len;
+		janus_dtls_srtp_incoming_msg(pc->dtls, buf, len);
+		/* Update stats */
+		pc->dtls_in_stats.info[0].packets++;
+		pc->dtls_in_stats.info[0].bytes += len;
 		return;
 	}
 	/* Not DTLS... RTP or RTCP? (http://tools.ietf.org/html/rfc5761#section-4) */
 	if(janus_is_rtp(buf, len)) {
 		/* This is RTP */
-		if(janus_is_webrtc_encryption_enabled() && (!component->dtls || !component->dtls->srtp_valid || !component->dtls->srtp_in)) {
+		if(janus_is_webrtc_encryption_enabled() && (!pc->dtls || !pc->dtls->srtp_valid || !pc->dtls->srtp_in)) {
 			JANUS_LOG(LOG_WARN, "[%"SCNu64"]     Missing valid SRTP session (packet arrived too early?), skipping...\n", handle->handle_id);
 		} else {
 			janus_rtp_header *header = (janus_rtp_header *)buf;
 			guint32 packet_ssrc = ntohl(header->ssrc);
-			/* Is this audio or video? */
+			/* Which medium does this refer to? Is this audio or video? */
 			int video = 0, vindex = 0, rtx = 0;
-			/* Bundled streams, check SSRC */
-			video = ((stream->video_ssrc_peer[0] == packet_ssrc
-				|| stream->video_ssrc_peer_rtx[0] == packet_ssrc
-				|| stream->video_ssrc_peer[1] == packet_ssrc
-				|| stream->video_ssrc_peer_rtx[1] == packet_ssrc
-				|| stream->video_ssrc_peer[2] == packet_ssrc
-				|| stream->video_ssrc_peer_rtx[2] == packet_ssrc) ? 1 : 0);
-			if(!video && stream->audio_ssrc_peer != packet_ssrc) {
-				/* Apparently we were not told the peer SSRCs, try the RTP mid extension (or payload types) */
-				gboolean found = FALSE;
-				if(handle->stream->mid_ext_id > 0) {
+			janus_handle_webrtc_medium *medium = g_hash_table_lookup(pc->media_byssrc, GINT_TO_POINTER(packet_ssrc));
+			if(medium == NULL) {
+				/* SSRC not found, try the mid/rid RTP extensions if in use */
+				if(pc->mid_ext_id > 0) {
 					char sdes_item[16];
-					if(janus_rtp_header_extension_parse_mid(buf, len, handle->stream->mid_ext_id, sdes_item, sizeof(sdes_item)) == 0) {
-						if(handle->audio_mid && !strcmp(handle->audio_mid, sdes_item)) {
-							/* It's audio */
-							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Unadvertized SSRC (%"SCNu32") is audio! (mid %s)\n", handle->handle_id, packet_ssrc, sdes_item);
-							video = 0;
-							stream->audio_ssrc_peer = packet_ssrc;
-							found = TRUE;
-						} else if(handle->video_mid && !strcmp(handle->video_mid, sdes_item)) {
-							/* It's video */
-							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Unadvertized SSRC (%"SCNu32") is video! (mid %s)\n", handle->handle_id, packet_ssrc, sdes_item);
-							video = 1;
+					if(janus_rtp_header_extension_parse_mid(buf, len, pc->mid_ext_id, sdes_item, sizeof(sdes_item)) == 0) {
+						medium = g_hash_table_lookup(pc->media_bymid, sdes_item);
+						if(medium != NULL) {
+							/* Found! Associate this SSRC to this stream */
+							JANUS_LOG(LOG_VERB, "[%"SCNu64"] SSRC %"SCNu32" is associated to mid %s\n",
+								handle->handle_id, packet_ssrc, medium->mid);
+							gboolean found = FALSE;
 							/* Check if simulcasting is involved */
-							if(stream->rid[0] == NULL || stream->rid_ext_id < 1) {
-								stream->video_ssrc_peer[0] = packet_ssrc;
+							if(medium->rid[0] == NULL || pc->rid_ext_id < 1) {
+								medium->ssrc_peer[0] = packet_ssrc;
 								found = TRUE;
 							} else {
-								if(janus_rtp_header_extension_parse_rid(buf, len, stream->rid_ext_id, sdes_item, sizeof(sdes_item)) == 0) {
+								if(janus_rtp_header_extension_parse_rid(buf, len, pc->rid_ext_id, sdes_item, sizeof(sdes_item)) == 0) {
 									/* Try the RTP stream ID */
-									if(stream->rid[2] != NULL && !strcmp(stream->rid[2], sdes_item)) {
+									if(medium->rid[2] != NULL && !strcmp(medium->rid[2], sdes_item)) {
 										JANUS_LOG(LOG_VERB, "[%"SCNu64"]  -- Simulcasting: rid=%s\n", handle->handle_id, sdes_item);
-										stream->video_ssrc_peer[0] = packet_ssrc;
-										vindex = 0;
+										medium->ssrc_peer[0] = packet_ssrc;
 										found = TRUE;
-									} else if(stream->rid[1] != NULL && !strcmp(stream->rid[1], sdes_item)) {
+									} else if(medium->rid[1] != NULL && !strcmp(medium->rid[1], sdes_item)) {
 										JANUS_LOG(LOG_VERB, "[%"SCNu64"]  -- Simulcasting #1: rid=%s\n", handle->handle_id, sdes_item);
-										stream->video_ssrc_peer[1] = packet_ssrc;
-										vindex = 1;
+										medium->ssrc_peer[1] = packet_ssrc;
 										found = TRUE;
-									} else if(stream->rid[0] != NULL && !strcmp(stream->rid[0], sdes_item)) {
+									} else if(medium->rid[0] != NULL && !strcmp(medium->rid[0], sdes_item)) {
 										JANUS_LOG(LOG_VERB, "[%"SCNu64"]  -- Simulcasting #2: rid=%s\n", handle->handle_id, sdes_item);
-										stream->video_ssrc_peer[2] = packet_ssrc;
-										vindex = 2;
+										medium->ssrc_peer[2] = packet_ssrc;
 										found = TRUE;
 									} else {
 										JANUS_LOG(LOG_WARN, "[%"SCNu64"]  -- Simulcasting: unknown rid %s..?\n", handle->handle_id, sdes_item);
 									}
-								} else if(stream->ridrtx_ext_id > 0 &&
-										janus_rtp_header_extension_parse_rid(buf, len, stream->ridrtx_ext_id, sdes_item, sizeof(sdes_item)) == 0) {
+								} else if(pc->ridrtx_ext_id > 0 &&
+										janus_rtp_header_extension_parse_rid(buf, len, pc->ridrtx_ext_id, sdes_item, sizeof(sdes_item)) == 0) {
 									/* Try the repaired RTP stream ID */
-									if(stream->rid[2] != NULL && !strcmp(stream->rid[2], sdes_item)) {
+									if(medium->rid[2] != NULL && !strcmp(medium->rid[2], sdes_item)) {
 										JANUS_LOG(LOG_VERB, "[%"SCNu64"]  -- Simulcasting: rid=%s (rtx)\n", handle->handle_id, sdes_item);
-										stream->video_ssrc_peer_rtx[0] = packet_ssrc;
-										vindex = 0;
-										rtx = 1;
+										medium->ssrc_peer_rtx[0] = packet_ssrc;
 										found = TRUE;
-									} else if(stream->rid[1] != NULL && !strcmp(stream->rid[1], sdes_item)) {
+									} else if(medium->rid[1] != NULL && !strcmp(medium->rid[1], sdes_item)) {
 										JANUS_LOG(LOG_VERB, "[%"SCNu64"]  -- Simulcasting #1: rid=%s (rtx)\n", handle->handle_id, sdes_item);
-										stream->video_ssrc_peer_rtx[1] = packet_ssrc;
-										vindex = 1;
-										rtx = 1;
+										medium->ssrc_peer_rtx[1] = packet_ssrc;
 										found = TRUE;
-									} else if(stream->rid[0] != NULL && !strcmp(stream->rid[0], sdes_item)) {
-										JANUS_LOG(LOG_VERB, "[%"SCNu64"]  -- Simulcasting #2: rid=%s (rtx)\n", handle->handle_id, sdes_item);
-										stream->video_ssrc_peer_rtx[2] = packet_ssrc;
-										vindex = 2;
-										rtx = 1;
+									} else if(medium->rid[0] != NULL && !strcmp(medium->rid[0], sdes_item)) {
+										JANUS_LOG(LOG_VERB, "[%"SCNu64"]  -- Simulcasting #1: rid=%s (rtx)\n", handle->handle_id, sdes_item);
+										medium->ssrc_peer_rtx[2] = packet_ssrc;
 										found = TRUE;
 									} else {
 										JANUS_LOG(LOG_WARN, "[%"SCNu64"]  -- Simulcasting: unknown rid %s..?\n", handle->handle_id, sdes_item);
 									}
 								}
 							}
+							if(found) {
+								g_hash_table_insert(pc->media_byssrc, GINT_TO_POINTER(packet_ssrc), medium);
+								janus_refcount_increase(&medium->ref);
+							} else {
+								medium = NULL;
+							}
 						}
 					}
 				}
-				if(!found) {
-					JANUS_LOG(LOG_WARN, "[%"SCNu64"] Not video and not audio? dropping (SSRC %"SCNu32")...\n", handle->handle_id, packet_ssrc);
-					return;
-				}
 			}
+			if(medium == NULL) {
+				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Unknown SSRC, dropping packet (SSRC %"SCNu32")...\n",
+					handle->handle_id, packet_ssrc);
+				return;
+			}
+			video = (medium->type == JANUS_MEDIA_VIDEO);
 			/* Make sure we're prepared to receive this media packet */
-			if((!video && !stream->audio_recv) || (video && !stream->video_recv))
+			if(!medium->recv)
 				return;
 			/* If this is video, check if this is simulcast and/or a retransmission using RFC4588 */
 			if(video) {
-				if(stream->video_ssrc_peer[1] == packet_ssrc) {
+				vindex = 0;
+				if(medium->ssrc_peer[1] == packet_ssrc) {
 					/* FIXME Simulcast (1) */
 					JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Simulcast #1 (SSRC %"SCNu32")...\n", handle->handle_id, packet_ssrc);
 					vindex = 1;
-				} else if(stream->video_ssrc_peer[2] == packet_ssrc) {
+				} else if(medium->ssrc_peer[2] == packet_ssrc) {
 					/* FIXME Simulcast (2) */
 					JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Simulcast #2 (SSRC %"SCNu32")...\n", handle->handle_id, packet_ssrc);
 					vindex = 2;
 				} else {
 					/* Maybe a video retransmission using RFC4588? */
-					if(stream->video_ssrc_peer_rtx[0] == packet_ssrc) {
+					if(medium->ssrc_peer_rtx[0] == packet_ssrc) {
 						rtx = 1;
 						vindex = 0;
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] RFC4588 rtx packet on video (SSRC %"SCNu32")...\n",
 							handle->handle_id, packet_ssrc);
-					} else if(stream->video_ssrc_peer_rtx[1] == packet_ssrc) {
+					} else if(medium->ssrc_peer_rtx[1] == packet_ssrc) {
 						rtx = 1;
 						vindex = 1;
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] RFC4588 rtx packet on video #%d (SSRC %"SCNu32")...\n",
 							handle->handle_id, vindex, packet_ssrc);
-					} else if(stream->video_ssrc_peer_rtx[2] == packet_ssrc) {
+					} else if(medium->ssrc_peer_rtx[2] == packet_ssrc) {
 						rtx = 1;
 						vindex = 2;
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] RFC4588 rtx packet on video #%d (SSRC %"SCNu32")...\n",
@@ -2312,7 +2284,7 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 
 			int buflen = len;
 			srtp_err_status_t res = janus_is_webrtc_encryption_enabled() ?
-				srtp_unprotect(component->dtls->srtp_in, buf, &buflen) : srtp_err_status_ok;
+				srtp_unprotect(pc->dtls->srtp_in, buf, &buflen) : srtp_err_status_ok;
 			if(res != srtp_err_status_ok) {
 				if(res != srtp_err_status_replay_fail && res != srtp_err_status_replay_old) {
 					/* Only print the error if it's not a 'replay fail' or 'replay old' (which is probably just the result of us NACKing a packet) */
@@ -2321,16 +2293,12 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 					JANUS_LOG(LOG_ERR, "[%"SCNu64"]     SRTP unprotect error: %s (len=%d-->%d, ts=%"SCNu32", seq=%"SCNu16")\n", handle->handle_id, janus_srtp_error_str(res), len, buflen, timestamp, seq);
 				}
 			} else {
-				if(video) {
-					if(stream->video_ssrc_peer[0] == 0) {
-						stream->video_ssrc_peer[0] = ntohl(header->ssrc);
-						JANUS_LOG(LOG_VERB, "[%"SCNu64"]     Peer video SSRC: %u\n", handle->handle_id, stream->video_ssrc_peer[0]);
-					}
-				} else {
-					if(stream->audio_ssrc_peer == 0) {
-						stream->audio_ssrc_peer = ntohl(header->ssrc);
-						JANUS_LOG(LOG_VERB, "[%"SCNu64"]     Peer audio SSRC: %u\n", handle->handle_id, stream->audio_ssrc_peer);
-					}
+				if(medium->ssrc_peer[0] == 0) {
+					medium->ssrc_peer[0] = ntohl(header->ssrc);
+					JANUS_LOG(LOG_VERB, "[%"SCNu64"]     Peer #%d (%s) SSRC: %u\n",
+						handle->handle_id, medium->mindex,
+						medium->type == JANUS_MEDIA_VIDEO ? "video" : "audio",
+						medium->ssrc_peer[0]);
 				}
 				/* Do we need to dump this packet for debugging? */
 				if(g_atomic_int_get(&handle->dump_packets))
@@ -2346,8 +2314,8 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 				if(rtx) {
 					/* The original sequence number is in the first two bytes of the payload */
 					/* Rewrite the header with the info from the original packet (payload type, SSRC, sequence number) */
-					header->type = stream->video_payload_type;
-					packet_ssrc = stream->video_ssrc_peer[vindex];
+					header->type = medium->payload_type;
+					packet_ssrc = medium->ssrc_peer[vindex];
 					header->ssrc = htonl(packet_ssrc);
 					if(plen > 0) {
 						memcpy(&header->seq_number, payload, 2);
@@ -2360,50 +2328,50 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 						buf += 2;
 						payload +=2;
 						header = (janus_rtp_header *)buf;
-						if(stream->rid_ext_id > 1 && stream->ridrtx_ext_id > 1) {
+						if(pc->rid_ext_id > 1 && pc->ridrtx_ext_id > 1) {
 							/* Replace the 'repaired' extension ID as well with the 'regular' one */
-							janus_rtp_header_extension_replace_id(buf, buflen, stream->ridrtx_ext_id, stream->rid_ext_id);
+							janus_rtp_header_extension_replace_id(buf, buflen, pc->ridrtx_ext_id, pc->rid_ext_id);
 						}
 					}
 				}
 				/* Check if we need to handle transport wide cc */
-				if(stream->do_transport_wide_cc) {
+				if(pc->do_transport_wide_cc) {
 					guint16 transport_seq_num;
 					/* Get transport wide seq num */
-					if(janus_rtp_header_extension_parse_transport_wide_cc(buf, buflen, stream->transport_wide_cc_ext_id, &transport_seq_num)==0) {
+					if(janus_rtp_header_extension_parse_transport_wide_cc(buf, buflen, pc->transport_wide_cc_ext_id, &transport_seq_num)==0) {
 						/* Get current timestamp */
 						struct timeval now;
 						gettimeofday(&now,0);
 						/* Create <seq num, time> pair */
 						janus_rtcp_transport_wide_cc_stats *stats = g_malloc0(sizeof(janus_rtcp_transport_wide_cc_stats));
 						/* Check if we have a sequence wrap */
-						if(transport_seq_num<0x0FFF && (stream->transport_wide_cc_last_seq_num&0xFFFF)>0xF000) {
+						if(transport_seq_num<0x0FFF && (pc->transport_wide_cc_last_seq_num&0xFFFF)>0xF000) {
 							/* Increase cycles */
-							stream->transport_wide_cc_cycles++;
+							pc->transport_wide_cc_cycles++;
 						}
 						/* Get extended value */
-						guint32 transport_ext_seq_num = stream->transport_wide_cc_cycles<<16 | transport_seq_num;
+						guint32 transport_ext_seq_num = pc->transport_wide_cc_cycles<<16 | transport_seq_num;
 						/* Store last received transport seq num */
-						stream->transport_wide_cc_last_seq_num = transport_seq_num;
+						pc->transport_wide_cc_last_seq_num = transport_seq_num;
 						/* Set stats values */
 						stats->transport_seq_num = transport_ext_seq_num;
 						stats->timestamp = (((guint64)now.tv_sec)*1E6+now.tv_usec);
 						/* Lock and append to received list */
-						janus_mutex_lock(&stream->mutex);
-						stream->transport_wide_received_seq_nums = g_slist_prepend(stream->transport_wide_received_seq_nums, stats);
-						janus_mutex_unlock(&stream->mutex);
+						janus_mutex_lock(&pc->mutex);
+						pc->transport_wide_received_seq_nums = g_slist_prepend(pc->transport_wide_received_seq_nums, stats);
+						janus_mutex_unlock(&pc->mutex);
 					}
 				}
-				if(video) {
+				if(medium->do_nacks) {
 					/* Check if this packet is a duplicate: can happen with RFC4588 */
 					guint16 seqno = ntohs(header->seq_number);
-					int nstate = stream->rtx_nacked[vindex] ?
-						GPOINTER_TO_INT(g_hash_table_lookup(stream->rtx_nacked[vindex], GUINT_TO_POINTER(seqno))) : 0;
+					int nstate = medium->rtx_nacked[vindex] ?
+						GPOINTER_TO_INT(g_hash_table_lookup(medium->rtx_nacked[vindex], GUINT_TO_POINTER(seqno))) : 0;
 					if(nstate == 1) {
 						/* Packet was NACKed and this is the first time we receive it: change state to received */
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Received NACKed packet %"SCNu16" (SSRC %"SCNu32", vindex %d)...\n",
 							handle->handle_id, seqno, packet_ssrc, vindex);
-						g_hash_table_insert(stream->rtx_nacked[vindex], GUINT_TO_POINTER(seqno), GUINT_TO_POINTER(2));
+						g_hash_table_insert(medium->rtx_nacked[vindex], GUINT_TO_POINTER(seqno), GUINT_TO_POINTER(2));
 					} else if(nstate == 2) {
 						/* We already received this packet: drop it */
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Detected duplicate packet %"SCNu16" (SSRC %"SCNu32", vindex %d)...\n",
@@ -2423,45 +2391,31 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 				}
 				/* Backup the RTP header before passing it to the proper RTP switching context */
 				janus_rtp_header backup = *header;
-				if(!video) {
-					if(stream->audio_ssrc_peer_orig == 0)
-						stream->audio_ssrc_peer_orig = packet_ssrc;
-					janus_rtp_header_update(header, &stream->rtp_ctx[0], FALSE, 0);
-					header->ssrc = htonl(stream->audio_ssrc_peer_orig);
-				} else {
-					if(stream->video_ssrc_peer_orig[vindex] == 0)
-						stream->video_ssrc_peer_orig[vindex] = packet_ssrc;
-					janus_rtp_header_update(header, &stream->rtp_ctx[vindex], TRUE, 0);
-					header->ssrc = htonl(stream->video_ssrc_peer_orig[vindex]);
-				}
+				if(medium->ssrc_peer_orig[vindex] == 0)
+					medium->ssrc_peer_orig[vindex] = packet_ssrc;
+				janus_rtp_header_update(header, &medium->rtp_ctx[vindex], medium->type == JANUS_MEDIA_VIDEO);
+				header->ssrc = htonl(medium->ssrc_peer_orig[vindex]);
 				/* Keep track of payload types too */
-				if(!video && stream->audio_payload_type < 0) {
-					stream->audio_payload_type = header->type;
-					if(stream->audio_codec == NULL) {
-						const char *codec = janus_get_codec_from_pt(handle->local_sdp, stream->audio_payload_type);
-						if(codec != NULL)
-							stream->audio_codec = g_strdup(codec);
-					}
-				} else if(video && stream->video_payload_type < 0) {
-					stream->video_payload_type = header->type;
-					if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX) &&
-							stream->rtx_payload_types && g_hash_table_size(stream->rtx_payload_types) > 0) {
-						stream->video_rtx_payload_type = GPOINTER_TO_INT(g_hash_table_lookup(stream->rtx_payload_types, GINT_TO_POINTER(stream->video_payload_type)));
+				if(medium->payload_type < 0) {
+					medium->payload_type = header->type;
+					if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX) &&
+							medium->rtx_payload_types && g_hash_table_size(medium->rtx_payload_types) > 0) {
+						medium->rtx_payload_type = GPOINTER_TO_INT(g_hash_table_lookup(medium->rtx_payload_types, GINT_TO_POINTER(medium->payload_type)));
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Retransmissions will have payload type %d\n",
-							handle->handle_id, stream->video_rtx_payload_type);
+							handle->handle_id, medium->rtx_payload_type);
 					}
-					if(stream->video_codec == NULL) {
-						const char *codec = janus_get_codec_from_pt(handle->local_sdp, stream->video_payload_type);
+					if(medium->codec == NULL) {
+						const char *codec = janus_get_codec_from_pt(handle->local_sdp, medium->payload_type);
 						if(codec != NULL)
-							stream->video_codec = g_strdup(codec);
+							medium->codec = g_strdup(codec);
 					}
-					if(stream->video_is_keyframe == NULL && stream->video_codec != NULL) {
-						if(!strcasecmp(stream->video_codec, "vp8"))
-							stream->video_is_keyframe = &janus_vp8_is_keyframe;
-						else if(!strcasecmp(stream->video_codec, "vp9"))
-							stream->video_is_keyframe = &janus_vp9_is_keyframe;
-						else if(!strcasecmp(stream->video_codec, "h264"))
-							stream->video_is_keyframe = &janus_h264_is_keyframe;
+					if(medium->type == JANUS_MEDIA_VIDEO && medium->video_is_keyframe == NULL && medium->codec != NULL) {
+						if(!strcasecmp(medium->codec, "vp8"))
+							medium->video_is_keyframe = &janus_vp8_is_keyframe;
+						else if(!strcasecmp(medium->codec, "vp9"))
+							medium->video_is_keyframe = &janus_vp9_is_keyframe;
+						else if(!strcasecmp(medium->codec, "h264"))
+							medium->video_is_keyframe = &janus_h264_is_keyframe;
 					}
 				}
 				/* Pass the data to the responsible plugin */
@@ -2469,62 +2423,39 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 				if(plugin && plugin->incoming_rtp && handle->app_handle &&
 						!g_atomic_int_get(&handle->app_handle->stopped) &&
 						!g_atomic_int_get(&handle->destroyed))
-					plugin->incoming_rtp(handle->app_handle, video, buf, buflen);
+					plugin->incoming_rtp(handle->app_handle, medium->mindex, video, buf, buflen);
 				/* Restore the header for the stats (plugins may have messed with it) */
 				*header = backup;
 				/* Update stats (overall data received, and data received in the last second) */
 				if(buflen > 0) {
 					gint64 now = janus_get_monotonic_time();
-					if(!video) {
-						if(component->in_stats.audio.bytes == 0 || component->in_stats.audio.notified_lastsec) {
-							/* We either received our first audio packet, or we started receiving it again after missing more than a second */
-							component->in_stats.audio.notified_lastsec = FALSE;
-							janus_ice_notify_media(handle, FALSE, TRUE);
-						}
-						/* Overall audio data */
-						component->in_stats.audio.packets++;
-						component->in_stats.audio.bytes += buflen;
-						/* Last second audio data */
-						if(component->in_stats.audio.updated == 0)
-							component->in_stats.audio.updated = now;
-						if(now > component->in_stats.audio.updated &&
-								now - component->in_stats.audio.updated >= G_USEC_PER_SEC) {
-							component->in_stats.audio.bytes_lastsec = component->in_stats.audio.bytes_lastsec_temp;
-							component->in_stats.audio.bytes_lastsec_temp = 0;
-							component->in_stats.audio.updated = now;
-						}
-						component->in_stats.audio.bytes_lastsec_temp += buflen;
-					} else {
-						if(component->in_stats.video[vindex].bytes == 0 || component->in_stats.video[vindex].notified_lastsec) {
-							/* We either received our first video packet, or we started receiving it again after missing more than a second */
-							component->in_stats.video[vindex].notified_lastsec = FALSE;
-							janus_ice_notify_media(handle, TRUE, TRUE);
-						}
-						/* Overall video data for this SSRC */
-						component->in_stats.video[vindex].packets++;
-						component->in_stats.video[vindex].bytes += buflen;
-						/* Last second video data for this SSRC */
-						if(component->in_stats.video[vindex].updated == 0)
-							component->in_stats.video[vindex].updated = now;
-						if(now > component->in_stats.video[vindex].updated &&
-								now - component->in_stats.video[vindex].updated >= G_USEC_PER_SEC) {
-							component->in_stats.video[vindex].bytes_lastsec = component->in_stats.video[vindex].bytes_lastsec_temp;
-							component->in_stats.video[vindex].bytes_lastsec_temp = 0;
-							component->in_stats.video[vindex].updated = now;
-						}
-						component->in_stats.video[vindex].bytes_lastsec_temp += buflen;
+					if(medium->in_stats.info[vindex].bytes == 0 || medium->in_stats.info[vindex].notified_lastsec) {
+						/* We either received our first packet, or we started receiving it again after missing more than a second */
+						medium->in_stats.info[vindex].notified_lastsec = FALSE;
+						janus_ice_notify_media(handle, medium->mid, medium->type == JANUS_MEDIA_VIDEO, TRUE);
+					}
+					/* Overall video data for this SSRC */
+					medium->in_stats.info[vindex].packets++;
+					medium->in_stats.info[vindex].bytes += buflen;
+					/* Last second video data for this SSRC */
+					if(medium->in_stats.info[vindex].updated == 0)
+						medium->in_stats.info[vindex].updated = now;
+					if(now > medium->in_stats.info[vindex].updated &&
+							now - medium->in_stats.info[vindex].updated >= G_USEC_PER_SEC) {
+						medium->in_stats.info[vindex].bytes_lastsec = medium->in_stats.info[vindex].bytes_lastsec_temp;
+						medium->in_stats.info[vindex].bytes_lastsec_temp = 0;
+						medium->in_stats.info[vindex].updated = now;
 					}
+					medium->in_stats.info[vindex].bytes_lastsec_temp += buflen;
 				}
 
 				/* Update the RTCP context as well */
-				rtcp_context *rtcp_ctx = video ? stream->video_rtcp_ctx[vindex] : stream->audio_rtcp_ctx;
-				gboolean retransmissions_disabled = (!video && !component->do_audio_nacks) || (video && !component->do_video_nacks);
+				rtcp_context *rtcp_ctx = medium->rtcp_ctx[vindex];
+				gboolean retransmissions_disabled = !medium->do_nacks;
 				janus_rtcp_process_incoming_rtp(rtcp_ctx, buf, buflen,
-						(video && rtx) ? TRUE : FALSE,
-						(video && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX)),
-						retransmissions_disabled
-				);
-
+					(video && rtx),
+					(video && janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX)),
+					retransmissions_disabled);
 				/* Keep track of RTP sequence numbers, in case we need to NACK them */
 				/* 	Note: unsigned int overflow/underflow wraps (defined behavior) */
 				if(retransmissions_disabled) {
@@ -2533,18 +2464,18 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 				}
 				guint16 new_seqn = ntohs(header->seq_number);
 				/* If this is video, check if this is a keyframe: if so, we empty our NACK queue */
-				if(video && stream->video_is_keyframe) {
-					if(stream->video_is_keyframe(payload, plen)) {
+				if(video && medium->video_is_keyframe) {
+					if(medium->video_is_keyframe(payload, plen)) {
 						if(rtcp_ctx && (int16_t)(new_seqn - rtcp_ctx->max_seq_nr) > 0) {
 							JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Keyframe received with a highest sequence number, resetting NACK queue\n", handle->handle_id);
-							janus_seq_list_free(&component->last_seqs_video[vindex]);
+							janus_seq_list_free(&medium->last_seqs[vindex]);
 						}
 					}
 				}
 				guint16 cur_seqn;
 				int last_seqs_len = 0;
-				janus_mutex_lock(&component->mutex);
-				janus_seq_info **last_seqs = video ? &component->last_seqs_video[vindex] : &component->last_seqs_audio;
+				janus_mutex_lock(&medium->mutex);
+				janus_seq_info **last_seqs = &medium->last_seqs[vindex];
 				janus_seq_info *cur_seq = *last_seqs;
 				if(cur_seq) {
 					cur_seq = cur_seq->prev;
@@ -2591,20 +2522,20 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 								handle->handle_id, cur_seq->seq, video ? "video" : "audio", vindex);
 							nacks = g_slist_prepend(nacks, GUINT_TO_POINTER(cur_seq->seq));
 							cur_seq->state = SEQ_NACKED;
-							if(video && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX)) {
+							if(video && janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX)) {
 								/* Keep track of this sequence number, we need to avoid duplicates */
 								JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Tracking NACKed packet %"SCNu16" (SSRC %"SCNu32", vindex %d)...\n",
 									handle->handle_id, cur_seq->seq, packet_ssrc, vindex);
-								if(stream->rtx_nacked[vindex] == NULL)
-									stream->rtx_nacked[vindex] = g_hash_table_new(NULL, NULL);
-								g_hash_table_insert(stream->rtx_nacked[vindex], GUINT_TO_POINTER(cur_seq->seq), GINT_TO_POINTER(1));
+								if(medium->rtx_nacked[vindex] == NULL)
+									medium->rtx_nacked[vindex] = g_hash_table_new(NULL, NULL);
+								g_hash_table_insert(medium->rtx_nacked[vindex], GUINT_TO_POINTER(cur_seq->seq), GINT_TO_POINTER(1));
 								/* We don't track it forever, though: add a timed source to remove it in a few seconds */
-								janus_ice_nacked_packet *np = g_malloc(sizeof(janus_ice_nacked_packet));
-								np->handle = handle;
+								janus_nacked_packet *np = g_malloc(sizeof(janus_nacked_packet));
+								np->medium = medium;
 								np->seq_number = cur_seq->seq;
 								np->vindex = vindex;
 								GSource *timeout_source = g_timeout_source_new_seconds(5);
-								g_source_set_callback(timeout_source, janus_ice_nacked_packet_cleanup, np, (GDestroyNotify)g_free);
+								g_source_set_callback(timeout_source, janus_nacked_packet_cleanup, np, (GDestroyNotify)g_free);
 								g_source_attach(timeout_source, handle->mainctx);
 								g_source_unref(timeout_source);
 							}
@@ -2637,26 +2568,21 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 					if(res > 0) {
 						/* Set the right local and remote SSRC in the RTCP packet */
 						janus_rtcp_fix_ssrc(NULL, nackbuf, res, 1,
-							video ? stream->video_ssrc : stream->audio_ssrc,
-							video ? stream->video_ssrc_peer[vindex] : stream->audio_ssrc_peer);
-						janus_ice_relay_rtcp_internal(handle, video, nackbuf, res, FALSE);
+							medium->ssrc, medium->ssrc_peer[vindex]);
+						janus_ice_relay_rtcp_internal(handle, medium->mindex, video, nackbuf, res, FALSE);
 					}
 					/* Update stats */
-					component->nack_sent_recent_cnt += nacks_count;
-					if(video) {
-						component->out_stats.video[vindex].nacks += nacks_count;
-					} else {
-						component->out_stats.audio.nacks += nacks_count;
-					}
+					medium->nack_sent_recent_cnt += nacks_count;
+					medium->out_stats.info[vindex].nacks += nacks_count;
 				}
-				if(component->nack_sent_recent_cnt &&
-						(now - component->nack_sent_log_ts) > 5*G_USEC_PER_SEC) {
+				if(medium->nack_sent_recent_cnt &&
+						(now - medium->nack_sent_log_ts) > 5*G_USEC_PER_SEC) {
 					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Sent NACKs for %u missing packets (%s stream #%d)\n",
-						handle->handle_id, component->nack_sent_recent_cnt, video ? "video" : "audio", vindex);
-					component->nack_sent_recent_cnt = 0;
-					component->nack_sent_log_ts = now;
+						handle->handle_id, medium->nack_sent_recent_cnt, video ? "video" : "audio", vindex);
+					medium->nack_sent_recent_cnt = 0;
+					medium->nack_sent_log_ts = now;
 				}
-				janus_mutex_unlock(&component->mutex);
+				janus_mutex_unlock(&medium->mutex);
 				g_slist_free(nacks);
 				nacks = NULL;
 			}
@@ -2665,12 +2591,12 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 	} else if(janus_is_rtcp(buf, len)) {
 		/* This is RTCP */
 		JANUS_LOG(LOG_HUGE, "[%"SCNu64"]  Got an RTCP packet\n", handle->handle_id);
-		if(janus_is_webrtc_encryption_enabled() && (!component->dtls || !component->dtls->srtp_valid || !component->dtls->srtp_in)) {
+		if(janus_is_webrtc_encryption_enabled() && (!pc->dtls || !pc->dtls->srtp_valid || !pc->dtls->srtp_in)) {
 			JANUS_LOG(LOG_WARN, "[%"SCNu64"]     Missing valid SRTP session (packet arrived too early?), skipping...\n", handle->handle_id);
 		} else {
 			int buflen = len;
 			srtp_err_status_t res = janus_is_webrtc_encryption_enabled() ?
-				srtp_unprotect_rtcp(component->dtls->srtp_in, buf, &buflen) : srtp_err_status_ok;
+				srtp_unprotect_rtcp(pc->dtls->srtp_in, buf, &buflen) : srtp_err_status_ok;
 			if(res != srtp_err_status_ok) {
 				JANUS_LOG(LOG_ERR, "[%"SCNu64"]     SRTCP unprotect error: %s (len=%d-->%d)\n", handle->handle_id, janus_srtp_error_str(res), len, buflen);
 			} else {
@@ -2682,71 +2608,38 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 				if(janus_rtcp_has_bye(buf, buflen)) {
 					/* Note: we used to use this as a trigger to close the PeerConnection, but not anymore
 					 * Discussion here, https://groups.google.com/forum/#!topic/meetecho-janus/4XtfbYB7Jvc */
-					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Got RTCP BYE on stream %u (component %u)\n", handle->handle_id, stream->stream_id, component->component_id);
+					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Got RTCP BYE on stream %u (component %u)\n", handle->handle_id, pc->stream_id, pc->component_id);
 				}
 				/* Is this audio or video? */
 				int video = 0, vindex = 0;
-				/* Bundled streams, should we check the SSRCs? */
-				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO)) {
-					/* No audio has been negotiated, definitely video */
-					JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Incoming RTCP, bundling: this is video (no audio has been negotiated)\n", handle->handle_id);
-					video = 1;
-				} else if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO)) {
-					/* No video has been negotiated, definitely audio */
-					JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Incoming RTCP, bundling: this is audio (no video has been negotiated)\n", handle->handle_id);
-					video = 0;
-				} else {
-					if(stream->audio_ssrc_peer == 0 || stream->video_ssrc_peer[0] == 0) {
-						/* We don't know the remote SSRC: this can happen for recvonly clients
-						 * (see https://groups.google.com/forum/#!topic/discuss-webrtc/5yuZjV7lkNc)
-						 * Check the local SSRC, compare it to what we have */
-						guint32 rtcp_ssrc = janus_rtcp_get_receiver_ssrc(buf, buflen);
-						if(rtcp_ssrc == 0) {
-							/* No SSRC, maybe an empty RR? */
-							return;
-						}
-						if(rtcp_ssrc == stream->audio_ssrc) {
-							video = 0;
-						} else if(rtcp_ssrc == stream->video_ssrc) {
-							video = 1;
-						} else if(janus_rtcp_has_fir(buf, buflen) || janus_rtcp_has_pli(buf, buflen) || janus_rtcp_get_remb(buf, buflen)) {
-							/* Mh, no SR or RR? Try checking if there's any FIR, PLI or REMB */
-							video = 1;
-						} else {
-							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Dropping RTCP packet with unknown SSRC (%"SCNu32")\n", handle->handle_id, rtcp_ssrc);
-							return;
+				guint32 rtcp_ssrc = janus_rtcp_get_sender_ssrc(buf, buflen);
+				janus_handle_webrtc_medium *medium = g_hash_table_lookup(pc->media_byssrc, GINT_TO_POINTER(rtcp_ssrc));
+				if(medium == NULL) {
+					/* We don't know the remote SSRC: this can happen for recvonly clients
+					 * (see https://groups.google.com/forum/#!topic/discuss-webrtc/5yuZjV7lkNc)
+					 * Check the local SSRC, compare it to what we have */
+					rtcp_ssrc = janus_rtcp_get_receiver_ssrc(buf, buflen);
+					medium = g_hash_table_lookup(pc->media_byssrc, GINT_TO_POINTER(rtcp_ssrc));
+					if(medium == NULL) {
+						if(rtcp_ssrc > 0) {
+							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Unknown SSRC, dropping RTCP packet (SSRC %"SCNu32")...\n",
+								handle->handle_id, rtcp_ssrc);
 						}
-						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Incoming RTCP, bundling: this is %s (local SSRC: video=%"SCNu32", audio=%"SCNu32", got %"SCNu32")\n",
-							handle->handle_id, video ? "video" : "audio", stream->video_ssrc, stream->audio_ssrc, rtcp_ssrc);
-					} else {
-						/* Check the remote SSRC, compare it to what we have: in case
-						 * we're simulcasting, let's compare to the other SSRCs too */
-						guint32 rtcp_ssrc = janus_rtcp_get_sender_ssrc(buf, buflen);
-						if(rtcp_ssrc == 0) {
-							/* No SSRC, maybe an empty RR? */
-							return;
-						}
-						if(rtcp_ssrc == stream->audio_ssrc_peer) {
-							video = 0;
-						} else if(rtcp_ssrc == stream->video_ssrc_peer[0]) {
-							video = 1;
-						} else if(stream->video_ssrc_peer[1] && rtcp_ssrc == stream->video_ssrc_peer[1]) {
-							video = 1;
-							vindex = 1;
-						} else if(stream->video_ssrc_peer[2] && rtcp_ssrc == stream->video_ssrc_peer[2]) {
-							video = 1;
-							vindex = 2;
-						} else {
-							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Dropping RTCP packet with unknown SSRC (%"SCNu32")\n", handle->handle_id, rtcp_ssrc);
-							return;
-						}
-						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Incoming RTCP, bundling: this is %s (remote SSRC: video=%"SCNu32" #%d, audio=%"SCNu32", got %"SCNu32")\n",
-							handle->handle_id, video ? "video" : "audio", stream->video_ssrc_peer[vindex], vindex, stream->audio_ssrc_peer, rtcp_ssrc);
+						return;
+					}
+				}
+				video = (medium->type == JANUS_MEDIA_VIDEO);
+				/* If this is video, check if this is simulcast */
+				if(video) {
+					if(medium->ssrc_peer[1] == rtcp_ssrc) {
+						vindex = 1;
+					} else if(medium->ssrc_peer[2] == rtcp_ssrc) {
+						vindex = 2;
 					}
 				}
 
 				/* Let's process this RTCP (compound?) packet, and update the RTCP context for this stream in case */
-				rtcp_context *rtcp_ctx = video ? stream->video_rtcp_ctx[vindex] : stream->audio_rtcp_ctx;
+				rtcp_context *rtcp_ctx = medium->rtcp_ctx[vindex];
 				if (janus_rtcp_parse(rtcp_ctx, buf, buflen) < 0) {
 					/* Drop the packet if the parsing function returns with an error */
 					return;
@@ -2757,13 +2650,13 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 				gint64 now = janus_get_monotonic_time();
 				GSList *nacks = janus_rtcp_get_nacks(buf, buflen);
 				guint nacks_count = g_slist_length(nacks);
-				if(nacks_count && ((!video && component->do_audio_nacks) || (video && component->do_video_nacks))) {
+				if(nacks_count && (medium->do_nacks)) {
 					/* Handle NACK */
 					JANUS_LOG(LOG_HUGE, "[%"SCNu64"]     Just got some NACKS (%d) we should handle...\n", handle->handle_id, nacks_count);
-					GHashTable *retransmit_seqs = (video ? component->video_retransmit_seqs : component->audio_retransmit_seqs);
+					GHashTable *retransmit_seqs = medium->retransmit_seqs;
 					GSList *list = (retransmit_seqs != NULL ? nacks : NULL);
 					int retransmits_cnt = 0;
-					janus_mutex_lock(&component->mutex);
+					janus_mutex_lock(&medium->mutex);
 					while(list) {
 						unsigned int seqnr = GPOINTER_TO_UINT(list->data);
 						JANUS_LOG(LOG_DBG, "[%"SCNu64"]   >> %u\n", handle->handle_id, seqnr);
@@ -2785,6 +2678,7 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 							retransmits_cnt++;
 							/* Enqueue it */
 							janus_ice_queued_packet *pkt = g_malloc(sizeof(janus_ice_queued_packet));
+							pkt->mindex = medium->mindex;
 							pkt->data = g_malloc(p->length+SRTP_MAX_TAG_LEN);
 							memcpy(pkt->data, p->data, p->length);
 							pkt->length = p->length;
@@ -2794,17 +2688,17 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 							pkt->label = NULL;
 							pkt->added = janus_get_monotonic_time();
 							/* What to send and how depends on whether we're doing RFC4588 or not */
-							if(!video || !janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX)) {
+							if(!video || !janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX)) {
 								/* We're not: just clarify the packet was already encrypted before */
 								pkt->encrypted = TRUE;
 							} else {
 								/* We are: overwrite the RTP header (which means we'll need a new SRTP encrypt) */
 								pkt->encrypted = FALSE;
 								janus_rtp_header *header = (janus_rtp_header *)pkt->data;
-								header->type = stream->video_rtx_payload_type;
-								header->ssrc = htonl(stream->video_ssrc_rtx);
-								component->rtx_seq_number++;
-								header->seq_number = htons(component->rtx_seq_number);
+								header->type = medium->rtx_payload_type;
+								header->ssrc = htonl(medium->ssrc_rtx);
+								medium->rtx_seq_number++;
+								header->seq_number = htons(medium->rtx_seq_number);
 							}
 							if(handle->queued_packets != NULL) {
 #if GLIB_CHECK_VERSION(2, 46, 0)
@@ -2820,34 +2714,30 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 						}
 						list = list->next;
 					}
-					component->retransmit_recent_cnt += retransmits_cnt;
+					medium->retransmit_recent_cnt += retransmits_cnt;
 					/* FIXME Remove the NACK compound packet, we've handled it */
 					buflen = janus_rtcp_remove_nacks(buf, buflen);
 					/* Update stats */
-					if(video) {
-						component->in_stats.video[vindex].nacks += nacks_count;
-					} else {
-						component->in_stats.audio.nacks += nacks_count;
-					}
-					janus_mutex_unlock(&component->mutex);
+					medium->in_stats.info[vindex].nacks += nacks_count;
+					janus_mutex_unlock(&medium->mutex);
 					g_slist_free(nacks);
 					nacks = NULL;
 				}
-				if(component->retransmit_recent_cnt &&
-						now - component->retransmit_log_ts > 5*G_USEC_PER_SEC) {
+				if(medium->retransmit_recent_cnt &&
+						now - medium->retransmit_log_ts > 5*G_USEC_PER_SEC) {
 					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Retransmitted %u packets due to NACK (%s stream #%d)\n",
-						handle->handle_id, component->retransmit_recent_cnt, video ? "video" : "audio", vindex);
-					component->retransmit_recent_cnt = 0;
-					component->retransmit_log_ts = now;
+						handle->handle_id, medium->retransmit_recent_cnt, video ? "video" : "audio", vindex);
+					medium->retransmit_recent_cnt = 0;
+					medium->retransmit_log_ts = now;
 				}
 
 				/* Fix packet data for RTCP SR and RTCP RR */
-				janus_rtp_switching_context *rtp_ctx = video ? &stream->rtp_ctx[vindex] : &stream->rtp_ctx[0];
-				uint32_t base_ts = video ? rtp_ctx->v_base_ts : rtp_ctx->a_base_ts;
-				uint32_t base_ts_prev = video ? rtp_ctx->v_base_ts_prev : rtp_ctx->a_base_ts_prev;
-				uint32_t ssrc_peer = video ? stream->video_ssrc_peer_orig[vindex] : stream->audio_ssrc_peer_orig;
-				uint32_t ssrc_local = video ? stream->video_ssrc : stream->audio_ssrc;
-				uint32_t ssrc_expected = video ? rtp_ctx->v_last_ssrc : rtp_ctx->a_last_ssrc;
+				janus_rtp_switching_context *rtp_ctx = &medium->rtp_ctx[vindex];
+				uint32_t base_ts = rtp_ctx->base_ts;
+				uint32_t base_ts_prev = rtp_ctx->base_ts_prev;
+				uint32_t ssrc_peer = medium->ssrc_peer_orig[vindex];
+				uint32_t ssrc_local = medium->ssrc;
+				uint32_t ssrc_expected = rtp_ctx->last_ssrc;
 				if (janus_rtcp_fix_report_data(buf, buflen, base_ts, base_ts_prev, ssrc_peer, ssrc_local, ssrc_expected, video) < 0) {
 					/* Drop packet in case of parsing error or SSRC different from the one expected. */
 					/* This might happen at the very beginning of the communication or early after */
@@ -2859,23 +2749,23 @@ static void janus_ice_cb_nice_recv(NiceAgent *agent, guint stream_id, guint comp
 				if(plugin && plugin->incoming_rtcp && handle->app_handle &&
 						!g_atomic_int_get(&handle->app_handle->stopped) &&
 						!g_atomic_int_get(&handle->destroyed))
-					plugin->incoming_rtcp(handle->app_handle, video, buf, buflen);
+					plugin->incoming_rtcp(handle->app_handle, medium->mindex, video, buf, buflen);
 			}
 		}
 		return;
 	} else {
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Not RTP and not RTCP... may these be data channels?\n", handle->handle_id);
-		janus_dtls_srtp_incoming_msg(component->dtls, buf, len);
+		janus_dtls_srtp_incoming_msg(pc->dtls, buf, len);
 		/* Update stats (only overall data received) */
 		if(len > 0) {
-			component->in_stats.data.packets++;
-			component->in_stats.data.bytes += len;
+			pc->dtls_in_stats.info[0].packets++;
+			pc->dtls_in_stats.info[0].bytes += len;
 		}
 		return;
 	}
 }
 
-void janus_ice_incoming_data(janus_ice_handle *handle, char *label, char *buffer, int length) {
+void janus_ice_incoming_data(janus_handle *handle, char *label, char *buffer, int length) {
 	if(handle == NULL || buffer == NULL || length <= 0)
 		return;
 	janus_plugin *plugin = (janus_plugin *)handle->app;
@@ -2887,15 +2777,12 @@ void janus_ice_incoming_data(janus_ice_handle *handle, char *label, char *buffer
 
 
 /* Helper: encoding local candidates to string/SDP */
-static int janus_ice_candidate_to_string(janus_ice_handle *handle, NiceCandidate *c, char *buffer, int buflen, gboolean log_candidate) {
+static int janus_ice_candidate_to_string(janus_handle *handle, NiceCandidate *c, char *buffer, int buflen, gboolean log_candidate) {
 	if(!handle || !handle->agent || !c || !buffer || buflen < 1)
 		return -1;
-	janus_ice_stream *stream = handle->stream;
-	if(!stream)
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc)
 		return -2;
-	janus_ice_component *component = stream->component;
-	if(!component)
-		return -3;
 	char *host_ip = NULL;
 	if(nat_1_1_enabled) {
 		/* A 1:1 NAT mapping was specified, overwrite all the host addresses with the public IP */
@@ -3030,40 +2917,35 @@ static int janus_ice_candidate_to_string(janus_ice_handle *handle, NiceCandidate
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"]     %s\n", handle->handle_id, buffer);
 	if(log_candidate) {
 		/* Save for the summary, in case we need it */
-		component->local_candidates = g_slist_append(component->local_candidates, g_strdup(buffer));
+		pc->local_candidates = g_slist_append(pc->local_candidates, g_strdup(buffer));
 		/* Notify event handlers */
 		if(janus_events_is_enabled()) {
 			janus_session *session = (janus_session *)handle->session;
 			json_t *info = json_object();
 			json_object_set_new(info, "local-candidate", json_string(buffer));
-			json_object_set_new(info, "stream_id", json_integer(stream->stream_id));
-			json_object_set_new(info, "component_id", json_integer(component->component_id));
+			json_object_set_new(info, "stream_id", json_integer(pc->stream_id));
+			json_object_set_new(info, "component_id", json_integer(pc->component_id));
 			janus_events_notify_handlers(JANUS_EVENT_TYPE_WEBRTC, session->session_id, handle->handle_id, handle->opaque_id, info);
 		}
 	}
 	return 0;
 }
 
-void janus_ice_candidates_to_sdp(janus_ice_handle *handle, janus_sdp_mline *mline, guint stream_id, guint component_id) {
+void janus_handle_candidates_to_sdp(janus_handle *handle, janus_sdp_mline *mline, guint stream_id, guint component_id) {
 	if(!handle || !handle->agent || !mline)
 		return;
-	janus_ice_stream *stream = handle->stream;
-	if(!stream || stream->stream_id != stream_id) {
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc || pc->stream_id != stream_id) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No stream %d??\n", handle->handle_id, stream_id);
 		return;
 	}
-	janus_ice_component *component = stream->component;
-	if(!component || component->component_id != component_id) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"]     No component %d in stream %d??\n", handle->handle_id, component_id, stream_id);
-		return;
-	}
 	NiceAgent *agent = handle->agent;
 	/* Iterate on all */
 	gchar buffer[200];
 	GSList *candidates, *i;
 	candidates = nice_agent_get_local_candidates (agent, stream_id, component_id);
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] We have %d candidates for Stream #%d, Component #%d\n", handle->handle_id, g_slist_length(candidates), stream_id, component_id);
-	gboolean log_candidates = (component->local_candidates == NULL);
+	gboolean log_candidates = (pc->local_candidates == NULL);
 	for(i = candidates; i; i = i->next) {
 		NiceCandidate *c = (NiceCandidate *) i->data;
 		if(janus_ice_candidate_to_string(handle, c, buffer, sizeof(buffer), log_candidates) == 0) {
@@ -3081,35 +2963,30 @@ void janus_ice_candidates_to_sdp(janus_ice_handle *handle, janus_sdp_mline *mlin
 	g_slist_free(candidates);
 }
 
-void janus_ice_setup_remote_candidates(janus_ice_handle *handle, guint stream_id, guint component_id) {
+void janus_handle_setup_remote_candidates(janus_handle *handle, guint stream_id, guint component_id) {
 	if(!handle || !handle->agent)
 		return;
-	janus_ice_stream *stream = handle->stream;
-	if(!stream || stream->stream_id != stream_id) {
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc || pc->stream_id != stream_id) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] No such stream %d: cannot setup remote candidates for component %d\n", handle->handle_id, stream_id, component_id);
 		return;
 	}
-	janus_ice_component *component = stream->component;
-	if(!component || component->component_id != component_id) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"] No such component %d in stream %d: cannot setup remote candidates\n", handle->handle_id, component_id, stream_id);
-		return;
-	}
-	if(component->process_started) {
+	if(pc->process_started) {
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Component %d in stream %d has already been set up\n", handle->handle_id, component_id, stream_id);
 		return;
 	}
-	if(!component->candidates || !component->candidates->data) {
-		if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE)
-				|| janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALL_TRICKLES)) {
+	if(!pc->candidates || !pc->candidates->data) {
+		if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE)
+				|| janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALL_TRICKLES)) {
 			JANUS_LOG(LOG_ERR, "[%"SCNu64"] No remote candidates for component %d in stream %d: was the remote SDP parsed?\n", handle->handle_id, component_id, stream_id);
 		}
 		return;
 	}
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] ## Setting remote candidates: stream %d, component %d (%u in the list)\n",
-		handle->handle_id, stream_id, component_id, g_slist_length(component->candidates));
+		handle->handle_id, stream_id, component_id, g_slist_length(pc->candidates));
 	/* Add all candidates */
 	NiceCandidate *c = NULL;
-	GSList *gsc = component->candidates;
+	GSList *gsc = pc->candidates;
 	gchar *rufrag = NULL, *rpwd = NULL;
 	while(gsc) {
 		c = (NiceCandidate *) gsc->data;
@@ -3128,61 +3005,44 @@ void janus_ice_setup_remote_candidates(janus_ice_handle *handle, guint stream_id
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"]   Password:   %s\n", handle->handle_id, c->password);
 		gsc = gsc->next;
 	}
-	gint added = nice_agent_set_remote_candidates(handle->agent, stream_id, component_id, component->candidates);
-	if(added < 0 || (guint)added < g_slist_length(component->candidates)) {
+	gint added = nice_agent_set_remote_candidates(handle->agent, stream_id, component_id, pc->candidates);
+	if(added < 0 || (guint)added < g_slist_length(pc->candidates)) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to set remote candidates :-( (added %u, expected %u)\n",
-			handle->handle_id, (guint)added, g_slist_length(component->candidates));
+			handle->handle_id, (guint)added, g_slist_length(pc->candidates));
 	} else {
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Remote candidates set!\n", handle->handle_id);
-		component->process_started = TRUE;
+		pc->process_started = TRUE;
 	}
 }
 
-int janus_ice_setup_local(janus_ice_handle *handle, int offer, int audio, int video, int data, int trickle) {
+int janus_handle_setup_local(janus_handle *handle, gboolean offer, gboolean trickle) {
 	if(!handle)
 		return -1;
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AGENT)) {
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_HAS_AGENT)) {
 		JANUS_LOG(LOG_WARN, "[%"SCNu64"] Agent already exists?\n", handle->handle_id);
 		return -2;
 	}
-	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Setting ICE locally: got %s (%d audios, %d videos)\n", handle->handle_id, offer ? "OFFER" : "ANSWER", audio, video);
+	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Setting ICE locally: got %s)\n",
+		handle->handle_id, offer ? "OFFER" : "ANSWER");
 	g_atomic_int_set(&handle->closepc, 0);
-	janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AGENT);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_START);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEGOTIATED);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ICE_RESTART);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RESEND_TRICKLES);
-
-	/* Note: in case this is not an OFFER, we don't know whether any medium are supported on the other side or not yet */
-	if(audio) {
-		janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO);
-	} else {
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO);
-	}
-	if(video) {
-		janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO);
-	} else {
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO);
-	}
-	if(data) {
-		janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS);
-	} else {
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS);
-	}
+	janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_HAS_AGENT);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_START);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEGOTIATED);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ICE_RESTART);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RESEND_TRICKLES);
+
 	/* Note: in case this is not an OFFER, we don't know whether ICE trickling is supported on the other side or not yet */
 	if(offer && trickle) {
-		janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE);
+		janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE);
 	} else {
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE);
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE);
 	}
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALL_TRICKLES);
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE_SYNCED);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALL_TRICKLES);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE_SYNCED);
 
 	/* Note: NICE_COMPATIBILITY_RFC5245 is only available in more recent versions of libnice */
 	handle->controlling = janus_ice_lite_enabled ? FALSE : !offer;
@@ -3312,56 +3172,16 @@ int janus_ice_setup_local(janus_ice_handle *handle, int offer, int audio, int vi
 
 	handle->cdone = 0;
 	handle->stream_id = 0;
-	/* If this is our first offer, let's generate some mids */
-	if(!offer) {
-		if(audio) {
-			if(handle->audio_mid == NULL)
-				handle->audio_mid = g_strdup("audio");
-			if(handle->stream_mid == NULL)
-				handle->stream_mid = handle->audio_mid;
-		}
-		if(video) {
-			if(handle->video_mid == NULL)
-				handle->video_mid = g_strdup("video");
-			if(handle->stream_mid == NULL)
-				handle->stream_mid = handle->video_mid;
-		}
-#ifdef HAVE_SCTP
-		if(data) {
-			if(handle->data_mid == NULL)
-				handle->data_mid = g_strdup("data");
-			if(handle->stream_mid == NULL)
-				handle->stream_mid = handle->data_mid;
-		}
-#endif
-	}
 	/* Now create an ICE stream for all the media we'll handle */
 	handle->stream_id = nice_agent_add_stream(handle->agent, 1);
-	janus_ice_stream *stream = g_malloc0(sizeof(janus_ice_stream));
-	janus_refcount_init(&stream->ref, janus_ice_stream_free);
+	janus_handle_webrtc *pc = g_malloc0(sizeof(janus_handle_webrtc));
+	janus_refcount_init(&pc->ref, janus_handle_webrtc_free);
 	janus_refcount_increase(&handle->ref);
-	stream->stream_id = handle->stream_id;
-	stream->handle = handle;
-	stream->audio_payload_type = -1;
-	stream->video_payload_type = -1;
-	stream->video_rtx_payload_type = -1;
+	pc->stream_id = handle->stream_id;
+	pc->handle = handle;
 	/* FIXME By default, if we're being called we're DTLS clients, but this may be changed by ICE... */
-	stream->dtls_role = offer ? JANUS_DTLS_ROLE_CLIENT : JANUS_DTLS_ROLE_ACTPASS;
-	if(audio) {
-		stream->audio_ssrc = janus_random_uint32();	/* FIXME Should we look for conflicts? */
-		stream->audio_rtcp_ctx = g_malloc0(sizeof(janus_rtcp_context));
-		stream->audio_rtcp_ctx->tb = 48000;	/* May change later */
-	}
-	if(video) {
-		stream->video_ssrc = janus_random_uint32();	/* FIXME Should we look for conflicts? */
-		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX)) {
-			/* Create an SSRC for RFC4588 as well */
-			stream->video_ssrc_rtx = janus_random_uint32();	/* FIXME Should we look for conflicts? */
-		}
-		stream->video_rtcp_ctx[0] = g_malloc0(sizeof(janus_rtcp_context));
-		stream->video_rtcp_ctx[0]->tb = 90000;
-	}
-	janus_mutex_init(&stream->mutex);
+	pc->dtls_role = offer ? JANUS_DTLS_ROLE_CLIENT : JANUS_DTLS_ROLE_ACTPASS;
+	janus_mutex_init(&pc->mutex);
 	if(!have_turnrest_credentials) {
 		/* No TURN REST API server and credentials, any static ones? */
 		if(janus_turn_server != NULL) {
@@ -3391,27 +3211,19 @@ int janus_ice_setup_local(janus_ice_handle *handle, int offer, int audio, int vi
 		}
 #endif
 	}
-	handle->stream = stream;
-	janus_ice_component *component = g_malloc0(sizeof(janus_ice_component));
-	janus_refcount_init(&component->ref, janus_ice_component_free);
-	component->stream = stream;
-	janus_refcount_increase(&stream->ref);
-	component->stream_id = stream->stream_id;
-	component->component_id = 1;
-	janus_mutex_init(&component->mutex);
-	stream->component = component;
+	handle->pc = pc;
 #ifdef HAVE_PORTRANGE
 	/* FIXME: libnice supports this since 0.1.0, but the 0.1.3 on Fedora fails with an undefined reference! */
 	nice_agent_set_port_range(handle->agent, handle->stream_id, 1, rtp_range_min, rtp_range_max);
 #endif
 	if(!nice_agent_gather_candidates(handle->agent, handle->stream_id)) {
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error gathering candidates...\n", handle->handle_id);
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AGENT);
-		janus_ice_webrtc_hangup(handle, "Gathering error");
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_HAS_AGENT);
+		janus_handle_webrtc_hangup(handle, "Gathering error");
 		return -1;
 	}
 	nice_agent_attach_recv(handle->agent, handle->stream_id, 1, g_main_loop_get_context(handle->mainloop),
-		janus_ice_cb_nice_recv, component);
+		janus_ice_cb_nice_recv, pc);
 #ifdef HAVE_TURNRESTAPI
 	if(turnrest_credentials != NULL) {
 		janus_turnrest_response_destroy(turnrest_credentials);
@@ -3419,45 +3231,48 @@ int janus_ice_setup_local(janus_ice_handle *handle, int offer, int audio, int vi
 	}
 #endif
 	/* Create DTLS-SRTP context, at last */
-	component->dtls = janus_dtls_srtp_create(component, stream->dtls_role);
-	if(!component->dtls) {
+	pc->dtls = janus_dtls_srtp_create(pc, pc->dtls_role);
+	if(!pc->dtls) {
 		/* FIXME We should clear some resources... */
 		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error creating DTLS-SRTP stack...\n", handle->handle_id);
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AGENT);
-		janus_ice_webrtc_hangup(handle, "DTLS-SRTP stack error");
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_HAS_AGENT);
+		janus_handle_webrtc_hangup(handle, "DTLS-SRTP stack error");
 		return -1;
 	}
-	janus_refcount_increase(&component->dtls->ref);
+	janus_refcount_increase(&pc->dtls->ref);
+	/* Create the media instances we need */
+	pc->media = g_hash_table_new_full(NULL, NULL, NULL, (GDestroyNotify)janus_handle_webrtc_medium_destroy);
+	pc->media_byssrc = g_hash_table_new_full(NULL, NULL, NULL, (GDestroyNotify)janus_handle_webrtc_medium_dereference);
+	pc->media_bymid = g_hash_table_new_full(g_str_hash, g_str_equal,
+		(GDestroyNotify)g_free, (GDestroyNotify)janus_handle_webrtc_medium_dereference);
+	pc->media_bytype = g_hash_table_new_full(NULL, NULL, NULL, (GDestroyNotify)janus_handle_webrtc_medium_dereference);
 	return 0;
 }
 
-void janus_ice_restart(janus_ice_handle *handle) {
-	if(!handle || !handle->agent || !handle->stream)
+void janus_handle_ice_restart(janus_handle *handle) {
+	if(!handle || !handle->agent || !handle->pc)
 		return;
 	/* Restart ICE */
 	if(nice_agent_restart(handle->agent) == FALSE) {
 		JANUS_LOG(LOG_WARN, "[%"SCNu64"] ICE restart failed...\n", handle->handle_id);
 	}
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ICE_RESTART);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ICE_RESTART);
 }
 
-void janus_ice_resend_trickles(janus_ice_handle *handle) {
+void janus_handle_resend_trickles(janus_handle *handle) {
 	if(!handle || !handle->agent)
 		return;
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RESEND_TRICKLES);
-	janus_ice_stream *stream = handle->stream;
-	if(!stream)
-		return;
-	janus_ice_component *component = stream->component;
-	if(!component)
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RESEND_TRICKLES);
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc)
 		return;
 	NiceAgent *agent = handle->agent;
 	/* Iterate on all existing local candidates */
 	gchar buffer[200];
 	GSList *candidates, *i;
-	candidates = nice_agent_get_local_candidates (agent, stream->stream_id, component->component_id);
+	candidates = nice_agent_get_local_candidates (agent, pc->stream_id, pc->component_id);
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] We have %d candidates for Stream #%d, Component #%d\n",
-		handle->handle_id, g_slist_length(candidates), stream->stream_id, component->component_id);
+		handle->handle_id, g_slist_length(candidates), pc->stream_id, pc->component_id);
 	for(i = candidates; i; i = i->next) {
 		NiceCandidate *c = (NiceCandidate *) i->data;
 		if(c->type == NICE_CANDIDATE_TYPE_PEER_REFLEXIVE)
@@ -3476,35 +3291,36 @@ static gint rtcp_transport_wide_cc_stats_comparator(gconstpointer item1, gconstp
 	return ((rtcp_transport_wide_cc_stats*)item1)->transport_seq_num - ((rtcp_transport_wide_cc_stats*)item2)->transport_seq_num;
 }
 static gboolean janus_ice_outgoing_transport_wide_cc_feedback(gpointer user_data) {
-	janus_ice_handle *handle = (janus_ice_handle *)user_data;
-	janus_ice_stream *stream = handle->stream;
-	if(stream && stream->do_transport_wide_cc) {
+	janus_handle *handle = (janus_handle *)user_data;
+	janus_handle_webrtc *pc = handle->pc;
+	janus_handle_webrtc_medium *medium = pc ? g_hash_table_lookup(pc->media_bytype, GINT_TO_POINTER(JANUS_MEDIA_VIDEO)) : NULL;
+	if(pc && pc->do_transport_wide_cc && medium) {
 		/* Create a transport wide feedback message */
 		size_t size = 1300;
 		char rtcpbuf[1300];
 		/* Order packet list */
-		stream->transport_wide_received_seq_nums = g_slist_sort(stream->transport_wide_received_seq_nums,
+		pc->transport_wide_received_seq_nums = g_slist_sort(pc->transport_wide_received_seq_nums,
 			rtcp_transport_wide_cc_stats_comparator);
 		/* Create full stats queue */
 		GQueue *packets = g_queue_new();
 		/* For all packets */
 		GSList *it = NULL;
-		for(it = stream->transport_wide_received_seq_nums; it; it = it->next) {
+		for(it = pc->transport_wide_received_seq_nums; it; it = it->next) {
 			/* Get stat */
 			janus_rtcp_transport_wide_cc_stats *stats = (janus_rtcp_transport_wide_cc_stats *)it->data;
 			/* Get transport seq */
 			guint32 transport_seq_num = stats->transport_seq_num;
 			/* Check if it is an out of order  */
-			if(transport_seq_num < stream->transport_wide_cc_last_feedback_seq_num) {
+			if(transport_seq_num < pc->transport_wide_cc_last_feedback_seq_num) {
 				/* Skip, it was already reported as lost */
 				g_free(stats);
 				continue;
 			}
 			/* If not first */
-			if(stream->transport_wide_cc_last_feedback_seq_num) {
+			if(pc->transport_wide_cc_last_feedback_seq_num) {
 				/* For each lost */
 				guint32 i = 0;
-				for(i = stream->transport_wide_cc_last_feedback_seq_num+1; i<transport_seq_num; ++i) {
+				for(i = pc->transport_wide_cc_last_feedback_seq_num+1; i<transport_seq_num; ++i) {
 					/* Create new stat */
 					janus_rtcp_transport_wide_cc_stats *missing = g_malloc(sizeof(janus_rtcp_transport_wide_cc_stats));
 					/* Add missing packet */
@@ -3515,13 +3331,13 @@ static gboolean janus_ice_outgoing_transport_wide_cc_feedback(gpointer user_data
 				}
 			}
 			/* Store last */
-			stream->transport_wide_cc_last_feedback_seq_num = transport_seq_num;
+			pc->transport_wide_cc_last_feedback_seq_num = transport_seq_num;
 			/* Add this one */
 			g_queue_push_tail(packets, stats);
 		}
 		/* Free and reset stats list */
-		g_slist_free(stream->transport_wide_received_seq_nums);
-		stream->transport_wide_received_seq_nums = NULL;
+		g_slist_free(pc->transport_wide_received_seq_nums);
+		pc->transport_wide_received_seq_nums = NULL;
 		/* Create and enqueue RTCP packets */
 		guint packets_len = 0;
 		while((packets_len = g_queue_get_length(packets)) > 0) {
@@ -3544,12 +3360,12 @@ static gboolean janus_ice_outgoing_transport_wide_cc_feedback(gpointer user_data
 				packets_to_process = packets;
 			}
 			/* Get feedback packet count and increase it for next one */
-			guint8 feedback_packet_count = stream->transport_wide_cc_feedback_count++;
+			guint8 feedback_packet_count = pc->transport_wide_cc_feedback_count++;
 			/* Create RTCP packet */
 			int len = janus_rtcp_transport_wide_cc_feedback(rtcpbuf, size,
-				stream->video_ssrc, stream->video_ssrc_peer[0], feedback_packet_count, packets_to_process);
+				medium->ssrc, medium->ssrc_peer[0], feedback_packet_count, packets_to_process);
 			/* Enqueue it, we'll send it later */
-			janus_ice_relay_rtcp_internal(handle, 1, rtcpbuf, len, FALSE);
+			janus_ice_relay_rtcp_internal(handle, medium->mindex, TRUE, rtcpbuf, len, FALSE);
 			if(packets_to_process != packets) {
 				g_queue_free(packets_to_process);
 			}
@@ -3561,131 +3377,81 @@ static gboolean janus_ice_outgoing_transport_wide_cc_feedback(gpointer user_data
 }
 
 static gboolean janus_ice_outgoing_rtcp_handle(gpointer user_data) {
-	janus_ice_handle *handle = (janus_ice_handle *)user_data;
-	janus_ice_stream *stream = handle->stream;
-	/* Audio */
-	if(stream && stream->component && stream->component->out_stats.audio.packets > 0) {
-		/* Create a SR/SDES compound */
-		int srlen = 28;
-		int sdeslen = 16;
-		char rtcpbuf[srlen+sdeslen];
-		memset(rtcpbuf, 0, sizeof(rtcpbuf));
-		rtcp_sr *sr = (rtcp_sr *)&rtcpbuf;
-		sr->header.version = 2;
-		sr->header.type = RTCP_SR;
-		sr->header.rc = 0;
-		sr->header.length = htons((srlen/4)-1);
-		sr->ssrc = htonl(stream->audio_ssrc);
-		struct timeval tv;
-		gettimeofday(&tv, NULL);
-		uint32_t s = tv.tv_sec + 2208988800u;
-		uint32_t u = tv.tv_usec;
-		uint32_t f = (u << 12) + (u << 8) - ((u * 3650) >> 6);
-		sr->si.ntp_ts_msw = htonl(s);
-		sr->si.ntp_ts_lsw = htonl(f);
-		/* Compute an RTP timestamp coherent with the NTP one */
-		rtcp_context *rtcp_ctx = stream->audio_rtcp_ctx;
-		if(rtcp_ctx == NULL) {
-			sr->si.rtp_ts = htonl(stream->audio_last_ts);	/* FIXME */
-		} else {
-			int64_t ntp = tv.tv_sec*G_USEC_PER_SEC + tv.tv_usec;
-			uint32_t rtp_ts = ((ntp-stream->audio_first_ntp_ts)*(rtcp_ctx->tb))/1000000 + stream->audio_first_rtp_ts;
-			sr->si.rtp_ts = htonl(rtp_ts);
-		}
-		sr->si.s_packets = htonl(stream->component->out_stats.audio.packets);
-		sr->si.s_octets = htonl(stream->component->out_stats.audio.bytes);
-		rtcp_sdes *sdes = (rtcp_sdes *)&rtcpbuf[28];
-		janus_rtcp_sdes_cname((char *)sdes, sdeslen, "janus", 5);
-		sdes->chunk.ssrc = htonl(stream->audio_ssrc);
-		/* Enqueue it, we'll send it later */
-		janus_ice_relay_rtcp_internal(handle, 0, rtcpbuf, srlen+sdeslen, FALSE);
-		/* Check if we detected too many losses, and send a slowlink event in case */
-		guint lost = janus_rtcp_context_get_lost_all(rtcp_ctx, TRUE);
-		janus_slow_link_update(stream->component, handle, FALSE, TRUE, lost);
-	}
-	if(stream && stream->audio_recv) {
-		/* Create a RR too */
-		int rrlen = 32;
-		char rtcpbuf[32];
-		memset(rtcpbuf, 0, sizeof(rtcpbuf));
-		rtcp_rr *rr = (rtcp_rr *)&rtcpbuf;
-		rr->header.version = 2;
-		rr->header.type = RTCP_RR;
-		rr->header.rc = 1;
-		rr->header.length = htons((rrlen/4)-1);
-		rr->ssrc = htonl(stream->audio_ssrc);
-		janus_rtcp_report_block(stream->audio_rtcp_ctx, &rr->rb[0]);
-		rr->rb[0].ssrc = htonl(stream->audio_ssrc_peer);
-		/* Enqueue it, we'll send it later */
-		janus_ice_relay_rtcp_internal(handle, 0, rtcpbuf, 32, FALSE);
-		/* Check if we detected too many losses, and send a slowlink event in case */
-		guint lost = janus_rtcp_context_get_lost_all(stream->audio_rtcp_ctx, FALSE);
-		janus_slow_link_update(stream->component, handle, FALSE, FALSE, lost);
-	}
-	/* Now do the same for video */
-	if(stream && stream->component && stream->component->out_stats.video[0].packets > 0) {
-		/* Create a SR/SDES compound */
-		int srlen = 28;
-		int sdeslen = 16;
-		char rtcpbuf[srlen+sdeslen];
-		memset(rtcpbuf, 0, sizeof(rtcpbuf));
-		rtcp_sr *sr = (rtcp_sr *)&rtcpbuf;
-		sr->header.version = 2;
-		sr->header.type = RTCP_SR;
-		sr->header.rc = 0;
-		sr->header.length = htons((srlen/4)-1);
-		sr->ssrc = htonl(stream->video_ssrc);
-		struct timeval tv;
-		gettimeofday(&tv, NULL);
-		uint32_t s = tv.tv_sec + 2208988800u;
-		uint32_t u = tv.tv_usec;
-		uint32_t f = (u << 12) + (u << 8) - ((u * 3650) >> 6);
-		sr->si.ntp_ts_msw = htonl(s);
-		sr->si.ntp_ts_lsw = htonl(f);
-		/* Compute an RTP timestamp coherent with the NTP one */
-		rtcp_context *rtcp_ctx = stream->video_rtcp_ctx[0];
-		if(rtcp_ctx == NULL) {
-			sr->si.rtp_ts = htonl(stream->video_last_ts);	/* FIXME */
-		} else {
-			int64_t ntp = tv.tv_sec*G_USEC_PER_SEC + tv.tv_usec;
-			uint32_t rtp_ts = ((ntp-stream->video_first_ntp_ts[0])*(rtcp_ctx->tb))/1000000 + stream->video_first_rtp_ts[0];
-			sr->si.rtp_ts = htonl(rtp_ts);
+	janus_handle *handle = (janus_handle *)user_data;
+	janus_handle_webrtc *pc = handle->pc;
+	/* Iterate on all media */
+	janus_handle_webrtc_medium *medium = NULL;
+	uint mi=0;
+	for(mi=0; mi<g_hash_table_size(pc->media); mi++) {
+		medium = g_hash_table_lookup(pc->media, GUINT_TO_POINTER(mi));
+		if(!medium || (medium->type != JANUS_MEDIA_AUDIO && medium->type != JANUS_MEDIA_VIDEO))
+			continue;
+		if(medium->out_stats.info[0].packets > 0) {
+			/* Create a SR/SDES compound */
+			int srlen = 28;
+			int sdeslen = 16;
+			char rtcpbuf[srlen+sdeslen];
+			memset(rtcpbuf, 0, sizeof(rtcpbuf));
+			rtcp_sr *sr = (rtcp_sr *)&rtcpbuf;
+			sr->header.version = 2;
+			sr->header.type = RTCP_SR;
+			sr->header.rc = 0;
+			sr->header.length = htons((srlen/4)-1);
+			sr->ssrc = htonl(medium->ssrc);
+			struct timeval tv;
+			gettimeofday(&tv, NULL);
+			uint32_t s = tv.tv_sec + 2208988800u;
+			uint32_t u = tv.tv_usec;
+			uint32_t f = (u << 12) + (u << 8) - ((u * 3650) >> 6);
+			sr->si.ntp_ts_msw = htonl(s);
+			sr->si.ntp_ts_lsw = htonl(f);
+			/* Compute an RTP timestamp coherent with the NTP one */
+			rtcp_context *rtcp_ctx = medium->rtcp_ctx[0];
+			if(rtcp_ctx == NULL) {
+				sr->si.rtp_ts = htonl(medium->last_ts);	/* FIXME */
+			} else {
+				int64_t ntp = tv.tv_sec*G_USEC_PER_SEC + tv.tv_usec;
+				uint32_t rtp_ts = ((ntp - medium->first_ntp_ts[0])*(rtcp_ctx->tb))/1000000 + medium->first_rtp_ts[0];
+				sr->si.rtp_ts = htonl(rtp_ts);
+			}
+			sr->si.s_packets = htonl(medium->out_stats.info[0].packets);
+			sr->si.s_octets = htonl(medium->out_stats.info[0].bytes);
+			rtcp_sdes *sdes = (rtcp_sdes *)&rtcpbuf[28];
+			janus_rtcp_sdes_cname((char *)sdes, sdeslen, "janus", 5);
+			sdes->chunk.ssrc = htonl(medium->ssrc);
+			/* Enqueue it, we'll send it later */
+			janus_ice_relay_rtcp_internal(handle, medium->mindex, medium->type == JANUS_MEDIA_VIDEO, rtcpbuf, srlen+sdeslen, FALSE);
+			/* Check if we detected too many losses, and send a slowlink event in case */
+			guint lost = janus_rtcp_context_get_lost_all(rtcp_ctx, TRUE);
+			janus_slow_link_update(medium, handle, TRUE, lost);
 		}
-		sr->si.s_packets = htonl(stream->component->out_stats.video[0].packets);
-		sr->si.s_octets = htonl(stream->component->out_stats.video[0].bytes);
-		rtcp_sdes *sdes = (rtcp_sdes *)&rtcpbuf[28];
-		janus_rtcp_sdes_cname((char *)sdes, sdeslen, "janus", 5);
-		sdes->chunk.ssrc = htonl(stream->video_ssrc);
-		/* Enqueue it, we'll send it later */
-		janus_ice_relay_rtcp_internal(handle, 1, rtcpbuf, srlen+sdeslen, FALSE);
-		/* Check if we detected too many losses, and send a slowlink event in case */
-		guint lost = janus_rtcp_context_get_lost_all(rtcp_ctx, TRUE);
-		janus_slow_link_update(stream->component, handle, TRUE, TRUE, lost);
-	}
-	if(stream && stream->video_recv) {
-		/* Create a RR too (for each SSRC, if we're simulcasting) */
-		int vindex=0;
-		for(vindex=0; vindex<3; vindex++) {
-			if(stream->video_rtcp_ctx[vindex] && stream->video_rtcp_ctx[vindex]->rtp_recvd) {
-				/* Create a RR */
-				int rrlen = 32;
-				char rtcpbuf[32];
-				memset(rtcpbuf, 0, sizeof(rtcpbuf));
-				rtcp_rr *rr = (rtcp_rr *)&rtcpbuf;
-				rr->header.version = 2;
-				rr->header.type = RTCP_RR;
-				rr->header.rc = 1;
-				rr->header.length = htons((rrlen/4)-1);
-				rr->ssrc = htonl(stream->video_ssrc);
-				janus_rtcp_report_block(stream->video_rtcp_ctx[vindex], &rr->rb[0]);
-				rr->rb[0].ssrc = htonl(stream->video_ssrc_peer[vindex]);
-				/* Enqueue it, we'll send it later */
-				janus_ice_relay_rtcp_internal(handle, 1, rtcpbuf, 32, FALSE);
+		if(medium->recv) {
+			/* Create a RR too (for each SSRC, if we're simulcasting) */
+			int vindex=0;
+			for(vindex=0; vindex<3; vindex++) {
+				if(medium->rtcp_ctx[vindex] && medium->rtcp_ctx[vindex]->rtp_recvd) {
+					/* Create a RR */
+					int rrlen = 32;
+					char rtcpbuf[32];
+					memset(rtcpbuf, 0, sizeof(rtcpbuf));
+					rtcp_rr *rr = (rtcp_rr *)&rtcpbuf;
+					rr->header.version = 2;
+					rr->header.type = RTCP_RR;
+					rr->header.rc = 1;
+					rr->header.length = htons((rrlen/4)-1);
+					rr->ssrc = htonl(medium->ssrc);
+					janus_rtcp_report_block(medium->rtcp_ctx[vindex], &rr->rb[0]);
+					rr->rb[0].ssrc = htonl(medium->ssrc_peer[vindex]);
+					/* Enqueue it, we'll send it later */
+					janus_ice_relay_rtcp_internal(handle, medium->mindex, medium->type == JANUS_MEDIA_VIDEO, rtcpbuf, 32, FALSE);
+					if(vindex == 0) {
+						/* Check if we detected too many losses, and send a slowlink event in case */
+						guint lost = janus_rtcp_context_get_lost_all(medium->rtcp_ctx[vindex], FALSE);
+						janus_slow_link_update(medium, handle, FALSE, lost);
+					}
+				}
 			}
 		}
-		/* Check if we detected too many losses, and send a slowlink event in case */
-		guint lost = janus_rtcp_context_get_lost_all(stream->video_rtcp_ctx[0], FALSE);
-		janus_slow_link_update(stream->component, handle, TRUE, FALSE, lost);
 	}
 	if(twcc_period == 1000) {
 		/* The Transport Wide CC feedback period is 1s as well, send it here */
@@ -3695,137 +3461,91 @@ static gboolean janus_ice_outgoing_rtcp_handle(gpointer user_data) {
 }
 
 static gboolean janus_ice_outgoing_stats_handle(gpointer user_data) {
-	janus_ice_handle *handle = (janus_ice_handle *)user_data;
+	janus_handle *handle = (janus_handle *)user_data;
 	/* This callback is for stats and other things we need to do on a regular basis (typically called once per second) */
 	janus_session *session = (janus_session *)handle->session;
 	gint64 now = janus_get_monotonic_time();
 	/* Reset the last second counters if too much time passed with no data in or out */
-	janus_ice_stream *stream = handle->stream;
-	if(stream == NULL || stream->component == NULL)
+	janus_handle_webrtc *pc = handle->pc;
+	if(pc == NULL)
 		return G_SOURCE_CONTINUE;
-	janus_ice_component *component = stream->component;
-	/* Audio */
-	gint64 last = component->in_stats.audio.updated;
-	if(last && now > last && now-last >= 2*G_USEC_PER_SEC && component->in_stats.audio.bytes_lastsec_temp > 0) {
-		component->in_stats.audio.bytes_lastsec = 0;
-		component->in_stats.audio.bytes_lastsec_temp = 0;
-	}
-	last = component->out_stats.audio.updated;
-	if(last && now > last && now-last >= 2*G_USEC_PER_SEC && component->out_stats.audio.bytes_lastsec_temp > 0) {
-		component->out_stats.audio.bytes_lastsec = 0;
-		component->out_stats.audio.bytes_lastsec_temp = 0;
-	}
-	/* Video */
-	int vindex = 0;
-	for(vindex=0; vindex < 3; vindex++) {
-		gint64 last = component->in_stats.video[vindex].updated;
-		if(last && now > last && now-last >= 2*G_USEC_PER_SEC && component->in_stats.video[vindex].bytes_lastsec_temp > 0) {
-			component->in_stats.video[vindex].bytes_lastsec = 0;
-			component->in_stats.video[vindex].bytes_lastsec_temp = 0;
-		}
-		last = component->out_stats.video[vindex].updated;
-		if(last && now > last && now-last >= 2*G_USEC_PER_SEC && component->out_stats.video[vindex].bytes_lastsec_temp > 0) {
-			component->out_stats.video[vindex].bytes_lastsec = 0;
-			component->out_stats.video[vindex].bytes_lastsec_temp = 0;
-		}
-	}
-	/* Now let's see if we need to notify the user about no incoming audio or video */
-	if(no_media_timer > 0 && component->dtls && component->dtls->dtls_connected > 0 && (now - component->dtls->dtls_connected >= G_USEC_PER_SEC)) {
-		/* Audio */
-		gint64 last = component->in_stats.audio.updated;
-		if(!component->in_stats.audio.notified_lastsec && last &&
-				!component->in_stats.audio.bytes_lastsec && !component->in_stats.audio.bytes_lastsec_temp &&
-					now-last >= (gint64)no_media_timer*G_USEC_PER_SEC) {
-			/* We missed more than no_second_timer seconds of audio! */
-			component->in_stats.audio.notified_lastsec = TRUE;
-			JANUS_LOG(LOG_WARN, "[%"SCNu64"] Didn't receive audio for more than %d seconds...\n", handle->handle_id, no_media_timer);
-			janus_ice_notify_media(handle, FALSE, FALSE);
-		}
-		/* Video */
-		last = component->in_stats.video[0].updated;
-		if(!component->in_stats.video[0].notified_lastsec && last &&
-				!component->in_stats.video[0].bytes_lastsec && !component->in_stats.video[0].bytes_lastsec_temp &&
-					now-last >= (gint64)no_media_timer*G_USEC_PER_SEC) {
-			/* We missed more than no_second_timer seconds of video! */
-			component->in_stats.video[0].notified_lastsec = TRUE;
-			JANUS_LOG(LOG_WARN, "[%"SCNu64"] Didn't receive video for more than a second...\n", handle->handle_id);
-			janus_ice_notify_media(handle, TRUE, FALSE);
+	/* Iterate on all media */
+	janus_handle_webrtc_medium *medium = NULL;
+	uint mi=0;
+	for(mi=0; mi<g_hash_table_size(pc->media); mi++) {
+		medium = g_hash_table_lookup(pc->media, GUINT_TO_POINTER(mi));
+		if(!medium)
+			continue;
+		int vindex = 0;
+		for(vindex=0; vindex < 3; vindex++) {
+			gint64 last = medium->in_stats.info[vindex].updated;
+			if(last && now > last && now-last >= 2*G_USEC_PER_SEC && medium->in_stats.info[vindex].bytes_lastsec_temp > 0) {
+				medium->in_stats.info[vindex].bytes_lastsec = 0;
+				medium->in_stats.info[vindex].bytes_lastsec_temp = 0;
+			}
+			last = medium->out_stats.info[vindex].updated;
+			if(last && now > last && now-last >= 2*G_USEC_PER_SEC && medium->out_stats.info[vindex].bytes_lastsec_temp > 0) {
+				medium->out_stats.info[vindex].bytes_lastsec = 0;
+				medium->out_stats.info[vindex].bytes_lastsec_temp = 0;
+			}
 		}
-	}
-	/* We also send live stats to event handlers every tot-seconds (configurable) */
-	handle->last_event_stats++;
-	if(janus_ice_event_stats_period > 0 && handle->last_event_stats >= janus_ice_event_stats_period) {
-		handle->last_event_stats = 0;
-		/* Audio */
-		if(janus_events_is_enabled() && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO)) {
-			if(stream && stream->audio_rtcp_ctx) {
-				json_t *info = json_object();
-				json_object_set_new(info, "media", json_string("audio"));
-				json_object_set_new(info, "base", json_integer(stream->audio_rtcp_ctx->tb));
-				json_object_set_new(info, "rtt", json_integer(janus_rtcp_context_get_rtt(stream->audio_rtcp_ctx)));
-				json_object_set_new(info, "lost", json_integer(janus_rtcp_context_get_lost_all(stream->audio_rtcp_ctx, FALSE)));
-				json_object_set_new(info, "lost-by-remote", json_integer(janus_rtcp_context_get_lost_all(stream->audio_rtcp_ctx, TRUE)));
-				json_object_set_new(info, "jitter-local", json_integer(janus_rtcp_context_get_jitter(stream->audio_rtcp_ctx, FALSE)));
-				json_object_set_new(info, "jitter-remote", json_integer(janus_rtcp_context_get_jitter(stream->audio_rtcp_ctx, TRUE)));
-				json_object_set_new(info, "in-link-quality", json_integer(janus_rtcp_context_get_in_link_quality(stream->audio_rtcp_ctx)));
-				json_object_set_new(info, "in-media-link-quality", json_integer(janus_rtcp_context_get_in_media_link_quality(stream->audio_rtcp_ctx)));
-				json_object_set_new(info, "out-link-quality", json_integer(janus_rtcp_context_get_out_link_quality(stream->audio_rtcp_ctx)));
-				json_object_set_new(info, "out-media-link-quality", json_integer(janus_rtcp_context_get_out_media_link_quality(stream->audio_rtcp_ctx)));
-				if(stream->component) {
-					json_object_set_new(info, "packets-received", json_integer(stream->component->in_stats.audio.packets));
-					json_object_set_new(info, "packets-sent", json_integer(stream->component->out_stats.audio.packets));
-					json_object_set_new(info, "bytes-received", json_integer(stream->component->in_stats.audio.bytes));
-					json_object_set_new(info, "bytes-sent", json_integer(stream->component->out_stats.audio.bytes));
-					json_object_set_new(info, "bytes-received-lastsec", json_integer(stream->component->in_stats.audio.bytes_lastsec));
-					json_object_set_new(info, "bytes-sent-lastsec", json_integer(stream->component->out_stats.audio.bytes_lastsec));
-					json_object_set_new(info, "nacks-received", json_integer(stream->component->in_stats.audio.nacks));
-					json_object_set_new(info, "nacks-sent", json_integer(stream->component->out_stats.audio.nacks));
-					json_object_set_new(info, "retransmissions-received", json_integer(stream->audio_rtcp_ctx->retransmitted));
-				}
-				janus_events_notify_handlers(JANUS_EVENT_TYPE_MEDIA, session->session_id, handle->handle_id, handle->opaque_id, info);
+		/* Now let's see if we need to notify the user about no incoming audio or video */
+		if(no_media_timer > 0 && pc->dtls && pc->dtls->dtls_connected > 0 && (now - pc->dtls->dtls_connected >= G_USEC_PER_SEC)) {
+			gint64 last = medium->in_stats.info[0].updated;
+			if(!medium->in_stats.info[0].notified_lastsec && last &&
+					!medium->in_stats.info[0].bytes_lastsec && !medium->in_stats.info[0].bytes_lastsec_temp &&
+						now-last >= (gint64)no_media_timer*G_USEC_PER_SEC) {
+				/* We missed more than no_second_timer seconds of video! */
+				medium->in_stats.info[0].notified_lastsec = TRUE;
+				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Didn't receive video for more than a second...\n", handle->handle_id);
+				janus_ice_notify_media(handle, medium->mid, medium->type == JANUS_MEDIA_VIDEO, FALSE);
 			}
 		}
-		/* Do the same for video */
-		if(janus_events_is_enabled() && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO)) {
-			int vindex=0;
-			for(vindex=0; vindex<3; vindex++) {
-				if(stream && stream->video_rtcp_ctx[vindex]) {
-					json_t *info = json_object();
-					if(vindex == 0)
-						json_object_set_new(info, "media", json_string("video"));
-					else if(vindex == 1)
-						json_object_set_new(info, "media", json_string("video-sim1"));
-					else
-						json_object_set_new(info, "media", json_string("video-sim2"));
-					json_object_set_new(info, "base", json_integer(stream->video_rtcp_ctx[vindex]->tb));
-					if(vindex == 0)
-						json_object_set_new(info, "rtt", json_integer(janus_rtcp_context_get_rtt(stream->video_rtcp_ctx[vindex])));
-					json_object_set_new(info, "lost", json_integer(janus_rtcp_context_get_lost_all(stream->video_rtcp_ctx[vindex], FALSE)));
-					json_object_set_new(info, "lost-by-remote", json_integer(janus_rtcp_context_get_lost_all(stream->video_rtcp_ctx[vindex], TRUE)));
-					json_object_set_new(info, "jitter-local", json_integer(janus_rtcp_context_get_jitter(stream->video_rtcp_ctx[vindex], FALSE)));
-					json_object_set_new(info, "jitter-remote", json_integer(janus_rtcp_context_get_jitter(stream->video_rtcp_ctx[vindex], TRUE)));
-					json_object_set_new(info, "in-link-quality", json_integer(janus_rtcp_context_get_in_link_quality(stream->video_rtcp_ctx[vindex])));
-					json_object_set_new(info, "in-media-link-quality", json_integer(janus_rtcp_context_get_in_media_link_quality(stream->video_rtcp_ctx[vindex])));
-					json_object_set_new(info, "out-link-quality", json_integer(janus_rtcp_context_get_out_link_quality(stream->video_rtcp_ctx[vindex])));
-					json_object_set_new(info, "out-media-link-quality", json_integer(janus_rtcp_context_get_out_media_link_quality(stream->video_rtcp_ctx[vindex])));
-					if(stream->component) {
-						json_object_set_new(info, "packets-received", json_integer(stream->component->in_stats.video[vindex].packets));
-						json_object_set_new(info, "packets-sent", json_integer(stream->component->out_stats.video[vindex].packets));
-						json_object_set_new(info, "bytes-received", json_integer(stream->component->in_stats.video[vindex].bytes));
-						json_object_set_new(info, "bytes-sent", json_integer(stream->component->out_stats.video[vindex].bytes));
-						json_object_set_new(info, "bytes-received-lastsec", json_integer(stream->component->in_stats.video[vindex].bytes_lastsec));
-						json_object_set_new(info, "bytes-sent-lastsec", json_integer(stream->component->out_stats.video[vindex].bytes_lastsec));
-						json_object_set_new(info, "nacks-received", json_integer(stream->component->in_stats.video[vindex].nacks));
-						json_object_set_new(info, "nacks-sent", json_integer(stream->component->out_stats.video[vindex].nacks));
-						json_object_set_new(info, "retransmissions-received", json_integer(stream->video_rtcp_ctx[vindex]->retransmitted));
+		/* We also send live stats to event handlers every tot-seconds (configurable) */
+		handle->last_event_stats++;
+		if(janus_ice_event_stats_period > 0 && handle->last_event_stats >= janus_ice_event_stats_period) {
+			handle->last_event_stats = 0;
+			if(janus_events_is_enabled()) {
+				int vindex=0;
+				for(vindex=0; vindex<3; vindex++) {
+					if(medium->rtcp_ctx[vindex]) {
+						json_t *info = json_object();
+						json_object_set_new(info, "mid", json_string(medium->mid));
+						json_object_set_new(info, "mindex", json_integer(medium->mindex));
+						if(vindex == 0)
+							json_object_set_new(info, "media", json_string(medium->type == JANUS_MEDIA_VIDEO ? "video" : "audio"));
+						else if(vindex == 1)
+							json_object_set_new(info, "media", json_string("video-sim1"));
+						else
+							json_object_set_new(info, "media", json_string("video-sim2"));
+						json_object_set_new(info, "base", json_integer(medium->rtcp_ctx[vindex]->tb));
+						if(vindex == 0)
+							json_object_set_new(info, "rtt", json_integer(janus_rtcp_context_get_rtt(medium->rtcp_ctx[vindex])));
+						json_object_set_new(info, "lost", json_integer(janus_rtcp_context_get_lost_all(medium->rtcp_ctx[vindex], FALSE)));
+						json_object_set_new(info, "lost-by-remote", json_integer(janus_rtcp_context_get_lost_all(medium->rtcp_ctx[vindex], TRUE)));
+						json_object_set_new(info, "jitter-local", json_integer(janus_rtcp_context_get_jitter(medium->rtcp_ctx[vindex], FALSE)));
+						json_object_set_new(info, "jitter-remote", json_integer(janus_rtcp_context_get_jitter(medium->rtcp_ctx[vindex], TRUE)));
+						json_object_set_new(info, "in-link-quality", json_integer(janus_rtcp_context_get_in_link_quality(medium->rtcp_ctx[vindex])));
+						json_object_set_new(info, "in-media-link-quality", json_integer(janus_rtcp_context_get_in_media_link_quality(medium->rtcp_ctx[vindex])));
+						json_object_set_new(info, "out-link-quality", json_integer(janus_rtcp_context_get_out_link_quality(medium->rtcp_ctx[vindex])));
+						json_object_set_new(info, "out-media-link-quality", json_integer(janus_rtcp_context_get_out_media_link_quality(medium->rtcp_ctx[vindex])));
+						json_object_set_new(info, "packets-received", json_integer(medium->in_stats.info[vindex].packets));
+						json_object_set_new(info, "packets-sent", json_integer(medium->out_stats.info[vindex].packets));
+						json_object_set_new(info, "bytes-received", json_integer(medium->in_stats.info[vindex].bytes));
+						json_object_set_new(info, "bytes-sent", json_integer(medium->out_stats.info[vindex].bytes));
+						json_object_set_new(info, "bytes-received-lastsec", json_integer(medium->in_stats.info[vindex].bytes_lastsec));
+						json_object_set_new(info, "bytes-sent-lastsec", json_integer(medium->out_stats.info[vindex].bytes_lastsec));
+						json_object_set_new(info, "nacks-received", json_integer(medium->in_stats.info[vindex].nacks));
+						json_object_set_new(info, "nacks-sent", json_integer(medium->out_stats.info[vindex].nacks));
+						json_object_set_new(info, "retransmissions-received", json_integer(medium->rtcp_ctx[vindex]->retransmitted));
+						janus_events_notify_handlers(JANUS_EVENT_TYPE_MEDIA, session->session_id, handle->handle_id, handle->opaque_id, info);
 					}
-					janus_events_notify_handlers(JANUS_EVENT_TYPE_MEDIA, session->session_id, handle->handle_id, handle->opaque_id, info);
 				}
 			}
 		}
 	}
 	/* Should we clean up old NACK buffers for any of the streams? */
-	janus_cleanup_nack_buffer(now, handle->stream, TRUE, TRUE);
+	janus_cleanup_nack_buffer(now, handle->pc, TRUE, TRUE);
 	/* Check if we should also print a summary of SRTP-related errors */
 	handle->last_srtp_summary++;
 	if(handle->last_srtp_summary == 0 || handle->last_srtp_summary == 2) {
@@ -3840,31 +3560,30 @@ static gboolean janus_ice_outgoing_stats_handle(gpointer user_data) {
 	return G_SOURCE_CONTINUE;
 }
 
-static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janus_ice_queued_packet *pkt) {
+static gboolean janus_ice_outgoing_traffic_handle(janus_handle *handle, janus_ice_queued_packet *pkt) {
 	janus_session *session = (janus_session *)handle->session;
-	janus_ice_stream *stream = handle->stream;
-	janus_ice_component *component = stream ? stream->component : NULL;
+	janus_handle_webrtc *pc = handle->pc;
 	if(pkt == &janus_ice_dtls_handshake) {
 		if(!janus_is_webrtc_encryption_enabled()) {
 			JANUS_LOG(LOG_WARN, "[%"SCNu64"] WebRTC encryption disabled, skipping DTLS handshake\n", handle->handle_id);
-			janus_ice_dtls_handshake_done(handle, component);
+			janus_handle_dtls_handshake_done(handle);
 			return G_SOURCE_CONTINUE;
-		} else if(!component) {
-			JANUS_LOG(LOG_WARN, "[%"SCNu64"] ICE component not initialized, aborting DTLS handshake\n", handle->handle_id);
+		} else if(!pc) {
+			JANUS_LOG(LOG_WARN, "[%"SCNu64"] PeerConnection not initialized, aborting DTLS handshake\n", handle->handle_id);
 			return G_SOURCE_CONTINUE;
 		}
 		/* Start the DTLS handshake */
-		janus_dtls_srtp_handshake(component->dtls);
+		janus_dtls_srtp_handshake(pc->dtls);
 		/* Create retransmission timer */
-		component->dtlsrt_source = g_timeout_source_new(50);
-		g_source_set_callback(component->dtlsrt_source, janus_dtls_retry, component->dtls, NULL);
-		guint id = g_source_attach(component->dtlsrt_source, handle->mainctx);
+		pc->dtlsrt_source = g_timeout_source_new(50);
+		g_source_set_callback(pc->dtlsrt_source, janus_dtls_retry, pc->dtls, NULL);
+		guint id = g_source_attach(pc->dtlsrt_source, handle->mainctx);
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Creating retransmission timer with ID %u\n", handle->handle_id, id);
 		return G_SOURCE_CONTINUE;
 	} else if(pkt == &janus_ice_hangup_peerconnection) {
 		/* The media session is over, send an alert on all streams and components */
-		if(handle->stream && handle->stream->component && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY)) {
-			janus_dtls_srtp_send_alert(handle->stream->component->dtls);
+		if(handle->pc && janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY)) {
+			janus_dtls_srtp_send_alert(handle->pc->dtls);
 		}
 		/* Notify the plugin about the fact this PeerConnection has just gone */
 		janus_plugin *plugin = (janus_plugin *)handle->app;
@@ -3929,14 +3648,14 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 				plugin ? plugin->get_package() : NULL, handle->opaque_id);
 		return G_SOURCE_REMOVE;
 	}
-	if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY)) {
+	if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY)) {
 		janus_ice_free_queued_packet(pkt);
 		return G_SOURCE_CONTINUE;
 	}
 	/* Now let's get on with the packet */
 	if(pkt == NULL)
 		return G_SOURCE_CONTINUE;
-	if(pkt->data == NULL || stream == NULL) {
+	if(pkt->data == NULL) {
 		janus_ice_free_queued_packet(pkt);
 		return G_SOURCE_CONTINUE;
 	}
@@ -3946,31 +3665,38 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 		janus_ice_free_queued_packet(pkt);
 		return G_SOURCE_CONTINUE;
 	}
-	if(!stream->cdone) {
-		if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT) && !stream->noerrorlog) {
+	if(!pc->cdone) {
+		if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT) && !pc->noerrorlog) {
 			JANUS_LOG(LOG_ERR, "[%"SCNu64"] No candidates not gathered yet for stream??\n", handle->handle_id);
-			stream->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
+			pc->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
 		}
 		janus_ice_free_queued_packet(pkt);
 		return G_SOURCE_CONTINUE;
 	}
+	/* Find the right medium instance */
+	janus_handle_webrtc_medium *medium = g_hash_table_lookup(pc->media, GINT_TO_POINTER(pkt->mindex));
+	if(!medium) {
+		JANUS_LOG(LOG_ERR, "[%"SCNu64"] No medium #%d associated to this packet??\n", handle->handle_id, pkt->mindex);
+		janus_ice_free_queued_packet(pkt);
+		return G_SOURCE_CONTINUE;
+	}
 	if(pkt->control) {
 		/* RTCP */
 		int video = (pkt->type == JANUS_ICE_PACKET_VIDEO);
-		stream->noerrorlog = FALSE;
-		if(janus_is_webrtc_encryption_enabled() && (!component->dtls || !component->dtls->srtp_valid || !component->dtls->srtp_out)) {
-			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT) && !component->noerrorlog) {
+		pc->noerrorlog = FALSE;
+		if(janus_is_webrtc_encryption_enabled() && (!pc->dtls || !pc->dtls->srtp_valid || !pc->dtls->srtp_out)) {
+			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT) && !medium->noerrorlog) {
 				JANUS_LOG(LOG_WARN, "[%"SCNu64"] %s stream (#%u) component has no valid SRTP session (yet?)\n",
-					handle->handle_id, video ? "video" : "audio", stream->stream_id);
-				component->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
+					handle->handle_id, video ? "video" : "audio", pc->stream_id);
+				medium->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
 			}
 			janus_ice_free_queued_packet(pkt);
 			return G_SOURCE_CONTINUE;
 		}
-		component->noerrorlog = FALSE;
+		medium->noerrorlog = FALSE;
 		if(pkt->encrypted) {
 			/* Already SRTCP */
-			int sent = nice_agent_send(handle->agent, stream->stream_id, component->component_id, pkt->length, (const gchar *)pkt->data);
+			int sent = nice_agent_send(handle->agent, pc->stream_id, pc->component_id, pkt->length, (const gchar *)pkt->data);
 			if(sent < pkt->length) {
 				JANUS_LOG(LOG_ERR, "[%"SCNu64"] ... only sent %d bytes? (was %d)\n", handle->handle_id, sent, pkt->length);
 			}
@@ -3986,16 +3712,15 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 				rr->header.type = RTCP_RR;
 				rr->header.rc = 0;
 				rr->header.length = htons((rrlen/4)-1);
-				janus_ice_stream *stream = handle->stream;
 				/* Append REMB */
 				memcpy(rtcpbuf+rrlen, pkt->data, pkt->length);
 				/* If we're simulcasting, set the extra SSRCs (the first one will be set by janus_rtcp_fix_ssrc) */
-				if(stream->video_ssrc_peer[1] && pkt->length >= 28) {
+				if(medium->ssrc_peer[1] && pkt->length >= 28) {
 					rtcp_fb *rtcpfb = (rtcp_fb *)(rtcpbuf+rrlen);
 					rtcp_remb *remb = (rtcp_remb *)rtcpfb->fci;
-					remb->ssrc[1] = htonl(stream->video_ssrc_peer[1]);
-					if(stream->video_ssrc_peer[2] && pkt->length >= 32) {
-						remb->ssrc[2] = htonl(stream->video_ssrc_peer[2]);
+					remb->ssrc[1] = htonl(medium->ssrc_peer[1]);
+					if(medium->ssrc_peer[2] && pkt->length >= 32) {
+						remb->ssrc[2] = htonl(medium->ssrc_peer[2]);
 					}
 				}
 				/* Free old packet and update */
@@ -4011,7 +3736,7 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 			/* Encrypt SRTCP */
 			int protected = pkt->length;
 			int res = janus_is_webrtc_encryption_enabled() ?
-				srtp_protect_rtcp(component->dtls->srtp_out, pkt->data, &protected) : srtp_err_status_ok;
+				srtp_protect_rtcp(pc->dtls->srtp_out, pkt->data, &protected) : srtp_err_status_ok;
 			if(res != srtp_err_status_ok) {
 				/* We don't spam the logs for every SRTP error: just take note of this, and print a summary later */
 				handle->srtp_errors_count++;
@@ -4020,7 +3745,7 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 				JANUS_LOG(LOG_DBG, "[%"SCNu64"] ... SRTCP protect error... %s (len=%d-->%d)...\n", handle->handle_id, janus_srtp_error_str(res), pkt->length, protected);
 			} else {
 				/* Shoot! */
-				int sent = nice_agent_send(handle->agent, stream->stream_id, component->component_id, protected, pkt->data);
+				int sent = nice_agent_send(handle->agent, pc->stream_id, pc->component_id, protected, pkt->data);
 				if(sent < protected) {
 					JANUS_LOG(LOG_ERR, "[%"SCNu64"] ... only sent %d bytes? (was %d)\n", handle->handle_id, sent, protected);
 				}
@@ -4032,25 +3757,25 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 		if(pkt->type == JANUS_ICE_PACKET_AUDIO || pkt->type == JANUS_ICE_PACKET_VIDEO) {
 			/* RTP */
 			int video = (pkt->type == JANUS_ICE_PACKET_VIDEO);
-			if((!video && !stream->audio_send) || (video && !stream->video_send)) {
+			if(!medium->send) {
 				janus_ice_free_queued_packet(pkt);
 				return G_SOURCE_CONTINUE;
 			}
-			if(janus_is_webrtc_encryption_enabled() && (!component->dtls || !component->dtls->srtp_valid || !component->dtls->srtp_out)) {
-				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT) && !component->noerrorlog) {
+			if(janus_is_webrtc_encryption_enabled() && (!pc->dtls || !pc->dtls->srtp_valid || !pc->dtls->srtp_out)) {
+				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT) && !medium->noerrorlog) {
 					JANUS_LOG(LOG_WARN, "[%"SCNu64"] %s stream component has no valid SRTP session (yet?)\n",
 						handle->handle_id, video ? "video" : "audio");
-					component->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
+					medium->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
 				}
 				janus_ice_free_queued_packet(pkt);
 				return G_SOURCE_CONTINUE;
 			}
-			component->noerrorlog = FALSE;
+			medium->noerrorlog = FALSE;
 			if(pkt->encrypted) {
 				/* Already RTP (probably a retransmission?) */
 				janus_rtp_header *header = (janus_rtp_header *)pkt->data;
 				JANUS_LOG(LOG_HUGE, "[%"SCNu64"] ... Retransmitting seq.nr %"SCNu16"\n\n", handle->handle_id, ntohs(header->seq_number));
-				int sent = nice_agent_send(handle->agent, stream->stream_id, component->component_id, pkt->length, (const gchar *)pkt->data);
+				int sent = nice_agent_send(handle->agent, pc->stream_id, pc->component_id, pkt->length, (const gchar *)pkt->data);
 				if(sent < pkt->length) {
 					JANUS_LOG(LOG_ERR, "[%"SCNu64"] ... only sent %d bytes? (was %d)\n", handle->handle_id, sent, pkt->length);
 				}
@@ -4059,36 +3784,29 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 				janus_rtp_header *header = (janus_rtp_header *)pkt->data;
 				if(!pkt->retransmission) {
 					/* ... but only if this isn't a retransmission (for those we already set it before) */
-					header->ssrc = htonl(video ? stream->video_ssrc : stream->audio_ssrc);
+					header->ssrc = htonl(medium->ssrc);
 				}
 				/* Keep track of payload types too */
-				if(!video && stream->audio_payload_type < 0) {
-					stream->audio_payload_type = header->type;
-					if(stream->audio_codec == NULL) {
-						const char *codec = janus_get_codec_from_pt(handle->local_sdp, stream->audio_payload_type);
-						if(codec != NULL)
-							stream->audio_codec = g_strdup(codec);
-					}
-				} else if(video && stream->video_payload_type < 0) {
-					stream->video_payload_type = header->type;
-					if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX) &&
-							stream->rtx_payload_types && g_hash_table_size(stream->rtx_payload_types) > 0) {
-						stream->video_rtx_payload_type = GPOINTER_TO_INT(g_hash_table_lookup(stream->rtx_payload_types, GINT_TO_POINTER(stream->video_payload_type)));
+				if(medium->payload_type < 0) {
+					medium->payload_type = header->type;
+					if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX) &&
+							medium->rtx_payload_types && g_hash_table_size(medium->rtx_payload_types) > 0) {
+						medium->rtx_payload_type = GPOINTER_TO_INT(g_hash_table_lookup(medium->rtx_payload_types, GINT_TO_POINTER(medium->payload_type)));
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Retransmissions will have payload type %d\n",
-							handle->handle_id, stream->video_rtx_payload_type);
+							handle->handle_id, medium->rtx_payload_type);
 					}
-					if(stream->video_codec == NULL) {
-						const char *codec = janus_get_codec_from_pt(handle->local_sdp, stream->video_payload_type);
+					if(medium->codec == NULL) {
+						const char *codec = janus_get_codec_from_pt(handle->local_sdp, medium->payload_type);
 						if(codec != NULL)
-							stream->video_codec = g_strdup(codec);
+							medium->codec = g_strdup(codec);
 					}
-					if(stream->video_is_keyframe == NULL && stream->video_codec != NULL) {
-						if(!strcasecmp(stream->video_codec, "vp8"))
-							stream->video_is_keyframe = &janus_vp8_is_keyframe;
-						else if(!strcasecmp(stream->video_codec, "vp9"))
-							stream->video_is_keyframe = &janus_vp9_is_keyframe;
-						else if(!strcasecmp(stream->video_codec, "h264"))
-							stream->video_is_keyframe = &janus_h264_is_keyframe;
+					if(medium->video_is_keyframe == NULL && medium->codec != NULL) {
+						if(!strcasecmp(medium->codec, "vp8"))
+							medium->video_is_keyframe = &janus_vp8_is_keyframe;
+						else if(!strcasecmp(medium->codec, "vp9"))
+							medium->video_is_keyframe = &janus_vp9_is_keyframe;
+						else if(!strcasecmp(medium->codec, "h264"))
+							medium->video_is_keyframe = &janus_h264_is_keyframe;
 					}
 				}
 				/* Do we need to dump this packet for debugging? */
@@ -4096,18 +3814,18 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 					janus_text2pcap_dump(handle->text2pcap, JANUS_TEXT2PCAP_RTP, FALSE, pkt->data, pkt->length,
 						"[session=%"SCNu64"][handle=%"SCNu64"]", session->session_id, handle->handle_id);
 				/* If this is video, check if this is a keyframe: if so, we empty our retransmit buffer for incoming NACKs */
-				if(video && stream->video_is_keyframe) {
+				if(video && medium->video_is_keyframe) {
 					int plen = 0;
 					char *payload = janus_rtp_payload(pkt->data, pkt->length, &plen);
-					if(stream->video_is_keyframe(payload, plen)) {
+					if(medium->video_is_keyframe(payload, plen)) {
 						JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Keyframe sent, cleaning retransmit buffer\n", handle->handle_id);
-						janus_cleanup_nack_buffer(0, stream, FALSE, TRUE);
+						janus_cleanup_nack_buffer(0, pc, FALSE, TRUE);
 					}
 				}
 				/* Before encrypting, check if we need to copy the unencrypted payload (e.g., for rtx/90000) */
 				janus_rtp_packet *p = NULL;
-				if(max_nack_queue > 0 && !pkt->retransmission && pkt->type == JANUS_ICE_PACKET_VIDEO && component->do_video_nacks &&
-						janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX)) {
+				if(max_nack_queue > 0 && !pkt->retransmission && pkt->type == JANUS_ICE_PACKET_VIDEO && medium->do_nacks &&
+						janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX)) {
 					/* Save the packet for retransmissions that may be needed later: start by
 					 * making room for two more bytes to store the original sequence number */
 					p = g_malloc(sizeof(janus_rtp_packet));
@@ -4120,7 +3838,7 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 					char *payload = janus_rtp_payload(pkt->data, pkt->length, &plen);
 					if(plen == 0) {
 						JANUS_LOG(LOG_WARN, "[%"SCNu64"] Discarding outgoing empty RTP packet\n", handle->handle_id);
-						janus_ice_free_rtp_packet(p);
+						janus_rtp_packet_free(p);
 						janus_ice_free_queued_packet(pkt);
 						return G_SOURCE_CONTINUE;
 					}
@@ -4135,7 +3853,7 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 				/* Encrypt SRTP */
 				int protected = pkt->length;
 				int res = janus_is_webrtc_encryption_enabled() ?
-					srtp_protect(component->dtls->srtp_out, pkt->data, &protected) : srtp_err_status_ok;
+					srtp_protect(pc->dtls->srtp_out, pkt->data, &protected) : srtp_err_status_ok;
 				if(res != srtp_err_status_ok) {
 					/* We don't spam the logs for every SRTP error: just take note of this, and print a summary later */
 					handle->srtp_errors_count++;
@@ -4146,10 +3864,10 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 					guint16 seq = ntohs(header->seq_number);
 					JANUS_LOG(LOG_DBG, "[%"SCNu64"] ... SRTP protect error... %s (len=%d-->%d, ts=%"SCNu32", seq=%"SCNu16")...\n",
 						handle->handle_id, janus_srtp_error_str(res), pkt->length, protected, timestamp, seq);
-					janus_ice_free_rtp_packet(p);
+					janus_rtp_packet_free(p);
 				} else {
 					/* Shoot! */
-					int sent = nice_agent_send(handle->agent, stream->stream_id, component->component_id, protected, pkt->data);
+					int sent = nice_agent_send(handle->agent, pc->stream_id, pc->component_id, protected, pkt->data);
 					if(sent < protected) {
 						JANUS_LOG(LOG_ERR, "[%"SCNu64"] ... only sent %d bytes? (was %d)\n", handle->handle_id, sent, protected);
 					}
@@ -4158,63 +3876,41 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 						/* Update the RTCP context as well */
 						janus_rtp_header *header = (janus_rtp_header *)pkt->data;
 						guint32 timestamp = ntohl(header->timestamp);
-						if(pkt->type == JANUS_ICE_PACKET_AUDIO) {
-							component->out_stats.audio.packets++;
-							component->out_stats.audio.bytes += pkt->length;
-							/* Last second outgoing audio */
-							gint64 now = janus_get_monotonic_time();
-							if(component->out_stats.audio.updated == 0)
-								component->out_stats.audio.updated = now;
-							if(now > component->out_stats.audio.updated &&
-									now - component->out_stats.audio.updated >= G_USEC_PER_SEC) {
-								component->out_stats.audio.bytes_lastsec = component->out_stats.audio.bytes_lastsec_temp;
-								component->out_stats.audio.bytes_lastsec_temp = 0;
-								component->out_stats.audio.updated = now;
-							}
-							component->out_stats.audio.bytes_lastsec_temp += pkt->length;
-							stream->audio_last_ts = timestamp;
-							if(stream->audio_first_ntp_ts == 0) {
-								struct timeval tv;
-								gettimeofday(&tv, NULL);
-								stream->audio_first_ntp_ts = (gint64)tv.tv_sec*G_USEC_PER_SEC + tv.tv_usec;
-								stream->audio_first_rtp_ts = timestamp;
-							}
-							/* Let's check if this was G.711: in case we may need to change the timestamp base */
-							rtcp_context *rtcp_ctx = stream->audio_rtcp_ctx;
-							int pt = header->type;
-							if((pt == 0 || pt == 8) && rtcp_ctx && (rtcp_ctx->tb == 48000))
-								rtcp_ctx->tb = 8000;
-						} else if(pkt->type == JANUS_ICE_PACKET_VIDEO) {
-							component->out_stats.video[0].packets++;
-							component->out_stats.video[0].bytes += pkt->length;
-							/* Last second outgoing video */
-							gint64 now = janus_get_monotonic_time();
-							if(component->out_stats.video[0].updated == 0)
-								component->out_stats.video[0].updated = now;
-							if(now > component->out_stats.video[0].updated &&
-									now - component->out_stats.video[0].updated >= G_USEC_PER_SEC) {
-								component->out_stats.video[0].bytes_lastsec = component->out_stats.video[0].bytes_lastsec_temp;
-								component->out_stats.video[0].bytes_lastsec_temp = 0;
-								component->out_stats.video[0].updated = now;
-							}
-							component->out_stats.video[0].bytes_lastsec_temp += pkt->length;
-							stream->video_last_ts = timestamp;
-							if(stream->video_first_ntp_ts[0] == 0) {
-								struct timeval tv;
-								gettimeofday(&tv, NULL);
-								stream->video_first_ntp_ts[0] = (gint64)tv.tv_sec*G_USEC_PER_SEC + tv.tv_usec;
-								stream->video_first_rtp_ts[0] = timestamp;
-							}
+						medium->out_stats.info[0].packets++;
+						medium->out_stats.info[0].bytes += pkt->length;
+						/* Last second outgoing audio */
+						gint64 now = janus_get_monotonic_time();
+						if(medium->out_stats.info[0].updated == 0)
+							medium->out_stats.info[0].updated = now;
+						if(now > medium->out_stats.info[0].updated &&
+								now - medium->out_stats.info[0].updated >= G_USEC_PER_SEC) {
+							medium->out_stats.info[0].bytes_lastsec = medium->out_stats.info[0].bytes_lastsec_temp;
+							medium->out_stats.info[0].bytes_lastsec_temp = 0;
+							medium->out_stats.info[0].updated = now;
+						}
+						medium->out_stats.info[0].bytes_lastsec_temp += pkt->length;
+						medium->last_ts = timestamp;
+						if(medium->first_ntp_ts[0] == 0) {
+							struct timeval tv;
+							gettimeofday(&tv, NULL);
+							medium->first_ntp_ts[0] = (gint64)tv.tv_sec*G_USEC_PER_SEC + tv.tv_usec;
+							medium->first_rtp_ts[0] = timestamp;
 						}
-						/* Update sent packets counter */
-						rtcp_context *rtcp_ctx = video ? stream->video_rtcp_ctx[0] : stream->audio_rtcp_ctx;
-						if(rtcp_ctx)
+						rtcp_context *rtcp_ctx = medium->rtcp_ctx[0];
+						if(rtcp_ctx) {
+							if(pkt->type == JANUS_ICE_PACKET_AUDIO) {
+								/* Let's check if this was G.711: in case we may need to change the timestamp base */
+								int pt = header->type;
+								if((pt == 0 || pt == 8) && (rtcp_ctx->tb == 48000))
+									rtcp_ctx->tb = 8000;
+							}
+							/* Update sent packets counter */
 							g_atomic_int_inc(&rtcp_ctx->sent_packets_since_last_rr);
+						}
 					}
 					if(max_nack_queue > 0 && !pkt->retransmission) {
 						/* Save the packet for retransmissions that may be needed later */
-						if((pkt->type == JANUS_ICE_PACKET_AUDIO && !component->do_audio_nacks) ||
-								(pkt->type == JANUS_ICE_PACKET_VIDEO && !component->do_video_nacks)) {
+						if(!medium->do_nacks) {
 							/* ... unless NACKs are disabled for this medium */
 							janus_ice_free_queued_packet(pkt);
 							return G_SOURCE_CONTINUE;
@@ -4230,64 +3926,54 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 						p->last_retransmit = 0;
 						janus_rtp_header *header = (janus_rtp_header *)pkt->data;
 						guint16 seq = ntohs(header->seq_number);
-						if(!video) {
-							if(component->audio_retransmit_buffer == NULL) {
-								component->audio_retransmit_buffer = g_queue_new();
-								component->audio_retransmit_seqs = g_hash_table_new(NULL, NULL);
-							}
-							g_queue_push_tail(component->audio_retransmit_buffer, p);
-							/* Insert in the table too, for quick lookup */
-							g_hash_table_insert(component->audio_retransmit_seqs, GUINT_TO_POINTER(seq), p);
-						} else {
-							if(component->video_retransmit_buffer == NULL) {
-								component->video_retransmit_buffer = g_queue_new();
-								component->video_retransmit_seqs = g_hash_table_new(NULL, NULL);
-							}
-							g_queue_push_tail(component->video_retransmit_buffer, p);
-							/* Insert in the table too, for quick lookup */
-							g_hash_table_insert(component->video_retransmit_seqs, GUINT_TO_POINTER(seq), p);
+						if(medium->retransmit_buffer == NULL) {
+							medium->retransmit_buffer = g_queue_new();
+							medium->retransmit_seqs = g_hash_table_new(NULL, NULL);
 						}
+						g_queue_push_tail(medium->retransmit_buffer, p);
+						/* Insert in the table too, for quick lookup */
+						g_hash_table_insert(medium->retransmit_seqs, GUINT_TO_POINTER(seq), p);
 					} else {
-						janus_ice_free_rtp_packet(p);
+						janus_rtp_packet_free(p);
 					}
 				}
 			}
 		} else if(pkt->type == JANUS_ICE_PACKET_DATA) {
 			/* Data */
-			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS)) {
+			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_DATA_CHANNELS)) {
 				janus_ice_free_queued_packet(pkt);
 				return G_SOURCE_CONTINUE;
 			}
 #ifdef HAVE_SCTP
-			if(!component->dtls) {
-				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT) && !component->noerrorlog) {
+			if(!pc->dtls) {
+				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT) && !medium->noerrorlog) {
 					JANUS_LOG(LOG_WARN, "[%"SCNu64"] SCTP stream component has no valid DTLS session (yet?)\n", handle->handle_id);
-					component->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
+					medium->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
 				}
 				janus_ice_free_queued_packet(pkt);
 				return G_SOURCE_CONTINUE;
 			}
-			component->noerrorlog = FALSE;
-			janus_dtls_wrap_sctp_data(component->dtls, pkt->label, pkt->data, pkt->length);
+			medium->noerrorlog = FALSE;
+			janus_dtls_wrap_sctp_data(pc->dtls, pkt->label, pkt->data, pkt->length);
 #endif
 		} else if(pkt->type == JANUS_ICE_PACKET_SCTP) {
 			/* SCTP data to push */
-			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS)) {
+			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_DATA_CHANNELS)) {
 				janus_ice_free_queued_packet(pkt);
 				return G_SOURCE_CONTINUE;
 			}
 #ifdef HAVE_SCTP
 			/* Encapsulate this data in DTLS and send it */
-			if(!component->dtls) {
-				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT) && !component->noerrorlog) {
+			if(!pc->dtls) {
+				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT) && !medium->noerrorlog) {
 					JANUS_LOG(LOG_WARN, "[%"SCNu64"] SCTP stream component has no valid DTLS session (yet?)\n", handle->handle_id);
-					component->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
+					medium->noerrorlog = TRUE;	/* Don't flood with the same error all over again */
 				}
 				janus_ice_free_queued_packet(pkt);
 				return G_SOURCE_CONTINUE;
 			}
-			component->noerrorlog = FALSE;
-			janus_dtls_send_sctp_data(component->dtls, pkt->data, pkt->length);
+			medium->noerrorlog = FALSE;
+			janus_dtls_send_sctp_data(pc->dtls, pkt->data, pkt->length);
 #endif
 		} else {
 			JANUS_LOG(LOG_WARN, "[%"SCNu64"] Unsupported packet type %d\n", handle->handle_id, pkt->type);
@@ -4297,7 +3983,7 @@ static gboolean janus_ice_outgoing_traffic_handle(janus_ice_handle *handle, janu
 	return G_SOURCE_CONTINUE;
 }
 
-static void janus_ice_queue_packet(janus_ice_handle *handle, janus_ice_queued_packet *pkt) {
+static void janus_ice_queue_packet(janus_handle *handle, janus_ice_queued_packet *pkt) {
 	/* TODO: There is a potential race condition where the "queued_packets"
 	 * could get released between the condition and pushing the packet. */
 	if(handle->queued_packets != NULL) {
@@ -4308,14 +3994,19 @@ static void janus_ice_queue_packet(janus_ice_handle *handle, janus_ice_queued_pa
 	}
 }
 
-void janus_ice_relay_rtp(janus_ice_handle *handle, int video, char *buf, int len) {
-	if(!handle || handle->queued_packets == NULL || buf == NULL || len < 1)
+void janus_ice_relay_rtp(janus_handle *handle, int mindex, gboolean video, char *buf, int len) {
+	if(!handle || !handle->pc || handle->queued_packets == NULL || buf == NULL || len < 1)
 		return;
-	if((!video && !janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO))
-			|| (video && !janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO)))
+	/* Find the right medium instance */
+	janus_handle_webrtc_medium *medium = (mindex != -1 ?
+			g_hash_table_lookup(handle->pc->media, GINT_TO_POINTER(mindex)) :
+			g_hash_table_lookup(handle->pc->media_bytype,
+				GINT_TO_POINTER(video ? JANUS_MEDIA_VIDEO : JANUS_MEDIA_AUDIO)));
+	if(!medium)
 		return;
 	/* Queue this packet */
 	janus_ice_queued_packet *pkt = g_malloc(sizeof(janus_ice_queued_packet));
+	pkt->mindex = medium->mindex;
 	pkt->data = g_malloc(len+SRTP_MAX_TAG_LEN);
 	memcpy(pkt->data, buf, len);
 	pkt->length = len;
@@ -4328,8 +4019,15 @@ void janus_ice_relay_rtp(janus_ice_handle *handle, int video, char *buf, int len
 	janus_ice_queue_packet(handle, pkt);
 }
 
-void janus_ice_relay_rtcp_internal(janus_ice_handle *handle, int video, char *buf, int len, gboolean filter_rtcp) {
-	if(!handle || handle->queued_packets == NULL || buf == NULL || len < 1)
+static void janus_ice_relay_rtcp_internal(janus_handle *handle, int mindex, gboolean video, char *buf, int len, gboolean filter_rtcp) {
+	if(!handle || !handle->pc || handle->queued_packets == NULL || buf == NULL || len < 1)
+		return;
+	/* Find the right medium instance */
+	janus_handle_webrtc_medium *medium = (mindex != -1 ?
+			g_hash_table_lookup(handle->pc->media, GINT_TO_POINTER(mindex)) :
+			g_hash_table_lookup(handle->pc->media_bytype,
+				GINT_TO_POINTER(video ? JANUS_MEDIA_VIDEO : JANUS_MEDIA_AUDIO)));
+	if(!medium)
 		return;
 	/* We use this internal method to check whether we need to filter RTCP (e.g., to make
 	 * sure we don't just forward any SR/RR from peers/plugins, but use our own) or it has
@@ -4338,9 +4036,6 @@ void janus_ice_relay_rtcp_internal(janus_ice_handle *handle, int video, char *bu
 	int rtcp_len = len;
 	if(filter_rtcp) {
 		/* FIXME Strip RR/SR/SDES/NACKs/etc. */
-		janus_ice_stream *stream = handle->stream;
-		if(stream == NULL)
-			return;
 		rtcp_buf = janus_rtcp_filter(buf, len, &rtcp_len);
 		if(rtcp_buf == NULL || rtcp_len < 1)
 			return;
@@ -4348,14 +4043,13 @@ void janus_ice_relay_rtcp_internal(janus_ice_handle *handle, int video, char *bu
 		 * leg. Note that this is only needed for RTCP packets coming from plugins: the
 		 * ones created by the core already have the right SSRCs in the right place */
 		JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Fixing SSRCs (local %u, peer %u)\n", handle->handle_id,
-			video ? stream->video_ssrc : stream->audio_ssrc,
-			video ? stream->video_ssrc_peer[0] : stream->audio_ssrc_peer);
+			medium->ssrc, medium->ssrc_peer[0]);
 		janus_rtcp_fix_ssrc(NULL, rtcp_buf, rtcp_len, 1,
-			video ? stream->video_ssrc : stream->audio_ssrc,
-			video ? stream->video_ssrc_peer[0] : stream->audio_ssrc_peer);
+			medium->ssrc, medium->ssrc_peer[0]);
 	}
 	/* Queue this packet */
 	janus_ice_queued_packet *pkt = g_malloc(sizeof(janus_ice_queued_packet));
+	pkt->mindex = medium->mindex;
 	pkt->data = g_malloc(rtcp_len+SRTP_MAX_TAG_LEN+4);
 	memcpy(pkt->data, rtcp_buf, rtcp_len);
 	pkt->length = rtcp_len;
@@ -4372,38 +4066,48 @@ void janus_ice_relay_rtcp_internal(janus_ice_handle *handle, int video, char *bu
 	}
 }
 
-void janus_ice_relay_rtcp(janus_ice_handle *handle, int video, char *buf, int len) {
-	janus_ice_relay_rtcp_internal(handle, video, buf, len, TRUE);
+void janus_ice_relay_rtcp(janus_handle *handle, int mindex, gboolean video, char *buf, int len) {
+	janus_ice_relay_rtcp_internal(handle, mindex, video, buf, len, TRUE);
 	/* If this is a PLI and we're simulcasting, send a PLI on other layers as well */
-	if(janus_rtcp_has_pli(buf, len)) {
-		janus_ice_stream *stream = handle->stream;
-		if(stream == NULL)
+	if(video && janus_rtcp_has_pli(buf, len)) {
+		/* Find the right medium instance */
+		janus_handle_webrtc_medium *medium = (mindex != -1 ?
+				g_hash_table_lookup(handle->pc->media, GINT_TO_POINTER(mindex)) :
+				g_hash_table_lookup(handle->pc->media_bytype,
+					GINT_TO_POINTER(video ? JANUS_MEDIA_VIDEO : JANUS_MEDIA_AUDIO)));
+		if(!medium)
 			return;
-		if(stream->video_ssrc_peer[1]) {
+		if(medium->ssrc_peer[1]) {
 			char plibuf[12];
 			memset(plibuf, 0, 12);
 			janus_rtcp_pli((char *)&plibuf, 12);
 			janus_rtcp_fix_ssrc(NULL, plibuf, sizeof(plibuf), 1,
-				stream->video_ssrc, stream->video_ssrc_peer[1]);
-			janus_ice_relay_rtcp_internal(handle, 1, plibuf, sizeof(plibuf), FALSE);
+				medium->ssrc, medium->ssrc_peer[1]);
+			janus_ice_relay_rtcp_internal(handle, mindex, TRUE, plibuf, sizeof(plibuf), FALSE);
 		}
-		if(stream->video_ssrc_peer[2]) {
+		if(medium->ssrc_peer[2]) {
 			char plibuf[12];
 			memset(plibuf, 0, 12);
 			janus_rtcp_pli((char *)&plibuf, 12);
 			janus_rtcp_fix_ssrc(NULL, plibuf, sizeof(plibuf), 1,
-				stream->video_ssrc, stream->video_ssrc_peer[2]);
-			janus_ice_relay_rtcp_internal(handle, 1, plibuf, sizeof(plibuf), FALSE);
+				medium->ssrc, medium->ssrc_peer[2]);
+			janus_ice_relay_rtcp_internal(handle, mindex, TRUE, plibuf, sizeof(plibuf), FALSE);
 		}
 	}
 }
 
 #ifdef HAVE_SCTP
-void janus_ice_relay_data(janus_ice_handle *handle, char *label, char *buf, int len) {
-	if(!handle || handle->queued_packets == NULL || buf == NULL || len < 1)
+void janus_ice_relay_data(janus_handle *handle, char *label, char *buf, int len) {
+	if(!handle || !handle->pc || handle->queued_packets == NULL || buf == NULL || len < 1)
+		return;
+	/* Find the right medium instance */
+	janus_handle_webrtc_medium *medium = g_hash_table_lookup(handle->pc->media_bytype,
+		GINT_TO_POINTER(JANUS_MEDIA_DATA));
+	if(!medium)
 		return;
 	/* Queue this packet */
 	janus_ice_queued_packet *pkt = g_malloc(sizeof(janus_ice_queued_packet));
+	pkt->mindex = medium->mindex;
 	pkt->data = g_malloc(len);
 	memcpy(pkt->data, buf, len);
 	pkt->length = len;
@@ -4417,12 +4121,18 @@ void janus_ice_relay_data(janus_ice_handle *handle, char *label, char *buf, int
 }
 #endif
 
-void janus_ice_relay_sctp(janus_ice_handle *handle, char *buffer, int length) {
+void janus_ice_relay_sctp(janus_handle *handle, char *buffer, int length) {
 #ifdef HAVE_SCTP
-	if(!handle || handle->queued_packets == NULL || buffer == NULL || length < 1)
+	if(!handle || !handle->pc || handle->queued_packets == NULL || buffer == NULL || length < 1)
+		return;
+	/* Find the right medium instance */
+	janus_handle_webrtc_medium *medium = g_hash_table_lookup(handle->pc->media_bytype,
+		GINT_TO_POINTER(JANUS_MEDIA_DATA));
+	if(!medium)
 		return;
 	/* Queue this packet */
 	janus_ice_queued_packet *pkt = g_malloc(sizeof(janus_ice_queued_packet));
+	pkt->mindex = medium->mindex;
 	pkt->data = g_malloc(length);
 	memcpy(pkt->data, buffer, length);
 	pkt->length = length;
@@ -4436,27 +4146,18 @@ void janus_ice_relay_sctp(janus_ice_handle *handle, char *buffer, int length) {
 #endif
 }
 
-void janus_ice_dtls_handshake_done(janus_ice_handle *handle, janus_ice_component *component) {
-	if(!handle || !component)
+void janus_handle_dtls_handshake_done(janus_handle *handle) {
+	if(!handle)
 		return;
 	JANUS_LOG(LOG_VERB, "[%"SCNu64"] The DTLS handshake for the component %d in stream %d has been completed\n",
-		handle->handle_id, component->component_id, component->stream_id);
-	/* Check if all components are ready */
+		handle->handle_id, handle->pc->component_id, handle->pc->stream_id);
 	janus_mutex_lock(&handle->mutex);
-	if(handle->stream && janus_is_webrtc_encryption_enabled()) {
-		if(handle->stream->component && (!handle->stream->component->dtls ||
-				!handle->stream->component->dtls->srtp_valid)) {
-			/* Still waiting for this component to become ready */
-			janus_mutex_unlock(&handle->mutex);
-			return;
-		}
-	}
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY)) {
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY)) {
 		/* Already notified */
 		janus_mutex_unlock(&handle->mutex);
 		return;
 	}
-	janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY);
+	janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY);
 	/* Create a source for RTCP and one for stats */
 	handle->rtcp_source = g_timeout_source_new_seconds(1);
 	g_source_set_priority(handle->rtcp_source, G_PRIORITY_DEFAULT);
diff --git a/ice.h b/ice.h
index 76a520841c..ba0e612b12 100644
--- a/ice.h
+++ b/ice.h
@@ -1,9 +1,13 @@
 /*! \file    ice.h
  * \author   Lorenzo Miniero <lorenzo@meetecho.com>
  * \copyright GNU General Public License v3
- * \brief    ICE/STUN/TURN processing (headers)
- * \details  Implementation (based on libnice) of the ICE process. The
- * code handles the whole ICE process, from the gathering of candidates
+ * \brief    Janus handles and ICE/STUN/TURN processing (headers)
+ * \details  A Janus handle represents an abstraction of the communication
+ * between a user and a specific plugin, within a Janus session. This is
+ * particularly important in terms of media connectivity, as each handle
+ * can be associated with a single WebRTC PeerConnection. This code also
+ * contains the implementation (based on libnice) of a WebRTC PeerConnection.
+ * The code handles the whole ICE process, from the gathering of candidates
  * to the final setup of a virtual channel RTP and RTCP can be transported
  * on. Incoming RTP and RTCP packets from peers are relayed to the associated
  * plugins by means of the incoming_rtp and incoming_rtcp callbacks. Packets
@@ -178,40 +182,46 @@ gboolean janus_is_opaqueid_in_api_enabled(void);
 const gchar *janus_get_ice_state_name(gint state);
 
 
-/*! \brief Janus ICE handle/session */
-typedef struct janus_ice_handle janus_ice_handle;
-/*! \brief Janus ICE stream */
-typedef struct janus_ice_stream janus_ice_stream;
-/*! \brief Janus ICE component */
-typedef struct janus_ice_component janus_ice_component;
+/*! \brief Janus handle in a session */
+typedef struct janus_handle janus_handle;
+/*! \brief Janus handle WebRTC PeerConnection */
+typedef struct janus_handle_webrtc janus_handle_webrtc;
+/*! \brief A single medium (i.e., m-line) in a Janus handle PeerConnection: can be bidirectional */
+typedef struct janus_handle_webrtc_medium janus_handle_webrtc_medium;
 /*! \brief Helper to handle pending trickle candidates (e.g., when we're still waiting for an offer) */
-typedef struct janus_ice_trickle janus_ice_trickle;
-
-#define JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER	(1 << 0)
-#define JANUS_ICE_HANDLE_WEBRTC_START				(1 << 1)
-#define JANUS_ICE_HANDLE_WEBRTC_READY				(1 << 2)
-#define JANUS_ICE_HANDLE_WEBRTC_STOP				(1 << 3)
-#define JANUS_ICE_HANDLE_WEBRTC_ALERT				(1 << 4)
-#define JANUS_ICE_HANDLE_WEBRTC_NEGOTIATED			(1 << 5)
-#define JANUS_ICE_HANDLE_WEBRTC_TRICKLE				(1 << 7)
-#define JANUS_ICE_HANDLE_WEBRTC_ALL_TRICKLES		(1 << 8)
-#define JANUS_ICE_HANDLE_WEBRTC_TRICKLE_SYNCED		(1 << 9)
-#define JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS		(1 << 10)
-#define JANUS_ICE_HANDLE_WEBRTC_CLEANING			(1 << 11)
-#define JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO			(1 << 12)
-#define JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO			(1 << 13)
-#define JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER			(1 << 14)
-#define JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER			(1 << 15)
-#define JANUS_ICE_HANDLE_WEBRTC_HAS_AGENT			(1 << 16)
-#define JANUS_ICE_HANDLE_WEBRTC_ICE_RESTART			(1 << 17)
-#define JANUS_ICE_HANDLE_WEBRTC_RESEND_TRICKLES		(1 << 18)
-#define JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX			(1 << 19)
-#define JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP	(1 << 20)
-
+typedef struct janus_trickle janus_trickle;
+
+#define JANUS_HANDLE_WEBRTC_PROCESSING_OFFER	(1 << 0)
+#define JANUS_HANDLE_WEBRTC_START				(1 << 1)
+#define JANUS_HANDLE_WEBRTC_READY				(1 << 2)
+#define JANUS_HANDLE_WEBRTC_STOP				(1 << 3)
+#define JANUS_HANDLE_WEBRTC_ALERT				(1 << 4)
+#define JANUS_HANDLE_WEBRTC_NEGOTIATED			(1 << 5)
+#define JANUS_HANDLE_WEBRTC_TRICKLE				(1 << 7)
+#define JANUS_HANDLE_WEBRTC_ALL_TRICKLES		(1 << 8)
+#define JANUS_HANDLE_WEBRTC_TRICKLE_SYNCED		(1 << 9)
+#define JANUS_HANDLE_WEBRTC_DATA_CHANNELS		(1 << 10)
+#define JANUS_HANDLE_WEBRTC_CLEANING			(1 << 11)
+#define JANUS_HANDLE_WEBRTC_GOT_OFFER			(1 << 14)
+#define JANUS_HANDLE_WEBRTC_GOT_ANSWER			(1 << 15)
+#define JANUS_HANDLE_WEBRTC_HAS_AGENT			(1 << 16)
+#define JANUS_HANDLE_WEBRTC_ICE_RESTART			(1 << 17)
+#define JANUS_HANDLE_WEBRTC_RESEND_TRICKLES		(1 << 18)
+#define JANUS_HANDLE_WEBRTC_RFC4588_RTX			(1 << 19)
+#define JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP	(1 << 20)
+
+
+/*! \brief Janus media types */
+typedef enum janus_media_type {
+	JANUS_MEDIA_UNKNOWN = 0,
+	JANUS_MEDIA_AUDIO,
+	JANUS_MEDIA_VIDEO,
+	JANUS_MEDIA_DATA
+} janus_media_type;
 
 /*! \brief Janus media statistics
  * \note To improve with more stuff */
-typedef struct janus_ice_stats_info {
+typedef struct janus_media_stats_info {
 	/*! \brief Packets sent or received */
 	guint32 packets;
 	/*! \brief Bytes sent or received */
@@ -224,27 +234,21 @@ typedef struct janus_ice_stats_info {
 	gboolean notified_lastsec;
 	/*! \brief Number of NACKs sent or received */
 	guint32 nacks;
-} janus_ice_stats_info;
+} janus_media_stats_info;
 
 /*! \brief Janus media statistics container
  * \note To improve with more stuff */
-typedef struct janus_ice_stats {
-	/*! \brief Audio info */
-	janus_ice_stats_info audio;
-	/*! \brief Video info (considering we may be simulcasting) */
-	janus_ice_stats_info video[3];
-	/*! \brief Data info */
-	janus_ice_stats_info data;
-	/*! \brief Last known count of lost audio packets (for slow_link) */
-	guint sl_lost_count_audio;
-	/*! \brief Last known count of lost video packets (for slow_link) */
-	guint sl_lost_count_video;
-} janus_ice_stats;
+typedef struct janus_media_stats {
+	/*! \brief Media stats info (considering we may be simulcasting) */
+	janus_media_stats_info info[3];
+	/*! \brief Last known count of lost packets (for slow_link) */
+	guint sl_lost_count;
+} janus_media_stats;
 
 /*! \brief Quick helper method to notify a WebRTC hangup through the Janus API
- * @param handle The janus_ice_handle instance this event refers to
+ * @param handle The janus_handle instance this event refers to
  * @param reason A description of why this happened */
-void janus_ice_notify_hangup(janus_ice_handle *handle, const char *reason);
+void janus_ice_notify_hangup(janus_handle *handle, const char *reason);
 
 
 /*! \brief Quick helper method to check if a plugin session associated with a Janus handle is still valid
@@ -270,8 +274,8 @@ enum {
 };
 
 
-/*! \brief Janus ICE handle */
-struct janus_ice_handle {
+/*! \brief Janus handle */
+struct janus_handle {
 	/*! \brief Opaque pointer to the core/peer session */
 	void *session;
 	/*! \brief Handle identifier, guaranteed to be non-zero */
@@ -302,18 +306,12 @@ struct janus_ice_handle {
 	gint64 agent_created;
 	/*! \brief ICE role (controlling or controlled) */
 	gboolean controlling;
-	/*! \brief Audio mid (media ID) */
-	gchar *audio_mid;
-	/*! \brief Video mid (media ID) */
-	gchar *video_mid;
-	/*! \brief Data channel mid (media ID) */
-	gchar *data_mid;
-	/*! \brief Main mid (will be a pointer to one of the above) */
-	gchar *stream_mid;
+	/*! \brief Main mid */
+	gchar *pc_mid;
 	/*! \brief ICE Stream ID */
 	guint stream_id;
-	/*! \brief ICE stream */
-	janus_ice_stream *stream;
+	/*! \brief WebRTC PeerConnection, if any */
+	janus_handle_webrtc *pc;
 	/*! \brief RTP profile set by caller (so that we can match it) */
 	gchar *rtp_profile;
 	/*! \brief SDP generated locally (just for debugging purposes) */
@@ -346,71 +344,45 @@ struct janus_ice_handle {
 	janus_refcount ref;
 };
 
-/*! \brief Janus ICE stream */
-struct janus_ice_stream {
-	/*! \brief Janus ICE handle this stream belongs to */
-	janus_ice_handle *handle;
+/*! \brief Janus handle WebRTC PeerConnection */
+struct janus_handle_webrtc {
+	/*! \brief Janus handle this stream belongs to */
+	janus_handle *handle;
 	/*! \brief libnice ICE stream ID */
 	guint stream_id;
+	/*! \brief libnice ICE component ID */
+	guint component_id;
 	/*! \brief Whether this stream is ready to be used */
 	gint cdone:1;
-	/*! \brief Audio SSRC of the server for this stream */
-	guint32 audio_ssrc;
-	/*! \brief Video SSRC of the server for this stream */
-	guint32 video_ssrc;
-	/*! \brief Video retransmission SSRC of the peer for this stream */
-	guint32 video_ssrc_rtx;
-	/*! \brief Audio SSRC of the peer for this stream */
-	guint32 audio_ssrc_peer, audio_ssrc_peer_new, audio_ssrc_peer_orig;
-	/*! \brief Video SSRC(s) of the peer for this stream (may be simulcasting) */
-	guint32 video_ssrc_peer[3], video_ssrc_peer_new[3], video_ssrc_peer_orig[3], video_ssrc_peer_temp;
-	/*! \brief Video retransmissions SSRC(s) of the peer for this stream */
-	guint32 video_ssrc_peer_rtx[3], video_ssrc_peer_rtx_new[3], video_ssrc_peer_rtx_orig[3];
-	/*! \brief Array of RTP Stream IDs (for Firefox simulcasting, if enabled) */
-	char *rid[3];
-	/*! \brief Whether we should use the legacy simulcast syntax (a=simulcast:recv rid=..) or the proper one (a=simulcast:recv ..) */
-	gboolean legacy_rid;
-	/*! \brief RTP switching context(s) in case of renegotiations (audio+video and/or simulcast) */
-	janus_rtp_switching_context rtp_ctx[3];
-	/*! \brief List of payload types we can expect for audio */
-	GList *audio_payload_types;
-	/*! \brief List of payload types we can expect for video */
-	GList *video_payload_types;
-	/*! \brief Mapping of rtx payload types to actual media-related packet types */
-	GHashTable *rtx_payload_types;
-	/*! \brief RTP payload types of this stream */
-	gint audio_payload_type, video_payload_type, video_rtx_payload_type;
-	/*! \brief Codecs used by this stream */
-	char *audio_codec, *video_codec;
-	/*! \brief Pointer to function to check if a packet is a keyframe (depends on negotiated codec) */
-	gboolean (* video_is_keyframe)(const char* buffer, int len);
-	/*! \brief Media direction */
-	gboolean audio_send, audio_recv, video_send, video_recv;
-	/*! \brief RTCP context for the audio stream */
-	janus_rtcp_context *audio_rtcp_ctx;
-	/*! \brief RTCP context(s) for the video stream (may be simulcasting) */
-	janus_rtcp_context *video_rtcp_ctx[3];
-	/*! \brief Map(s) of the NACKed packets (to track retransmissions and avoid duplicates) */
-	GHashTable *rtx_nacked[3];
-	/*! \brief First received audio NTP timestamp */
-	gint64 audio_first_ntp_ts;
-	/*! \brief First received audio RTP timestamp */
-	guint32 audio_first_rtp_ts;
-	/*! \brief First received video NTP timestamp (for all simulcast video streams) */
-	gint64 video_first_ntp_ts[3];
-	/*! \brief First received video NTP RTP timestamp (for all simulcast video streams) */
-	guint32 video_first_rtp_ts[3];
-	/*! \brief Last sent audio RTP timestamp */
-	guint32 audio_last_ts;
-	/*! \brief Last sent video RTP timestamp */
-	guint32 video_last_ts;
+	/*! \brief libnice ICE component state */
+	guint state;
+	/*! \brief Monotonic time of when ICE has successfully connected */
+	gint64 ice_connected;
+	/*! \brief GLib list of libnice remote candidates for this component */
+	GSList *candidates;
+	/*! \brief GLib list of local candidates for this component (summary) */
+	GSList *local_candidates;
+	/*! \brief GLib list of remote candidates for this component (summary) */
+	GSList *remote_candidates;
+	/*! \brief String representation of the selected pair as notified by libnice (foundations) */
+	gchar *selected_pair;
+	/*! \brief Whether the setup of remote candidates for this component has started or not */
+	gboolean process_started;
+	/*! \brief Timer to check when we should consider ICE as failed */
+	GSource *icestate_source;
+	/*! \brief Time of when we first detected an ICE failed (we'll need this for the timer above) */
+	gint64 icefailed_detected;
+	/*! \brief Re-transmission timer for DTLS */
+	GSource *dtlsrt_source;
+	/*! \brief DTLS-SRTP stack */
+	janus_dtls_srtp *dtls;
 	/*! \brief SDES mid RTP extension ID */
 	gint mid_ext_id;
 	/*! \brief RTP Stream extension ID, and the related rtx one */
 	gint rid_ext_id, ridrtx_ext_id;
 	/*! \brief Frame marking extension ID */
 	gint framemarking_ext_id;
-	/*! \brief Whether we do transport wide cc for video */
+	/*! \brief Whether we do transport wide cc */
 	gboolean do_transport_wide_cc;
 	/*! \brief Transport wide cc rtp ext ID */
 	gint transport_wide_cc_ext_id;
@@ -426,6 +398,8 @@ struct janus_ice_stream {
 	GSList *transport_wide_received_seq_nums;
 	/*! \brief DTLS role of the server for this stream */
 	janus_dtls_role dtls_role;
+	/*! \brief Data exchanged for DTLS handshakes and messages */
+	janus_media_stats dtls_in_stats, dtls_out_stats;
 	/*! \brief Hashing algorhitm used by the peer for the DTLS certificate (e.g., "SHA-256") */
 	gchar *remote_hashing;
 	/*! \brief Hashed fingerprint of the peer's certificate, as parsed in SDP */
@@ -434,10 +408,15 @@ struct janus_ice_stream {
 	gchar *ruser;
 	/*! \brief The ICE password for this stream */
 	gchar *rpass;
-	/*! \brief GLib hash table of components (IDs are the keys) */
-	GHashTable *components;
-	/*! \brief ICE component */
-	janus_ice_component *component;
+	/*! \brief GLib hash table of media (m-line indexes are the keys) */
+	GHashTable *media;
+	/*! \brief GLib hash table of media (SSRCs are the keys) */
+	GHashTable *media_byssrc;
+	/*! \brief GLib hash table of media (mids are the keys) */
+	GHashTable *media_bymid;
+	/*! \brief GLib hash table of media (media types are the keys)
+	 * @note Temporary! Will go away very soon*/
+	GHashTable *media_bytype;
 	/*! \brief Helper flag to avoid flooding the console with the same error all over again */
 	gboolean noerrorlog;
 	/*! \brief Mutex to lock/unlock this stream */
@@ -449,44 +428,58 @@ struct janus_ice_stream {
 };
 
 #define LAST_SEQS_MAX_LEN 160
-/*! \brief Janus ICE component */
-struct janus_ice_component {
-	/*! \brief Janus ICE stream this component belongs to */
-	janus_ice_stream *stream;
-	/*! \brief libnice ICE stream ID */
-	guint stream_id;
-	/*! \brief libnice ICE component ID */
-	guint component_id;
-	/*! \brief libnice ICE component state */
-	guint state;
-	/*! \brief Monotonic time of when this component has successfully connected */
-	gint64 component_connected;
-	/*! \brief GLib list of libnice remote candidates for this component */
-	GSList *candidates;
-	/*! \brief GLib list of local candidates for this component (summary) */
-	GSList *local_candidates;
-	/*! \brief GLib list of remote candidates for this component (summary) */
-	GSList *remote_candidates;
-	/*! \brief String representation of the selected pair as notified by libnice (foundations) */
-	gchar *selected_pair;
-	/*! \brief Whether the setup of remote candidates for this component has started or not */
-	gboolean process_started;
-	/*! \brief Timer to check when we should consider ICE as failed */
-	GSource *icestate_source;
-	/*! \brief Time of when we first detected an ICE failed (we'll need this for the timer above) */
-	gint64 icefailed_detected;
-	/*! \brief Re-transmission timer for DTLS */
-	GSource *dtlsrt_source;
-	/*! \brief DTLS-SRTP stack */
-	janus_dtls_srtp *dtls;
-	/*! \brief Whether we should do NACKs (in or out) for audio */
-	gboolean do_audio_nacks;
-	/*! \brief Whether we should do NACKs (in or out) for video */
-	gboolean do_video_nacks;
+/*! \brief A single media in a PeerConnection */
+struct janus_handle_webrtc_medium {
+	/*! \brief WebRTC PeerConnection this m-line belongs to */
+	janus_handle_webrtc *pc;
+	/*! \brief Type of this medium */
+	janus_media_type type;
+	/*! \brief Index of this medium in the media list */
+	int mindex;
+	/*! \brief Media ID */
+	char *mid;
+	/*! \brief SSRC of the server for this medium */
+	guint32 ssrc;
+	/*! \brief Retransmission SSRC of the server for this medium */
+	guint32 ssrc_rtx;
+	/*! \brief SSRC(s) of the peer for this medium (may be simulcasting) */
+	guint32 ssrc_peer[3], ssrc_peer_new[3], ssrc_peer_orig[3], ssrc_peer_temp;
+	/*! \brief Retransmissions SSRC(s) of the peer for this medium (may be simulcasting) */
+	guint32 ssrc_peer_rtx[3], ssrc_peer_rtx_new[3], ssrc_peer_rtx_orig[3];
+	/*! \brief Array of RTP Stream IDs (for Firefox simulcasting, if enabled) */
+	char *rid[3];
+	/*! \brief Whether we should use the legacy simulcast syntax (a=simulcast:recv rid=..) or the proper one (a=simulcast:recv ..) */
+	gboolean legacy_rid;
+	/*! \brief RTP switching context(s) in case of renegotiations (audio+video and/or simulcast) */
+	janus_rtp_switching_context rtp_ctx[3];
+	/*! \brief List of payload types we can expect */
+	GList *payload_types;
+	/*! \brief Mapping of rtx payload types to actual media-related packet types */
+	GHashTable *rtx_payload_types;
+	/*! \brief RTP payload types for this medium */
+	gint payload_type, rtx_payload_type;
+	/*! \brief Codec used in this medium */
+	char *codec;
+	/*! \brief Pointer to function to check if a packet is a keyframe (depends on negotiated codec; video only) */
+	gboolean (* video_is_keyframe)(const char* buffer, int len);
+	/*! \brief Media direction */
+	gboolean send, recv;
+	/*! \brief RTCP context(s) for the medium (may be simulcasting) */
+	janus_rtcp_context *rtcp_ctx[3];
+	/*! \brief Map(s) of the NACKed packets (to track retransmissions and avoid duplicates) */
+	GHashTable *rtx_nacked[3];
+	/*! \brief First received NTP timestamp */
+	gint64 first_ntp_ts[3];
+	/*! \brief First received RTP timestamp */
+	guint32 first_rtp_ts[3];
+	/*! \brief Last sent RTP timestamp */
+	guint32 last_ts;
+	/*! \brief Whether we should do NACKs (in or out) for this medium */
+	gboolean do_nacks;
 	/*! \brief List of previously sent janus_rtp_packet RTP packets, in case we receive NACKs */
-	GQueue *audio_retransmit_buffer, *video_retransmit_buffer;
+	GQueue *retransmit_buffer;
 	/*! \brief HashTable of retransmittable sequence numbers, in case we receive NACKs */
-	GHashTable *audio_retransmit_seqs, *video_retransmit_seqs;
+	GHashTable *retransmit_seqs;
 	/*! \brief Current sequence number for the RFC4588 rtx SSRC session */
 	guint16 rtx_seq_number;
 	/*! \brief Last time a log message about sending retransmits was printed */
@@ -497,28 +490,32 @@ struct janus_ice_component {
 	gint64 nack_sent_log_ts;
 	/*! \brief Number of NACKs sent since last log message */
 	guint nack_sent_recent_cnt;
-	/*! \brief List of recently received audio sequence numbers (as a support to NACK generation) */
-	janus_seq_info *last_seqs_audio;
-	/*! \brief List of recently received video sequence numbers (as a support to NACK generation, for each simulcast SSRC) */
-	janus_seq_info *last_seqs_video[3];
-	/*! \brief Stats for incoming data (audio/video/data) */
-	janus_ice_stats in_stats;
-	/*! \brief Stats for outgoing data (audio/video/data) */
-	janus_ice_stats out_stats;
+	/*! \brief List of recently received sequence numbers (as a support to NACK generation, for each simulcast SSRC if needed) */
+	janus_seq_info *last_seqs[3];
+	/*! \brief Stats for incoming data */
+	janus_media_stats in_stats;
+	/*! \brief Stats for outgoing data */
+	janus_media_stats out_stats;
 	/*! \brief Helper flag to avoid flooding the console with the same error all over again */
 	gboolean noerrorlog;
-	/*! \brief Mutex to lock/unlock this component */
+	/*! \brief Mutex to lock/unlock this medium */
 	janus_mutex mutex;
 	/*! \brief Atomic flag to check if this instance has been destroyed */
 	volatile gint destroyed;
 	/*! \brief Reference counter for this instance */
 	janus_refcount ref;
 };
+/*! \brief Method to quickly create a medium to be added to a handle PeerConnection
+ * @note This will autogenerate SSRCs, if needed
+ * @param[in] handle The Janus handle instance to add the medium to
+ * @param[in] type The medium type
+ * @returns A pointer to the new medium, if successful, or NULL otherwise */
+janus_handle_webrtc_medium *janus_handle_webrtc_medium_create(janus_handle *handle, janus_media_type type);
 
 /*! \brief Helper to handle pending trickle candidates (e.g., when we're still waiting for an offer) */
-struct janus_ice_trickle {
-	/*! \brief Janus ICE handle this trickle candidate belongs to */
-	janus_ice_handle *handle;
+struct janus_trickle {
+	/*! \brief Janus handle this trickle candidate belongs to */
+	janus_handle *handle;
 	/*! \brief Monotonic time of when this trickle candidate has been received */
 	gint64 received;
 	/*! \brief Janus API transaction ID of the original trickle request */
@@ -530,125 +527,120 @@ struct janus_ice_trickle {
 /** @name Janus ICE trickle candidates methods
  */
 ///@{
-/*! \brief Helper method to allocate a janus_ice_trickle instance
+/*! \brief Helper method to allocate a janus_trickle instance
  * @param[in] transaction The Janus API ID of the original trickle request
  * @param[in] candidate The trickle candidate, as a Jansson object
  * @returns a pointer to the new instance, if successful, NULL otherwise */
-janus_ice_trickle *janus_ice_trickle_new(const char *transaction, json_t *candidate);
+janus_trickle *janus_trickle_new(const char *transaction, json_t *candidate);
 /*! \brief Helper method to parse trickle candidates
- * @param[in] handle The Janus ICE handle this candidate belongs to
+ * @param[in] handle The Janus handle this candidate belongs to
  * @param[in] candidate The trickle candidate to parse, as a Jansson object
  * @param[in,out] error Error string describing the failure, if any
  * @returns 0 in case of success, any code from apierror.h in case of failure */
-gint janus_ice_trickle_parse(janus_ice_handle *handle, json_t *candidate, const char **error);
-/*! \brief Helper method to destroy a janus_ice_trickle instance
- * @param[in] trickle The janus_ice_trickle instance to destroy */
-void janus_ice_trickle_destroy(janus_ice_trickle *trickle);
+gint janus_trickle_parse(janus_handle *handle, json_t *candidate, const char **error);
+/*! \brief Helper method to destroy a janus_trickle instance
+ * @param[in] trickle The janus_trickle instance to destroy */
+void janus_trickle_destroy(janus_trickle *trickle);
 ///@}
 
 
-/** @name Janus ICE handle methods
+/** @name Janus handle methods
  */
 ///@{
-/*! \brief Method to create a new Janus ICE handle
+/*! \brief Method to create a new Janus handle
  * @param[in] core_session The core/peer session this ICE handle will belong to
  * @param[in] opaque_id The opaque identifier provided by the creator, if any (optional)
- * @returns The created Janus ICE handle if successful, NULL otherwise */
-janus_ice_handle *janus_ice_handle_create(void *core_session, const char *opaque_id);
-/*! \brief Method to attach a Janus ICE handle to a plugin
+ * @returns The created Janus handle if successful, NULL otherwise */
+janus_handle *janus_handle_create(void *core_session, const char *opaque_id);
+/*! \brief Method to attach a Janus handle to a plugin
  * \details This method is very important, as it allows plugins to send/receive media (RTP/RTCP) to/from a WebRTC peer.
  * @param[in] core_session The core/peer session this ICE handle belongs to
- * @param[in] handle The Janus ICE handle
+ * @param[in] handle The Janus handle
  * @param[in] plugin The plugin the ICE handle needs to be attached to
  * @returns 0 in case of success, a negative integer otherwise */
-gint janus_ice_handle_attach_plugin(void *core_session, janus_ice_handle *handle, janus_plugin *plugin);
-/*! \brief Method to destroy a Janus ICE handle
+gint janus_handle_attach_plugin(void *core_session, janus_handle *handle, janus_plugin *plugin);
+/*! \brief Method to destroy a Janus handle
  * @param[in] core_session The core/peer session this ICE handle belongs to
- * @param[in] handle The Janus ICE handle to destroy
+ * @param[in] handle The Janus handle to destroy
  * @returns 0 in case of success, a negative integer otherwise */
-gint janus_ice_handle_destroy(void *core_session, janus_ice_handle *handle);
-/*! \brief Method to only hangup (e.g., DTLS alert) the WebRTC PeerConnection allocated by a Janus ICE handle
- * @param[in] handle The Janus ICE handle instance managing the WebRTC PeerConnection to hangup
+gint janus_handle_destroy(void *core_session, janus_handle *handle);
+/*! \brief Method to only hangup (e.g., DTLS alert) the WebRTC PeerConnection allocated by a Janus handle
+ * @param[in] handle The Janus handle instance managing the WebRTC PeerConnection to hangup
  * @param[in] reason A description of why this happened */
-void janus_ice_webrtc_hangup(janus_ice_handle *handle, const char *reason);
-/*! \brief Method to only free resources related to a specific ICE stream allocated by a Janus ICE handle
- * @param[in] stream The Janus ICE stream instance to free */
-void janus_ice_stream_destroy(janus_ice_stream *stream);
-/*! \brief Method to only free resources related to a specific ICE component allocated by a Janus ICE handle
- * @param[in] component The Janus ICE component instance to free */
-void janus_ice_component_destroy(janus_ice_component *component);
+void janus_handle_webrtc_hangup(janus_handle *handle, const char *reason);
+/*! \brief Method to only free resources related to a specific PeerConnection allocated by a Janus handle
+ * @param[in] pc The WebRTC PeerConnection instance to free */
+void janus_handle_webrtc_destroy(janus_handle_webrtc *pc);
 ///@}
 
 
-/** @name Janus ICE media relaying callbacks
+/** @name Janus media relaying callbacks
  */
 ///@{
 /*! \brief Core RTP callback, called when a plugin has an RTP packet to send to a peer
- * @param[in] handle The Janus ICE handle associated with the peer
+ * @param[in] handle The Janus handle associated with the peer
+ * @param[in] mindex Index of the target stream (relative to the SDP), -1 for the first audio/video stream
  * @param[in] video Whether this is an audio or a video frame
  * @param[in] buf The packet data (buffer)
  * @param[in] len The buffer lenght */
-void janus_ice_relay_rtp(janus_ice_handle *handle, int video, char *buf, int len);
+void janus_ice_relay_rtp(janus_handle *handle, int mindex, gboolean video, char *buf, int len);
 /*! \brief Core RTCP callback, called when a plugin has an RTCP message to send to a peer
- * @param[in] handle The Janus ICE handle associated with the peer
+ * @param[in] handle The Janus handle associated with the peer
+ * @param[in] mindex Index of the target stream (relative to the SDP), -1 for the first audio/video stream
  * @param[in] video Whether this is related to an audio or a video stream
  * @param[in] buf The message data (buffer)
  * @param[in] len The buffer lenght */
-void janus_ice_relay_rtcp(janus_ice_handle *handle, int video, char *buf, int len);
+void janus_ice_relay_rtcp(janus_handle *handle, int mindex, gboolean video, char *buf, int len);
 /*! \brief Core SCTP/DataChannel callback, called when a plugin has data to send to a peer
- * @param[in] handle The Janus ICE handle associated with the peer
+ * @param[in] handle The Janus handle associated with the peer
  * @param[in] label The label of the data channel to use
  * @param[in] buf The message data (buffer)
  * @param[in] len The buffer lenght */
-void janus_ice_relay_data(janus_ice_handle *handle, char *label, char *buf, int len);
+void janus_ice_relay_data(janus_handle *handle, char *label, char *buf, int len);
 /*! \brief Plugin SCTP/DataChannel callback, called by the SCTP stack when when there's data for a plugin
- * @param[in] handle The Janus ICE handle associated with the peer
+ * @param[in] handle The Janus handle associated with the peer
  * @param[in] label The label of the data channel the message is from
  * @param[in] buffer The message data (buffer)
  * @param[in] length The buffer lenght */
-void janus_ice_incoming_data(janus_ice_handle *handle, char *label, char *buffer, int length);
+void janus_ice_incoming_data(janus_handle *handle, char *label, char *buffer, int length);
 /*! \brief Core SCTP/DataChannel callback, called by the SCTP stack when when there's data to send.
- * @param[in] handle The Janus ICE handle associated with the peer
+ * @param[in] handle The Janus handle associated with the peer
  * @param[in] buffer The message data (buffer)
  * @param[in] length The buffer lenght */
-void janus_ice_relay_sctp(janus_ice_handle *handle, char *buffer, int length);
+void janus_ice_relay_sctp(janus_handle *handle, char *buffer, int length);
 ///@}
 
 
-/** @name Janus ICE handle helpers
+/** @name Janus handle helpers
  */
 ///@{
 /*! \brief Method to locally set up the ICE candidates (initialization and gathering)
- * @param[in] handle The Janus ICE handle this method refers to
+ * @param[in] handle The Janus handle this method refers to
  * @param[in] offer Whether this is for an OFFER or an ANSWER
- * @param[in] audio Whether audio is enabled
- * @param[in] video Whether video is enabled
- * @param[in] data Whether SCTP data channels are enabled
  * @param[in] trickle Whether ICE trickling is supported or not
  * @returns 0 in case of success, a negative integer otherwise */
-int janus_ice_setup_local(janus_ice_handle *handle, int offer, int audio, int video, int data, int trickle);
+int janus_handle_setup_local(janus_handle *handle, gboolean offer, gboolean trickle);
 /*! \brief Method to add local candidates to a janus_sdp SDP object representation
- * @param[in] handle The Janus ICE handle this method refers to
+ * @param[in] handle The Janus handle this method refers to
  * @param[in] mline The Janus SDP m-line object to add candidates to
  * @param[in] stream_id The stream ID of the candidate to add to the SDP
  * @param[in] component_id The component ID of the candidate to add to the SDP */
-void janus_ice_candidates_to_sdp(janus_ice_handle *handle, janus_sdp_mline *mline, guint stream_id, guint component_id);
+void janus_handle_candidates_to_sdp(janus_handle *handle, janus_sdp_mline *mline, guint stream_id, guint component_id);
 /*! \brief Method to handle remote candidates and start the connectivity checks
- * @param[in] handle The Janus ICE handle this method refers to
+ * @param[in] handle The Janus handle this method refers to
  * @param[in] stream_id The stream ID of the candidate to add to the SDP
  * @param[in] component_id The component ID of the candidate to add to the SDP */
-void janus_ice_setup_remote_candidates(janus_ice_handle *handle, guint stream_id, guint component_id);
+void janus_handle_setup_remote_candidates(janus_handle *handle, guint stream_id, guint component_id);
 /*! \brief Callback to be notified when the DTLS handshake for a specific component has been completed
  * \details This method also decides when to notify attached plugins about the availability of a reliable PeerConnection
- * @param[in] handle The Janus ICE handle this callback refers to
- * @param[in] component The Janus ICE component that is now ready to be used */
-void janus_ice_dtls_handshake_done(janus_ice_handle *handle, janus_ice_component *component);
+ * @param[in] handle The Janus handle this callback refers to */
+void janus_handle_dtls_handshake_done(janus_handle *handle);
 /*! \brief Method to restart ICE and the connectivity checks
- * @param[in] handle The Janus ICE handle this method refers to */
-void janus_ice_restart(janus_ice_handle *handle);
+ * @param[in] handle The Janus handle this method refers to */
+void janus_handle_ice_restart(janus_handle *handle);
 /*! \brief Method to resend all the existing candidates via trickle (e.g., after an ICE restart)
- * @param[in] handle The Janus ICE handle this method refers to */
-void janus_ice_resend_trickles(janus_ice_handle *handle);
+ * @param[in] handle The Janus handle this method refers to */
+void janus_handle_resend_trickles(janus_handle *handle);
 ///@}
 
 
diff --git a/janus.c b/janus.c
index 4461fa27f4..d3a612aae8 100644
--- a/janus.c
+++ b/janus.c
@@ -86,41 +86,41 @@ static char *api_secret = NULL, *admin_api_secret = NULL;
 static int janus_process_error_string(janus_request *request, uint64_t session_id, const char *transaction, gint error, gchar *error_string);
 
 static struct janus_json_parameter incoming_request_parameters[] = {
-	{"transaction", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"janus", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"transaction", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"janus", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter attach_parameters[] = {
-	{"plugin", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"opaque_id", JSON_STRING, 0},
+	{"plugin", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"opaque_id", JANUS_JSON_STRING, 0},
 };
 static struct janus_json_parameter body_parameters[] = {
 	{"body", JSON_OBJECT, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter jsep_parameters[] = {
-	{"type", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"type", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"trickle", JANUS_JSON_BOOL, 0},
-	{"sdp", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"sdp", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter add_token_parameters[] = {
-	{"token", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"plugins", JSON_ARRAY, 0}
+	{"token", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"plugins", JANUS_JSON_ARRAY, 0}
 };
 static struct janus_json_parameter token_parameters[] = {
-	{"token", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"token", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter admin_parameters[] = {
-	{"transaction", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"janus", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"transaction", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"janus", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter debug_parameters[] = {
 	{"debug", JANUS_JSON_BOOL, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter timeout_parameters[] = {
-	{"timeout", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"timeout", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter level_parameters[] = {
-	{"level", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"level", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter timestamps_parameters[] = {
 	{"timestamps", JANUS_JSON_BOOL, JANUS_JSON_PARAM_REQUIRED}
@@ -129,10 +129,10 @@ static struct janus_json_parameter colors_parameters[] = {
 	{"colors", JANUS_JSON_BOOL, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter mnq_parameters[] = {
-	{"max_nack_queue", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"max_nack_queue", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter nmt_parameters[] = {
-	{"no_media_timer", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"no_media_timer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter st_parameters[] = {
 	{"slowlink_threshold", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
@@ -141,7 +141,7 @@ static struct janus_json_parameter ans_parameters[] = {
 	{"accept", JANUS_JSON_BOOL, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter queryhandler_parameters[] = {
-	{"handler", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"handler", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"request", JSON_OBJECT, 0}
 };
 static struct janus_json_parameter messageplugin_parameters[] = {
@@ -149,7 +149,7 @@ static struct janus_json_parameter messageplugin_parameters[] = {
 	{"request", JSON_OBJECT, 0}
 };
 static struct janus_json_parameter customevent_parameters[] = {
-	{"schema", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"schema", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"data", JSON_OBJECT, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter customlogline_parameters[] = {
@@ -157,9 +157,9 @@ static struct janus_json_parameter customlogline_parameters[] = {
 	{"level", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter text2pcap_parameters[] = {
-	{"folder", JSON_STRING, 0},
-	{"filename", JSON_STRING, 0},
-	{"truncate", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"folder", JANUS_JSON_STRING, 0},
+	{"filename", JANUS_JSON_STRING, 0},
+	{"truncate", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter handleinfo_parameters[] = {
 	{"plugin_only", JANUS_JSON_BOOL, 0}
@@ -174,8 +174,8 @@ static struct janus_json_parameter teststun_parameters[] = {
 };
 
 /* Admin/Monitor helpers */
-json_t *janus_admin_stream_summary(janus_ice_stream *stream);
-json_t *janus_admin_component_summary(janus_ice_component *component);
+json_t *janus_admin_webrtc_summary(janus_handle_webrtc *pc);
+json_t *janus_admin_webrtc_medium_summary(janus_handle_webrtc_medium *medium);
 
 
 /* IP addresses */
@@ -483,8 +483,8 @@ void janus_transport_task(gpointer data, gpointer user_data);
 ///@{
 int janus_plugin_push_event(janus_plugin_session *plugin_session, janus_plugin *plugin, const char *transaction, json_t *message, json_t *jsep);
 json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plugin *plugin, const char *sdp_type, const char *sdp, gboolean restart);
-void janus_plugin_relay_rtp(janus_plugin_session *plugin_session, int video, char *buf, int len);
-void janus_plugin_relay_rtcp(janus_plugin_session *plugin_session, int video, char *buf, int len);
+void janus_plugin_relay_rtp(janus_plugin_session *plugin_session, int mindex, gboolean video, char *buf, int len);
+void janus_plugin_relay_rtcp(janus_plugin_session *plugin_session, int mindex, gboolean video, char *buf, int len);
 void janus_plugin_relay_data(janus_plugin_session *plugin_session, char *label, char *buf, int len);
 void janus_plugin_close_pc(janus_plugin_session *plugin_session);
 void janus_plugin_end_session(janus_plugin_session *plugin_session);
@@ -513,7 +513,7 @@ static GHashTable *sessions = NULL;
 static GMainContext *sessions_watchdog_context = NULL;
 
 
-static void janus_ice_handle_dereference(janus_ice_handle *handle) {
+static void janus_handle_dereference(janus_handle *handle) {
 	if(handle)
 		janus_refcount_decrease(&handle->ref);
 }
@@ -521,9 +521,9 @@ static void janus_ice_handle_dereference(janus_ice_handle *handle) {
 static void janus_session_free(const janus_refcount *session_ref) {
 	janus_session *session = janus_refcount_containerof(session_ref, janus_session, ref);
 	/* This session can be destroyed, free all the resources */
-	if(session->ice_handles != NULL) {
-		g_hash_table_destroy(session->ice_handles);
-		session->ice_handles = NULL;
+	if(session->handles != NULL) {
+		g_hash_table_destroy(session->handles);
+		session->handles = NULL;
 	}
 	if(session->source != NULL) {
 		janus_request_destroy(session->source);
@@ -618,7 +618,7 @@ janus_session *janus_session_create(guint64 session_id) {
 	g_atomic_int_set(&session->timeout, 0);
 	g_atomic_int_set(&session->transport_gone, 0);
 	session->last_activity = janus_get_monotonic_time();
-	session->ice_handles = NULL;
+	session->handles = NULL;
 	janus_mutex_init(&session->mutex);
 	janus_mutex_lock(&sessions_mutex);
 	g_hash_table_insert(sessions, janus_uint64_dup(session->session_id), session);
@@ -663,11 +663,11 @@ gint janus_session_destroy(janus_session *session) {
 	return 0;
 }
 
-janus_ice_handle *janus_session_handles_find(janus_session *session, guint64 handle_id) {
+janus_handle *janus_session_handles_find(janus_session *session, guint64 handle_id) {
 	if(session == NULL)
 		return NULL;
 	janus_mutex_lock(&session->mutex);
-	janus_ice_handle *handle = session->ice_handles ? g_hash_table_lookup(session->ice_handles, &handle_id) : NULL;
+	janus_handle *handle = session->handles ? g_hash_table_lookup(session->handles, &handle_id) : NULL;
 	if(handle != NULL) {
 		/* A successful find automatically increases the reference counter:
 		 * it's up to the caller to decrease it again when done */
@@ -677,35 +677,35 @@ janus_ice_handle *janus_session_handles_find(janus_session *session, guint64 han
 	return handle;
 }
 
-void janus_session_handles_insert(janus_session *session, janus_ice_handle *handle) {
+void janus_session_handles_insert(janus_session *session, janus_handle *handle) {
 	janus_mutex_lock(&session->mutex);
-	if(session->ice_handles == NULL)
-		session->ice_handles = g_hash_table_new_full(g_int64_hash, g_int64_equal, (GDestroyNotify)g_free, (GDestroyNotify)janus_ice_handle_dereference);
+	if(session->handles == NULL)
+		session->handles = g_hash_table_new_full(g_int64_hash, g_int64_equal, (GDestroyNotify)g_free, (GDestroyNotify)janus_handle_dereference);
 	janus_refcount_increase(&handle->ref);
-	g_hash_table_insert(session->ice_handles, janus_uint64_dup(handle->handle_id), handle);
+	g_hash_table_insert(session->handles, janus_uint64_dup(handle->handle_id), handle);
 	janus_mutex_unlock(&session->mutex);
 }
 
-gint janus_session_handles_remove(janus_session *session, janus_ice_handle *handle) {
+gint janus_session_handles_remove(janus_session *session, janus_handle *handle) {
 	janus_mutex_lock(&session->mutex);
-	gint error = janus_ice_handle_destroy(session, handle);
-	g_hash_table_remove(session->ice_handles, &handle->handle_id);
+	gint error = janus_handle_destroy(session, handle);
+	g_hash_table_remove(session->handles, &handle->handle_id);
 	janus_mutex_unlock(&session->mutex);
 	return error;
 }
 
 void janus_session_handles_clear(janus_session *session) {
 	janus_mutex_lock(&session->mutex);
-	if(session->ice_handles != NULL && g_hash_table_size(session->ice_handles) > 0) {
+	if(session->handles != NULL && g_hash_table_size(session->handles) > 0) {
 		GHashTableIter iter;
 		gpointer value;
 		/* Remove all handles */
-		g_hash_table_iter_init(&iter, session->ice_handles);
+		g_hash_table_iter_init(&iter, session->handles);
 		while (g_hash_table_iter_next(&iter, NULL, &value)) {
-			janus_ice_handle *handle = value;
+			janus_handle *handle = value;
 			if(!handle)
 				continue;
-			janus_ice_handle_destroy(session, handle);
+			janus_handle_destroy(session, handle);
 			g_hash_table_iter_remove(&iter);
 		}
 	}
@@ -715,12 +715,12 @@ void janus_session_handles_clear(janus_session *session) {
 json_t *janus_session_handles_list_json(janus_session *session) {
 	json_t *list = json_array();
 	janus_mutex_lock(&session->mutex);
-	if(session->ice_handles != NULL && g_hash_table_size(session->ice_handles) > 0) {
+	if(session->handles != NULL && g_hash_table_size(session->handles) > 0) {
 		GHashTableIter iter;
 		gpointer value;
-		g_hash_table_iter_init(&iter, session->ice_handles);
+		g_hash_table_iter_init(&iter, session->handles);
 		while (g_hash_table_iter_next(&iter, NULL, &value)) {
-			janus_ice_handle *handle = value;
+			janus_handle *handle = value;
 			if(!handle)
 				continue;
 			json_array_append_new(list, json_integer(handle->handle_id));
@@ -784,9 +784,9 @@ static int janus_request_check_secret(janus_request *request, guint64 session_id
 	return 0;
 }
 
-static void janus_request_ice_handle_answer(janus_ice_handle *handle, int audio, int video, int data, char *jsep_sdp) {
+static void janus_request_handle_answer(janus_handle *handle, char *jsep_sdp) {
 	/* We got our answer */
-	janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+	janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 	/* Any pending trickles? */
 	if(handle->pending_trickles) {
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"]   -- Processing %d pending trickle candidates\n", handle->handle_id, g_list_length(handle->pending_trickles));
@@ -794,27 +794,27 @@ static void janus_request_ice_handle_answer(janus_ice_handle *handle, int audio,
 		while(handle->pending_trickles) {
 			temp = g_list_first(handle->pending_trickles);
 			handle->pending_trickles = g_list_remove_link(handle->pending_trickles, temp);
-			janus_ice_trickle *trickle = (janus_ice_trickle *)temp->data;
+			janus_trickle *trickle = (janus_trickle *)temp->data;
 			g_list_free(temp);
 			if(trickle == NULL)
 				continue;
 			if((janus_get_monotonic_time() - trickle->received) > candidates_timeout*G_USEC_PER_SEC) {
 				/* FIXME Candidate is too old, discard it */
 				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Discarding candidate (too old)\n", handle->handle_id);
-				janus_ice_trickle_destroy(trickle);
+				janus_trickle_destroy(trickle);
 				/* FIXME We should report that */
 				continue;
 			}
 			json_t *candidate = trickle->candidate;
 			if(candidate == NULL) {
-				janus_ice_trickle_destroy(trickle);
+				janus_trickle_destroy(trickle);
 				continue;
 			}
 			if(json_is_object(candidate)) {
 				/* We got a single candidate */
 				int error = 0;
 				const char *error_string = NULL;
-				if((error = janus_ice_trickle_parse(handle, candidate, &error_string)) != 0) {
+				if((error = janus_trickle_parse(handle, candidate, &error_string)) != 0) {
 					/* FIXME We should report the error parsing the trickle candidate */
 				}
 			} else if(json_is_array(candidate)) {
@@ -826,23 +826,23 @@ static void janus_request_ice_handle_answer(janus_ice_handle *handle, int audio,
 					for(i=0; i<json_array_size(candidate); i++) {
 						json_t *c = json_array_get(candidate, i);
 						/* FIXME We don't care if any trickle fails to parse */
-						janus_ice_trickle_parse(handle, c, NULL);
+						janus_trickle_parse(handle, c, NULL);
 					}
 				}
 			}
 			/* Done, free candidate */
-			janus_ice_trickle_destroy(trickle);
+			janus_trickle_destroy(trickle);
 		}
 	}
 
-	gboolean candidates_found = (handle->stream && handle->stream->component && g_slist_length(handle->stream->component->candidates) > 0);
+	gboolean candidates_found = (handle->pc && g_slist_length(handle->pc->candidates) > 0);
 	/* This was an answer, check if it's time to start ICE */
-	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE) && !candidates_found) {
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE) && !candidates_found) {
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"]   -- ICE Trickling is supported by the browser, waiting for remote candidates...\n", handle->handle_id);
-		janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_START);
+		janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_START);
 	} else {
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Done! Sending connectivity checks...\n", handle->handle_id);
-		janus_ice_setup_remote_candidates(handle, handle->stream_id, 1);
+		janus_handle_setup_remote_candidates(handle, handle->stream_id, 1);
 	}
 }
 
@@ -865,7 +865,7 @@ int janus_process_incoming_request(janus_request *request) {
 		handle_id = json_integer_value(h);
 
 	janus_session *session = NULL;
-	janus_ice_handle *handle = NULL;
+	janus_handle *handle = NULL;
 
 	/* Get transaction and message request */
 	JANUS_VALIDATE_JSON_OBJECT(root, incoming_request_parameters,
@@ -1030,7 +1030,7 @@ int janus_process_incoming_request(janus_request *request) {
 		json_t *opaque = json_object_get(root, "opaque_id");
 		const char *opaque_id = opaque ? json_string_value(opaque) : NULL;
 		/* Create handle */
-		handle = janus_ice_handle_create(session, opaque_id);
+		handle = janus_handle_create(session, opaque_id);
 		if(handle == NULL) {
 			ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_UNKNOWN, "Memory error");
 			goto jsondone;
@@ -1040,7 +1040,7 @@ int janus_process_incoming_request(janus_request *request) {
 		janus_refcount_increase(&handle->ref);
 		/* Attach to the plugin */
 		int error = 0;
-		if((error = janus_ice_handle_attach_plugin(session, handle, plugin_t)) != 0) {
+		if((error = janus_handle_attach_plugin(session, handle, plugin_t)) != 0) {
 			/* TODO Make error struct to pass verbose information */
 			janus_session_handles_remove(session, handle);
 			JANUS_LOG(LOG_ERR, "Couldn't attach to plugin '%s', error '%d'\n", plugin_text, error);
@@ -1108,7 +1108,7 @@ int janus_process_incoming_request(janus_request *request) {
 			ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_PLUGIN_DETACH, "No plugin attached");
 			goto jsondone;
 		}
-		janus_ice_webrtc_hangup(handle, "Janus API");
+		janus_handle_webrtc_hangup(handle, "Janus API");
 		/* Prepare JSON reply */
 		json_t *reply = janus_create_message("success", session_id, transaction_text);
 		/* Send the success reply */
@@ -1179,10 +1179,10 @@ int janus_process_incoming_request(janus_request *request) {
 			json_t *jsep_trickle = json_object_get(jsep, "trickle");
 			do_trickle = jsep_trickle ? json_is_true(jsep_trickle) : TRUE;
 			/* Are we still cleaning up from a previous media session? */
-			if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING)) {
+			if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING)) {
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Still cleaning up from a previous media session, let's wait a bit...\n", handle->handle_id);
 				gint64 waited = 0;
-				while(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING)) {
+				while(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING)) {
 					g_usleep(100000);
 					waited += 100000;
 					if(waited >= 3*G_USEC_PER_SEC) {
@@ -1193,25 +1193,25 @@ int janus_process_incoming_request(janus_request *request) {
 				}
 			}
 			/* Check if we're renegotiating (if we have an answer, we did an offer/answer round already) */
-			renegotiation = janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEGOTIATED);
+			renegotiation = janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEGOTIATED);
 			/* Check the JSEP type */
 			janus_mutex_lock(&handle->mutex);
-			int offer = 0;
+			gboolean offer = FALSE;
 			if(!strcasecmp(jsep_type, "offer")) {
-				offer = 1;
-				janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
-				janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER);
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER);
+				offer = TRUE;
+				janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
+				janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_OFFER);
+				janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER);
 			} else if(!strcasecmp(jsep_type, "answer")) {
-				janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER);
-				if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER))
-					janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEGOTIATED);
-				offer = 0;
+				janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER);
+				offer = FALSE;
+				if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_OFFER))
+					janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEGOTIATED);
 			} else {
 				/* TODO Handle other message types as well */
 				ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_JSEP_UNKNOWN_TYPE, "JSEP error: unknown message type '%s'", jsep_type);
 				g_free(jsep_type);
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+				janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 				janus_mutex_unlock(&handle->mutex);
 				goto jsondone;
 			}
@@ -1221,12 +1221,12 @@ int janus_process_incoming_request(janus_request *request) {
 			/* Is this valid SDP? */
 			char error_str[512];
 			int audio = 0, video = 0, data = 0;
-			janus_sdp *parsed_sdp = janus_sdp_preparse(handle, jsep_sdp, error_str, sizeof(error_str), &audio, &video, &data);
+			janus_sdp *parsed_sdp = janus_sdp_preparse(handle, jsep_sdp, &audio, &video, &data, error_str, sizeof(error_str));
 			if(parsed_sdp == NULL) {
 				/* Invalid SDP */
 				ret = janus_process_error_string(request, session_id, transaction_text, JANUS_ERROR_JSEP_INVALID_SDP, error_str);
 				g_free(jsep_type);
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+				janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 				janus_mutex_unlock(&handle->mutex);
 				goto jsondone;
 			}
@@ -1235,36 +1235,18 @@ int janus_process_incoming_request(janus_request *request) {
 				janus_events_notify_handlers(JANUS_EVENT_TYPE_JSEP,
 					session_id, handle_id, handle->opaque_id, "remote", jsep_type, jsep_sdp);
 			}
-			/* FIXME We're only handling single audio/video lines for now... */
-			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Audio %s been negotiated, Video %s been negotiated, SCTP/DataChannels %s been negotiated\n",
-			                    handle->handle_id,
-			                    audio ? "has" : "has NOT",
-			                    video ? "has" : "has NOT",
-			                    data ? "have" : "have NOT");
-			if(audio > 1) {
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"] More than one audio line? only going to negotiate one...\n", handle->handle_id);
-			}
-			if(video > 1) {
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"] More than one video line? only going to negotiate one...\n", handle->handle_id);
-			}
-			if(data > 1) {
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"] More than one data line? only going to negotiate one...\n", handle->handle_id);
-			}
-#ifndef HAVE_SCTP
-			if(data) {
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"]   -- DataChannels have been negotiated, but support for them has not been compiled...\n", handle->handle_id);
-			}
-#endif
+			JANUS_LOG(LOG_VERB, "[%"SCNu64"] There are %d audio, %d video and %d data m-lines\n",
+				handle->handle_id, audio, video, data);
 			/* We behave differently if it's a new session or an update... */
 			if(!renegotiation) {
 				/* New session */
 				if(offer) {
 					/* Setup ICE locally (we received an offer) */
-					if(janus_ice_setup_local(handle, offer, audio, video, data, do_trickle) < 0) {
+					if(janus_handle_setup_local(handle, offer, do_trickle) < 0) {
 						JANUS_LOG(LOG_ERR, "Error setting ICE locally\n");
 						janus_sdp_destroy(parsed_sdp);
 						g_free(jsep_type);
-						janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+						janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 						ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_UNKNOWN, "Error setting ICE locally");
 						janus_mutex_unlock(&handle->mutex);
 						goto jsondone;
@@ -1275,17 +1257,17 @@ int janus_process_incoming_request(janus_request *request) {
 						JANUS_LOG(LOG_ERR, "Unexpected ANSWER (did we offer?)\n");
 						janus_sdp_destroy(parsed_sdp);
 						g_free(jsep_type);
-						janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+						janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 						ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_UNEXPECTED_ANSWER, "Unexpected ANSWER (did we offer?)");
 						janus_mutex_unlock(&handle->mutex);
 						goto jsondone;
 					}
 				}
-				if(janus_sdp_process(handle, parsed_sdp, FALSE) < 0) {
+				if(janus_sdp_process_remote(handle, parsed_sdp, FALSE) < 0) {
 					JANUS_LOG(LOG_ERR, "Error processing SDP\n");
 					janus_sdp_destroy(parsed_sdp);
 					g_free(jsep_type);
-					janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+					janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 					ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_JSEP_INVALID_SDP, "Error processing SDP");
 					janus_mutex_unlock(&handle->mutex);
 					goto jsondone;
@@ -1293,65 +1275,64 @@ int janus_process_incoming_request(janus_request *request) {
 				if(!offer) {
 					/* Set remote candidates now (we received an answer) */
 					if(do_trickle) {
-						janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE);
+						janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE);
 					} else {
-						janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE);
+						janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE);
 					}
-					janus_request_ice_handle_answer(handle, audio, video, data, jsep_sdp);
+					janus_request_handle_answer(handle, jsep_sdp);
 				} else {
 					/* Check if the mid RTP extension is being negotiated */
-					handle->stream->mid_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_MID);
+					handle->pc->mid_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_MID);
 					/* Check if the RTP Stream ID extension is being negotiated */
-					handle->stream->rid_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_RID);
-					handle->stream->ridrtx_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_REPAIRED_RID);
+					handle->pc->rid_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_RID);
+					handle->pc->ridrtx_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_REPAIRED_RID);
 					/* Check if the frame marking ID extension is being negotiated */
-					handle->stream->framemarking_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_FRAME_MARKING);
+					handle->pc->framemarking_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_FRAME_MARKING);
 					/* Check if transport wide CC is supported */
 					int transport_wide_cc_ext_id = janus_rtp_header_extension_get_id(jsep_sdp, JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC);
-					handle->stream->do_transport_wide_cc = transport_wide_cc_ext_id > 0 ? TRUE : FALSE;
-					handle->stream->transport_wide_cc_ext_id = transport_wide_cc_ext_id;
+					handle->pc->do_transport_wide_cc = transport_wide_cc_ext_id > 0 ? TRUE : FALSE;
+					handle->pc->transport_wide_cc_ext_id = transport_wide_cc_ext_id;
 				}
 			} else {
 				/* FIXME This is a renegotiation: we can currently only handle simple changes in media
 				 * direction and ICE restarts: anything more complex than that will result in an error */
 				JANUS_LOG(LOG_INFO, "[%"SCNu64"] Negotiation update, checking what changed...\n", handle->handle_id);
-				if(janus_sdp_process(handle, parsed_sdp, TRUE) < 0) {
+				if(janus_sdp_process_remote(handle, parsed_sdp, TRUE) < 0) {
 					JANUS_LOG(LOG_ERR, "Error processing SDP\n");
 					janus_sdp_destroy(parsed_sdp);
 					g_free(jsep_type);
-					janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+					janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 					ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_UNEXPECTED_ANSWER, "Error processing SDP");
 					janus_mutex_unlock(&handle->mutex);
 					goto jsondone;
 				}
-				if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ICE_RESTART)) {
+				if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ICE_RESTART)) {
 					JANUS_LOG(LOG_INFO, "[%"SCNu64"] Restarting ICE...\n", handle->handle_id);
 					/* Update remote credentials for ICE */
-					if(handle->stream) {
-						nice_agent_set_remote_credentials(handle->agent, handle->stream->stream_id,
-							handle->stream->ruser, handle->stream->rpass);
+					if(handle->pc) {
+						nice_agent_set_remote_credentials(handle->agent, handle->pc->stream_id,
+							handle->pc->ruser, handle->pc->rpass);
 					}
 					/* FIXME We only need to do that for offers: if it's an answer, we did that already */
 					if(offer) {
-						janus_ice_restart(handle);
+						janus_handle_ice_restart(handle);
 					} else {
-						janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ICE_RESTART);
+						janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ICE_RESTART);
 					}
 					/* If we're full-trickling, we'll need to resend the candidates later */
 					if(janus_ice_is_full_trickle_enabled()) {
-						janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RESEND_TRICKLES);
+						janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RESEND_TRICKLES);
 					}
 				}
 #ifdef HAVE_SCTP
 				if(!offer) {
 					/* Were datachannels just added? */
-					if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS)) {
-						janus_ice_stream *stream = handle->stream;
-						if(stream != NULL && stream->component != NULL
-								&& stream->component->dtls != NULL && stream->component->dtls->sctp == NULL) {
+					if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_DATA_CHANNELS)) {
+						janus_handle_webrtc *pc = handle->pc;
+						if(pc->dtls != NULL && pc->dtls->sctp == NULL) {
 							/* Create SCTP association as well */
 							JANUS_LOG(LOG_WARN, "[%"SCNu64"] Creating datachannels...\n", handle->handle_id);
-							janus_dtls_srtp_create_sctp(stream->component->dtls);
+							janus_dtls_srtp_create_sctp(pc->dtls);
 						}
 					}
 				}
@@ -1367,13 +1348,13 @@ int janus_process_incoming_request(janus_request *request) {
 				ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_JSEP_INVALID_SDP, "JSEP error: invalid SDP");
 				janus_sdp_destroy(parsed_sdp);
 				g_free(jsep_type);
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+				janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 				goto jsondone;
 			}
 			jsep_sdp_stripped = janus_sdp_write(parsed_sdp);
 			janus_sdp_destroy(parsed_sdp);
 			sdp = NULL;
-			janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+			janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 		}
 
 		/* Make sure the app handle is still valid */
@@ -1381,7 +1362,7 @@ int janus_process_incoming_request(janus_request *request) {
 			ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_PLUGIN_MESSAGE, "No plugin to handle this message");
 			g_free(jsep_type);
 			g_free(jsep_sdp_stripped);
-			janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+			janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 			goto jsondone;
 		}
 
@@ -1390,34 +1371,45 @@ int janus_process_incoming_request(janus_request *request) {
 		json_t *body_jsep = NULL;
 		if(jsep_sdp_stripped) {
 			body_jsep = json_pack("{ssss}", "type", jsep_type, "sdp", jsep_sdp_stripped);
-			/* Check if simulcasting is enabled */
-			if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO)) {
-				if(handle->stream && (handle->stream->rid[0] || handle->stream->video_ssrc_peer[1])) {
-					json_t *simulcast = json_object();
+			/* Check if simulcasting is enabled in one of the media streams */
+			json_t *simulcast = NULL;
+			janus_handle_webrtc_medium *medium = NULL;
+			uint mi=0;
+			for(mi=0; mi<g_hash_table_size(handle->pc->media); mi++) {
+				medium = g_hash_table_lookup(handle->pc->media, GUINT_TO_POINTER(mi));
+				if(medium && (medium->ssrc_peer[1] || medium->rid[0])) {
+					if(simulcast == NULL)
+						simulcast = json_array();
+					json_t *msc = json_object();
+					json_object_set_new(msc, "mindex", json_integer(medium->mindex));
+					if(medium->mid != NULL)
+						json_object_set_new(msc, "rid", json_string(medium->mid));
 					/* If we have rids, pass those, otherwise pass the SSRCs */
-					if(handle->stream->rid[0]) {
+					if(medium->rid[0] && medium->pc->rid_ext_id > 0) {
 						json_t *rids = json_array();
-						json_array_append_new(rids, json_string(handle->stream->rid[0]));
-						if(handle->stream->rid[1])
-							json_array_append_new(rids, json_string(handle->stream->rid[1]));
-						if(handle->stream->rid[2])
-							json_array_append_new(rids, json_string(handle->stream->rid[2]));
-						json_object_set_new(simulcast, "rids", rids);
-						json_object_set_new(simulcast, "rid-ext", json_integer(handle->stream->rid_ext_id));
+						json_array_append_new(rids, json_string(medium->rid[0]));
+						if(medium->rid[1])
+							json_array_append_new(rids, json_string(medium->rid[1]));
+						if(medium->rid[2])
+							json_array_append_new(rids, json_string(medium->rid[2]));
+						json_object_set_new(msc, "rids", rids);
+						json_object_set_new(msc, "rid-ext", json_integer(handle->pc->rid_ext_id));
 					} else {
 						json_t *ssrcs = json_array();
-						json_array_append_new(ssrcs, json_integer(handle->stream->video_ssrc_peer[0]));
-						if(handle->stream->video_ssrc_peer[1])
-							json_array_append_new(ssrcs, json_integer(handle->stream->video_ssrc_peer[1]));
-						if(handle->stream->video_ssrc_peer[2])
-							json_array_append_new(ssrcs, json_integer(handle->stream->video_ssrc_peer[2]));
-						json_object_set_new(simulcast, "ssrcs", ssrcs);
+						json_array_append_new(ssrcs, json_integer(medium->ssrc_peer[0]));
+						if(medium->ssrc_peer[1])
+							json_array_append_new(ssrcs, json_integer(medium->ssrc_peer[1]));
+						if(medium->ssrc_peer[2])
+							json_array_append_new(ssrcs, json_integer(medium->ssrc_peer[2]));
+						json_object_set_new(msc, "ssrcs", ssrcs);
 					}
-					if(handle->stream->framemarking_ext_id > 0)
-						json_object_set_new(simulcast, "framemarking-ext", json_integer(handle->stream->framemarking_ext_id));
-					json_object_set_new(body_jsep, "simulcast", simulcast);
+					if(handle->pc->framemarking_ext_id > 0)
+						json_object_set_new(msc, "framemarking-ext", json_integer(handle->pc->framemarking_ext_id));
+					json_array_append_new(simulcast, msc);
 				}
 			}
+			if(simulcast)
+				json_object_set_new(body_jsep, "simulcast", simulcast);
 			/* Check if this is a renegotiation or update */
 			if(renegotiation)
 				json_object_set_new(body_jsep, "update", json_true());
@@ -1490,41 +1482,41 @@ int janus_process_incoming_request(janus_request *request) {
 			ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_INVALID_JSON, "Can't have both candidate and candidates");
 			goto jsondone;
 		}
-		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING)) {
+		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING)) {
 			JANUS_LOG(LOG_ERR, "[%"SCNu64"] Received a trickle, but still cleaning a previous session\n", handle->handle_id);
 			ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_WEBRTC_STATE, "Still cleaning a previous session");
 			goto jsondone;
 		}
 		janus_mutex_lock(&handle->mutex);
-		if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE)) {
+		if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE)) {
 			/* It looks like this peer supports Trickle, after all */
 			JANUS_LOG(LOG_VERB, "Handle %"SCNu64" supports trickle even if it didn't negotiate it...\n", handle->handle_id);
-			janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE);
+			janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE);
 		}
 		/* Is there any stream ready? this trickle may get here before the SDP it relates to */
-		if(handle->stream == NULL) {
+		if(handle->pc == NULL) {
 			JANUS_LOG(LOG_WARN, "[%"SCNu64"] No stream, queueing this trickle as it got here before the SDP...\n", handle->handle_id);
 			/* Enqueue this trickle candidate(s), we'll process this later */
-			janus_ice_trickle *early_trickle = janus_ice_trickle_new(transaction_text, candidate ? candidate : candidates);
+			janus_trickle *early_trickle = janus_trickle_new(transaction_text, candidate ? candidate : candidates);
 			handle->pending_trickles = g_list_append(handle->pending_trickles, early_trickle);
 			/* Send the ack right away, an event will tell the application if the candidate(s) failed */
 			goto trickledone;
 		}
 		/* Is the ICE stack ready already? */
-		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER) ||
-				!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER) ||
-				!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER)) {
+		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER) ||
+				!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_OFFER) ||
+				!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER)) {
 			const char *cause = NULL;
-			if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER))
+			if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER))
 				cause = "processing the offer";
-			else if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER))
+			else if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER))
 				cause = "waiting for the answer";
-			else if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER))
+			else if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_OFFER))
 				cause = "waiting for the offer";
 			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Still %s, queueing this trickle to wait until we're done there...\n",
 				handle->handle_id, cause);
 			/* Enqueue this trickle candidate(s), we'll process this later */
-			janus_ice_trickle *early_trickle = janus_ice_trickle_new(transaction_text, candidate ? candidate : candidates);
+			janus_trickle *early_trickle = janus_trickle_new(transaction_text, candidate ? candidate : candidates);
 			handle->pending_trickles = g_list_append(handle->pending_trickles, early_trickle);
 			/* Send the ack right away, an event will tell the application if the candidate(s) failed */
 			goto trickledone;
@@ -1533,7 +1525,7 @@ int janus_process_incoming_request(janus_request *request) {
 			/* We got a single candidate */
 			int error = 0;
 			const char *error_string = NULL;
-			if((error = janus_ice_trickle_parse(handle, candidate, &error_string)) != 0) {
+			if((error = janus_trickle_parse(handle, candidate, &error_string)) != 0) {
 				ret = janus_process_error(request, session_id, transaction_text, error, "%s", error_string);
 				janus_mutex_unlock(&handle->mutex);
 				goto jsondone;
@@ -1552,7 +1544,7 @@ int janus_process_incoming_request(janus_request *request) {
 				for(i=0; i<json_array_size(candidates); i++) {
 					json_t *c = json_array_get(candidates, i);
 					/* FIXME We don't care if any trickle fails to parse */
-					janus_ice_trickle_parse(handle, c, NULL);
+					janus_trickle_parse(handle, c, NULL);
 				}
 			}
 		}
@@ -1731,7 +1723,7 @@ int janus_process_incoming_admin_request(janus_request *request) {
 		handle_id = json_integer_value(h);
 
 	janus_session *session = NULL;
-	janus_ice_handle *handle = NULL;
+	janus_handle *handle = NULL;
 
 	/* Get transaction and message request */
 	JANUS_VALIDATE_JSON_OBJECT(root, admin_parameters,
@@ -2412,7 +2404,7 @@ int janus_process_incoming_admin_request(janus_request *request) {
 				ret = janus_process_error(request, session_id, transaction_text, JANUS_ERROR_PLUGIN_DETACH, "No plugin attached");
 				goto jsondone;
 			}
-			janus_ice_webrtc_hangup(handle, "Admin API");
+			janus_handle_webrtc_hangup(handle, "Admin API");
 			/* Prepare JSON reply */
 			json_t *reply = janus_create_message("success", session_id, transaction_text);
 			/* Send the success reply */
@@ -2518,25 +2510,23 @@ int janus_process_incoming_admin_request(janus_request *request) {
 		if(plugin_only)
 			goto info_done;
 		json_t *flags = json_object();
-		json_object_set_new(flags, "got-offer", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER) ? json_true() : json_false());
-		json_object_set_new(flags, "got-answer", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER) ? json_true() : json_false());
-		json_object_set_new(flags, "negotiated", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEGOTIATED) ? json_true() : json_false());
-		json_object_set_new(flags, "processing-offer", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER) ? json_true() : json_false());
-		json_object_set_new(flags, "starting", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_START) ? json_true() : json_false());
-		json_object_set_new(flags, "ice-restart", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ICE_RESTART) ? json_true() : json_false());
-		json_object_set_new(flags, "ready", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY) ? json_true() : json_false());
-		json_object_set_new(flags, "stopped", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP) ? json_true() : json_false());
-		json_object_set_new(flags, "alert", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT) ? json_true() : json_false());
-		json_object_set_new(flags, "trickle", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE) ? json_true() : json_false());
-		json_object_set_new(flags, "all-trickles", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALL_TRICKLES) ? json_true() : json_false());
-		json_object_set_new(flags, "resend-trickles", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RESEND_TRICKLES) ? json_true() : json_false());
-		json_object_set_new(flags, "trickle-synced", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_TRICKLE_SYNCED) ? json_true() : json_false());
-		json_object_set_new(flags, "data-channels", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS) ? json_true() : json_false());
-		json_object_set_new(flags, "has-audio", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO) ? json_true() : json_false());
-		json_object_set_new(flags, "has-video", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO) ? json_true() : json_false());
-		json_object_set_new(flags, "new-datachan-sdp", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP) ? json_true() : json_false());
-		json_object_set_new(flags, "rfc4588-rtx", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX) ? json_true() : json_false());
-		json_object_set_new(flags, "cleaning", janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING) ? json_true() : json_false());
+		json_object_set_new(flags, "got-offer", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_OFFER) ? json_true() : json_false());
+		json_object_set_new(flags, "got-answer", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER) ? json_true() : json_false());
+		json_object_set_new(flags, "negotiated", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEGOTIATED) ? json_true() : json_false());
+		json_object_set_new(flags, "processing-offer", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER) ? json_true() : json_false());
+		json_object_set_new(flags, "starting", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_START) ? json_true() : json_false());
+		json_object_set_new(flags, "ice-restart", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ICE_RESTART) ? json_true() : json_false());
+		json_object_set_new(flags, "ready", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY) ? json_true() : json_false());
+		json_object_set_new(flags, "stopped", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP) ? json_true() : json_false());
+		json_object_set_new(flags, "alert", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT) ? json_true() : json_false());
+		json_object_set_new(flags, "trickle", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE) ? json_true() : json_false());
+		json_object_set_new(flags, "all-trickles", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALL_TRICKLES) ? json_true() : json_false());
+		json_object_set_new(flags, "resend-trickles", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RESEND_TRICKLES) ? json_true() : json_false());
+		json_object_set_new(flags, "trickle-synced", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_TRICKLE_SYNCED) ? json_true() : json_false());
+		json_object_set_new(flags, "data-channels", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_DATA_CHANNELS) ? json_true() : json_false());
+		json_object_set_new(flags, "new-datachan-sdp", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP) ? json_true() : json_false());
+		json_object_set_new(flags, "rfc4588-rtx", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX) ? json_true() : json_false());
+		json_object_set_new(flags, "cleaning", janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING) ? json_true() : json_false());
 		json_object_set_new(info, "flags", flags);
 		if(handle->agent) {
 			json_object_set_new(info, "agent-created", json_integer(handle->agent_created));
@@ -2564,13 +2554,11 @@ int janus_process_incoming_admin_request(janus_request *request) {
 				json_object_set_new(info, "pcap-file", json_string(handle->text2pcap->filename));
 			}
 		}
-		json_t *streams = json_array();
-		if(handle->stream) {
-			json_t *s = janus_admin_stream_summary(handle->stream);
-			if(s)
-				json_array_append_new(streams, s);
+		if(handle->pc) {
+			json_t *w = janus_admin_webrtc_summary(handle->pc);
+			if(w)
+				json_object_set_new(info, "webrtc", w);
 		}
-		json_object_set_new(info, "streams", streams);
 info_done:
 		janus_mutex_unlock(&handle->mutex);
 		/* Prepare JSON reply */
@@ -2637,177 +2625,54 @@ int janus_process_error(janus_request *request, uint64_t session_id, const char
 }
 
 /* Admin/monitor helpers */
-json_t *janus_admin_stream_summary(janus_ice_stream *stream) {
-	if(stream == NULL)
+json_t *janus_admin_webrtc_summary(janus_handle_webrtc *pc) {
+	if(pc == NULL)
 		return NULL;
-	json_t *s = json_object();
-	json_object_set_new(s, "id", json_integer(stream->stream_id));
-	json_object_set_new(s, "ready", json_integer(stream->cdone));
-	json_t *ss = json_object();
-	if(stream->audio_ssrc)
-		json_object_set_new(ss, "audio", json_integer(stream->audio_ssrc));
-	if(stream->video_ssrc)
-		json_object_set_new(ss, "video", json_integer(stream->video_ssrc));
-	if(stream->video_ssrc_rtx)
-		json_object_set_new(ss, "video-rtx", json_integer(stream->video_ssrc_rtx));
-	if(stream->audio_ssrc_peer)
-		json_object_set_new(ss, "audio-peer", json_integer(stream->audio_ssrc_peer));
-	if(stream->video_ssrc_peer[0])
-		json_object_set_new(ss, "video-peer", json_integer(stream->video_ssrc_peer[0]));
-	if(stream->video_ssrc_peer[1])
-		json_object_set_new(ss, "video-peer-sim-1", json_integer(stream->video_ssrc_peer[1]));
-	if(stream->video_ssrc_peer[2])
-		json_object_set_new(ss, "video-peer-sim-2", json_integer(stream->video_ssrc_peer[2]));
-	if(stream->video_ssrc_peer_rtx[0])
-		json_object_set_new(ss, "video-peer-rtx", json_integer(stream->video_ssrc_peer_rtx[0]));
-	if(stream->video_ssrc_peer_rtx[1])
-		json_object_set_new(ss, "video-peer-sim-1-rtx", json_integer(stream->video_ssrc_peer_rtx[1]));
-	if(stream->video_ssrc_peer_rtx[2])
-		json_object_set_new(ss, "video-peer-sim-2-rtx", json_integer(stream->video_ssrc_peer_rtx[2]));
-	json_object_set_new(s, "ssrc", ss);
-	if(stream->rid[0] && stream->rid_ext_id > 0) {
-		json_t *sr = json_object();
-		json_t *rid = json_array();
-		json_array_append_new(rid, json_string(stream->rid[0]));
-		if(stream->rid[1])
-			json_array_append_new(rid, json_string(stream->rid[1]));
-		if(stream->rid[2])
-			json_array_append_new(rid, json_string(stream->rid[2]));
-		json_object_set_new(sr, "rid", rid);
-		json_object_set_new(sr, "rid-ext-id", json_integer(stream->rid_ext_id));
-		if(stream->ridrtx_ext_id > 0)
-			json_object_set_new(sr, "ridrtx-ext-id", json_integer(stream->ridrtx_ext_id));
-		if(stream->legacy_rid)
-			json_object_set_new(sr, "rid-syntax", json_string("legacy"));
-		json_object_set_new(s, "rid-simulcast", sr);
-	}
-	json_t *sd = json_object();
-	json_object_set_new(sd, "audio-send", stream->audio_send ? json_true() : json_false());
-	json_object_set_new(sd, "audio-recv", stream->audio_recv ? json_true() : json_false());
-	json_object_set_new(sd, "video-send", stream->video_send ? json_true() : json_false());
-	json_object_set_new(sd, "video-recv", stream->video_recv ? json_true() : json_false());
-	json_object_set_new(s, "direction", sd);
-	if(stream->audio_payload_type > -1 || stream->video_payload_type > -1) {
-		json_t *sc = json_object();
-		if(stream->audio_payload_type > -1)
-			json_object_set_new(sc, "audio-pt", json_integer(stream->audio_payload_type));
-		if(stream->audio_codec != NULL)
-			json_object_set_new(sc, "audio-codec", json_string(stream->audio_codec));
-		if(stream->video_payload_type > -1)
-			json_object_set_new(sc, "video-pt", json_integer(stream->video_payload_type));
-		if(stream->video_rtx_payload_type > -1)
-			json_object_set_new(sc, "video-rtx-pt", json_integer(stream->video_rtx_payload_type));
-		if(stream->video_codec != NULL)
-			json_object_set_new(sc, "video-codec", json_string(stream->video_codec));
-		json_object_set_new(s, "codecs", sc);
-	}
-	json_t *bwe = json_object();
-	json_object_set_new(bwe, "twcc", stream->do_transport_wide_cc ? json_true() : json_false());
-	if(stream->transport_wide_cc_ext_id > 0)
-		json_object_set_new(bwe, "twcc-ext-id", json_integer(stream->transport_wide_cc_ext_id));
-	json_object_set_new(s, "bwe", bwe);
-	json_t *components = json_array();
-	if(stream->component) {
-		json_t *c = janus_admin_component_summary(stream->component);
-		if(c)
-			json_array_append_new(components, c);
-	}
-	json_t *rtcp_stats = NULL;
-	if(stream->audio_rtcp_ctx != NULL) {
-		rtcp_stats = json_object();
-		json_t *audio_rtcp_stats = json_object();
-		json_object_set_new(audio_rtcp_stats, "base", json_integer(stream->audio_rtcp_ctx->tb));
-		json_object_set_new(audio_rtcp_stats, "rtt", json_integer(janus_rtcp_context_get_rtt(stream->audio_rtcp_ctx)));
-		json_object_set_new(audio_rtcp_stats, "lost", json_integer(janus_rtcp_context_get_lost_all(stream->audio_rtcp_ctx, FALSE)));
-		json_object_set_new(audio_rtcp_stats, "lost-by-remote", json_integer(janus_rtcp_context_get_lost_all(stream->audio_rtcp_ctx, TRUE)));
-		json_object_set_new(audio_rtcp_stats, "jitter-local", json_integer(janus_rtcp_context_get_jitter(stream->audio_rtcp_ctx, FALSE)));
-		json_object_set_new(audio_rtcp_stats, "jitter-remote", json_integer(janus_rtcp_context_get_jitter(stream->audio_rtcp_ctx, TRUE)));
-		json_object_set_new(audio_rtcp_stats, "in-link-quality", json_integer(janus_rtcp_context_get_in_link_quality(stream->audio_rtcp_ctx)));
-		json_object_set_new(audio_rtcp_stats, "in-media-link-quality", json_integer(janus_rtcp_context_get_in_media_link_quality(stream->audio_rtcp_ctx)));
-		json_object_set_new(audio_rtcp_stats, "out-link-quality", json_integer(janus_rtcp_context_get_out_link_quality(stream->audio_rtcp_ctx)));
-		json_object_set_new(audio_rtcp_stats, "out-media-link-quality", json_integer(janus_rtcp_context_get_out_media_link_quality(stream->audio_rtcp_ctx)));
-		json_object_set_new(rtcp_stats, "audio", audio_rtcp_stats);
-	}
-	int vindex=0;
-	for(vindex=0; vindex<3; vindex++) {
-		if(stream->video_rtcp_ctx[vindex] != NULL) {
-			if(rtcp_stats == NULL)
-				rtcp_stats = json_object();
-			json_t *video_rtcp_stats = json_object();
-			json_object_set_new(video_rtcp_stats, "base", json_integer(stream->video_rtcp_ctx[vindex]->tb));
-			if(vindex == 0)
-				json_object_set_new(video_rtcp_stats, "rtt", json_integer(janus_rtcp_context_get_rtt(stream->video_rtcp_ctx[vindex])));
-			json_object_set_new(video_rtcp_stats, "lost", json_integer(janus_rtcp_context_get_lost_all(stream->video_rtcp_ctx[vindex], FALSE)));
-			json_object_set_new(video_rtcp_stats, "lost-by-remote", json_integer(janus_rtcp_context_get_lost_all(stream->video_rtcp_ctx[vindex], TRUE)));
-			json_object_set_new(video_rtcp_stats, "jitter-local", json_integer(janus_rtcp_context_get_jitter(stream->video_rtcp_ctx[vindex], FALSE)));
-			json_object_set_new(video_rtcp_stats, "jitter-remote", json_integer(janus_rtcp_context_get_jitter(stream->video_rtcp_ctx[vindex], TRUE)));
-			json_object_set_new(video_rtcp_stats, "in-link-quality", json_integer(janus_rtcp_context_get_in_link_quality(stream->video_rtcp_ctx[vindex])));
-			json_object_set_new(video_rtcp_stats, "in-media-link-quality", json_integer(janus_rtcp_context_get_in_media_link_quality(stream->video_rtcp_ctx[vindex])));
-			json_object_set_new(video_rtcp_stats, "out-link-quality", json_integer(janus_rtcp_context_get_out_link_quality(stream->video_rtcp_ctx[vindex])));
-			json_object_set_new(video_rtcp_stats, "out-media-link-quality", json_integer(janus_rtcp_context_get_out_media_link_quality(stream->video_rtcp_ctx[vindex])));
-			if(vindex == 0)
-				json_object_set_new(rtcp_stats, "video", video_rtcp_stats);
-			else if(vindex == 1)
-				json_object_set_new(rtcp_stats, "video-sim1", video_rtcp_stats);
-			else
-				json_object_set_new(rtcp_stats, "video-sim2", video_rtcp_stats);
-		}
-	}
-	if(rtcp_stats != NULL)
-		json_object_set_new(s, "rtcp_stats", rtcp_stats);
-	json_object_set_new(s, "components", components);
-	return s;
-}
-
-json_t *janus_admin_component_summary(janus_ice_component *component) {
-	if(component == NULL)
-		return NULL;
-	janus_ice_handle *handle = component->stream ? component->stream->handle : NULL;
-	json_t *c = json_object();
-	json_object_set_new(c, "id", json_integer(component->component_id));
-	json_object_set_new(c, "state", json_string(janus_get_ice_state_name(component->state)));
-	if(component->icefailed_detected) {
-		json_object_set_new(c, "failed-detected", json_integer(component->icefailed_detected));
-		json_object_set_new(c, "icetimer-started", component->icestate_source ? json_true() : json_false());
-	}
-	if(component->component_connected > 0)
-		json_object_set_new(c, "connected", json_integer(component->component_connected));
-	if(component->local_candidates) {
+	json_t *w = json_object();
+	json_t *i = json_object();
+	json_object_set_new(i, "stream_id", json_integer(pc->stream_id));
+	json_object_set_new(i, "component_id", json_integer(pc->component_id));
+	json_object_set_new(i, "state", json_string(janus_get_ice_state_name(pc->state)));
+	if(pc->icefailed_detected) {
+		json_object_set_new(i, "failed-detected", json_integer(pc->icefailed_detected));
+		json_object_set_new(i, "icetimer-started", pc->icestate_source ? json_true() : json_false());
+	}
+	if(pc->ice_connected > 0)
+		json_object_set_new(i, "connected", json_integer(pc->ice_connected));
+	if(pc->local_candidates) {
 		json_t *cs = json_array();
-		GSList *candidates = component->local_candidates, *i = NULL;
-		for (i = candidates; i; i = i->next) {
-			gchar *lc = (gchar *) i->data;
+		GSList *candidates = pc->local_candidates, *c = NULL;
+		for (c = candidates; c; c = c->next) {
+			gchar *lc = (gchar *) c->data;
 			if(lc)
 				json_array_append_new(cs, json_string(lc));
 		}
-		json_object_set_new(c, "local-candidates", cs);
+		json_object_set_new(i, "local-candidates", cs);
 	}
-	if(component->remote_candidates) {
+	if(pc->remote_candidates) {
 		json_t *cs = json_array();
-		GSList *candidates = component->remote_candidates, *i = NULL;
-		for (i = candidates; i; i = i->next) {
-			gchar *rc = (gchar *) i->data;
+		GSList *candidates = pc->remote_candidates, *c = NULL;
+		for (c = candidates; c; c = c->next) {
+			gchar *rc = (gchar *) c->data;
 			if(rc)
 				json_array_append_new(cs, json_string(rc));
 		}
-		json_object_set_new(c, "remote-candidates", cs);
+		json_object_set_new(i, "remote-candidates", cs);
 	}
-	if(component->selected_pair) {
-		json_object_set_new(c, "selected-pair", json_string(component->selected_pair));
+	if(pc->selected_pair) {
+		json_object_set_new(i, "selected-pair", json_string(pc->selected_pair));
 	}
+	json_object_set_new(i, "ready", json_integer(pc->cdone));
+	json_object_set_new(w, "ice", i);
 	json_t *d = json_object();
-	json_t *in_stats = json_object();
-	json_t *out_stats = json_object();
-	if(component->dtls) {
-		janus_dtls_srtp *dtls = component->dtls;
+	if(pc->dtls) {
+		janus_dtls_srtp *dtls = pc->dtls;
 		json_object_set_new(d, "fingerprint", json_string(janus_dtls_get_local_fingerprint()));
-		if(component->stream) {
-			if(component->stream->remote_fingerprint)
-				json_object_set_new(d, "remote-fingerprint", json_string(component->stream->remote_fingerprint));
-			if(component->stream->remote_hashing)
-				json_object_set_new(d, "remote-fingerprint-hash", json_string(component->stream->remote_hashing));
-			json_object_set_new(d, "dtls-role", json_string(janus_get_dtls_srtp_role(component->stream->dtls_role)));
-		}
+		if(pc->remote_fingerprint)
+			json_object_set_new(d, "remote-fingerprint", json_string(pc->remote_fingerprint));
+		if(pc->remote_hashing)
+			json_object_set_new(d, "remote-fingerprint-hash", json_string(pc->remote_hashing));
+		json_object_set_new(d, "dtls-role", json_string(janus_get_dtls_srtp_role(pc->dtls_role)));
 		json_object_set_new(d, "dtls-state", json_string(janus_get_dtls_srtp_state(dtls->dtls_state)));
 		json_object_set_new(d, "retransmissions", json_integer(dtls->retransmissions));
 		json_object_set_new(d, "valid", dtls->srtp_valid ? json_true() : json_false());
@@ -2818,64 +2683,184 @@ json_t *janus_admin_component_summary(janus_ice_component *component) {
 			json_object_set_new(d, "handshake-started", json_integer(dtls->dtls_started));
 		if(dtls->dtls_connected > 0)
 			json_object_set_new(d, "connected", json_integer(dtls->dtls_connected));
-		if(handle && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO)) {
-			json_object_set_new(in_stats, "audio_packets", json_integer(component->in_stats.audio.packets));
-			json_object_set_new(in_stats, "audio_bytes", json_integer(component->in_stats.audio.bytes));
-			json_object_set_new(in_stats, "audio_bytes_lastsec", json_integer(component->in_stats.audio.bytes_lastsec));
-			json_object_set_new(in_stats, "do_audio_nacks", component->do_audio_nacks ? json_true() : json_false());
-			if(component->do_audio_nacks) {
-				json_object_set_new(in_stats, "audio_nacks", json_integer(component->in_stats.audio.nacks));
-				if(component->stream && component->stream->audio_rtcp_ctx)
-					json_object_set_new(in_stats, "audio_retransmissions", json_integer(component->stream->audio_rtcp_ctx->retransmitted));
-			}
-		}
-		if(handle && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO)) {
-			int vindex=0;
-			for(vindex=0; vindex<3; vindex++) {
-				if(vindex > 0 && component->stream->video_ssrc_peer[vindex] == 0)
-					continue;
-				json_t *container = (vindex == 0 ? in_stats : json_object());
-				json_object_set_new(container, "video_packets", json_integer(component->in_stats.video[vindex].packets));
-				json_object_set_new(container, "video_bytes", json_integer(component->in_stats.video[vindex].bytes));
-				json_object_set_new(container, "video_bytes_lastsec", json_integer(component->in_stats.video[vindex].bytes_lastsec));
-				if(vindex == 0)
-					json_object_set_new(container, "do_video_nacks", component->do_video_nacks ? json_true() : json_false());
-				if(component->do_video_nacks) {
-					json_object_set_new(container, "video_nacks", json_integer(component->in_stats.video[vindex].nacks));
-					if(component->stream && component->stream->video_rtcp_ctx[vindex])
-						json_object_set_new(in_stats, "video_retransmissions", json_integer(component->stream->video_rtcp_ctx[vindex]->retransmitted));
-				}
-				if(vindex == 1)
-					json_object_set_new(in_stats, "video-simulcast-1", container);
-				else if(vindex == 2)
-					json_object_set_new(in_stats, "video-simulcast-2", container);
-			}
-		}
-		json_object_set_new(in_stats, "data_packets", json_integer(component->in_stats.data.packets));
-		json_object_set_new(in_stats, "data_bytes", json_integer(component->in_stats.data.bytes));
-		if(handle && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO)) {
-			json_object_set_new(out_stats, "audio_packets", json_integer(component->out_stats.audio.packets));
-			json_object_set_new(out_stats, "audio_bytes", json_integer(component->out_stats.audio.bytes));
-			json_object_set_new(out_stats, "audio_bytes_lastsec", json_integer(component->out_stats.audio.bytes_lastsec));
-			json_object_set_new(out_stats, "audio_nacks", json_integer(component->out_stats.audio.nacks));
-		}
-		if(handle && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO)) {
-			json_object_set_new(out_stats, "video_packets", json_integer(component->out_stats.video[0].packets));
-			json_object_set_new(out_stats, "video_bytes", json_integer(component->out_stats.video[0].bytes));
-			json_object_set_new(out_stats, "video_bytes_lastsec", json_integer(component->out_stats.video[0].bytes_lastsec));
-			json_object_set_new(out_stats, "video_nacks", json_integer(component->out_stats.video[0].nacks));
-		}
-		json_object_set_new(out_stats, "data_packets", json_integer(component->out_stats.data.packets));
-		json_object_set_new(out_stats, "data_bytes", json_integer(component->out_stats.data.bytes));
 #ifdef HAVE_SCTP
 		/* FIXME Actually check if this succeeded? */
 		json_object_set_new(d, "sctp-association", dtls->sctp ? json_true() : json_false());
 #endif
+		json_t *stats = json_object();
+		json_t *in_stats = json_object();
+		json_object_set_new(in_stats, "packets", json_integer(pc->dtls_in_stats.info[0].packets));
+		json_object_set_new(in_stats, "bytes", json_integer(pc->dtls_in_stats.info[0].bytes));
+		json_t *out_stats = json_object();
+		json_object_set_new(out_stats, "packets", json_integer(pc->dtls_out_stats.info[0].packets));
+		json_object_set_new(out_stats, "bytes", json_integer(pc->dtls_out_stats.info[0].bytes));
+		json_object_set_new(stats, "in", in_stats);
+		json_object_set_new(stats, "out", out_stats);
+		json_object_set_new(d, "stats", stats);
+	}
+	json_object_set_new(w, "dtls", d);
+	json_t *bwe = json_object();
+	json_object_set_new(bwe, "twcc", pc->do_transport_wide_cc ? json_true() : json_false());
+	if(pc->transport_wide_cc_ext_id >= 0)
+		json_object_set_new(bwe, "twcc-ext-id", json_integer(pc->transport_wide_cc_ext_id));
+	json_object_set_new(w, "bwe", bwe);
+	if(pc->mid_ext_id > 0 || pc->rid_ext_id > 0) {
+		json_t *ext = json_object();
+		if(pc->mid_ext_id > 0)
+			json_object_set_new(ext, "mid-ext-id", json_integer(pc->mid_ext_id));
+		if(pc->rid_ext_id > 0)
+			json_object_set_new(ext, "rid-ext-id", json_integer(pc->rid_ext_id));
+		if(pc->ridrtx_ext_id > 0)
+			json_object_set_new(ext, "ridrtx-ext-id", json_integer(pc->ridrtx_ext_id));
+		json_object_set_new(w, "spec-ext", ext);
+	}
+	json_t *media = json_object();
+	/* Iterate on all media */
+	janus_handle_webrtc_medium *medium = NULL;
+	uint mi=0;
+	for(mi=0; mi<g_hash_table_size(pc->media); mi++) {
+		medium = g_hash_table_lookup(pc->media, GUINT_TO_POINTER(mi));
+		json_t *m = janus_admin_webrtc_medium_summary(medium);
+		if(m)
+			json_object_set_new(media, medium->mid, m);
+	}
+	json_object_set_new(w, "media", media);
+	return w;
+}
+
+json_t *janus_admin_webrtc_medium_summary(janus_handle_webrtc_medium *medium) {
+	if(medium == NULL)
+		return NULL;
+	/* SSRCs */
+	json_t *m = json_object();
+	if(medium->type == JANUS_MEDIA_AUDIO)
+		json_object_set_new(m, "type", json_string("audio"));
+	else if(medium->type == JANUS_MEDIA_VIDEO)
+		json_object_set_new(m, "type", json_string("video"));
+	else if(medium->type == JANUS_MEDIA_DATA)
+		json_object_set_new(m, "type", json_string("data"));
+	json_object_set_new(m, "mindex", json_integer(medium->mindex));
+	json_object_set_new(m, "mid", json_string(medium->mid));
+	if(medium->type != JANUS_MEDIA_DATA)
+		json_object_set_new(m, "do_nacks", medium->do_nacks ? json_true() : json_false());
+	if(medium->type != JANUS_MEDIA_DATA) {
+		json_t *ms = json_object();
+		if(medium->ssrc)
+			json_object_set_new(ms, "ssrc", json_integer(medium->ssrc));
+		if(medium->ssrc_rtx)
+			json_object_set_new(ms, "ssrc-rtx", json_integer(medium->ssrc_rtx));
+		if(medium->ssrc_peer[0])
+			json_object_set_new(ms, "ssrc-peer", json_integer(medium->ssrc_peer[0]));
+		if(medium->ssrc_peer[1])
+			json_object_set_new(ms, "ssrc-peer-sim-1", json_integer(medium->ssrc_peer[1]));
+		if(medium->ssrc_peer[2])
+			json_object_set_new(ms, "ssrc-peer-sim-2", json_integer(medium->ssrc_peer[2]));
+		if(medium->ssrc_peer_rtx[0])
+			json_object_set_new(ms, "ssrc-peer-rtx", json_integer(medium->ssrc_peer_rtx[0]));
+		if(medium->ssrc_peer_rtx[1])
+			json_object_set_new(ms, "ssrc-peer-sim-1-rtx", json_integer(medium->ssrc_peer_rtx[1]));
+		if(medium->ssrc_peer_rtx[2])
+			json_object_set_new(ms, "ssrc-peer-sim-2-rtx", json_integer(medium->ssrc_peer_rtx[2]));
+		json_object_set_new(m, "ssrc", ms);
+		if(medium->rid[0] && medium->pc->rid_ext_id > 0) {
+			json_t *mr = json_object();
+			json_t *rid = json_array();
+			json_array_append_new(rid, json_string(medium->rid[0]));
+			if(medium->rid[1])
+				json_array_append_new(rid, json_string(medium->rid[1]));
+			if(medium->rid[2])
+				json_array_append_new(rid, json_string(medium->rid[2]));
+			json_object_set_new(mr, "rid", rid);
+			json_object_set_new(mr, "rid-ext-id", json_integer(medium->pc->rid_ext_id));
+			if(medium->pc->ridrtx_ext_id > 0)
+				json_object_set_new(mr, "ridrtx-ext-id", json_integer(medium->pc->ridrtx_ext_id));
+			if(medium->legacy_rid)
+				json_object_set_new(mr, "rid-syntax", json_string("legacy"));
+			json_object_set_new(m, "rid-simulcast", mr);
+		}
+	}
+	/* Media direction */
+	if(medium->type != JANUS_MEDIA_DATA) {
+		json_t *md = json_object();
+		json_object_set_new(md, "send", medium->send ? json_true() : json_false());
+		json_object_set_new(md, "recv", medium->recv ? json_true() : json_false());
+		json_object_set_new(m, "direction", md);
+	}
+	/* Payload type and codec */
+	if(medium->type != JANUS_MEDIA_DATA && (medium->payload_type > -1 || medium->payload_type > -1)) {
+		json_t *sc = json_object();
+		if(medium->payload_type > -1)
+			json_object_set_new(sc, "pt", json_integer(medium->payload_type));
+		if(medium->rtx_payload_type > -1)
+			json_object_set_new(sc, "rtx-pt", json_integer(medium->rtx_payload_type));
+		if(medium->codec != NULL)
+			json_object_set_new(sc, "codec", json_string(medium->codec));
+		json_object_set_new(m, "codecs", sc);
+	}
+	/* RTCP stats */
+	int vindex=0;
+	if(medium->type != JANUS_MEDIA_DATA) {
+		json_t *rtcp_stats = NULL;
+		for(vindex=0; vindex<3; vindex++) {
+			if(medium->rtcp_ctx[vindex] != NULL) {
+				if(rtcp_stats == NULL)
+					rtcp_stats = json_object();
+				json_t *medium_rtcp_stats = json_object();
+				json_object_set_new(medium_rtcp_stats, "base", json_integer(medium->rtcp_ctx[vindex]->tb));
+				if(vindex == 0)
+					json_object_set_new(medium_rtcp_stats, "rtt", json_integer(janus_rtcp_context_get_rtt(medium->rtcp_ctx[vindex])));
+				json_object_set_new(medium_rtcp_stats, "lost", json_integer(janus_rtcp_context_get_lost_all(medium->rtcp_ctx[vindex], FALSE)));
+				json_object_set_new(medium_rtcp_stats, "lost-by-remote", json_integer(janus_rtcp_context_get_lost_all(medium->rtcp_ctx[vindex], TRUE)));
+				json_object_set_new(medium_rtcp_stats, "jitter-local", json_integer(janus_rtcp_context_get_jitter(medium->rtcp_ctx[vindex], FALSE)));
+				json_object_set_new(medium_rtcp_stats, "jitter-remote", json_integer(janus_rtcp_context_get_jitter(medium->rtcp_ctx[vindex], TRUE)));
+				json_object_set_new(medium_rtcp_stats, "in-link-quality", json_integer(janus_rtcp_context_get_in_link_quality(medium->rtcp_ctx[vindex])));
+				json_object_set_new(medium_rtcp_stats, "in-media-link-quality", json_integer(janus_rtcp_context_get_in_media_link_quality(medium->rtcp_ctx[vindex])));
+				json_object_set_new(medium_rtcp_stats, "out-link-quality", json_integer(janus_rtcp_context_get_out_link_quality(medium->rtcp_ctx[vindex])));
+				json_object_set_new(medium_rtcp_stats, "out-media-link-quality", json_integer(janus_rtcp_context_get_out_media_link_quality(medium->rtcp_ctx[vindex])));
+				if(vindex == 0)
+					json_object_set_new(rtcp_stats, "main", medium_rtcp_stats);
+				else if(vindex == 1)
+					json_object_set_new(rtcp_stats, "sim1", medium_rtcp_stats);
+				else
+					json_object_set_new(rtcp_stats, "sim2", medium_rtcp_stats);
+			}
+		}
+		if(rtcp_stats != NULL)
+			json_object_set_new(m, "rtcp", rtcp_stats);
 	}
-	json_object_set_new(c, "dtls", d);
-	json_object_set_new(c, "in_stats", in_stats);
-	json_object_set_new(c, "out_stats", out_stats);
-	return c;
+	/* Media stats */
+	json_t *stats = json_object();
+	json_t *in_stats = json_object();
+	json_t *out_stats = json_object();
+	for(vindex=0; vindex<3; vindex++) {
+		if(vindex > 0 && medium->ssrc_peer[vindex] == 0)
+			continue;
+		json_t *container = (vindex == 0 ? in_stats : json_object());
+		json_object_set_new(container, "packets", json_integer(medium->in_stats.info[vindex].packets));
+		json_object_set_new(container, "bytes", json_integer(medium->in_stats.info[vindex].bytes));
+		if(medium->type != JANUS_MEDIA_DATA) {
+			json_object_set_new(container, "bytes_lastsec", json_integer(medium->in_stats.info[vindex].bytes_lastsec));
+			if(medium->do_nacks) {
+				json_object_set_new(container, "nacks", json_integer(medium->in_stats.info[vindex].nacks));
+				if(medium->rtcp_ctx[vindex])
+					json_object_set_new(in_stats, "retransmissions", json_integer(medium->rtcp_ctx[vindex]->retransmitted));
+			}
+			if(vindex == 1)
+				json_object_set_new(in_stats, "video-simulcast-1", container);
+			else if(vindex == 2)
+				json_object_set_new(in_stats, "video-simulcast-2", container);
+		}
+	}
+	json_object_set_new(out_stats, "packets", json_integer(medium->out_stats.info[0].packets));
+	json_object_set_new(out_stats, "bytes", json_integer(medium->out_stats.info[0].bytes));
+	if(medium->type != JANUS_MEDIA_DATA) {
+		json_object_set_new(out_stats, "bytes_lastsec", json_integer(medium->out_stats.info[0].bytes_lastsec));
+		json_object_set_new(out_stats, "nacks", json_integer(medium->out_stats.info[0].nacks));
+	}
+	json_object_set_new(stats, "in", in_stats);
+	json_object_set_new(stats, "out", out_stats);
+	json_object_set_new(m, "stats", stats);
+	return m;
 }
 
 
@@ -3071,23 +3056,23 @@ int janus_plugin_push_event(janus_plugin_session *plugin_session, janus_plugin *
 	if(!janus_plugin_session_is_alive(plugin_session))
 		return -2;
 	janus_refcount_increase(&plugin_session->ref);
-	janus_ice_handle *ice_handle = (janus_ice_handle *)plugin_session->gateway_handle;
-	if(!ice_handle || janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP)) {
+	janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+	if(!handle || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP)) {
 		janus_refcount_decrease(&plugin_session->ref);
 		return JANUS_ERROR_SESSION_NOT_FOUND;
 	}
-	janus_refcount_increase(&ice_handle->ref);
-	janus_session *session = ice_handle->session;
+	janus_refcount_increase(&handle->ref);
+	janus_session *session = handle->session;
 	if(!session || g_atomic_int_get(&session->destroyed)) {
 		janus_refcount_decrease(&plugin_session->ref);
-		janus_refcount_decrease(&ice_handle->ref);
+		janus_refcount_decrease(&handle->ref);
 		return JANUS_ERROR_SESSION_NOT_FOUND;
 	}
 	/* Make sure this is a JSON object */
 	if(!json_is_object(message)) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Cannot push event (JSON error: not an object)\n", ice_handle->handle_id);
+		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Cannot push event (JSON error: not an object)\n", handle->handle_id);
 		janus_refcount_decrease(&plugin_session->ref);
-		janus_refcount_decrease(&ice_handle->ref);
+		janus_refcount_decrease(&handle->ref);
 		return JANUS_ERROR_INVALID_JSON_OBJECT;
 	}
 	/* Attach JSEP if possible? */
@@ -3098,16 +3083,16 @@ int janus_plugin_push_event(janus_plugin_session *plugin_session, janus_plugin *
 	if(sdp_type != NULL && sdp != NULL) {
 		merged_jsep = janus_plugin_handle_sdp(plugin_session, plugin, sdp_type, sdp, restart);
 		if(merged_jsep == NULL) {
-			if(ice_handle == NULL || janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP)
-					|| janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT)) {
-				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Cannot push event (handle not available anymore or negotiation stopped)\n", ice_handle->handle_id);
+			if(handle == NULL || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP)
+					|| janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT)) {
+				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Cannot push event (handle not available anymore or negotiation stopped)\n", handle->handle_id);
 				janus_refcount_decrease(&plugin_session->ref);
-				janus_refcount_decrease(&ice_handle->ref);
+				janus_refcount_decrease(&handle->ref);
 				return JANUS_ERROR_HANDLE_NOT_FOUND;
 			} else {
-				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Cannot push event (JSON error: problem with the SDP)\n", ice_handle->handle_id);
+				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Cannot push event (JSON error: problem with the SDP)\n", handle->handle_id);
 				janus_refcount_decrease(&plugin_session->ref);
-				janus_refcount_decrease(&ice_handle->ref);
+				janus_refcount_decrease(&handle->ref);
 				return JANUS_ERROR_JSEP_INVALID_SDP;
 			}
 		}
@@ -3116,9 +3101,9 @@ int janus_plugin_push_event(janus_plugin_session *plugin_session, janus_plugin *
 	json_incref(message);
 	/* Prepare JSON event */
 	json_t *event = janus_create_message("event", session->session_id, transaction);
-	json_object_set_new(event, "sender", json_integer(ice_handle->handle_id));
-	if(janus_is_opaqueid_in_api_enabled() && ice_handle->opaque_id != NULL)
-		json_object_set_new(event, "opaque_id", json_string(ice_handle->opaque_id));
+	json_object_set_new(event, "sender", json_integer(handle->handle_id));
+	if(janus_is_opaqueid_in_api_enabled() && handle->opaque_id != NULL)
+		json_object_set_new(event, "opaque_id", json_string(handle->opaque_id));
 	json_t *plugin_data = json_object();
 	json_object_set_new(plugin_data, "plugin", json_string(plugin->get_package()));
 	json_object_set_new(plugin_data, "data", message);
@@ -3131,21 +3116,21 @@ int janus_plugin_push_event(janus_plugin_session *plugin_session, janus_plugin *
 			const char *merged_sdp = json_string_value(json_object_get(merged_jsep, "sdp"));
 			/* Notify event handlers as well */
 			janus_events_notify_handlers(JANUS_EVENT_TYPE_JSEP,
-				session->session_id, ice_handle->handle_id, ice_handle->opaque_id, "local", merged_sdp_type, merged_sdp);
+				session->session_id, handle->handle_id, handle->opaque_id, "local", merged_sdp_type, merged_sdp);
 		}
 	}
 	/* Send the event */
-	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Sending event to transport...\n", ice_handle->handle_id);
+	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Sending event to transport...\n", handle->handle_id);
 	janus_session_notify_event(session, event);
 
-	if((restart || janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RESEND_TRICKLES))
+	if((restart || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RESEND_TRICKLES))
 			&& janus_ice_is_full_trickle_enabled()) {
 		/* We're restarting ICE, send our trickle candidates again */
-		janus_ice_resend_trickles(ice_handle);
+		janus_handle_resend_trickles(handle);
 	}
 
 	janus_refcount_decrease(&plugin_session->ref);
-	janus_refcount_decrease(&ice_handle->ref);
+	janus_refcount_decrease(&handle->ref);
 	return JANUS_OK;
 }
 
@@ -3155,9 +3140,9 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 		JANUS_LOG(LOG_ERR, "Invalid arguments\n");
 		return NULL;
 	}
-	janus_ice_handle *ice_handle = (janus_ice_handle *)plugin_session->gateway_handle;
-	//~ if(ice_handle == NULL || janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY)) {
-	if(ice_handle == NULL) {
+	janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+	//~ if(handle == NULL || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY)) {
+	if(handle == NULL) {
 		JANUS_LOG(LOG_ERR, "Invalid ICE handle\n");
 		return NULL;
 	}
@@ -3165,13 +3150,13 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 	if(!strcasecmp(sdp_type, "offer")) {
 		/* This is an offer from a plugin */
 		offer = 1;
-		janus_flags_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER);
-		janus_flags_clear(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER);
+		janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_OFFER);
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER);
 	} else if(!strcasecmp(sdp_type, "answer")) {
 		/* This is an answer from a plugin */
-		janus_flags_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_ANSWER);
-		if(janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_GOT_OFFER))
-			janus_flags_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEGOTIATED);
+		janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_ANSWER);
+		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_GOT_OFFER))
+			janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEGOTIATED);
 	} else {
 		/* TODO Handle other messages */
 		JANUS_LOG(LOG_ERR, "Unknown type '%s'\n", sdp_type);
@@ -3180,94 +3165,61 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 	/* Is this valid SDP? */
 	char error_str[512];
 	int audio = 0, video = 0, data = 0;
-	janus_sdp *parsed_sdp = janus_sdp_preparse(ice_handle, sdp, error_str, sizeof(error_str), &audio, &video, &data);
+	janus_sdp *parsed_sdp = janus_sdp_preparse(handle, sdp, &audio, &video, &data, error_str, sizeof(error_str));
 	if(parsed_sdp == NULL) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Couldn't parse SDP... %s\n", ice_handle->handle_id, error_str);
+		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Couldn't parse SDP... %s\n", handle->handle_id, error_str);
 		return NULL;
 	}
+	JANUS_LOG(LOG_VERB, "[%"SCNu64"] There are %d audio, %d video and %d data m-lines\n",
+		handle->handle_id, audio, video, data);
 	gboolean updating = FALSE;
 	if(offer) {
-		/* We may still not have a local ICE setup */
-		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Audio %s been negotiated\n", ice_handle->handle_id, audio ? "has" : "has NOT");
-		if(audio > 1) {
-			JANUS_LOG(LOG_ERR, "[%"SCNu64"] More than one audio line? only going to negotiate one...\n", ice_handle->handle_id);
-		}
-		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Video %s been negotiated\n", ice_handle->handle_id, video ? "has" : "has NOT");
-		if(video > 1) {
-			JANUS_LOG(LOG_ERR, "[%"SCNu64"] More than one video line? only going to negotiate one...\n", ice_handle->handle_id);
-		}
-		JANUS_LOG(LOG_VERB, "[%"SCNu64"] SCTP/DataChannels %s been negotiated\n", ice_handle->handle_id, data ? "have" : "have NOT");
-		if(data > 1) {
-			JANUS_LOG(LOG_ERR, "[%"SCNu64"] More than one data line? only going to negotiate one...\n", ice_handle->handle_id);
-		}
-#ifndef HAVE_SCTP
-		if(data) {
-			JANUS_LOG(LOG_WARN, "[%"SCNu64"]   -- DataChannels have been negotiated, but support for them has not been compiled...\n", ice_handle->handle_id);
-		}
-#endif
 		/* Are we still cleaning up from a previous media session? */
-		if(janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING)) {
-			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Still cleaning up from a previous media session, let's wait a bit...\n", ice_handle->handle_id);
+		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING)) {
+			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Still cleaning up from a previous media session, let's wait a bit...\n", handle->handle_id);
 			gint64 waited = 0;
-			while(janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_CLEANING)) {
-				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Still cleaning up from a previous media session, let's wait a bit...\n", ice_handle->handle_id);
+			while(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_CLEANING)) {
+				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Still cleaning up from a previous media session, let's wait a bit...\n", handle->handle_id);
 				g_usleep(100000);
 				waited += 100000;
 				if(waited >= 3*G_USEC_PER_SEC) {
-					JANUS_LOG(LOG_VERB, "[%"SCNu64"]   -- Waited 3 seconds, that's enough!\n", ice_handle->handle_id);
-					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Still cleaning a previous session\n", ice_handle->handle_id);
+					JANUS_LOG(LOG_VERB, "[%"SCNu64"]   -- Waited 3 seconds, that's enough!\n", handle->handle_id);
+					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Still cleaning a previous session\n", handle->handle_id);
 					janus_sdp_destroy(parsed_sdp);
 					return NULL;
 				}
 			}
 		}
-		if(ice_handle->agent == NULL) {
+		if(handle->agent == NULL) {
 			if(janus_is_rfc4588_enabled()) {
 				/* We still need to configure the WebRTC stuff: negotiate RFC4588 by default */
-				janus_flags_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX);
+				janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX);
 			}
 			/* Process SDP in order to setup ICE locally (this is going to result in an answer from the browser) */
-			janus_mutex_lock(&ice_handle->mutex);
-			if(janus_ice_setup_local(ice_handle, 0, audio, video, data, 1) < 0) {
-				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error setting ICE locally\n", ice_handle->handle_id);
+			janus_mutex_lock(&handle->mutex);
+			if(janus_handle_setup_local(handle, FALSE, TRUE) < 0) {
+				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error setting ICE locally\n", handle->handle_id);
+				janus_sdp_destroy(parsed_sdp);
+				janus_mutex_unlock(&handle->mutex);
+				return NULL;
+			}
+			/* Create medium instances */
+			if(janus_sdp_process_local(handle, parsed_sdp, FALSE) < 0) {
+				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error processing SDP\n", handle->handle_id);
 				janus_sdp_destroy(parsed_sdp);
-				janus_mutex_unlock(&ice_handle->mutex);
+				janus_mutex_unlock(&handle->mutex);
 				return NULL;
 			}
-			janus_mutex_unlock(&ice_handle->mutex);
+			janus_mutex_unlock(&handle->mutex);
 		} else {
 			updating = TRUE;
-			JANUS_LOG(LOG_INFO, "[%"SCNu64"] Updating existing session\n", ice_handle->handle_id);
-			if(offer && ice_handle->stream) {
-				/* We might need some new properties set as well */
-				janus_ice_stream *stream = ice_handle->stream;
-				if(audio) {
-					if(!janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO)) {
-						janus_flags_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO);
-						stream->audio_ssrc = janus_random_uint32();	/* FIXME Should we look for conflicts? */
-						if(stream->audio_rtcp_ctx == NULL) {
-							stream->audio_rtcp_ctx = g_malloc0(sizeof(rtcp_context));
-							stream->audio_rtcp_ctx->tb = 48000;	/* May change later */
-						}
-					}
-					if(ice_handle->audio_mid == NULL)
-						ice_handle->audio_mid = g_strdup("audio");
-				}
-				if(video) {
-					if(!janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO)) {
-						janus_flags_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO);
-						stream->video_ssrc = janus_random_uint32();	/* FIXME Should we look for conflicts? */
-						if(stream->video_rtcp_ctx[0] == NULL) {
-							stream->video_rtcp_ctx[0] = g_malloc0(sizeof(rtcp_context));
-							stream->video_rtcp_ctx[0]->tb = 90000;	/* May change later */
-						}
-					}
-					if(ice_handle->video_mid == NULL)
-						ice_handle->video_mid = g_strdup("video");
-				}
-				if(data) {
-					if(ice_handle->data_mid == NULL)
-						ice_handle->data_mid = g_strdup("data");
+			JANUS_LOG(LOG_INFO, "[%"SCNu64"] Updating existing session\n", handle->handle_id);
+			if(offer && handle->pc) {
+				/* Check what changed */
+				if(janus_sdp_process_local(handle, parsed_sdp, TRUE) < 0) {
+					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error processing SDP\n", handle->handle_id);
+					janus_sdp_destroy(parsed_sdp);
+					return NULL;
 				}
 			}
 		}
@@ -3292,57 +3244,68 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 		}
 	} else {
 		/* Check if the answer does contain the mid/rid/repaired-rid attributes */
+		uint mindex = 0;
 		gboolean do_mid = FALSE, do_rid = FALSE, do_repaired_rid = FALSE;
 		GList *temp = parsed_sdp->m_lines;
 		while(temp) {
 			janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+			gboolean have_mid = FALSE, have_rid = FALSE, have_repaired_rid = FALSE;
 			GList *tempA = m->attributes;
 			while(tempA) {
 				janus_sdp_attribute *a = (janus_sdp_attribute *)tempA->data;
 				if(a->name && a->value) {
 					if(strstr(a->value, JANUS_RTP_EXTMAP_MID))
-						do_mid = TRUE;
+						have_mid = TRUE;
 					else if(strstr(a->value, JANUS_RTP_EXTMAP_RID))
-						do_rid = TRUE;
+						have_rid = TRUE;
 					else if(strstr(a->value, JANUS_RTP_EXTMAP_REPAIRED_RID))
-						do_repaired_rid = TRUE;
+						have_repaired_rid = TRUE;
 				}
 				tempA = tempA->next;
 			}
+			if(!have_rid) {
+				janus_handle_webrtc_medium *medium = g_hash_table_lookup(handle->pc->media, GUINT_TO_POINTER(mindex));
+				if(medium != NULL) {
+					g_free(medium->rid[0]);
+					medium->rid[0] = NULL;
+					g_free(medium->rid[1]);
+					medium->rid[1] = NULL;
+					g_free(medium->rid[2]);
+					medium->rid[2] = NULL;
+					if(medium->ssrc_peer_temp > 0) {
+						medium->ssrc_peer[0] = medium->ssrc_peer_temp;
+						medium->ssrc_peer_temp = 0;
+					}
+				}
+			}
+			do_mid = do_mid || have_mid;
+			do_rid = do_rid || have_rid;
+			do_repaired_rid = do_repaired_rid || have_repaired_rid;
+			mindex++;
 			temp = temp->next;
 		}
-		if(!do_mid && ice_handle->stream)
-			ice_handle->stream->mid_ext_id = 0;
-		if(!do_rid && ice_handle->stream) {
-			ice_handle->stream->rid_ext_id = 0;
-			ice_handle->stream->ridrtx_ext_id = 0;
-			g_free(ice_handle->stream->rid[0]);
-			ice_handle->stream->rid[0] = NULL;
-			g_free(ice_handle->stream->rid[1]);
-			ice_handle->stream->rid[1] = NULL;
-			g_free(ice_handle->stream->rid[2]);
-			ice_handle->stream->rid[2] = NULL;
-			if(ice_handle->stream->video_ssrc_peer_temp > 0) {
-				ice_handle->stream->video_ssrc_peer[0] = ice_handle->stream->video_ssrc_peer_temp;
-				ice_handle->stream->video_ssrc_peer_temp = 0;
-			}
+		if(!do_mid && handle->pc)
+			handle->pc->mid_ext_id = 0;
+		if(!do_rid && handle->pc) {
+			handle->pc->rid_ext_id = 0;
+			handle->pc->ridrtx_ext_id = 0;
 		}
-		if(!do_repaired_rid && ice_handle->stream)
-			ice_handle->stream->ridrtx_ext_id = 0;
+		if(!do_repaired_rid && handle->pc)
+			handle->pc->ridrtx_ext_id = 0;
 	}
 	if(!updating && !janus_ice_is_full_trickle_enabled()) {
 		/* Wait for candidates-done callback */
-		while(ice_handle->cdone < 1) {
-			if(ice_handle == NULL || janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP)
-					|| janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT)) {
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Handle detached or PC closed, giving up...!\n", ice_handle ? ice_handle->handle_id : 0);
+		while(handle->cdone < 1) {
+			if(handle == NULL || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP)
+					|| janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT)) {
+				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Handle detached or PC closed, giving up...!\n", handle ? handle->handle_id : 0);
 				janus_sdp_destroy(parsed_sdp);
 				return NULL;
 			}
-			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Waiting for candidates-done callback...\n", ice_handle->handle_id);
+			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Waiting for candidates-done callback...\n", handle->handle_id);
 			g_usleep(10000);
-			if(ice_handle->cdone < 0) {
-				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error gathering candidates!\n", ice_handle->handle_id);
+			if(handle->cdone < 0) {
+				JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error gathering candidates!\n", handle->handle_id);
 				janus_sdp_destroy(parsed_sdp);
 				return NULL;
 			}
@@ -3351,63 +3314,70 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 	/* Anonymize SDP */
 	if(janus_sdp_anonymize(parsed_sdp) < 0) {
 		/* Invalid SDP */
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Invalid SDP\n", ice_handle->handle_id);
+		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Invalid SDP\n", handle->handle_id);
 		janus_sdp_destroy(parsed_sdp);
 		return NULL;
 	}
 
 	/* Check if this is a renegotiation and we need an ICE restart */
 	if(offer && restart)
-		janus_ice_restart(ice_handle);
+		janus_handle_ice_restart(handle);
 	/* Add our details */
-	janus_mutex_lock(&ice_handle->mutex);
-	janus_ice_stream *stream = ice_handle->stream;
-	if (stream == NULL) {
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error stream not found\n", ice_handle->handle_id);
-		janus_mutex_unlock(&ice_handle->mutex);
+	janus_mutex_lock(&handle->mutex);
+	janus_handle_webrtc *pc = handle->pc;
+	if (pc == NULL) {
+		JANUS_LOG(LOG_ERR, "[%"SCNu64"] No WebRTC PeerConnection\n", handle->handle_id);
+		janus_mutex_unlock(&handle->mutex);
 		return NULL;
 	}
-	if(janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX) &&
-			stream->rtx_payload_types == NULL) {
-		/* Make sure we have a list of rtx payload types to generate, if needed */
-		janus_sdp_mline *m = janus_sdp_mline_find(parsed_sdp, JANUS_SDP_VIDEO);
-		if(m && m->ptypes) {
-			stream->rtx_payload_types = g_hash_table_new(NULL, NULL);
-			GList *ptypes = g_list_copy(m->ptypes), *tempP = ptypes;
-			GList *rtx_ptypes = g_hash_table_get_values(stream->rtx_payload_types);
-			while(tempP) {
-				int ptype = GPOINTER_TO_INT(tempP->data);
-				int rtx_ptype = ptype+1;
-				if(rtx_ptype > 127)
-					rtx_ptype = 96;
-				while(g_list_find(m->ptypes, GINT_TO_POINTER(rtx_ptype))
-						|| g_list_find(rtx_ptypes, GINT_TO_POINTER(rtx_ptype))) {
-					rtx_ptype++;
+	/* Iterate on all media */
+	janus_handle_webrtc_medium *medium = NULL;
+	uint mi=0;
+	for(mi=0; mi<g_hash_table_size(pc->media); mi++) {
+		medium = g_hash_table_lookup(pc->media, GUINT_TO_POINTER(mi));
+		if(medium && medium->type == JANUS_MEDIA_VIDEO &&
+				janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX) &&
+				medium->rtx_payload_types == NULL) {
+			/* Make sure we have a list of rtx payload types to generate, if needed */
+			janus_sdp_mline *m = janus_sdp_mline_find_by_index(parsed_sdp, medium->mindex);
+			if(m && m->ptypes) {
+				medium->rtx_payload_types = g_hash_table_new(NULL, NULL);
+				GList *ptypes = g_list_copy(m->ptypes), *tempP = ptypes;
+				GList *rtx_ptypes = g_hash_table_get_values(medium->rtx_payload_types);
+				while(tempP) {
+					int ptype = GPOINTER_TO_INT(tempP->data);
+					int rtx_ptype = ptype+1;
 					if(rtx_ptype > 127)
 						rtx_ptype = 96;
-					if(rtx_ptype == ptype) {
-						/* We did a whole round? should never happen... */
-						rtx_ptype = -1;
-						break;
+					while(g_list_find(m->ptypes, GINT_TO_POINTER(rtx_ptype))
+							|| g_list_find(rtx_ptypes, GINT_TO_POINTER(rtx_ptype))) {
+						rtx_ptype++;
+						if(rtx_ptype > 127)
+							rtx_ptype = 96;
+						if(rtx_ptype == ptype) {
+							/* We did a whole round? should never happen... */
+							rtx_ptype = -1;
+							break;
+						}
 					}
+					if(rtx_ptype > 0)
+						g_hash_table_insert(medium->rtx_payload_types, GINT_TO_POINTER(ptype), GINT_TO_POINTER(rtx_ptype));
+					g_list_free(rtx_ptypes);
+					rtx_ptypes = g_hash_table_get_values(medium->rtx_payload_types);
+					tempP = tempP->next;
 				}
-				if(rtx_ptype > 0)
-					g_hash_table_insert(stream->rtx_payload_types, GINT_TO_POINTER(ptype), GINT_TO_POINTER(rtx_ptype));
+				g_list_free(ptypes);
 				g_list_free(rtx_ptypes);
-				rtx_ptypes = g_hash_table_get_values(stream->rtx_payload_types);
-				tempP = tempP->next;
 			}
-			g_list_free(ptypes);
-			g_list_free(rtx_ptypes);
 		}
 	}
 	/* Enrich the SDP the plugin gave us with all the WebRTC related stuff */
-	char *sdp_merged = janus_sdp_merge(ice_handle, parsed_sdp, offer ? TRUE : FALSE);
+	char *sdp_merged = janus_sdp_merge(handle, parsed_sdp, offer ? TRUE : FALSE);
 	if(sdp_merged == NULL) {
 		/* Couldn't merge SDP */
-		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error merging SDP\n", ice_handle->handle_id);
+		JANUS_LOG(LOG_ERR, "[%"SCNu64"] Error merging SDP\n", handle->handle_id);
 		janus_sdp_destroy(parsed_sdp);
-		janus_mutex_unlock(&ice_handle->mutex);
+		janus_mutex_unlock(&handle->mutex);
 		return NULL;
 	}
 	janus_sdp_destroy(parsed_sdp);
@@ -3415,22 +3385,21 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 	if(!updating) {
 		if(offer) {
 			/* We set the flag to wait for an answer before handling trickle candidates */
-			janus_flags_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER);
+			janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER);
 		} else {
-			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Sending answer, ready to setup remote candidates and send connectivity checks...\n", ice_handle->handle_id);
-			janus_request_ice_handle_answer(ice_handle, audio, video, data, NULL);
+			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Sending answer, ready to setup remote candidates and send connectivity checks...\n", handle->handle_id);
+			janus_request_handle_answer(handle, NULL);
 		}
 	}
 #ifdef HAVE_SCTP
-	if(!offer && janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_READY)) {
+	if(!offer && janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_READY)) {
 		/* Renegotiation: check if datachannels were just added on an existing PeerConnection */
-		if(janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS)) {
-			janus_ice_stream *stream = ice_handle->stream;
-			if(stream != NULL && stream->component != NULL &&
-					stream->component->dtls != NULL && stream->component->dtls->sctp == NULL) {
+		if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_DATA_CHANNELS)) {
+			janus_handle_webrtc *pc = handle->pc;
+			if(pc != NULL && pc->dtls != NULL && pc->dtls->sctp == NULL) {
 				/* Create SCTP association as well */
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Creating datachannels...\n", ice_handle->handle_id);
-				janus_dtls_srtp_create_sctp(stream->component->dtls);
+				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Creating datachannels...\n", handle->handle_id);
+				janus_dtls_srtp_create_sctp(pc->dtls);
 			}
 		}
 	}
@@ -3440,39 +3409,39 @@ json_t *janus_plugin_handle_sdp(janus_plugin_session *plugin_session, janus_plug
 	json_t *jsep = json_object();
 	json_object_set_new(jsep, "type", json_string(sdp_type));
 	json_object_set_new(jsep, "sdp", json_string(sdp_merged));
-	char *tmp = ice_handle->local_sdp;
-	ice_handle->local_sdp = sdp_merged;
-	janus_mutex_unlock(&ice_handle->mutex);
+	char *tmp = handle->local_sdp;
+	handle->local_sdp = sdp_merged;
+	janus_mutex_unlock(&handle->mutex);
 	g_free(tmp);
 	return jsep;
 }
 
-void janus_plugin_relay_rtp(janus_plugin_session *plugin_session, int video, char *buf, int len) {
+void janus_plugin_relay_rtp(janus_plugin_session *plugin_session, int mindex, gboolean video, char *buf, int len) {
 	if((plugin_session < (janus_plugin_session *)0x1000) || g_atomic_int_get(&plugin_session->stopped) || buf == NULL || len < 1)
 		return;
-	janus_ice_handle *handle = (janus_ice_handle *)plugin_session->gateway_handle;
-	if(!handle || janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP)
-			|| janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT))
+	janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+	if(!handle || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP)
+			|| janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT))
 		return;
-	janus_ice_relay_rtp(handle, video, buf, len);
+	janus_ice_relay_rtp(handle, mindex, video, buf, len);
 }
 
-void janus_plugin_relay_rtcp(janus_plugin_session *plugin_session, int video, char *buf, int len) {
+void janus_plugin_relay_rtcp(janus_plugin_session *plugin_session, int mindex, gboolean video, char *buf, int len) {
 	if((plugin_session < (janus_plugin_session *)0x1000) || g_atomic_int_get(&plugin_session->stopped) || buf == NULL || len < 1)
 		return;
-	janus_ice_handle *handle = (janus_ice_handle *)plugin_session->gateway_handle;
-	if(!handle || janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP)
-			|| janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT))
+	janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+	if(!handle || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP)
+			|| janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT))
 		return;
-	janus_ice_relay_rtcp(handle, video, buf, len);
+	janus_ice_relay_rtcp(handle, mindex, video, buf, len);
 }
 
 void janus_plugin_relay_data(janus_plugin_session *plugin_session, char *label, char *buf, int len) {
 	if((plugin_session < (janus_plugin_session *)0x1000) || g_atomic_int_get(&plugin_session->stopped) || buf == NULL || len < 1)
 		return;
-	janus_ice_handle *handle = (janus_ice_handle *)plugin_session->gateway_handle;
-	if(!handle || janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP)
-			|| janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT))
+	janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+	if(!handle || janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP)
+			|| janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT))
 		return;
 #ifdef HAVE_SCTP
 	janus_ice_relay_data(handle, label, buf, len);
@@ -3484,23 +3453,23 @@ void janus_plugin_relay_data(janus_plugin_session *plugin_session, char *label,
 static gboolean janus_plugin_close_pc_internal(gpointer user_data) {
 	/* We actually enforce the close_pc here */
 	janus_plugin_session *plugin_session = (janus_plugin_session *) user_data;
-	janus_ice_handle *ice_handle = (janus_ice_handle *)plugin_session->gateway_handle;
-	if(!ice_handle || !g_atomic_int_compare_and_exchange(&ice_handle->closepc, 1, 0)) {
+	janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+	if(!handle || !g_atomic_int_compare_and_exchange(&handle->closepc, 1, 0)) {
 		janus_refcount_decrease(&plugin_session->ref);
 		return G_SOURCE_REMOVE;
 	}
-	janus_refcount_increase(&ice_handle->ref);
-	if(janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP)
-			|| janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALERT)) {
+	janus_refcount_increase(&handle->ref);
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP)
+			|| janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALERT)) {
 		janus_refcount_decrease(&plugin_session->ref);
-		janus_refcount_decrease(&ice_handle->ref);
+		janus_refcount_decrease(&handle->ref);
 		return G_SOURCE_REMOVE;
 	}
-	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Plugin asked to hangup PeerConnection: sending alert\n", ice_handle->handle_id);
+	JANUS_LOG(LOG_VERB, "[%"SCNu64"] Plugin asked to hangup PeerConnection: sending alert\n", handle->handle_id);
 	/* Send an alert on all the DTLS connections */
-	janus_ice_webrtc_hangup(ice_handle, "Close PC");
+	janus_handle_webrtc_hangup(handle, "Close PC");
 	janus_refcount_decrease(&plugin_session->ref);
-	janus_refcount_decrease(&ice_handle->ref);
+	janus_refcount_decrease(&handle->ref);
 
 	return G_SOURCE_REMOVE;
 }
@@ -3509,8 +3478,8 @@ void janus_plugin_close_pc(janus_plugin_session *plugin_session) {
 	/* A plugin asked to get rid of a PeerConnection: enqueue it as a timed source */
 	if(!janus_plugin_session_is_alive(plugin_session))
 		return;
-	janus_ice_handle *ice_handle = (janus_ice_handle *)plugin_session->gateway_handle;
-	if(!ice_handle || !g_atomic_int_compare_and_exchange(&ice_handle->closepc, 0, 1))
+	janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+	if(!handle || !g_atomic_int_compare_and_exchange(&handle->closepc, 0, 1))
 		return;
 	janus_refcount_increase(&plugin_session->ref);
 	GSource *timeout_source = g_timeout_source_new_seconds(0);
@@ -3522,28 +3491,28 @@ void janus_plugin_close_pc(janus_plugin_session *plugin_session) {
 static gboolean janus_plugin_end_session_internal(gpointer user_data) {
 	/* We actually enforce the end_session here */
 	janus_plugin_session *plugin_session = (janus_plugin_session *) user_data;
-	janus_ice_handle *ice_handle = (janus_ice_handle *)plugin_session->gateway_handle;
-	if(!ice_handle) {
+	janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+	if(!handle) {
 		janus_refcount_decrease(&plugin_session->ref);
 		return G_SOURCE_REMOVE;
 	}
-	janus_refcount_increase(&ice_handle->ref);
-	if(janus_flags_is_set(&ice_handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_STOP)) {
+	janus_refcount_increase(&handle->ref);
+	if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_STOP)) {
 		janus_refcount_decrease(&plugin_session->ref);
-		janus_refcount_decrease(&ice_handle->ref);
+		janus_refcount_decrease(&handle->ref);
 		return G_SOURCE_REMOVE;
 	}
-	janus_session *session = (janus_session *)ice_handle->session;
+	janus_session *session = (janus_session *)handle->session;
 	if(!session) {
 		janus_refcount_decrease(&plugin_session->ref);
-		janus_refcount_decrease(&ice_handle->ref);
+		janus_refcount_decrease(&handle->ref);
 		return G_SOURCE_REMOVE;
 	}
 	/* Destroy the handle */
-	janus_session_handles_remove(session, ice_handle);
+	janus_session_handles_remove(session, handle);
 
 	janus_refcount_decrease(&plugin_session->ref);
-	janus_refcount_decrease(&ice_handle->ref);
+	janus_refcount_decrease(&handle->ref);
 	return G_SOURCE_REMOVE;
 }
 
@@ -3569,14 +3538,14 @@ void janus_plugin_notify_event(janus_plugin *plugin, janus_plugin_session *plugi
 			json_decref(event);
 			return;
 		}
-		janus_ice_handle *ice_handle = (janus_ice_handle *)plugin_session->gateway_handle;
-		if(!ice_handle) {
+		janus_handle *handle = (janus_handle *)plugin_session->gateway_handle;
+		if(!handle) {
 			json_decref(event);
 			return;
 		}
-		handle_id = ice_handle->handle_id;
-		opaque_id = ice_handle->opaque_id;
-		janus_session *session = (janus_session *)ice_handle->session;
+		handle_id = handle->handle_id;
+		opaque_id = handle->opaque_id;
+		janus_session *session = (janus_session *)handle->session;
 		if(!session) {
 			json_decref(event);
 			return;
@@ -4388,6 +4357,7 @@ gint main(int argc, char *argv[])
 	GThread *watchdog = g_thread_try_new("timeout watchdog", &janus_sessions_watchdog, watchdog_loop, &error);
 	if(error != NULL) {
 		JANUS_LOG(LOG_FATAL, "Got error %d (%s) trying to start sessions timeout watchdog...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		exit(1);
 	}
 	/* Start the thread that will dispatch incoming requests */
@@ -4395,6 +4365,7 @@ gint main(int argc, char *argv[])
 	GThread *requests_thread = g_thread_try_new("sessions requests", &janus_transport_requests, NULL, &error);
 	if(error != NULL) {
 		JANUS_LOG(LOG_FATAL, "Got error %d (%s) trying to start requests thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		exit(1);
 	}
 	/* Create a thread pool to handle asynchronous requests, no matter what the transport */
@@ -4403,6 +4374,7 @@ gint main(int argc, char *argv[])
 	if(error != NULL) {
 		/* Something went wrong... */
 		JANUS_LOG(LOG_FATAL, "Got error %d (%s) trying to launch the request pool task thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		exit(1);
 	}
 	/* Wait 120 seconds before stopping idle threads to avoid the creation of too many threads for AddressSanitizer. */
diff --git a/janus.h b/janus.h
index d4ead3864e..43b9706c9b 100644
--- a/janus.h
+++ b/janus.h
@@ -46,7 +46,7 @@ typedef struct janus_session {
 	/*! \brief Janus Core-Client session ID */
 	guint64 session_id;
 	/*! \brief Map of handles this session is managing */
-	GHashTable *ice_handles;
+	GHashTable *handles;
 	/*! \brief Time of the last activity on the session */
 	gint64 last_activity;
 	/*! \brief Pointer to the request instance (and the transport that originated the session) */
@@ -83,24 +83,24 @@ void janus_session_notify_event(janus_session *session, json_t *event);
  * @param[in] session The Janus Core-Client session to destroy
  * @returns 0 in case of success, a negative integer otherwise */
 gint janus_session_destroy(janus_session *session);
-/*! \brief Method to find an existing Janus ICE handle from its ID
+/*! \brief Method to find an existing Janus handle from its ID
  * @param[in] session The Janus Core-Client session this ICE handle belongs to
- * @param[in] handle_id The Janus ICE handle ID
- * @returns The Janus ICE handle if successful, NULL otherwise */
-janus_ice_handle *janus_session_handles_find(janus_session *session, guint64 handle_id);
-/*! \brief Method to insert a Janus ICE handle in a session
+ * @param[in] handle_id The Janus handle ID
+ * @returns The Janus handle if successful, NULL otherwise */
+janus_handle *janus_session_handles_find(janus_session *session, guint64 handle_id);
+/*! \brief Method to insert a Janus handle in a session
  * @param[in] session The Janus Core-Client session
- * @param[in] handle The Janus ICE handle */
-void janus_session_handles_insert(janus_session *session, janus_ice_handle *handle);
-/*! \brief Method to remove a Janus ICE handle from a session
+ * @param[in] handle The Janus handle */
+void janus_session_handles_insert(janus_session *session, janus_handle *handle);
+/*! \brief Method to remove a Janus handle from a session
  * @param[in] session The Janus Core-Client session
- * @param[in] handle The Janus ICE handle
- * @returns The error code of janus_ice_handle_destroy */
-gint janus_session_handles_remove(janus_session *session, janus_ice_handle *handle);
-/*! \brief Method to remove all Janus ICE handles from a session
+ * @param[in] handle The Janus handle
+ * @returns The error code of janus_handle_destroy */
+gint janus_session_handles_remove(janus_session *session, janus_handle *handle);
+/*! \brief Method to remove all Janus handles from a session
  * @param[in] session The Janus Core-Client session */
 void janus_session_handles_clear(janus_session *session);
-/*! \brief Method to list the IDs of all Janus ICE handles of a session as JSON
+/*! \brief Method to list the IDs of all Janus handles of a session as JSON
  * @param[in] session The Janus Core-Client session
  * @returns The JSON array */
 json_t *janus_session_handles_list_json(janus_session *session);
diff --git a/mainpage.dox b/mainpage.dox
index cb47903601..1373b651f7 100644
--- a/mainpage.dox
+++ b/mainpage.dox
@@ -351,8 +351,8 @@ var janus = new Janus(
  * possibly react accordingly (e.g., decrease the bitrate if most of our
  * packets are getting lost);
  * 		- \c onmessage: a message/event has been received from the plugin;
- * 		- \c onlocalstream: a local \c MediaStream is available and ready to be displayed;
- * 		- \c onremotestream: a remote \c MediaStream is available and ready to be displayed;
+ * 		- \c onlocaltrack: a local \c MediaStreamTrack is available and ready to be displayed;
+ * 		- \c onremotetrack: a remote \c MediaStreamTrack is available and ready to be displayed;
  * 		- \c ondataopen: a Data Channel is available and ready to be used;
  * 		- \c ondata: data has been received through the Data Channel;
  * 		- \c oncleanup: the WebRTC PeerConnection with the plugin was closed;
@@ -379,11 +379,11 @@ janus.attach(
 			// We got a message/event (msg) from the plugin
 			// If jsep is not null, this involves a WebRTC negotiation
 		},
-		onlocalstream: function(stream) {
-			// We have a local stream (getUserMedia worked!) to display
+		onlocaltrack: function(track, added) {
+			// A local track to display has just been added (getUserMedia worked!) or removed
 		},
-		onremotestream: function(stream) {
-			// We have a remote stream (working PeerConnection!) to display
+		onremotetrack: function(track, mid, added) {
+			// A remote track (working PeerConnection!) with a specific mid has just been added or removed
 		},
 		oncleanup: function() {
 			// PeerConnection with the plugin closed, clean the UI
@@ -413,7 +413,8 @@ janus.attach(
  * - \c handleRemoteJsep(callbacks): asks the library to handle an incoming WebRTC compliant session description;
  * - \c dtmf(parameters): sends a DTMF tone on the PeerConnection;
  * - \c data(parameters): sends data through the Data Channel, if available;
- * - \c getBitrate(): gets a verbose description of the currently received stream bitrate;
+ * - \c getBitrate(mid): gets a verbose description of the currently received
+ * video stream bitrate (optional mid to specify the stream, first video stream if missing);
  * - \c hangup(sendRequest): tells the library to close the PeerConnection; if the optional \c sendRequest argument is
  * set to \c true, then a \c hangup Janus API request is sent to Janus as well (disabled by default, Janus can usually
  * figure this out via DTLS alerts and the like but it may be useful to enable it sometimes);
@@ -439,8 +440,8 @@ janus.attach(
  * <code>send({msg, jsep})</code> to close the loop with the plugin), otherwise use
  * \c handleRemoteJsep ;
  * -# whether you took the initiative to set up a PeerConnection or the plugin did,
- * the \c onlocalstream and/or the \c onremotestream callbacks will provide
- * you with a stream you can display in your page;
+ * the \c onlocaltrack and/or the \c onremotetrack callbacks will provide
+ * you with info on media tracks you can display or play in your page;
  * -# each plugin may allow you to manipulate what should flow through the
  * PeerConnection channel: the \c send method and \c onmessage callback
  * will allow you to handle this interaction (e.g., to tell the plugin
@@ -545,13 +546,13 @@ janus.attach(
 			}
 		},
 		[..]
-		onlocalstream: function(stream) {
+		onlocaltrack: function(track, added) {
 			// Invoked after createOffer
-			// This is our video
+			// This is info on a local track: when added, we can choose to render
 		},
-		onremotestream: function(stream) {
+		onremotetrack: function(track, mid, added) {
 			// Invoked after handleRemoteJsep has got us a PeerConnection
-			// This is the remote video
+			// This is info on a remote track: when added, we can choose to render
 		},
 		[..]
   \endverbatim
@@ -559,7 +560,7 @@ janus.attach(
  * This, instead, is an example of how to use \c createAnswer, taken from the Streaming demo page:
  *
  \verbatim
-// Attach to echo test plugin
+// Attach to Streaming plugin
 janus.attach(
 	{
 		plugin: "janus.plugin.streaming",
@@ -591,12 +592,12 @@ janus.attach(
 			}
 		},
 		[..]
-		onlocalstream: function(stream) {
+		onlocaltrack: function(track, added) {
 			// This will NOT be invoked, we chose recvonly
 		},
-		onremotestream: function(stream) {
+		onremotetrack: function(track, mid, added) {
 			// Invoked after send has got us a PeerConnection
-			// This is the remote video
+			// This is info on a remote track: when added, we can choose to render
 		},
 		[..]
   \endverbatim
@@ -727,7 +728,7 @@ echotest.createOffer(
  \endverbatim
  *
  * Notice that renegotiations involving media changes (both local and remote)
- * will likely result in new calls to the \c onlocalstream and \c onremotestream
+ * will likely result in new calls to the \c onlocaltrack and \c onremotetrack
  * application callbacks: as such, be prepared to see those callbacks called
  * for the same PeerConnection more than once during the course of a media session.
  *
diff --git a/plugins/janus_audiobridge.c b/plugins/janus_audiobridge.c
index 37f13fc854..4ea9ff34bc 100644
--- a/plugins/janus_audiobridge.c
+++ b/plugins/janus_audiobridge.c
@@ -668,8 +668,8 @@ room-<unique room ID>: {
 
 
 /* Plugin information */
-#define JANUS_AUDIOBRIDGE_VERSION			10
-#define JANUS_AUDIOBRIDGE_VERSION_STRING	"0.0.10"
+#define JANUS_AUDIOBRIDGE_VERSION			11
+#define JANUS_AUDIOBRIDGE_VERSION_STRING	"0.0.11"
 #define JANUS_AUDIOBRIDGE_DESCRIPTION		"This is a plugin implementing an audio conference bridge for Janus, mixing Opus streams."
 #define JANUS_AUDIOBRIDGE_NAME				"JANUS AudioBridge plugin"
 #define JANUS_AUDIOBRIDGE_AUTHOR			"Meetecho s.r.l."
@@ -693,8 +693,8 @@ void janus_audiobridge_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_audiobridge_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 json_t *janus_audiobridge_handle_admin_message(json_t *message);
 void janus_audiobridge_setup_media(janus_plugin_session *handle);
-void janus_audiobridge_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_audiobridge_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_audiobridge_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_audiobridge_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_audiobridge_hangup_media(janus_plugin_session *handle);
 void janus_audiobridge_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_audiobridge_query_session(janus_plugin_session *handle);
@@ -732,51 +732,51 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter adminkey_parameters[] = {
-	{"admin_key", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"admin_key", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter create_parameters[] = {
-	{"description", JSON_STRING, 0},
-	{"secret", JSON_STRING, 0},
-	{"pin", JSON_STRING, 0},
+	{"description", JANUS_JSON_STRING, 0},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"pin", JANUS_JSON_STRING, 0},
 	{"is_private", JANUS_JSON_BOOL, 0},
-	{"allowed", JSON_ARRAY, 0},
-	{"sampling_rate", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"sampling", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},	/* We keep this to be backwards compatible */
+	{"allowed", JANUS_JSON_ARRAY, 0},
+	{"sampling_rate", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"sampling", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},	/* We keep this to be backwards compatible */
 	{"record", JANUS_JSON_BOOL, 0},
-	{"record_file", JSON_STRING, 0},
+	{"record_file", JANUS_JSON_STRING, 0},
 	{"permanent", JANUS_JSON_BOOL, 0},
 	{"audiolevel_ext", JANUS_JSON_BOOL, 0},
 	{"audiolevel_event", JANUS_JSON_BOOL, 0},
-	{"audio_active_packets", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"audio_level_average", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"audio_active_packets", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audio_level_average", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter edit_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"new_description", JSON_STRING, 0},
-	{"new_secret", JSON_STRING, 0},
-	{"new_pin", JSON_STRING, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"new_description", JANUS_JSON_STRING, 0},
+	{"new_secret", JANUS_JSON_STRING, 0},
+	{"new_pin", JANUS_JSON_STRING, 0},
 	{"new_is_private", JANUS_JSON_BOOL, 0},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter destroy_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter allowed_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"action", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"allowed", JSON_ARRAY, 0}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"action", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"allowed", JANUS_JSON_ARRAY, 0}
 };
 static struct janus_json_parameter kick_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter mute_parameters[] = {
 	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
@@ -784,40 +784,40 @@ static struct janus_json_parameter mute_parameters[] = {
 	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
 };
 static struct janus_json_parameter room_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter join_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"display", JSON_STRING, 0},
-	{"token", JSON_STRING, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"display", JANUS_JSON_STRING, 0},
+	{"token", JANUS_JSON_STRING, 0},
 	{"muted", JANUS_JSON_BOOL, 0},
-	{"quality", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"volume", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"quality", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"volume", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter configure_parameters[] = {
 	{"muted", JANUS_JSON_BOOL, 0},
-	{"quality", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"volume", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"quality", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"volume", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"record", JANUS_JSON_BOOL, 0},
-	{"filename", JSON_STRING, 0},
-	{"display", JSON_STRING, 0},
+	{"filename", JANUS_JSON_STRING, 0},
+	{"display", JANUS_JSON_STRING, 0},
 	{"update", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter rtp_forward_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"ssrc", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"ptype", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"port", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"host", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"host_family", JSON_STRING, 0},
-	{"srtp_suite", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"srtp_crypto", JSON_STRING, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"ssrc", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"ptype", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"host", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"host_family", JANUS_JSON_STRING, 0},
+	{"srtp_suite", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"srtp_crypto", JANUS_JSON_STRING, 0},
 	{"always_on", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter stop_rtp_forward_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"stream_id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"stream_id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 
 /* Static configuration instance */
@@ -1557,6 +1557,7 @@ int janus_audiobridge_init(janus_callbacks *callback, const char *config_path) {
 				/* FIXME We should clear some resources... */
 				janus_refcount_decrease(&audiobridge->ref);
 				JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the mixer thread...\n", error->code, error->message ? error->message : "??");
+				g_error_free(error);
 			} else {
 				janus_mutex_lock(&rooms_mutex);
 				g_hash_table_insert(rooms, janus_uint64_dup(audiobridge->room_id), audiobridge);
@@ -1588,6 +1589,7 @@ int janus_audiobridge_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the AudioBridge handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		janus_config_destroy(config);
 		return -1;
 	}
@@ -1970,6 +1972,7 @@ static json_t *janus_audiobridge_process_synchronous_request(janus_audiobridge_s
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the mixer thread...\n", error->code, error->message ? error->message : "??");
 			error_code = JANUS_AUDIOBRIDGE_ERROR_UNKNOWN_ERROR;
 			g_snprintf(error_cause, 512, "Got error %d (%s) trying to launch the mixer thread", error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			janus_refcount_decrease(&audiobridge->ref);
 			g_hash_table_remove(rooms, &audiobridge->room_id);
 			janus_mutex_unlock(&rooms_mutex);
@@ -3167,7 +3170,7 @@ void janus_audiobridge_setup_media(janus_plugin_session *handle) {
 	janus_mutex_unlock(&rooms_mutex);
 }
 
-void janus_audiobridge_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_audiobridge_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	janus_audiobridge_session *session = (janus_audiobridge_session *)handle->plugin_handle;
@@ -3408,7 +3411,7 @@ void janus_audiobridge_incoming_rtp(janus_plugin_session *handle, int video, cha
 	}
 }
 
-void janus_audiobridge_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_audiobridge_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	/* FIXME Should we care? */
@@ -3784,6 +3787,7 @@ static void *janus_audiobridge_handler(void *data) {
 					janus_refcount_decrease(&session->ref);
 					/* FIXME We should fail here... */
 					JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the participant thread...\n", error->code, error->message ? error->message : "??");
+					g_error_free(error);
 				}
 			}
 
@@ -4417,7 +4421,7 @@ static void *janus_audiobridge_handler(void *data) {
 			}
 			/* What is the Opus payload type? */
 			janus_audiobridge_participant *participant = (janus_audiobridge_participant *)session->participant;
-			participant->opus_pt = janus_sdp_get_codec_pt(offer, "opus");
+			participant->opus_pt = janus_sdp_get_codec_pt(offer, -1, "opus");
 			if(participant->opus_pt < 0) {
 				/* TODO Handle this case */
 				JANUS_LOG(LOG_ERR, "Offer doesn't contain Opus..?\n");
@@ -4447,12 +4451,21 @@ static void *janus_audiobridge_handler(void *data) {
 				}
 				temp = temp->next;
 			}
-			janus_sdp *answer = janus_sdp_generate_answer(offer,
-				/* Reject video and data channels, if offered */
-				JANUS_SDP_OA_VIDEO, FALSE,
-				JANUS_SDP_OA_DATA, FALSE,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_AUDIO_LEVEL,
-				JANUS_SDP_OA_DONE);
+			janus_sdp *answer = janus_sdp_generate_answer(offer);
+			/* Only accept the first audio line, and reject everything else if offered */
+			temp = offer->m_lines;
+			gboolean accepted = FALSE;
+			while(temp) {
+				janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+				if(m->type == JANUS_SDP_AUDIO && !accepted) {
+					accepted = TRUE;
+					janus_sdp_generate_answer_mline(offer, answer, m,
+						JANUS_SDP_OA_MLINE, JANUS_SDP_AUDIO,
+						JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_AUDIO_LEVEL,
+						JANUS_SDP_OA_DONE);
+				}
+				temp = temp->next;
+			}
 			/* Replace the session name */
 			g_free(answer->s_name);
 			char s_name[100];
@@ -4933,9 +4946,9 @@ static void janus_audiobridge_relay_rtp_packet(gpointer data, gpointer user_data
 	/* Set the payload type */
 	packet->data->type = participant->opus_pt;
 	/* Fix sequence number and timestamp (room switching may be involved) */
-	janus_rtp_header_update(packet->data, &participant->context, FALSE, OPUS_SAMPLES);
+	janus_rtp_header_update(packet->data, &participant->context, FALSE);
 	if(gateway != NULL)
-		gateway->relay_rtp(session->handle, 0, (char *)packet->data, packet->length);
+		gateway->relay_rtp(session->handle, -1, FALSE, (char *)packet->data, packet->length);
 	/* Restore the timestamp and sequence number to what the publisher set them to */
 	packet->data->timestamp = htonl(packet->timestamp);
 	packet->data->seq_number = htons(packet->seq_number);
diff --git a/plugins/janus_duktape.c b/plugins/janus_duktape.c
index 5b899adcf6..21a9ce5c0a 100644
--- a/plugins/janus_duktape.c
+++ b/plugins/janus_duktape.c
@@ -191,8 +191,8 @@ timeCallback("resumeScheduler", null, 0);
 
 
 /* Plugin information */
-#define JANUS_DUKTAPE_VERSION			1
-#define JANUS_DUKTAPE_VERSION_STRING	"0.0.1"
+#define JANUS_DUKTAPE_VERSION			2
+#define JANUS_DUKTAPE_VERSION_STRING	"0.0.2"
 #define JANUS_DUKTAPE_DESCRIPTION		"A custom plugin for implementing the logic in JavaScript, via Duktape."
 #define JANUS_DUKTAPE_NAME				"Janus JavaScript plugin (Duktape)"
 #define JANUS_DUKTAPE_AUTHOR			"Meetecho s.r.l."
@@ -213,10 +213,10 @@ void janus_duktape_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_duktape_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 json_t *janus_duktape_handle_admin_message(json_t *message);
 void janus_duktape_setup_media(janus_plugin_session *handle);
-void janus_duktape_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_duktape_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_duktape_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_duktape_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_duktape_incoming_data(janus_plugin_session *handle, char *label, char *buf, int len);
-void janus_duktape_slow_link(janus_plugin_session *handle, int uplink, int video);
+void janus_duktape_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink);
 void janus_duktape_hangup_media(janus_plugin_session *handle);
 void janus_duktape_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_duktape_query_session(janus_plugin_session *handle);
@@ -562,6 +562,7 @@ static duk_ret_t janus_duktape_method_pushevent(duk_context *ctx) {
 		if(error != NULL) {
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Duktape pushevent thread...\n",
 				error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			json_decref(event);
 			json_decref(jsep);
 			g_free(asev->transaction);
@@ -868,7 +869,7 @@ static duk_ret_t janus_duktape_method_setbitrate(duk_context *ctx) {
 	if(session->bitrate > 0 && g_atomic_int_get(&session->started)) {
 		char rtcpbuf[24];
 		janus_rtcp_remb((char *)(&rtcpbuf), 24, session->bitrate);
-		janus_core->relay_rtcp(session->handle, 1, rtcpbuf, 24);
+		janus_core->relay_rtcp(session->handle, -1, TRUE, rtcpbuf, 24);
 	}
 	/* Done */
 	janus_refcount_decrease(&session->ref);
@@ -927,7 +928,7 @@ static duk_ret_t janus_duktape_method_sendpli(duk_context *ctx) {
 	char rtcpbuf[12];
 	janus_rtcp_pli((char *)&rtcpbuf, 12);
 	JANUS_LOG(LOG_HUGE, "Sending PLI to session %"SCNu32"\n", session->id);
-	janus_core->relay_rtcp(session->handle, 1, rtcpbuf, 12);
+	janus_core->relay_rtcp(session->handle, -1, TRUE, rtcpbuf, 12);
 	/* Done */
 	janus_refcount_decrease(&session->ref);
 	duk_push_int(ctx, 0);
@@ -974,7 +975,7 @@ static duk_ret_t janus_duktape_method_relayrtp(duk_context *ctx) {
 	}
 	janus_mutex_unlock(&duktape_sessions_mutex);
 	/* Send the RTP packet */
-	janus_core->relay_rtp(session->handle, is_video, (char *)payload, len);
+	janus_core->relay_rtp(session->handle, -1, is_video, (char *)payload, len);
 	duk_push_int(ctx, 0);
 	return 1;
 }
@@ -1019,7 +1020,7 @@ static duk_ret_t janus_duktape_method_relayrtcp(duk_context *ctx) {
 	}
 	janus_mutex_unlock(&duktape_sessions_mutex);
 	/* Send the RTCP packet */
-	janus_core->relay_rtcp(session->handle, is_video, (char *)payload, len);
+	janus_core->relay_rtcp(session->handle, -1, is_video, (char *)payload, len);
 	duk_push_int(ctx, 0);
 	return 1;
 }
@@ -1134,7 +1135,7 @@ static duk_ret_t janus_duktape_method_startrecording(duk_context *ctx) {
 		char rtcpbuf[12];
 		janus_rtcp_pli((char *)&rtcpbuf, 12);
 		JANUS_LOG(LOG_HUGE, "Sending PLI to session %"SCNu32"\n", session->id);
-		janus_core->relay_rtcp(session->handle, 1, rtcpbuf, 12);
+		janus_core->relay_rtcp(session->handle, -1, TRUE, rtcpbuf, 12);
 	}
 	if(drc) {
 		session->drc = drc;
@@ -1414,6 +1415,7 @@ int janus_duktape_init(janus_callbacks *callback, const char *config_path) {
 		g_atomic_int_set(&duktape_initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Duktape scheduler thread...\n",
 			error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		duk_destroy_heap(duktape_ctx);
 		g_free(duktape_folder);
 		g_free(duktape_file);
@@ -1757,7 +1759,8 @@ void janus_duktape_create_session(janus_plugin_session *handle, int *error) {
 	janus_duktape_session *session = (janus_duktape_session *)g_malloc0(sizeof(janus_duktape_session));
 	session->handle = handle;
 	session->id = id;
-	janus_rtp_switching_context_reset(&session->rtpctx);
+	janus_rtp_switching_context_reset(&session->artpctx);
+	janus_rtp_switching_context_reset(&session->vrtpctx);
 	g_atomic_int_set(&session->hangingup, 0);
 	g_atomic_int_set(&session->destroyed, 0);
 	janus_refcount_init(&session->ref, janus_duktape_session_free);
@@ -2040,7 +2043,7 @@ void janus_duktape_setup_media(janus_plugin_session *handle) {
 	janus_refcount_decrease(&session->ref);
 }
 
-void janus_duktape_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_duktape_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&duktape_stopping) || !g_atomic_int_get(&duktape_initialized))
 		return;
 	janus_duktape_session *session = (janus_duktape_session *)handle->plugin_handle;
@@ -2099,12 +2102,12 @@ void janus_duktape_incoming_rtp(janus_plugin_session *handle, int video, char *b
 			char rtcpbuf[12];
 			janus_rtcp_pli((char *)&rtcpbuf, 12);
 			JANUS_LOG(LOG_HUGE, "Sending PLI to session %"SCNu32"\n", session->id);
-			janus_core->relay_rtcp(handle, 1, rtcpbuf, 12);
+			janus_core->relay_rtcp(handle, -1, TRUE, rtcpbuf, 12);
 		}
 	}
 }
 
-void janus_duktape_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_duktape_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&duktape_stopping) || !g_atomic_int_get(&duktape_initialized))
 		return;
 	janus_duktape_session *session = (janus_duktape_session *)handle->plugin_handle;
@@ -2141,9 +2144,9 @@ void janus_duktape_incoming_rtcp(janus_plugin_session *handle, int video, char *
 		if(session->bitrate > 0) {
 			char rtcpbuf[24];
 			janus_rtcp_remb((char *)(&rtcpbuf), 24, session->bitrate);
-			janus_core->relay_rtcp(handle, 1, rtcpbuf, 24);
+			janus_core->relay_rtcp(handle, -1, TRUE, rtcpbuf, 24);
 		} else {
-			janus_core->relay_rtcp(handle, 1, buf, len);
+			janus_core->relay_rtcp(handle, -1, TRUE, buf, len);
 		}
 	}
 	/* If there's an incoming PLI, instead, relay it to the source of the media if any */
@@ -2155,7 +2158,7 @@ void janus_duktape_incoming_rtcp(janus_plugin_session *handle, int video, char *
 			char rtcpbuf[12];
 			janus_rtcp_pli((char *)&rtcpbuf, 12);
 			JANUS_LOG(LOG_HUGE, "Sending PLI to session %"SCNu32"\n", session->sender->id);
-			janus_core->relay_rtcp(session->sender->handle, 1, rtcpbuf, 12);
+			janus_core->relay_rtcp(session->sender->handle, -1, TRUE, rtcpbuf, 12);
 			janus_mutex_unlock_nodebug(&session->sender->recipients_mutex);
 		}
 	}
@@ -2212,7 +2215,7 @@ void janus_duktape_incoming_data(janus_plugin_session *handle, char *label, char
 	g_free(text);
 }
 
-void janus_duktape_slow_link(janus_plugin_session *handle, int uplink, int video) {
+void janus_duktape_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&duktape_stopping) || !g_atomic_int_get(&duktape_initialized))
 		return;
 	janus_mutex_lock(&duktape_sessions_mutex);
@@ -2281,7 +2284,8 @@ void janus_duktape_hangup_media(janus_plugin_session *handle) {
 	session->bitrate = 0;
 	session->pli_freq = 0;
 	session->pli_latest = 0;
-	janus_rtp_switching_context_reset(&session->rtpctx);
+	janus_rtp_switching_context_reset(&session->artpctx);
+	janus_rtp_switching_context_reset(&session->vrtpctx);
 
 	/* Get rid of the recipients */
 	janus_mutex_lock(&session->recipients_mutex);
@@ -2329,10 +2333,10 @@ static void janus_duktape_relay_rtp_packet(gpointer data, gpointer user_data) {
 		return;
 	}
 	/* Fix sequence number and timestamp (publisher switching may be involved) */
-	janus_rtp_header_update(packet->data, &session->rtpctx, packet->is_video, packet->is_video ? 4500 : 960);
+	janus_rtp_header_update(packet->data, video ? &session->vrtpctx : &session->artpctx, packet->is_video);
 	/* Send the packet */
 	if(janus_core != NULL)
-		janus_core->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+		janus_core->relay_rtp(session->handle, -1, packet->is_video, (char *)packet->data, packet->length);
 	/* Restore the timestamp and sequence number to what the publisher set them to */
 	packet->data->timestamp = htonl(packet->timestamp);
 	packet->data->seq_number = htons(packet->seq_number);
diff --git a/plugins/janus_duktape_data.h b/plugins/janus_duktape_data.h
index fd77c1d77d..aa31516b68 100644
--- a/plugins/janus_duktape_data.h
+++ b/plugins/janus_duktape_data.h
@@ -58,7 +58,7 @@ typedef struct janus_duktape_session {
 	gboolean send_audio;				/* Whether outgoing audio can be sent or must be dropped */
 	gboolean send_video;				/* Whether outgoing video can be sent or must be dropped */
 	gboolean send_data;					/* Whether outgoing data can be sent or must be dropped */
-	janus_rtp_switching_context rtpctx;	/* Needed in case the source changes (e.g., stale operator/customer) */
+	janus_rtp_switching_context artpctx, vrtpctx;
 	uint32_t bitrate;					/* Bitrate limit */
 	uint16_t pli_freq;					/* Regular PLI frequency (0=disabled) */
 	gint64 pli_latest;					/* Time of latest sent PLI (to avoid flooding) */
diff --git a/plugins/janus_echotest.c b/plugins/janus_echotest.c
index ed8b1c5f7f..3d7e67d3d2 100644
--- a/plugins/janus_echotest.c
+++ b/plugins/janus_echotest.c
@@ -120,8 +120,8 @@
 
 
 /* Plugin information */
-#define JANUS_ECHOTEST_VERSION			7
-#define JANUS_ECHOTEST_VERSION_STRING	"0.0.7"
+#define JANUS_ECHOTEST_VERSION			8
+#define JANUS_ECHOTEST_VERSION_STRING	"0.0.8"
 #define JANUS_ECHOTEST_DESCRIPTION		"This is a trivial EchoTest plugin for Janus, just used to showcase the plugin interface."
 #define JANUS_ECHOTEST_NAME				"JANUS EchoTest plugin"
 #define JANUS_ECHOTEST_AUTHOR			"Meetecho s.r.l."
@@ -142,10 +142,10 @@ void janus_echotest_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_echotest_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 json_t *janus_echotest_handle_admin_message(json_t *message);
 void janus_echotest_setup_media(janus_plugin_session *handle);
-void janus_echotest_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_echotest_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_echotest_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_echotest_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_echotest_incoming_data(janus_plugin_session *handle, char *label, char *buf, int len);
-void janus_echotest_slow_link(janus_plugin_session *handle, int uplink, int video);
+void janus_echotest_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink);
 void janus_echotest_hangup_media(janus_plugin_session *handle);
 void janus_echotest_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_echotest_query_session(janus_plugin_session *handle);
@@ -318,6 +318,7 @@ int janus_echotest_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the EchoTest handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_ECHOTEST_NAME);
@@ -476,8 +477,8 @@ json_t *janus_echotest_query_session(janus_plugin_session *handle) {
 		json_object_set_new(info, "recording", recording);
 	}
 	json_object_set_new(info, "slowlink_count", json_integer(session->slowlink_count));
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -531,7 +532,7 @@ void janus_echotest_setup_media(janus_plugin_session *handle) {
 	/* We really don't care, as we only send RTP/RTCP we get in the first place back anyway */
 }
 
-void janus_echotest_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_echotest_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	/* Simple echo test */
@@ -555,10 +556,11 @@ void janus_echotest_incoming_rtp(janus_plugin_session *handle, int video, char *
 				buf, len, session->ssrc, session->rid, session->vcodec, &session->context);
 			if(session->sim_context.need_pli) {
 				/* Send a PLI */
+				JANUS_LOG(LOG_VERB, "We need a PLI for the simulcast context\n");
 				char rtcpbuf[12];
 				memset(rtcpbuf, 0, 12);
 				janus_rtcp_pli((char *)&rtcpbuf, 12);
-				gateway->relay_rtcp(handle, 1, rtcpbuf, 12);
+				gateway->relay_rtcp(handle, -1, TRUE, rtcpbuf, 12);
 			}
 			/* Do we need to drop this? */
 			if(!relay)
@@ -583,7 +585,7 @@ void janus_echotest_incoming_rtp(janus_plugin_session *handle, int video, char *
 				json_decref(event);
 			}
 			/* If we got here, update the RTP header and send the packet */
-			janus_rtp_header_update(header, &session->context, TRUE, 0);
+			janus_rtp_header_update(header, &session->context, TRUE);
 			if(session->vcodec == JANUS_VIDEOCODEC_VP8) {
 				int plen = 0;
 				char *payload = janus_rtp_payload(buf, len, &plen);
@@ -593,7 +595,7 @@ void janus_echotest_incoming_rtp(janus_plugin_session *handle, int video, char *
 			header->ssrc = htonl(1);
 			janus_recorder_save_frame(session->vrc, buf, len);
 			/* Send the frame back */
-			gateway->relay_rtp(handle, video, buf, len);
+			gateway->relay_rtp(handle, mindex, video, buf, len);
 			/* Restore header or core statistics will be messed up */
 			header->ssrc = htonl(ssrc);
 			header->timestamp = htonl(timestamp);
@@ -603,13 +605,13 @@ void janus_echotest_incoming_rtp(janus_plugin_session *handle, int video, char *
 				/* Save the frame if we're recording */
 				janus_recorder_save_frame(video ? session->vrc : session->arc, buf, len);
 				/* Send the frame back */
-				gateway->relay_rtp(handle, video, buf, len);
+				gateway->relay_rtp(handle, mindex, video, buf, len);
 			}
 		}
 	}
 }
 
-void janus_echotest_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_echotest_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	/* Simple echo test */
@@ -629,10 +631,10 @@ void janus_echotest_incoming_rtcp(janus_plugin_session *handle, int video, char
 				janus_rtcp_cap_remb(buf, len, 10000000);
 			else
 				janus_rtcp_cap_remb(buf, len, session->bitrate);
-			gateway->relay_rtcp(handle, 1, buf, len);
+			gateway->relay_rtcp(handle, -1, TRUE, buf, len);
 			return;
 		}
-		gateway->relay_rtcp(handle, video, buf, len);
+		gateway->relay_rtcp(handle, mindex, video, buf, len);
 	}
 }
 
@@ -666,7 +668,7 @@ void janus_echotest_incoming_data(janus_plugin_session *handle, char *label, cha
 	}
 }
 
-void janus_echotest_slow_link(janus_plugin_session *handle, int uplink, int video) {
+void janus_echotest_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink) {
 	/* The core is informing us that our peer got or sent too many NACKs, are we pushing media too hard? */
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
@@ -835,14 +837,21 @@ static void *janus_echotest_handler(void *data) {
 		const char *msg_sdp_type = json_string_value(json_object_get(msg->jsep, "type"));
 		const char *msg_sdp = json_string_value(json_object_get(msg->jsep, "sdp"));
 		json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
-		if(msg_simulcast) {
-			JANUS_LOG(LOG_VERB, "EchoTest client is going to do simulcasting\n");
-			int rid_ext_id = -1, framemarking_ext_id = -1;
-			janus_rtp_simulcasting_prepare(msg_simulcast, &rid_ext_id, &framemarking_ext_id, session->ssrc, session->rid);
-			session->sim_context.rid_ext_id = rid_ext_id;
-			session->sim_context.framemarking_ext_id = framemarking_ext_id;
-			session->sim_context.substream_target = 2;	/* Let's aim for the highest quality */
-			session->sim_context.templayer_target = 2;	/* Let's aim for all temporal layers */
+		if(msg_simulcast && json_array_size(msg_simulcast) > 0) {
+			size_t i = 0;
+			for(i=0; i<json_array_size(msg_simulcast); i++) {
+				json_t *s = json_array_get(msg_simulcast, i);
+				int mindex = json_integer_value(json_object_get(s, "mindex"));
+				JANUS_LOG(LOG_WARN, "EchoTest client is going to do simulcasting (#%d)\n", mindex);
+				int rid_ext_id = -1, framemarking_ext_id = -1;
+				janus_rtp_simulcasting_prepare(s, &rid_ext_id, &framemarking_ext_id, session->ssrc, session->rid);
+				session->sim_context.rid_ext_id = rid_ext_id;
+				session->sim_context.framemarking_ext_id = framemarking_ext_id;
+				session->sim_context.substream_target = 2;	/* Let's aim for the highest quality */
+				session->sim_context.templayer_target = 2;	/* Let's aim for all temporal layers */
+				/* FIXME We're stopping at the first item, there may be more */
+				break;
+			}
 		}
 		json_t *audio = json_object_get(root, "audio");
 		if(audio && !json_is_boolean(audio)) {
@@ -919,7 +928,7 @@ static void *janus_echotest_handler(void *data) {
 				char buf[12];
 				memset(buf, 0, 12);
 				janus_rtcp_pli((char *)&buf, 12);
-				gateway->relay_rtcp(session->handle, 1, buf, 12);
+				gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 			}
 			session->video_active = json_is_true(video);
 			JANUS_LOG(LOG_VERB, "Setting video property: %s\n", session->video_active ? "true" : "false");
@@ -931,7 +940,7 @@ static void *janus_echotest_handler(void *data) {
 				char buf[24];
 				janus_rtcp_remb((char *)&buf, 24, session->bitrate);
 				JANUS_LOG(LOG_VERB, "Sending REMB\n");
-				gateway->relay_rtcp(session->handle, 1, buf, 24);
+				gateway->relay_rtcp(session->handle, -1, TRUE, buf, 24);
 			}
 		}
 		if(substream) {
@@ -952,7 +961,7 @@ static void *janus_echotest_handler(void *data) {
 				char buf[12];
 				memset(buf, 0, 12);
 				janus_rtcp_pli((char *)&buf, 12);
-				gateway->relay_rtcp(session->handle, 1, buf, 12);
+				gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 			}
 		}
 		if(temporal) {
@@ -973,7 +982,7 @@ static void *janus_echotest_handler(void *data) {
 				char buf[12];
 				memset(buf, 0, 12);
 				janus_rtcp_pli((char *)&buf, 12);
-				gateway->relay_rtcp(session->handle, 1, buf, 12);
+				gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 			}
 		}
 
@@ -1034,16 +1043,23 @@ static void *janus_echotest_handler(void *data) {
 				}
 				temp = temp->next;
 			}
-			janus_sdp *answer = janus_sdp_generate_answer(offer,
-				JANUS_SDP_OA_AUDIO_CODEC, json_string_value(audiocodec),
-				JANUS_SDP_OA_AUDIO_FMTP, opus_fec ? "useinbandfec=1" : NULL,
-				JANUS_SDP_OA_VIDEO_CODEC, json_string_value(videocodec),
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_RID,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_REPAIRED_RID,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_FRAME_MARKING,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC,
-				JANUS_SDP_OA_DONE);
+			janus_sdp *answer = janus_sdp_generate_answer(offer);
+			temp = offer->m_lines;
+			while(temp) {
+				janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+				janus_sdp_generate_answer_mline(offer, answer, m,
+					JANUS_SDP_OA_MLINE, m->type,
+					JANUS_SDP_OA_CODEC, (m->type == JANUS_SDP_AUDIO ? json_string_value(audiocodec) :
+						(m->type == JANUS_SDP_VIDEO ? json_string_value(videocodec) : NULL)),
+					JANUS_SDP_OA_FMTP, (m->type == JANUS_SDP_AUDIO && opus_fec ? "useinbandfec=1" : NULL),
+					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
+					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_RID,
+					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_REPAIRED_RID,
+					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_FRAME_MARKING,
+					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC,
+					JANUS_SDP_OA_DONE);
+				temp = temp->next;
+			}
 			/* If we ended up sendonly, switch to inactive (as we don't really send anything ourselves) */
 			janus_sdp_mline *m = janus_sdp_mline_find(answer, JANUS_SDP_AUDIO);
 			if(m && m->direction == JANUS_SDP_SENDONLY)
@@ -1053,9 +1069,10 @@ static void *janus_echotest_handler(void *data) {
 				m->direction = JANUS_SDP_INACTIVE;
 			/* Check which codecs we ended up with */
 			const char *acodec = NULL, *vcodec = NULL;
-			janus_sdp_find_first_codecs(answer, &acodec, &vcodec);
+			janus_sdp_find_first_codecs(answer, JANUS_SDP_AUDIO, -1, &acodec);
 			if(acodec)
 				session->acodec = janus_audiocodec_from_name(acodec);
+			janus_sdp_find_first_codecs(answer, JANUS_SDP_VIDEO, -1, &vcodec);
 			if(vcodec)
 				session->vcodec = janus_videocodec_from_name(vcodec);
 			session->has_audio = session->acodec != JANUS_AUDIOCODEC_NONE;
@@ -1142,7 +1159,7 @@ static void *janus_echotest_handler(void *data) {
 					char buf[12];
 					memset(buf, 0, 12);
 					janus_rtcp_pli((char *)&buf, 12);
-					gateway->relay_rtcp(session->handle, 1, buf, 12);
+					gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 				}
 				if(session->has_data) {
 					memset(filename, 0, 255);
diff --git a/plugins/janus_lua.c b/plugins/janus_lua.c
index ef576ede69..8eefd11341 100644
--- a/plugins/janus_lua.c
+++ b/plugins/janus_lua.c
@@ -192,8 +192,8 @@ timeCallback("resumeScheduler", nil, 0)
 
 
 /* Plugin information */
-#define JANUS_LUA_VERSION			1
-#define JANUS_LUA_VERSION_STRING	"0.0.1"
+#define JANUS_LUA_VERSION			2
+#define JANUS_LUA_VERSION_STRING	"0.0.2"
 #define JANUS_LUA_DESCRIPTION		"A custom plugin for the Lua framework."
 #define JANUS_LUA_NAME				"Janus Lua plugin"
 #define JANUS_LUA_AUTHOR			"Meetecho s.r.l."
@@ -214,10 +214,10 @@ void janus_lua_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_lua_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 json_t *janus_lua_handle_admin_message(json_t *message);
 void janus_lua_setup_media(janus_plugin_session *handle);
-void janus_lua_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_lua_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_lua_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_lua_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_lua_incoming_data(janus_plugin_session *handle, char *label, char *buf, int len);
-void janus_lua_slow_link(janus_plugin_session *handle, int uplink, int video);
+void janus_lua_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink);
 void janus_lua_hangup_media(janus_plugin_session *handle);
 void janus_lua_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_lua_query_session(janus_plugin_session *handle);
@@ -495,6 +495,7 @@ static int janus_lua_method_pushevent(lua_State *s) {
 		if(error != NULL) {
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Lua pushevent thread...\n",
 				error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			json_decref(event);
 			json_decref(jsep);
 			g_free(asev->transaction);
@@ -781,7 +782,7 @@ static int janus_lua_method_setbitrate(lua_State *s) {
 	if(session->bitrate > 0 && g_atomic_int_get(&session->started)) {
 		char rtcpbuf[24];
 		janus_rtcp_remb((char *)(&rtcpbuf), 24, session->bitrate);
-		janus_core->relay_rtcp(session->handle, 1, rtcpbuf, 24);
+		janus_core->relay_rtcp(session->handle, -1, TRUE, rtcpbuf, 24);
 	}
 	/* Done */
 	janus_refcount_decrease(&session->ref);
@@ -840,7 +841,7 @@ static int janus_lua_method_sendpli(lua_State *s) {
 	char rtcpbuf[12];
 	janus_rtcp_pli((char *)&rtcpbuf, 12);
 	JANUS_LOG(LOG_HUGE, "Sending PLI to session %"SCNu32"\n", session->id);
-	janus_core->relay_rtcp(session->handle, 1, rtcpbuf, 12);
+	janus_core->relay_rtcp(session->handle, -1, TRUE, rtcpbuf, 12);
 	/* Done */
 	janus_refcount_decrease(&session->ref);
 	lua_pushnumber(s, 0);
@@ -874,7 +875,7 @@ static int janus_lua_method_relayrtp(lua_State *s) {
 	}
 	janus_mutex_unlock(&lua_sessions_mutex);
 	/* Send the RTP packet */
-	janus_core->relay_rtp(session->handle, is_video, (char *)payload, len);
+	janus_core->relay_rtp(session->handle, -1, is_video, (char *)payload, len);
 	lua_pushnumber(s, 0);
 	return 1;
 }
@@ -906,7 +907,7 @@ static int janus_lua_method_relayrtcp(lua_State *s) {
 	}
 	janus_mutex_unlock(&lua_sessions_mutex);
 	/* Send the RTCP packet */
-	janus_core->relay_rtcp(session->handle, is_video, (char *)payload, len);
+	janus_core->relay_rtcp(session->handle, -1, is_video, (char *)payload, len);
 	lua_pushnumber(s, 0);
 	return 1;
 }
@@ -1012,7 +1013,7 @@ static int janus_lua_method_startrecording(lua_State *s) {
 		char rtcpbuf[12];
 		janus_rtcp_pli((char *)&rtcpbuf, 12);
 		JANUS_LOG(LOG_HUGE, "Sending PLI to session %"SCNu32"\n", session->id);
-		janus_core->relay_rtcp(session->handle, 1, rtcpbuf, 12);
+		janus_core->relay_rtcp(session->handle, -1, TRUE, rtcpbuf, 12);
 	}
 	if(drc) {
 		session->drc = drc;
@@ -1253,6 +1254,7 @@ int janus_lua_init(janus_callbacks *callback, const char *config_path) {
 		g_atomic_int_set(&lua_initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Lua scheduler thread...\n",
 			error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		lua_close(lua_state);
 		g_free(lua_folder);
 		g_free(lua_file);
@@ -1517,7 +1519,8 @@ void janus_lua_create_session(janus_plugin_session *handle, int *error) {
 	janus_lua_session *session = (janus_lua_session *)g_malloc0(sizeof(janus_lua_session));
 	session->handle = handle;
 	session->id = id;
-	janus_rtp_switching_context_reset(&session->rtpctx);
+	janus_rtp_switching_context_reset(&session->artpctx);
+	janus_rtp_switching_context_reset(&session->vrtpctx);
 	g_atomic_int_set(&session->hangingup, 0);
 	g_atomic_int_set(&session->destroyed, 0);
 	janus_refcount_init(&session->ref, janus_lua_session_free);
@@ -1755,7 +1758,7 @@ void janus_lua_setup_media(janus_plugin_session *handle) {
 	janus_refcount_decrease(&session->ref);
 }
 
-void janus_lua_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_lua_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&lua_stopping) || !g_atomic_int_get(&lua_initialized))
 		return;
 	janus_lua_session *session = (janus_lua_session *)handle->plugin_handle;
@@ -1808,12 +1811,12 @@ void janus_lua_incoming_rtp(janus_plugin_session *handle, int video, char *buf,
 			char rtcpbuf[12];
 			janus_rtcp_pli((char *)&rtcpbuf, 12);
 			JANUS_LOG(LOG_HUGE, "Sending PLI to session %"SCNu32"\n", session->id);
-			janus_core->relay_rtcp(handle, 1, rtcpbuf, 12);
+			janus_core->relay_rtcp(handle, -1, TRUE, rtcpbuf, 12);
 		}
 	}
 }
 
-void janus_lua_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_lua_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&lua_stopping) || !g_atomic_int_get(&lua_initialized))
 		return;
 	janus_lua_session *session = (janus_lua_session *)handle->plugin_handle;
@@ -1844,9 +1847,9 @@ void janus_lua_incoming_rtcp(janus_plugin_session *handle, int video, char *buf,
 		if(session->bitrate > 0) {
 			char rtcpbuf[24];
 			janus_rtcp_remb((char *)(&rtcpbuf), 24, session->bitrate);
-			janus_core->relay_rtcp(handle, 1, rtcpbuf, 24);
+			janus_core->relay_rtcp(handle, -1, TRUE, rtcpbuf, 24);
 		} else {
-			janus_core->relay_rtcp(handle, 1, buf, len);
+			janus_core->relay_rtcp(handle, -1, TRUE, buf, len);
 		}
 	}
 	/* If there's an incoming PLI, instead, relay it to the source of the media if any */
@@ -1858,7 +1861,7 @@ void janus_lua_incoming_rtcp(janus_plugin_session *handle, int video, char *buf,
 			char rtcpbuf[12];
 			janus_rtcp_pli((char *)&rtcpbuf, 12);
 			JANUS_LOG(LOG_HUGE, "Sending PLI to session %"SCNu32"\n", session->sender->id);
-			janus_core->relay_rtcp(session->sender->handle, 1, rtcpbuf, 12);
+			janus_core->relay_rtcp(session->sender->handle, -1, TRUE, rtcpbuf, 12);
 			janus_mutex_unlock_nodebug(&session->sender->recipients_mutex);
 		}
 	}
@@ -1909,7 +1912,7 @@ void janus_lua_incoming_data(janus_plugin_session *handle, char *label, char *bu
 	g_free(text);
 }
 
-void janus_lua_slow_link(janus_plugin_session *handle, int uplink, int video) {
+void janus_lua_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&lua_stopping) || !g_atomic_int_get(&lua_initialized))
 		return;
 	janus_mutex_lock(&lua_sessions_mutex);
@@ -1972,7 +1975,8 @@ void janus_lua_hangup_media(janus_plugin_session *handle) {
 	session->bitrate = 0;
 	session->pli_freq = 0;
 	session->pli_latest = 0;
-	janus_rtp_switching_context_reset(&session->rtpctx);
+	janus_rtp_switching_context_reset(&session->artpctx);
+	janus_rtp_switching_context_reset(&session->vrtpctx);
 
 	/* Get rid of the recipients */
 	janus_mutex_lock(&session->recipients_mutex);
@@ -2014,10 +2018,10 @@ static void janus_lua_relay_rtp_packet(gpointer data, gpointer user_data) {
 		return;
 	}
 	/* Fix sequence number and timestamp (publisher switching may be involved) */
-	janus_rtp_header_update(packet->data, &session->rtpctx, packet->is_video, packet->is_video ? 4500 : 960);
+	janus_rtp_header_update(packet->data, video ? &session->vrtpctx : &session->artpctx, packet->is_video);
 	/* Send the packet */
 	if(janus_core != NULL)
-		janus_core->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+		janus_core->relay_rtp(session->handle, -1, packet->is_video, (char *)packet->data, packet->length);
 	/* Restore the timestamp and sequence number to what the publisher set them to */
 	packet->data->timestamp = htonl(packet->timestamp);
 	packet->data->seq_number = htons(packet->seq_number);
diff --git a/plugins/janus_lua_data.h b/plugins/janus_lua_data.h
index dd5e9a13f6..5fc1996d5d 100644
--- a/plugins/janus_lua_data.h
+++ b/plugins/janus_lua_data.h
@@ -58,7 +58,7 @@ typedef struct janus_lua_session {
 	gboolean send_audio;				/* Whether outgoing audio can be sent or must be dropped */
 	gboolean send_video;				/* Whether outgoing video can be sent or must be dropped */
 	gboolean send_data;					/* Whether outgoing data can be sent or must be dropped */
-	janus_rtp_switching_context rtpctx;	/* Needed in case the source changes (e.g., stale operator/customer) */
+	janus_rtp_switching_context artpctx, vrtpctx;
 	uint32_t bitrate;					/* Bitrate limit */
 	uint16_t pli_freq;					/* Regular PLI frequency (0=disabled) */
 	gint64 pli_latest;					/* Time of latest sent PLI (to avoid flooding) */
diff --git a/plugins/janus_nosip.c b/plugins/janus_nosip.c
index 8277e65062..1897ca613f 100644
--- a/plugins/janus_nosip.c
+++ b/plugins/janus_nosip.c
@@ -173,8 +173,8 @@
 
 
 /* Plugin information */
-#define JANUS_NOSIP_VERSION			1
-#define JANUS_NOSIP_VERSION_STRING	"0.0.1"
+#define JANUS_NOSIP_VERSION			2
+#define JANUS_NOSIP_VERSION_STRING	"0.0.2"
 #define JANUS_NOSIP_DESCRIPTION		"This is a simple RTP bridging plugin that leaves signalling details (e.g., SIP) up to the application."
 #define JANUS_NOSIP_NAME			"JANUS NoSIP plugin"
 #define JANUS_NOSIP_AUTHOR			"Meetecho s.r.l."
@@ -194,8 +194,8 @@ const char *janus_nosip_get_package(void);
 void janus_nosip_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_nosip_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 void janus_nosip_setup_media(janus_plugin_session *handle);
-void janus_nosip_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_nosip_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_nosip_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_nosip_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_nosip_hangup_media(janus_plugin_session *handle);
 void janus_nosip_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_nosip_query_session(janus_plugin_session *handle);
@@ -232,29 +232,29 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter generate_parameters[] = {
-	{"info", JSON_STRING, 0},
-	{"srtp", JSON_STRING, 0},
-	{"srtp_profile", JSON_STRING, 0},
+	{"info", JANUS_JSON_STRING, 0},
+	{"srtp", JANUS_JSON_STRING, 0},
+	{"srtp_profile", JANUS_JSON_STRING, 0},
 	{"update", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter process_parameters[] = {
-	{"type", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"sdp", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"info", JSON_STRING, 0},
-	{"srtp", JSON_STRING, 0},
-	{"srtp_profile", JSON_STRING, 0},
+	{"type", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"sdp", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"info", JANUS_JSON_STRING, 0},
+	{"srtp", JANUS_JSON_STRING, 0},
+	{"srtp_profile", JANUS_JSON_STRING, 0},
 	{"update", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter recording_parameters[] = {
-	{"action", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"action", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"audio", JANUS_JSON_BOOL, 0},
 	{"video", JANUS_JSON_BOOL, 0},
 	{"peer_audio", JANUS_JSON_BOOL, 0},
 	{"peer_video", JANUS_JSON_BOOL, 0},
-	{"filename", JSON_STRING, 0}
+	{"filename", JANUS_JSON_STRING, 0}
 };
 
 /* Useful stuff */
@@ -310,7 +310,7 @@ typedef struct janus_nosip_media {
 	srtp_t video_srtp_in, video_srtp_out;
 	srtp_policy_t video_remote_policy, video_local_policy;
 	gboolean video_send;
-	janus_rtp_switching_context context;
+	janus_rtp_switching_context acontext, vcontext;
 	int pipefd[2];
 	gboolean updated;
 } janus_nosip_media;
@@ -602,7 +602,8 @@ void janus_nosip_media_reset(janus_nosip_session *session) {
 	session->media.video_pt = -1;
 	session->media.video_pt_name = NULL;	/* Immutable string, no need to free*/
 	session->media.video_send = TRUE;
-	janus_rtp_switching_context_reset(&session->media.context);
+	janus_rtp_switching_context_reset(&session->media.acontext);
+	janus_rtp_switching_context_reset(&session->media.vcontext);
 }
 
 
@@ -753,6 +754,7 @@ int janus_nosip_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the NoSIP handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_NOSIP_NAME);
@@ -863,7 +865,8 @@ void janus_nosip_create_session(janus_plugin_session *handle, int *error) {
 	session->media.video_pt_name = NULL;
 	session->media.video_send = TRUE;
 	/* Initialize the RTP context */
-	janus_rtp_switching_context_reset(&session->media.context);
+	janus_rtp_switching_context_reset(&session->media.acontext);
+	janus_rtp_switching_context_reset(&session->media.vcontext);
 	session->media.pipefd[0] = -1;
 	session->media.pipefd[1] = -1;
 	session->media.updated = FALSE;
@@ -937,8 +940,8 @@ json_t *janus_nosip_query_session(janus_plugin_session *handle) {
 			json_object_set_new(recording, "video-peer", json_string(session->vrc_peer->filename));
 		json_object_set_new(info, "recording", recording);
 	}
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -988,7 +991,7 @@ void janus_nosip_setup_media(janus_plugin_session *handle) {
 	janus_mutex_unlock(&sessions_mutex);
 }
 
-void janus_nosip_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_nosip_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	if(gateway) {
@@ -1065,7 +1068,7 @@ void janus_nosip_incoming_rtp(janus_plugin_session *handle, int video, char *buf
 	}
 }
 
-void janus_nosip_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_nosip_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	if(gateway) {
@@ -1401,9 +1404,16 @@ static void *janus_nosip_handler(void *data) {
 				json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
 				if(msg_simulcast) {
 					JANUS_LOG(LOG_WARN, "Client negotiated simulcasting which we don't do here, falling back to base substream...\n");
-					json_t *s = json_object_get(msg_simulcast, "ssrcs");
-					if(s && json_array_size(s) > 0)
-						session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+					size_t i = 0;
+					for(i=0; i<json_array_size(msg_simulcast); i++) {
+						json_t *sobj = json_array_get(msg_simulcast, i);
+						json_t *s = json_object_get(sobj, "ssrcs");
+						if(s && json_array_size(s) > 0)
+							session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+						session->media.simulcast_ssrc = json_integer_value(json_object_get(s, "ssrc-0"));
+						/* FIXME We're stopping at the first item, there may be more */
+						break;
+					}
 				}
 				/* Send the barebone SDP back */
 				result = json_object();
@@ -1473,6 +1483,7 @@ static void *janus_nosip_handler(void *data) {
 					session->media.ready = 0;
 					janus_refcount_decrease(&session->ref);
 					JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");
+					g_error_free(error);
 				}
 			}
 		} else if(!strcasecmp(request_text, "hangup")) {
@@ -1619,7 +1630,7 @@ static void *janus_nosip_handler(void *data) {
 						JANUS_LOG(LOG_VERB, "Recording video, sending a PLI to kickstart it\n");
 						char buf[12];
 						janus_rtcp_pli((char *)&buf, 12);
-						gateway->relay_rtcp(session->handle, 1, buf, 12);
+						gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 					}
 				}
 			} else {
@@ -2142,8 +2153,6 @@ static void *janus_nosip_relay_thread(void *data) {
 	/* Loop */
 	int num = 0;
 	gboolean goon = TRUE;
-	int astep = 0, vstep = 0;
-	guint32 ats = 0, vts = 0;
 
 	session->media.updated = TRUE; /* Connect UDP sockets upon loop entry */
 	gboolean have_audio_server_ip = TRUE;
@@ -2341,32 +2350,11 @@ static void *janus_nosip_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* Check if the SSRC changed (e.g., after a re-INVITE or UPDATE) */
-					guint32 timestamp = ntohl(header->timestamp);
-					janus_rtp_header_update(header, &session->media.context, video,
-						(video ? (vstep ? vstep : 4500) : (astep ? astep : 960)));
-					if(video) {
-						if(vts == 0) {
-							vts = timestamp;
-						} else if(vstep == 0) {
-							vstep = timestamp-vts;
-							if(vstep < 0) {
-								vstep = 0;
-							}
-						}
-					} else {
-						if(ats == 0) {
-							ats = timestamp;
-						} else if(astep == 0) {
-							astep = timestamp-ats;
-							if(astep < 0) {
-								astep = 0;
-							}
-						}
-					}
+					janus_rtp_header_update(header, video ? &session->media.vcontext : &session->media.acontext, video);
 					/* Save the frame if we're recording */
 					janus_recorder_save_frame(video ? session->vrc_peer : session->arc_peer, buffer, bytes);
 					/* Relay to browser */
-					gateway->relay_rtp(session->handle, video, buffer, bytes);
+					gateway->relay_rtp(session->handle, -1, video, buffer, bytes);
 					continue;
 				} else {
 					/* Audio or Video RTCP */
@@ -2387,7 +2375,7 @@ static void *janus_nosip_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* Relay to browser */
-					gateway->relay_rtcp(session->handle, video, buffer, bytes);
+					gateway->relay_rtcp(session->handle, -1, video, buffer, bytes);
 					continue;
 				}
 			}
diff --git a/plugins/janus_recordplay.c b/plugins/janus_recordplay.c
index eda808f57e..40d9933b8c 100644
--- a/plugins/janus_recordplay.c
+++ b/plugins/janus_recordplay.c
@@ -273,8 +273,8 @@
 
 
 /* Plugin information */
-#define JANUS_RECORDPLAY_VERSION			4
-#define JANUS_RECORDPLAY_VERSION_STRING		"0.0.4"
+#define JANUS_RECORDPLAY_VERSION			5
+#define JANUS_RECORDPLAY_VERSION_STRING		"0.0.5"
 #define JANUS_RECORDPLAY_DESCRIPTION		"This is a trivial Record&Play plugin for Janus, to record WebRTC sessions and replay them."
 #define JANUS_RECORDPLAY_NAME				"JANUS Record&Play plugin"
 #define JANUS_RECORDPLAY_AUTHOR				"Meetecho s.r.l."
@@ -295,9 +295,9 @@ void janus_recordplay_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_recordplay_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 json_t *janus_recordplay_handle_admin_message(json_t *message);
 void janus_recordplay_setup_media(janus_plugin_session *handle);
-void janus_recordplay_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_recordplay_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_recordplay_slow_link(janus_plugin_session *handle, int uplink, int video);
+void janus_recordplay_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_recordplay_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_recordplay_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink);
 void janus_recordplay_hangup_media(janus_plugin_session *handle);
 void janus_recordplay_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_recordplay_query_session(janus_plugin_session *handle);
@@ -336,20 +336,20 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter configure_parameters[] = {
-	{"video-bitrate-max", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video-keyframe-interval", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"video-bitrate-max", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video-keyframe-interval", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter record_parameters[] = {
-	{"name", JSON_STRING, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_NONEMPTY},
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"filename", JSON_STRING, 0},
+	{"name", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_NONEMPTY},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"filename", JANUS_JSON_STRING, 0},
 	{"update", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter play_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
 	{"restart", JANUS_JSON_BOOL, 0}
 };
 
@@ -473,7 +473,7 @@ void janus_recordplay_update_recordings_list(void);
 static void *janus_recordplay_playout_thread(void *data);
 
 /* Helper to send RTCP feedback back to recorders, if needed */
-void janus_recordplay_send_rtcp_feedback(janus_plugin_session *handle, int video, char *buf, int len);
+static void janus_recordplay_send_rtcp_feedback(janus_plugin_session *handle, gboolean video, char *buf, int len);
 
 /* To make things easier, we use static payload types for viewers (unless it's for G.711 or G.722) */
 #define AUDIO_PT		111
@@ -624,15 +624,16 @@ static int janus_recordplay_generate_offer(janus_recordplay_recording *rec) {
 	g_snprintf(s_name, sizeof(s_name), "Recording %"SCNu64, rec->id);
 	janus_sdp *offer = janus_sdp_generate_offer(
 		s_name, "1.1.1.1",
-		JANUS_SDP_OA_AUDIO, offer_audio,
-		JANUS_SDP_OA_AUDIO_CODEC, janus_audiocodec_name(rec->acodec),
-		JANUS_SDP_OA_AUDIO_PT, rec->audio_pt,
-		JANUS_SDP_OA_AUDIO_DIRECTION, JANUS_SDP_SENDONLY,
-		JANUS_SDP_OA_VIDEO, offer_video,
-		JANUS_SDP_OA_VIDEO_CODEC, janus_videocodec_name(rec->vcodec),
-		JANUS_SDP_OA_VIDEO_PT, rec->video_pt,
-		JANUS_SDP_OA_VIDEO_DIRECTION, JANUS_SDP_SENDONLY,
-		JANUS_SDP_OA_DATA, FALSE,
+		JANUS_SDP_OA_MLINE, JANUS_SDP_AUDIO,
+			JANUS_SDP_OA_ENABLED, offer_audio,
+			JANUS_SDP_OA_CODEC, janus_audiocodec_name(rec->acodec),
+			JANUS_SDP_OA_PT, rec->audio_pt,
+			JANUS_SDP_OA_DIRECTION, JANUS_SDP_SENDONLY,
+		JANUS_SDP_OA_MLINE, JANUS_SDP_VIDEO,
+			JANUS_SDP_OA_ENABLED, offer_video,
+			JANUS_SDP_OA_CODEC, janus_videocodec_name(rec->vcodec),
+			JANUS_SDP_OA_PT, rec->video_pt,
+			JANUS_SDP_OA_DIRECTION, JANUS_SDP_SENDONLY,
 		JANUS_SDP_OA_DONE);
 	g_free(rec->offer);
 	rec->offer = janus_sdp_write(offer);
@@ -747,6 +748,7 @@ int janus_recordplay_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Record&Play handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_RECORDPLAY_NAME);
@@ -890,8 +892,8 @@ json_t *janus_recordplay_query_session(janus_plugin_session *handle) {
 		json_object_set_new(info, "recording_name", json_string(session->recording->name));
 		janus_refcount_decrease(&session->recording->ref);
 	}
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -1123,14 +1125,15 @@ void janus_recordplay_setup_media(janus_plugin_session *handle) {
 			janus_refcount_decrease(&session->ref);
 			/* FIXME Should we notify this back to the user somehow? */
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Record&Play playout thread...\n", error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			gateway->close_pc(session->handle);
 		}
 	}
 	janus_refcount_decrease(&session->ref);
 }
 
-void janus_recordplay_send_rtcp_feedback(janus_plugin_session *handle, int video, char *buf, int len) {
-	if(video != 1)
+static void janus_recordplay_send_rtcp_feedback(janus_plugin_session *handle, gboolean video, char *buf, int len) {
+	if(!video)
 		return;	/* We just do this for video, for now */
 
 	janus_recordplay_session *session = (janus_recordplay_session *)handle->plugin_handle;
@@ -1153,7 +1156,7 @@ void janus_recordplay_send_rtcp_feedback(janus_plugin_session *handle, int video
 		/* Send a new REMB back */
 		char rtcpbuf[24];
 		janus_rtcp_remb((char *)(&rtcpbuf), 24, bitrate);
-		gateway->relay_rtcp(handle, video, rtcpbuf, 24);
+		gateway->relay_rtcp(handle, -1, video, rtcpbuf, 24);
 
 		session->video_remb_last = now;
 	}
@@ -1165,14 +1168,14 @@ void janus_recordplay_send_rtcp_feedback(janus_plugin_session *handle, int video
 	if(elapsed >= interval) {
 		/* Send both a FIR and a PLI, just to be sure */
 		janus_rtcp_fir((char *)&rtcpbuf, 20, &session->video_fir_seq);
-		gateway->relay_rtcp(handle, video, rtcpbuf, 20);
+		gateway->relay_rtcp(handle, -1, video, rtcpbuf, 20);
 		janus_rtcp_pli((char *)&rtcpbuf, 12);
-		gateway->relay_rtcp(handle, video, rtcpbuf, 12);
+		gateway->relay_rtcp(handle, -1, video, rtcpbuf, 12);
 		session->video_keyframe_request_last = now;
 	}
 }
 
-void janus_recordplay_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_recordplay_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	janus_recordplay_session *session = (janus_recordplay_session *)handle->plugin_handle;
@@ -1199,13 +1202,13 @@ void janus_recordplay_incoming_rtp(janus_plugin_session *handle, int video, char
 			char rtcpbuf[12];
 			memset(rtcpbuf, 0, 12);
 			janus_rtcp_pli((char *)&rtcpbuf, 12);
-			gateway->relay_rtcp(handle, 1, rtcpbuf, 12);
+			gateway->relay_rtcp(handle, -1, TRUE, rtcpbuf, 12);
 		}
 		/* Do we need to drop this? */
 		if(!save)
 			return;
 		/* If we got here, update the RTP header and save the packet */
-		janus_rtp_header_update(header, &session->context, TRUE, 0);
+		janus_rtp_header_update(header, &session->context, TRUE);
 		if(session->recording->vcodec == JANUS_VIDEOCODEC_VP8) {
 			int plen = 0;
 			char *payload = janus_rtp_payload(buf, len, &plen);
@@ -1228,12 +1231,12 @@ void janus_recordplay_incoming_rtp(janus_plugin_session *handle, int video, char
 	janus_recordplay_send_rtcp_feedback(handle, video, buf, len);
 }
 
-void janus_recordplay_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_recordplay_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 }
 
-void janus_recordplay_slow_link(janus_plugin_session *handle, int uplink, int video) {
+void janus_recordplay_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized) || !gateway)
 		return;
 
@@ -1523,7 +1526,9 @@ static void *janus_recordplay_handler(void *data) {
 			janus_mutex_init(&rec->mutex);
 			/* Check which codec we should record for audio and/or video */
 			const char *acodec = NULL, *vcodec = NULL;
-			janus_sdp_find_preferred_codecs(offer, &acodec, &vcodec);
+			janus_sdp_find_preferred_codec(offer, JANUS_SDP_AUDIO, -1, &acodec);
+			janus_sdp_find_preferred_codec(offer, JANUS_SDP_VIDEO, -1, &vcodec);
+			JANUS_LOG(LOG_WARN, "%s, %s\n", acodec, vcodec);
 			rec->acodec = janus_audiocodec_from_name(acodec);
 			rec->vcodec = janus_videocodec_from_name(vcodec);
 			/* We found preferred codecs: let's just make sure the direction is what we need */
@@ -1586,20 +1591,37 @@ static void *janus_recordplay_handler(void *data) {
 			janus_mutex_unlock(&recordings_mutex);
 			/* We need to prepare an answer */
 recdone:
-			answer = janus_sdp_generate_answer(offer,
-				JANUS_SDP_OA_AUDIO, audio,
-				JANUS_SDP_OA_AUDIO_CODEC, janus_audiocodec_name(rec->acodec),
-				JANUS_SDP_OA_AUDIO_DIRECTION, JANUS_SDP_RECVONLY,
-				JANUS_SDP_OA_VIDEO, video,
-				JANUS_SDP_OA_VIDEO_CODEC, janus_videocodec_name(rec->vcodec),
-				JANUS_SDP_OA_VIDEO_DIRECTION, JANUS_SDP_RECVONLY,
-				JANUS_SDP_OA_DATA, FALSE,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_RID,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_REPAIRED_RID,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_FRAME_MARKING,
-				JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC,
-				JANUS_SDP_OA_DONE);
+			answer = janus_sdp_generate_answer(offer);
+			gboolean audio_accepted = FALSE, video_accepted = FALSE;
+			GList *temp = offer->m_lines;
+			while(temp) {
+				janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+				if(m->type == JANUS_SDP_AUDIO && audio && !audio_accepted) {
+					audio_accepted = TRUE;
+					janus_sdp_generate_answer_mline(offer, answer, m,
+						JANUS_SDP_OA_MLINE, JANUS_SDP_AUDIO,
+							JANUS_SDP_OA_DIRECTION, JANUS_SDP_RECVONLY,
+							JANUS_SDP_OA_CODEC, janus_audiocodec_name(rec->acodec),
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_RID,
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_REPAIRED_RID,
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC,
+						JANUS_SDP_OA_DONE);
+				} else if(m->type == JANUS_SDP_VIDEO && video && !video_accepted) {
+					video_accepted = TRUE;
+					janus_sdp_generate_answer_mline(offer, answer, m,
+						JANUS_SDP_OA_MLINE, JANUS_SDP_VIDEO,
+							JANUS_SDP_OA_DIRECTION, JANUS_SDP_RECVONLY,
+							JANUS_SDP_OA_CODEC, janus_videocodec_name(rec->vcodec),
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_RID,
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_REPAIRED_RID,
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_FRAME_MARKING,
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC,
+						JANUS_SDP_OA_DONE);
+				}
+				temp = temp->next;
+			}
 			g_free(answer->s_name);
 			char s_name[100];
 			g_snprintf(s_name, sizeof(s_name), "Recording %"SCNu64, rec->id);
@@ -1614,22 +1636,30 @@ static void *janus_recordplay_handler(void *data) {
 			JANUS_LOG(LOG_VERB, "Going to answer this SDP:\n%s\n", sdp);
 			/* If the user negotiated simulcasting, prepare it accordingly */
 			json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
-			if(msg_simulcast) {
-				JANUS_LOG(LOG_VERB, "Recording client negotiated simulcasting\n");
-				int rid_ext_id = -1, framemarking_ext_id = -1;
-				janus_rtp_simulcasting_prepare(msg_simulcast, &rid_ext_id, &framemarking_ext_id, session->ssrc, session->rid);
-				session->sim_context.rid_ext_id = rid_ext_id;
-				session->sim_context.framemarking_ext_id = framemarking_ext_id;
-				session->sim_context.substream_target = 2;	/* Let's aim for the highest quality */
-				session->sim_context.templayer_target = 2;	/* Let's aim for all temporal layers */
-				if(rec->vcodec != JANUS_VIDEOCODEC_VP8 && rec->vcodec != JANUS_VIDEOCODEC_H264) {
-					/* VP8 r H.264 were not negotiated, if simulcasting was enabled then disable it here */
-					int i=0;
-					for(i=0; i<3; i++) {
-						session->ssrc[i] = 0;
-						g_free(session->rid[i]);
-						session->rid[i] = NULL;
+			if(msg_simulcast && json_array_size(msg_simulcast) > 0 &&
+					(rec->vcodec == JANUS_VIDEOCODEC_VP8 || rec->vcodec == JANUS_VIDEOCODEC_H264)) {
+				size_t i = 0;
+				for(i=0; i<json_array_size(msg_simulcast); i++) {
+					json_t *s = json_array_get(msg_simulcast, i);
+					int mindex = json_integer_value(json_object_get(s, "mindex"));
+					JANUS_LOG(LOG_VERB, "Recording client negotiated simulcasting (#%d)\n", mindex);
+					int rid_ext_id = -1, framemarking_ext_id = -1;
+					janus_rtp_simulcasting_prepare(s, &rid_ext_id, &framemarking_ext_id, session->ssrc, session->rid);
+					session->sim_context.rid_ext_id = rid_ext_id;
+					session->sim_context.framemarking_ext_id = framemarking_ext_id;
+					session->sim_context.substream_target = 2;	/* Let's aim for the highest quality */
+					session->sim_context.templayer_target = 2;	/* Let's aim for all temporal layers */
+					if(rec->vcodec != JANUS_VIDEOCODEC_VP8 && rec->vcodec != JANUS_VIDEOCODEC_H264) {
+						/* VP8 r H.264 were not negotiated, if simulcasting was enabled then disable it here */
+						int i=0;
+						for(i=0; i<3; i++) {
+							session->ssrc[i] = 0;
+							g_free(session->rid[0]);
+							session->rid[0] = NULL;
+						}
 					}
+					/* FIXME We're stopping at the first item, there may be more */
+					break;
 				}
 			}
 			/* Done! */
@@ -2405,7 +2435,7 @@ static void *janus_recordplay_playout_thread(void *data) {
 				/* Update payload type */
 				janus_rtp_header *rtp = (janus_rtp_header *)buffer;
 				rtp->type = audio_pt;
-				gateway->relay_rtp(session->handle, 0, (char *)buffer, bytes);
+				gateway->relay_rtp(session->handle, -1, FALSE, (char *)buffer, bytes);
 				gettimeofday(&now, NULL);
 				abefore.tv_sec = now.tv_sec;
 				abefore.tv_usec = now.tv_usec;
@@ -2445,7 +2475,7 @@ static void *janus_recordplay_playout_thread(void *data) {
 					/* Update payload type */
 					janus_rtp_header *rtp = (janus_rtp_header *)buffer;
 					rtp->type = audio_pt;
-					gateway->relay_rtp(session->handle, 0, (char *)buffer, bytes);
+					gateway->relay_rtp(session->handle, -1, FALSE, (char *)buffer, bytes);
 					asent = TRUE;
 					audio = audio->next;
 				}
@@ -2463,7 +2493,7 @@ static void *janus_recordplay_playout_thread(void *data) {
 					/* Update payload type */
 					janus_rtp_header *rtp = (janus_rtp_header *)buffer;
 					rtp->type = video_pt;
-					gateway->relay_rtp(session->handle, 1, (char *)buffer, bytes);
+					gateway->relay_rtp(session->handle, -1, TRUE, (char *)buffer, bytes);
 					video = video->next;
 				}
 				vsent = TRUE;
@@ -2507,7 +2537,7 @@ static void *janus_recordplay_playout_thread(void *data) {
 						/* Update payload type */
 						janus_rtp_header *rtp = (janus_rtp_header *)buffer;
 						rtp->type = video_pt;
-						gateway->relay_rtp(session->handle, 1, (char *)buffer, bytes);
+						gateway->relay_rtp(session->handle, -1, TRUE, (char *)buffer, bytes);
 						video = video->next;
 					}
 					vsent = TRUE;
diff --git a/plugins/janus_sip.c b/plugins/janus_sip.c
index 3e4509a76d..b03e9b19ff 100644
--- a/plugins/janus_sip.c
+++ b/plugins/janus_sip.c
@@ -641,8 +641,8 @@ const char *janus_sip_get_package(void);
 void janus_sip_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_sip_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 void janus_sip_setup_media(janus_plugin_session *handle);
-void janus_sip_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_sip_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_sip_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_sip_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_sip_hangup_media(janus_plugin_session *handle);
 void janus_sip_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_sip_query_session(janus_plugin_session *handle);
@@ -679,20 +679,20 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter register_parameters[] = {
-	{"type", JSON_STRING, 0},
+	{"type", JANUS_JSON_STRING, 0},
 	{"send_register", JANUS_JSON_BOOL, 0},
 	{"force_udp", JANUS_JSON_BOOL, 0},
 	{"force_tcp", JANUS_JSON_BOOL, 0},
 	{"sips", JANUS_JSON_BOOL, 0},
-	{"username", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"secret", JSON_STRING, 0},
-	{"ha1_secret", JSON_STRING, 0},
-	{"authuser", JSON_STRING, 0},
-	{"display_name", JSON_STRING, 0},
-	{"user_agent", JSON_STRING, 0},
+	{"username", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"ha1_secret", JANUS_JSON_STRING, 0},
+	{"authuser", JANUS_JSON_STRING, 0},
+	{"display_name", JANUS_JSON_STRING, 0},
+	{"user_agent", JANUS_JSON_STRING, 0},
 	{"headers", JSON_OBJECT, 0},
 	{"master_id", JANUS_JSON_INTEGER, 0},
 	{"refresh", JANUS_JSON_BOOL, 0}
@@ -703,24 +703,24 @@ static struct janus_json_parameter subscribe_parameters[] = {
 	{"accept", JSON_STRING, 0}
 };
 static struct janus_json_parameter proxy_parameters[] = {
-	{"proxy", JSON_STRING, 0},
-	{"outbound_proxy", JSON_STRING, 0}
+	{"proxy", JANUS_JSON_STRING, 0},
+	{"outbound_proxy", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter call_parameters[] = {
-	{"uri", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"uri", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"headers", JSON_OBJECT, 0},
-	{"srtp", JSON_STRING, 0},
-	{"srtp_profile", JSON_STRING, 0},
+	{"srtp", JANUS_JSON_STRING, 0},
+	{"srtp_profile", JANUS_JSON_STRING, 0},
 	{"autoaccept_reinvites", JANUS_JSON_BOOL, 0},
 	{"refer_id", JANUS_JSON_INTEGER, 0},
 	/* The following are only needed in case "guest" registrations
 	 * still need an authenticated INVITE for some reason */
-	{"secret", JSON_STRING, 0},
-	{"ha1_secret", JSON_STRING, 0},
-	{"authuser", JSON_STRING, 0}
+	{"secret", JANUS_JSON_STRING, 0},
+	{"ha1_secret", JANUS_JSON_STRING, 0},
+	{"authuser", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter accept_parameters[] = {
-	{"srtp", JSON_STRING, 0},
+	{"srtp", JANUS_JSON_STRING, 0},
 	{"headers", JSON_OBJECT, 0},
 	{"autoaccept_reinvites", JANUS_JSON_BOOL, 0}
 };
@@ -733,23 +733,23 @@ static struct janus_json_parameter transfer_parameters[] = {
 	{"call_id", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter recording_parameters[] = {
-	{"action", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"action", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"audio", JANUS_JSON_BOOL, 0},
 	{"video", JANUS_JSON_BOOL, 0},
 	{"peer_audio", JANUS_JSON_BOOL, 0},
 	{"peer_video", JANUS_JSON_BOOL, 0},
-	{"filename", JSON_STRING, 0}
+	{"filename", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter dtmf_info_parameters[] = {
-	{"digit", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"duration", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"digit", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"duration", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter info_parameters[] = {
-	{"type", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"content", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"type", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"content", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter sipmessage_parameters[] = {
-	{"content", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"content", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 
 /* Useful stuff */
@@ -928,7 +928,7 @@ typedef struct janus_sip_media {
 	srtp_policy_t video_remote_policy, video_local_policy;
 	gboolean video_send;
 	janus_sdp_mdirection pre_hold_video_dir;
-	janus_rtp_switching_context context;
+	janus_rtp_switching_context acontext, vcontext;
 	int pipefd[2];
 	gboolean updated;
 } janus_sip_media;
@@ -1333,7 +1333,8 @@ static void janus_sip_media_reset(janus_sip_session *session) {
 	session->media.video_pt_name = NULL;	/* Immutable string, no need to free*/
 	session->media.video_send = TRUE;
 	session->media.pre_hold_video_dir = JANUS_SDP_DEFAULT;
-	janus_rtp_switching_context_reset(&session->media.context);
+	janus_rtp_switching_context_reset(&session->media.acontext);
+	janus_rtp_switching_context_reset(&session->media.vcontext);
 }
 
 
@@ -1706,6 +1707,7 @@ int janus_sip_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the SIP handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_SIP_NAME);
@@ -1856,7 +1858,8 @@ void janus_sip_create_session(janus_plugin_session *handle, int *error) {
 	session->media.video_send = TRUE;
 	session->media.pre_hold_video_dir = JANUS_SDP_DEFAULT;
 	/* Initialize the RTP context */
-	janus_rtp_switching_context_reset(&session->media.context);
+	janus_rtp_switching_context_reset(&session->media.acontext);
+	janus_rtp_switching_context_reset(&session->media.vcontext);
 	session->media.pipefd[0] = -1;
 	session->media.pipefd[1] = -1;
 	session->media.updated = FALSE;
@@ -1998,10 +2001,10 @@ json_t *janus_sip_query_session(janus_plugin_session *handle) {
 			json_object_set_new(recording, "video-peer", json_string(session->vrc_peer->filename));
 		json_object_set_new(info, "recording", recording);
 	}
-	json_object_set_new(info, "establishing", json_integer(g_atomic_int_get(&session->establishing)));
-	json_object_set_new(info, "established", json_integer(g_atomic_int_get(&session->established)));
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "establishing", g_atomic_int_get(&session->establishing) ? json_true() : json_false());
+	json_object_set_new(info, "established", g_atomic_int_get(&session->established) ? json_true() : json_false());
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -2053,7 +2056,7 @@ void janus_sip_setup_media(janus_plugin_session *handle) {
 	/* TODO Only relay RTP/RTCP when we get this event */
 }
 
-void janus_sip_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_sip_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	if(gateway) {
@@ -2171,7 +2174,7 @@ void janus_sip_incoming_rtp(janus_plugin_session *handle, int video, char *buf,
 	}
 }
 
-void janus_sip_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_sip_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	if(gateway) {
@@ -2733,6 +2736,7 @@ static void *janus_sip_handler(void *data) {
 					JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the SIP Sofia thread...\n", error->code, error->message ? error->message : "??");
 					error_code = JANUS_SIP_ERROR_UNKNOWN_ERROR;
 					g_snprintf(error_cause, 512, "Got error %d (%s) trying to launch the SIP Sofia thread", error->code, error->message ? error->message : "??");
+					g_error_free(error);
 					goto error;
 				}
 				long int timeout = 0;
@@ -3175,11 +3179,18 @@ static void *janus_sip_handler(void *data) {
 			}
 			/* If the user negotiated simulcasting, just stick with the base substream */
 			json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
-			if(msg_simulcast) {
+			if(msg_simulcast && json_array_size(msg_simulcast) > 0) {
 				JANUS_LOG(LOG_WARN, "Client negotiated simulcasting which we don't do here, falling back to base substream...\n");
-				json_t *s = json_object_get(msg_simulcast, "ssrcs");
-				if(s && json_array_size(s) > 0)
-					session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+				size_t i = 0;
+				for(i=0; i<json_array_size(msg_simulcast); i++) {
+					json_t *sobj = json_array_get(msg_simulcast, i);
+					json_t *s = json_object_get(sobj, "ssrcs");
+					if(s && json_array_size(s) > 0)
+						session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+					session->media.simulcast_ssrc = json_integer_value(json_object_get(s, "ssrc-0"));
+					/* FIXME We're stopping at the first item, there may be more */
+					break;
+				}
 			}
 			/* Check if there are new credentials to authenticate the INVITE */
 			if(authuser) {
@@ -3361,11 +3372,17 @@ static void *janus_sip_handler(void *data) {
 			JANUS_LOG(LOG_VERB, "Prepared SDP for 200 OK:\n%s", sdp);
 			/* If the user negotiated simulcasting, just stick with the base substream */
 			json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
-			if(msg_simulcast) {
+			if(msg_simulcast && json_array_size(msg_simulcast) > 0) {
 				JANUS_LOG(LOG_WARN, "Client negotiated simulcasting which we don't do here, falling back to base substream...\n");
-				json_t *s = json_object_get(msg_simulcast, "ssrcs");
-				if(s && json_array_size(s) > 0)
-					session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+				size_t i = 0;
+				for(i=0; i<json_array_size(msg_simulcast); i++) {
+					json_t *sobj = json_array_get(msg_simulcast, i);
+					json_t *s = json_object_get(sobj, "ssrcs");
+					if(s && json_array_size(s) > 0)
+						session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+					/* FIXME We're stopping at the first item, there may be more */
+					break;
+				}
 			}
 			/* Also notify event handlers */
 			if(notify_events && gateway->events_is_enabled()) {
@@ -3413,6 +3430,7 @@ static void *janus_sip_handler(void *data) {
 					session->media.ready = FALSE;
 					janus_refcount_decrease(&session->ref);
 					JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");
+					g_error_free(error);
 				}
 			}
 		} else if(!strcasecmp(request_text, "update")) {
@@ -3936,7 +3954,7 @@ static void *janus_sip_handler(void *data) {
 						JANUS_LOG(LOG_VERB, "Recording video, sending a PLI to kickstart it\n");
 						char buf[12];
 						janus_rtcp_pli((char *)&buf, 12);
-						gateway->relay_rtcp(session->handle, 1, buf, 12);
+						gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 					}
 				}
 			} else {
@@ -4890,6 +4908,7 @@ void janus_sip_sofia_callback(nua_event_t event, int status, char const *phrase,
 					session->media.ready = FALSE;
 					janus_refcount_decrease(&session->ref);
 					JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");
+					g_error_free(error);
 				}
 			}
 			/* Check if there's an isfocus feature parameter in the Contact header */
@@ -5655,8 +5674,6 @@ static void *janus_sip_relay_thread(void *data) {
 	/* Loop */
 	int num = 0;
 	gboolean goon = TRUE;
-	int astep = 0, vstep = 0;
-	guint32 ats = 0, vts = 0;
 
 	session->media.updated = TRUE; /* Connect UDP sockets upon loop entry */
 	gboolean have_audio_server_ip = TRUE;
@@ -5844,19 +5861,11 @@ static void *janus_sip_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* Check if the SSRC changed (e.g., after a re-INVITE or UPDATE) */
-					guint32 timestamp = ntohl(header->timestamp);
-					janus_rtp_header_update(header, &session->media.context, FALSE, astep ? astep : 960);
-					if(ats == 0) {
-						ats = timestamp;
-					} else if(astep == 0) {
-						astep = timestamp-ats;
-						if(astep < 0)
-							astep = 0;
-					}
+					janus_rtp_header_update(header, &session->media.acontext, FALSE);
 					/* Save the frame if we're recording */
 					janus_recorder_save_frame(session->arc_peer, buffer, bytes);
 					/* Relay to application */
-					gateway->relay_rtp(session->handle, 0, buffer, bytes);
+					gateway->relay_rtp(session->handle, -1, FALSE, buffer, bytes);
 					continue;
 				} else if(session->media.audio_rtcp_fd != -1 && fds[i].fd == session->media.audio_rtcp_fd) {
 					/* Got something audio (RTCP) */
@@ -5879,7 +5888,7 @@ static void *janus_sip_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* Relay to application */
-					gateway->relay_rtcp(session->handle, 0, buffer, bytes);
+					gateway->relay_rtcp(session->handle, -1, FALSE, buffer, bytes);
 					continue;
 				} else if(session->media.video_rtp_fd != -1 && fds[i].fd == session->media.video_rtp_fd) {
 					/* Got something video (RTP) */
@@ -5909,19 +5918,11 @@ static void *janus_sip_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* Check if the SSRC changed (e.g., after a re-INVITE or UPDATE) */
-					janus_rtp_header_update(header, &session->media.context, TRUE, vstep ? vstep : 4500);
-					guint32 timestamp = ntohl(header->timestamp);
-					if(vts == 0) {
-						vts = timestamp;
-					} else if(vstep == 0) {
-						vstep = timestamp-vts;
-						if(vstep < 0)
-							vstep = 0;
-					}
+					janus_rtp_header_update(header, &session->media.vcontext, TRUE);
 					/* Save the frame if we're recording */
 					janus_recorder_save_frame(session->vrc_peer, buffer, bytes);
 					/* Relay to application */
-					gateway->relay_rtp(session->handle, 1, buffer, bytes);
+					gateway->relay_rtp(session->handle, -1, TRUE, buffer, bytes);
 					continue;
 				} else if(session->media.video_rtcp_fd != -1 && fds[i].fd == session->media.video_rtcp_fd) {
 					/* Got something video (RTCP) */
@@ -5944,7 +5945,7 @@ static void *janus_sip_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* Relay to application */
-					gateway->relay_rtcp(session->handle, 1, buffer, bytes);
+					gateway->relay_rtcp(session->handle, -1, TRUE, buffer, bytes);
 					continue;
 				}
 			}
diff --git a/plugins/janus_sipre.c b/plugins/janus_sipre.c
index 46993159c4..2d223fa3d3 100644
--- a/plugins/janus_sipre.c
+++ b/plugins/janus_sipre.c
@@ -89,8 +89,8 @@
 
 
 /* Plugin information */
-#define JANUS_SIPRE_VERSION			1
-#define JANUS_SIPRE_VERSION_STRING	"0.0.1"
+#define JANUS_SIPRE_VERSION			2
+#define JANUS_SIPRE_VERSION_STRING	"0.0.2"
 #define JANUS_SIPRE_DESCRIPTION		"This is a simple SIP plugin for Janus (based on libre instead of Sofia), allowing WebRTC peers to register at a SIP server and call SIP user agents through a Janus instance."
 #define JANUS_SIPRE_NAME			"JANUS SIPre plugin"
 #define JANUS_SIPRE_AUTHOR			"Meetecho s.r.l."
@@ -110,8 +110,8 @@ const char *janus_sipre_get_package(void);
 void janus_sipre_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_sipre_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 void janus_sipre_setup_media(janus_plugin_session *handle);
-void janus_sipre_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_sipre_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_sipre_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_sipre_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_sipre_hangup_media(janus_plugin_session *handle);
 void janus_sipre_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_sipre_query_session(janus_plugin_session *handle);
@@ -163,18 +163,18 @@ static struct janus_json_parameter register_parameters[] = {
 	{"refresh", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter proxy_parameters[] = {
-	{"proxy", JSON_STRING, 0},
-	{"outbound_proxy", JSON_STRING, 0}
+	{"proxy", JANUS_JSON_STRING, 0},
+	{"outbound_proxy", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter call_parameters[] = {
 	{"uri", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"headers", JANUS_JSON_OBJECT, 0},
 	{"srtp", JANUS_JSON_STRING, 0},
-	{"srtp_profile", JSON_STRING, 0},
+	{"srtp_profile", JANUS_JSON_STRING, 0},
 	/* The following are only needed in case "guest" registrations
 	 * still need an authenticated INVITE for some reason */
-	{"secret", JSON_STRING, 0},
-	{"authuser", JSON_STRING, 0}
+	{"secret", JANUS_JSON_STRING, 0},
+	{"authuser", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter accept_parameters[] = {
 	{"srtp", JANUS_JSON_STRING, 0},
@@ -193,11 +193,11 @@ static struct janus_json_parameter dtmf_info_parameters[] = {
 	{"duration", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter info_parameters[] = {
-	{"type", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"content", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"type", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"content", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter sipmessage_parameters[] = {
-	{"content", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"content", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 
 /* Useful stuff */
@@ -488,7 +488,7 @@ typedef struct janus_sipre_media {
 	srtp_policy_t video_remote_policy, video_local_policy;
 	gboolean video_send;
 	janus_sdp_mdirection pre_hold_video_dir;
-	janus_rtp_switching_context context;
+	janus_rtp_switching_context acontext, vcontext;
 	int pipefd[2];
 	gboolean updated;
 } janus_sipre_media;
@@ -867,7 +867,8 @@ static void janus_sipre_media_reset(janus_sipre_session *session) {
 	session->media.video_pt_name = NULL;	/* Immutable string, no need to free*/
 	session->media.video_send = TRUE;
 	session->media.pre_hold_video_dir = JANUS_SDP_DEFAULT;
-	janus_rtp_switching_context_reset(&session->media.context);
+	janus_rtp_switching_context_reset(&session->media.acontext);
+	janus_rtp_switching_context_reset(&session->media.vcontext);
 }
 
 
@@ -1162,6 +1163,7 @@ int janus_sipre_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the SIPre handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	/* Launch the thread that will handle the libre initialization and event loop */
@@ -1170,6 +1172,7 @@ int janus_sipre_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the SIPre loop thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	/* Let's wait for the libre initialization to complete */
@@ -1328,7 +1331,8 @@ void janus_sipre_create_session(janus_plugin_session *handle, int *error) {
 	session->media.video_send = TRUE;
 	session->media.pre_hold_video_dir = JANUS_SDP_DEFAULT;
 	/* Initialize the RTP context */
-	janus_rtp_switching_context_reset(&session->media.context);
+	janus_rtp_switching_context_reset(&session->media.acontext);
+	janus_rtp_switching_context_reset(&session->media.vcontext);
 	session->media.pipefd[0] = -1;
 	session->media.pipefd[1] = -1;
 	session->media.updated = FALSE;
@@ -1416,10 +1420,10 @@ json_t *janus_sipre_query_session(janus_plugin_session *handle) {
 			json_object_set_new(recording, "video-peer", json_string(session->vrc_peer->filename));
 		json_object_set_new(info, "recording", recording);
 	}
-	json_object_set_new(info, "establishing", json_integer(g_atomic_int_get(&session->establishing)));
-	json_object_set_new(info, "established", json_integer(g_atomic_int_get(&session->established)));
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "establishing", g_atomic_int_get(&session->establishing) ? json_true() : json_false());
+	json_object_set_new(info, "established", g_atomic_int_get(&session->established) ? json_true() : json_false());
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -1471,7 +1475,7 @@ void janus_sipre_setup_media(janus_plugin_session *handle) {
 	/* TODO Only relay RTP/RTCP when we get this event */
 }
 
-void janus_sipre_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_sipre_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	if(gateway) {
@@ -1552,7 +1556,7 @@ void janus_sipre_incoming_rtp(janus_plugin_session *handle, int video, char *buf
 	}
 }
 
-void janus_sipre_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_sipre_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	if(gateway) {
@@ -2158,9 +2162,16 @@ static void *janus_sipre_handler(void *data) {
 			json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
 			if(msg_simulcast) {
 				JANUS_LOG(LOG_WARN, "Client negotiated simulcasting which we don't do here, falling back to base substream...\n");
-				json_t *s = json_object_get(msg_simulcast, "ssrcs");
-				if(s && json_array_size(s) > 0)
-					session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+				size_t i = 0;
+				for(i=0; i<json_array_size(msg_simulcast); i++) {
+					json_t *sobj = json_array_get(msg_simulcast, i);
+					json_t *s = json_object_get(sobj, "ssrcs");
+					if(s && json_array_size(s) > 0)
+						session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+					session->media.simulcast_ssrc = json_integer_value(json_object_get(s, "ssrc-0"));
+					/* FIXME We're stopping at the first item, there may be more */
+					break;
+				}
 			}
 			/* Check if there are new credentials to authenticate the INVITE */
 			if(authuser) {
@@ -2300,9 +2311,16 @@ static void *janus_sipre_handler(void *data) {
 			json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
 			if(msg_simulcast) {
 				JANUS_LOG(LOG_WARN, "Client negotiated simulcasting which we don't do here, falling back to base substream...\n");
-				json_t *s = json_object_get(msg_simulcast, "ssrcs");
-				if(s && json_array_size(s) > 0)
-					session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+				size_t i = 0;
+				for(i=0; i<json_array_size(msg_simulcast); i++) {
+					json_t *sobj = json_array_get(msg_simulcast, i);
+					json_t *s = json_object_get(sobj, "ssrcs");
+					if(s && json_array_size(s) > 0)
+						session->media.simulcast_ssrc = json_integer_value(json_array_get(s, 0));
+					session->media.simulcast_ssrc = json_integer_value(json_object_get(s, "ssrc-0"));
+					/* FIXME We're stopping at the first item, there may be more */
+					break;
+				}
 			}
 			/* Also notify event handlers */
 			if(notify_events && gateway->events_is_enabled()) {
@@ -2345,6 +2363,7 @@ static void *janus_sipre_handler(void *data) {
 					session->media.ready = FALSE;
 					janus_refcount_decrease(&session->ref);
 					JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");
+					g_error_free(error);
 				}
 			}
 		} else if(!strcasecmp(request_text, "update")) {
@@ -2720,7 +2739,7 @@ static void *janus_sipre_handler(void *data) {
 						char buf[12];
 						memset(buf, 0, 12);
 						janus_rtcp_pli((char *)&buf, 12);
-						gateway->relay_rtcp(session->handle, 1, buf, 12);
+						gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 					}
 				}
 			} else {
@@ -3323,8 +3342,6 @@ static void *janus_sipre_relay_thread(void *data) {
 	/* Loop */
 	int num = 0;
 	gboolean goon = TRUE;
-	int astep = 0, vstep = 0;
-	guint32 ats = 0, vts = 0;
 
 	session->media.updated = TRUE; /* Connect UDP sockets upon loop entry */
 	gboolean have_audio_server_ip = TRUE;
@@ -3530,32 +3547,11 @@ static void *janus_sipre_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* Check if the SSRC changed (e.g., after a re-INVITE or UPDATE) */
-					guint32 timestamp = ntohl(header->timestamp);
-					janus_rtp_header_update(header, &session->media.context, video,
-						(video ? (vstep ? vstep : 4500) : (astep ? astep : 960)));
-					if(video) {
-						if(vts == 0) {
-							vts = timestamp;
-						} else if(vstep == 0) {
-							vstep = timestamp-vts;
-							if(vstep < 0) {
-								vstep = 0;
-							}
-						}
-					} else {
-						if(ats == 0) {
-							ats = timestamp;
-						} else if(astep == 0) {
-							astep = timestamp-ats;
-							if(astep < 0) {
-								astep = 0;
-							}
-						}
-					}
+					janus_rtp_header_update(header, video ? &session->media.vcontext : &session->media.acontext, video);
 					/* Save the frame if we're recording */
 					janus_recorder_save_frame(video ? session->vrc_peer : session->arc_peer, buffer, bytes);
 					/* Relay to browser */
-					gateway->relay_rtp(session->handle, video, buffer, bytes);
+					gateway->relay_rtp(session->handle, -1, video, buffer, bytes);
 					continue;
 				} else {
 					/* Audio or Video RTCP */
@@ -3577,7 +3573,7 @@ static void *janus_sipre_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* Relay to browser */
-					gateway->relay_rtcp(session->handle, video, buffer, bytes);
+					gateway->relay_rtcp(session->handle, -1, video, buffer, bytes);
 					continue;
 				}
 			}
@@ -4017,6 +4013,7 @@ int janus_sipre_cb_answer(const struct sip_msg *msg, void *arg) {
 			session->media.ready = FALSE;
 			janus_refcount_decrease(&session->ref);
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTP/RTCP thread...\n", error->code, error->message ? error->message : "??");
+			g_error_free(error);
 		}
 	}
 	/* Send event back to the browser */
diff --git a/plugins/janus_streaming.c b/plugins/janus_streaming.c
index 1749076ea9..dcb2f00fc4 100644
--- a/plugins/janus_streaming.c
+++ b/plugins/janus_streaming.c
@@ -26,7 +26,7 @@
  * as well.
  *
  * For what concerns type 3., instead, the plugin is configured
- * to listen on a couple of ports for RTP: this means that the plugin
+ * to listen on a few ports for RTP: this means that the plugin
  * is implemented to receive RTP on those ports and relay them to all
  * peers attached to that stream. Any tool that can generate audio/video
  * RTP streams and specify a destination is good for the purpose: the
@@ -44,7 +44,7 @@
  *
  * Streams to make available are listed in the plugin configuration file.
  * A pre-filled configuration file is provided in \c conf/janus.plugin.streaming.jcfg
- * and includes a stream of every type.
+ * and includes some samples you can start from.
  *
  * To add more streams or modify the existing ones, you can use the following
  * syntax:
@@ -106,6 +106,7 @@ collision = in case of collision (more than one SSRC hitting the same port), the
 	will discard incoming RTP packets with a new SSRC unless this many milliseconds
 	passed, which would then change the current SSRC (0=disabled)
 dataport = local port for receiving data messages to relay
+datamcast = multicast group for receiving data frames, if any
 dataiface = network interface or IP address to bind to, if any (binds to all otherwise)
 databuffermsg = true|false (whether the plugin should store the latest
 	message and send it immediately for new viewers)
@@ -129,6 +130,64 @@ rtsp_pwd = RTSP authorization password, if needed
 rtsp_failcheck = whether an error should be returned if connecting to the RTSP server fails (default=true)
 rtspiface = network interface IP address or device name to listen on when receiving RTSP streams
 \endverbatim
+ *
+ * Notice that attributes like \c audioport or \c videopt only make sense
+ * when you're creating a mountpoint with a single audio and/or video stream,
+ * as the plugin in that case assumes that limitation is fine by you. In
+ * case you're interested in creating multistream mountpoints, that is
+ * mountpoints that can contain more than one audio and/or video stream
+ * at the same time, you HAVE to use a different syntax. Specifically,
+ * you'll need to use a \c media array/list, containing the different
+ * streams, in the right order, that you want to make available: each
+ * stream will then need to contain the related info, e.g., port to bind
+ * to, type of media, rtpmap and so on. An example is provided below:
+ *
+\verbatim
+multistream-test: {
+	type = "rtp"
+	id = 123
+	description = "Multistream test (1 audio, 2 video)"
+	media = (
+		{
+			type = "audio"
+			mid = "a"
+			label = "Audio stream"
+			port = 5102
+			pt = 111
+			rtpmap = "opus/48000/2"
+		},
+		{
+			type = "video"
+			mid = "v1"
+			label = "Video stream #1"
+			port = 5104
+			pt = 100
+			rtpmap = "VP8/90000"
+		},
+		{
+			type = "video"
+			mid = "v2"
+			label = "Video stream #2"
+			port = 5106
+			pt = 100
+			rtpmap = "VP8/90000"
+		}
+	)
+}
+\endverbatim
+ *
+ * In the above example, we're creating a mountpoint with a single audio
+ * stream and two different video streams: each stream has a unique \c mid
+ * (that you MUST provide) which is what will be used for the SDP offer
+ * to send to viewers, and their unique configuration properties. As you
+ * can see, it's much cleaner in the way you create and configure
+ * mountpoints: there's no hardcoded audio/video prefix for the name of
+ * properties, you configure media streams the same way and just add them
+ * to a list. Notice that of course this also works with the simple one
+ * audio/one video mountpoints you've used so far, and that has been
+ * documented before: as such, you're encouraged to start using this
+ * new approach as soon as possible, since in the next versions we
+ * might deprecate the old one.
  *
  * \section streamapi Streaming API
  *
@@ -192,15 +251,23 @@ rtspiface = network interface IP address or device name to listen on when receiv
 			"id" : <unique ID of mountpoint #1>,
 			"description" : "<description of mountpoint #1>",
 			"type" : "<type of mountpoint #1, in line with the types introduced above>",
-			"audio_age_ms" : <how much time passed since we last received audio; optional, available for RTP mountpoints only>,
-			"video_age_ms" : <how much time passed since we last received video; optional, available for RTP mountpoints only>
+			"media" : [
+				{
+					"mid" : "<unique mid of this stream>",
+					"label" : "<unique text label of this stream>",
+					"type" : "<audio|video|data">,
+					"age_ms" : <how much time passed since we last received media for this stream; optional>,
+				},
+				{
+					// Other streams, if available
+				}
+			]
 		},
 		{
 			"id" : <unique ID of mountpoint #2>,
 			"description" : "<description of mountpoint #2>",
 			"type" : "<type of mountpoint #2, in line with the types introduced above>",
-			"audio_age_ms" : <how much time passed since we last received audio; optional, available for RTP mountpoints only>,
-			"video_age_ms" : <how much time passed since we last received video; optional, available for RTP mountpoints only>
+			"media" : [..]
 		},
 		...
 	]
@@ -208,7 +275,7 @@ rtspiface = network interface IP address or device name to listen on when receiv
 \endverbatim
  *
  * As you can see, the \c list request only returns very generic info on
- * each mounpoint. In case you're interested in learning more details about
+ * each mountpoint. In case you're interested in learning more details about
  * a specific mountpoint, you can use the \c info request instead, which
  * returns more information, or all of it if the mountpoint secret is
  * provided in the request. An \c info request must be formatted like this:
@@ -235,15 +302,23 @@ rtspiface = network interface IP address or device name to listen on when receiv
 		"pin" : "<PIN to access mountpoint; only available if a valid secret was provided>",
 		"is_private" : <true|false, depending on whether the mountpoint is listable; only available if a valid secret was provided>,
 		"enabled" : <true|false, depending on whether the mountpoint is currently enabled or not>,
-		"audio" : <true, only present if the mountpoint contains audio>,
-		"audiopt" : <audio payload type, only present if configured and the mountpoint contains audio>,
-		"audiortpmap" : "<audio SDP rtpmap value, only present if configured and the mountpoint contains audio>",
-		"audiofmtp" : "<audio SDP fmtp value, only present if configured and the mountpoint contains audio>",
-		"video" : <true, only present if the mountpoint contains video>,
-		"videopt" : <video payload type, only present if configured and the mountpoint contains video>,
-		"videortpmap" : "<video SDP rtpmap value, only present if configured and the mountpoint contains video>",
-		"videofmtp" : "<video SDP fmtp value, only present if configured and the mountpoint contains video>",
-		...
+		"type" : "<type of mountpoint>",
+		"media" : [
+			{
+				"mid" : "<unique mid of this stream>",
+				"mindex" : "<unique mindex of this stream>",
+				"type" : "<audio|video|data">,
+				"label" : "<unique text label of this stream>",
+				"age_ms" : <how much time passed since we last received media for this stream; optional>,
+				"pt" : <payload type, only present if RTP and configured>,
+				"rtpmap" : "<SDP rtpmap value, only present if RTP and configured>",
+				"fmtp" : "<audio SDP fmtp value, only present if RTP and configured>",
+				...
+			},
+			{
+				// Other streams, if available
+			}
+		]
 	}
 }
 \endverbatim
@@ -289,6 +364,14 @@ rtspiface = network interface IP address or device name to listen on when receiv
  * NOT have to escape semicolons with a trailing slash, in those properties
  * where a semicolon might be needed (e.g., \c audiofmtp or \c videofmtp ).
  *
+ * Of course, just as we introduced the possibility of configuring multistream
+ * mountpoints statically with a \c media array, the same applies when using
+ * the API to create them: just add a \c media JSON array containing the
+ * list of streams to create and the related properties as you would do
+ * statically (that is, using generic properties like \c port, \c fmtp,
+ * etc., rather than the hardcoded \c audioport and the like), and it
+ * will work for dynamically created mountpoints as well.
+ *
  * A successful \c create will result in a \c created response:
  *
 \verbatim
@@ -301,6 +384,16 @@ rtspiface = network interface IP address or device name to listen on when receiv
 		"type" : "<type of the just created mountpoint>",
 		"description" : "<description of the just created mountpoint>",
 		"is_private" : <true|false, depending on whether the new mountpoint is listable>,
+		"ports" : [		// Only for RTP mountpoints
+			{
+				"type" : "<audio|video|data>",
+				"mid" : "<unique mid of stream #1>",
+				"port" : <port the plugin is listening on for this stream's media>
+			},
+			{
+				// Other streams, if available
+			}
+		]
 		...
 	}
 }
@@ -419,7 +512,10 @@ rtspiface = network interface IP address or device name to listen on when receiv
  * using the \c recording request. The same request can also be used to
  * stop recording. Although the same request is used in both cases, though,
  * the syntax for the two use cases differs a bit, namely in terms of the
- * type of some properties.
+ * type of some properties. Notice that, while for backwards compatibility
+ * you can still use the old \c audio, \c video and \c data named properties,
+ * they're now deprecated and so you're highly encouraged to use the new drill-down
+ * \c media list instead.
  *
  * To start recording a new mountpoint, the request should be formatted
  * like this:
@@ -429,12 +525,15 @@ rtspiface = network interface IP address or device name to listen on when receiv
 	"request" : "recording",
 	"action" : "start",
 	"id" : <unique ID of the mountpoint to manipulate; mandatory>,
-	"audio" : "<enable audio recording, and use this base path/filename; optional>",
-	"video" : "<enable video recording, and use this base path/filename; optional>",
-	"data" : "<enable data recording, and use this base path/filename; optional>",
-	"audio" : <true|false; whether or not audio should be recorded>,
-	"video" : <true|false; whether or not video should be recorded>,
-	"data" : <true|false; whether or not datachannel messages should be recorded>
+	"media" : [		// Drill-down recording controls
+		{
+			"mid" : "<mid of the stream to start recording>",
+			"filename" : "<base path/filename to use for the recording>"
+		},
+		{
+			// Recording controls for other streams, if provided
+		}
+	]
 }
 \endverbatim
  *
@@ -445,16 +544,23 @@ rtspiface = network interface IP address or device name to listen on when receiv
 	"request" : "recording",
 	"action" : "stop",
 	"id" : <unique ID of the mountpoint to manipulate; mandatory>,
-	"audio" : <true|false; whether or not audio recording should be stopped>,
-	"video" : <true|false; whether or not video recording should be stopped>,
-	"data" : <true|false; whether or not datachannel recording should be stopped>
+	"media" : [		// Drill-down recording controls
+		{
+			"mid" : "<mid of the stream to stop recording>"
+		},
+		{
+			// Recording controls for other streams, if provided
+		}
+	]
 }
 \endverbatim
  *
- * As you can notice, when you want to start a recording the \c audio ,
+ * When using the deprecated properties, when starting a recording the \c audio ,
  * \c video and \c data properties are strings, and specify the base path
  * to use for the recording filename; when stopping a recording, instead,
- * they're interpreted as boolean properties. Notice that, as with all
+ * they're interpreted as boolean properties. This is one more reason why
+ * you should migrate to the new \c media list instead, as it doesn't have
+ * this ambiguity between the two different requests. Notice that, as with all
  * APIs that wrap .mjr recordings, the filename you specify here is not
  * the actual filename: an \c .mjr extension is always going to be added
  * by the Janus core, so you should take this into account when tracking
@@ -490,24 +596,40 @@ rtspiface = network interface IP address or device name to listen on when receiv
 	"request" : "watch",
 	"id" : <unique ID of the mountpoint to subscribe to; mandatory>,
 	"pin" : "<PIN required to access the mountpoint; mandatory if configured>",
-	"offer_audio" : <true|false; whether or not audio should be negotiated; true by default if the mountpoint has audio>,
-	"offer_video" : <true|false; whether or not video should be negotiated; true by default if the mountpoint has video>,
-	"offer_data" : <true|false; whether or not datachannels should be negotiated; true by default if the mountpoint has datachannels>
+	"media" : [
+		<array of mids to subscribe to, as strings; optional, missing or empty array subscribes to all mids>
+	]
+	"offer_audio" : <true|false; deprecated; whether or not audio should be negotiated; true by default if the mountpoint has audio>,
+	"offer_video" : <true|false; deprecated; whether or not video should be negotiated; true by default if the mountpoint has video>,
+	"offer_data" : <true|false; deprecated; whether or not datachannels should be negotiated; true by default if the mountpoint has datachannels>
 }
 \endverbatim
  *
  * As you can see, it's just a matter of specifying the ID of the mountpoint to
  * subscribe to and, if needed, the PIN to access the mountpoint in case
- * it's protected. The \c offer_audio , \c offer_video and \c offer_data are
- * also particularly interesting, though, as they allow you to only subscribe
- * to a subset of the mountpoint media. By default, in fact, a \c watch
- * request will result in the plugin preparing a new SDP offer trying to
- * negotiate all the media streams available in the mountpoint; in case
- * the viewer knows they don't support one of the mountpoint codecs, though
- * (e.g., the video in the mountpoint is VP8, but they only support H.264),
- * or are not interested in getting all the media (e.g., they're ok with
- * just audio and not video, or don't have enough bandwidth for both),
- * they can use those properties to shape the SDP offer to their needs.
+ * it's protected. The \c media array is particularly interesting, as it
+ * allows you to only subscribe to a subset of the mountpoint media, which
+ * you can address by the related \c mid property of each stream. By default,
+ * in fact, a \c watch request will result in the plugin preparing a new
+ * SDP offer trying to negotiate all the media streams available in the
+ * mountpoint; in case the viewer knows they don't support one of the
+ * mountpoint codecs, though (e.g., the video in the mountpoint is VP8,
+ * but they only support H.264), or are not interested in getting all the
+ * media (e.g., they're ok with just audio and not video, or don't have
+ * enough bandwidth for both), they can use those properties to shape the
+ * SDP offer to their needs. The \c media array is optional, as a missing
+ * or empty array will simply be interpreted as a willingness to subscribe
+ * to all the streams in the mountpoint, which is the default behaviour.
+ * Notice that the order of the mids in the \c media array is irrelevant,
+ * as is how many times the same mid is listed in the array: the presence
+ * of a mid is just interpreted as an "on" switch for that stream, meaning
+ * it will be offered in the SDP.
+ *
+ * \note For backwards compatibility, the deprecated \c offer_audio ,
+ * \c offer_video and \c offer_data properties are also available. They
+ * also allow you to only subscribe to a subset of the mountpoint media,
+ * but with a more crude approach: specifically, they dictate whether or
+ * not any audio, video or data stream should be offered or not.
  *
  * As anticipated, if successful this request will generate a new JSEP SDP
  * offer, which will be attached to a \c preparing status event:
@@ -588,9 +710,8 @@ rtspiface = network interface IP address or device name to listen on when receiv
 \verbatim
 {
 	"request" : "configure",
-	"audio" : <true|false, depending on whether audio should be relayed or not; optional>,
-	"video" : <true|false, depending on whether video should be relayed or not; optional>,
-	"data" : <true|false, depending on whether datachannel messages should be relayed or not; optional>,
+	"mid" : <mid of the m-line to refer to for this configure request; optional>,
+	"send" : <true|false, depending on whether the media addressed by the above mid should be relayed or not; optional>,
 	"substream" : <substream to receive (0-2), in case simulcasting is enabled; optional>,
 	"temporal" : <temporal layers to receive (0-2), in case simulcasting is enabled; optional>,
 	"spatial_layer" : <spatial layer to receive (0-1), in case VP9-SVC is enabled; optional>,
@@ -598,16 +719,20 @@ rtspiface = network interface IP address or device name to listen on when receiv
 }
 \endverbatim
  *
- * As you can see, the \c audio , \c video and \c data properties can be
- * used as a media-level pause/resume functionality, whereas \c pause
- * and \c start simply pause and resume all streams at the same time.
- * The \c substream and \c temporal properties, instead, only make sense
+ * While the deprecated \c audio , \c video and \c data properties can still be
+ * used as a media-level pause/resume functionality, a better option is to
+ * specify the \c mid of the stream instead, and a \c send boolean property
+ * to specify if this specific stream should be relayed or not. The \c pause
+ * and \c start requests instead pause and resume all streams at the same time.
+ * The \c substream and \c temporal properties, finally, only make sense
  * when the mountpoint is configured with video simulcasting support, and
  * as such the viewer is interested in receiving a specific substream
  * or temporal layer, rather than any other of the available ones.
  * The \c spatial_layer and \c temporal_layer have exactly the same meaning,
  * but within the context of VP9-SVC mountpoints, and will have no effect
- * on mountpoints involving a different video codec.
+ * on mountpoints involving a different video codec. In both cases, make
+ * sure you specify the \c mid of the stream in case multiple videos are
+ * available in a mountpoint, or the request may have no effect.
  *
  * Another interesting feature in the Streaming plugin is the so-called
  * mountpoint "switching". Basically, when subscribed to a specific
@@ -699,8 +824,8 @@ rtspiface = network interface IP address or device name to listen on when receiv
 
 
 /* Plugin information */
-#define JANUS_STREAMING_VERSION			8
-#define JANUS_STREAMING_VERSION_STRING	"0.0.8"
+#define JANUS_STREAMING_VERSION			9
+#define JANUS_STREAMING_VERSION_STRING	"0.0.9"
 #define JANUS_STREAMING_DESCRIPTION		"This is a streaming plugin for Janus, allowing WebRTC peers to watch/listen to pre-recorded files or media generated by an external source."
 #define JANUS_STREAMING_NAME			"JANUS Streaming plugin"
 #define JANUS_STREAMING_AUTHOR			"Meetecho s.r.l."
@@ -721,8 +846,8 @@ void janus_streaming_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_streaming_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 json_t *janus_streaming_handle_admin_message(json_t *message);
 void janus_streaming_setup_media(janus_plugin_session *handle);
-void janus_streaming_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_streaming_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_streaming_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_streaming_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_streaming_hangup_media(janus_plugin_session *handle);
 void janus_streaming_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_streaming_query_session(janus_plugin_session *handle);
@@ -761,154 +886,186 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter id_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter watch_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"pin", JSON_STRING, 0},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"pin", JANUS_JSON_STRING, 0},
+	{"media", JANUS_JSON_ARRAY, 0},
+	{"restart", JANUS_JSON_BOOL, 0},
+	/* Deprecated parameters: still there for backwards compatibility */
 	{"offer_audio", JANUS_JSON_BOOL, 0},
 	{"offer_video", JANUS_JSON_BOOL, 0},
-	{"offer_data", JANUS_JSON_BOOL, 0},
-	{"restart", JANUS_JSON_BOOL, 0}
+	{"offer_data", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter adminkey_parameters[] = {
-	{"admin_key", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"admin_key", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter edit_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"new_description", JSON_STRING, 0},
-	{"new_secret", JSON_STRING, 0},
-	{"new_pin", JSON_STRING, 0},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"new_description", JANUS_JSON_STRING, 0},
+	{"new_secret", JANUS_JSON_STRING, 0},
+	{"new_pin", JANUS_JSON_STRING, 0},
 	{"new_is_private", JANUS_JSON_BOOL, 0},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter create_parameters[] = {
-	{"type", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"secret", JSON_STRING, 0},
-	{"pin", JSON_STRING, 0},
+	{"type", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"pin", JANUS_JSON_STRING, 0},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter rtp_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"name", JSON_STRING, 0},
-	{"description", JSON_STRING, 0},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"name", JANUS_JSON_STRING, 0},
+	{"description", JANUS_JSON_STRING, 0},
 	{"is_private", JANUS_JSON_BOOL, 0},
-	{"audio", JANUS_JSON_BOOL, 0},
-	{"video", JANUS_JSON_BOOL, 0},
-	{"data", JANUS_JSON_BOOL, 0},
-	{"collision", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"threads", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"srtpsuite", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"srtpcrypto", JSON_STRING, 0}
+	{"media", JANUS_JSON_ARRAY, 0},
+	{"audio", JANUS_JSON_BOOL, 0},		/* Deprecated */
+	{"video", JANUS_JSON_BOOL, 0},		/* Deprecated */
+	{"data", JANUS_JSON_BOOL, 0},		/* Deprecated */
+	{"collision", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"threads", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"srtpsuite", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"srtpcrypto", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter live_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"name", JSON_STRING, 0},
-	{"description", JSON_STRING, 0},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"name", JANUS_JSON_STRING, 0},
+	{"description", JANUS_JSON_STRING, 0},
 	{"is_private", JANUS_JSON_BOOL, 0},
-	{"filename", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"filename", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"audio", JANUS_JSON_BOOL, 0},
 	{"video", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter ondemand_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"name", JSON_STRING, 0},
-	{"description", JSON_STRING, 0},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"name", JANUS_JSON_STRING, 0},
+	{"description", JANUS_JSON_STRING, 0},
 	{"is_private", JANUS_JSON_BOOL, 0},
-	{"filename", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"filename", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
 	{"audio", JANUS_JSON_BOOL, 0},
 	{"video", JANUS_JSON_BOOL, 0}
 };
 #ifdef HAVE_LIBCURL
 static struct janus_json_parameter rtsp_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"name", JSON_STRING, 0},
-	{"description", JSON_STRING, 0},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"name", JANUS_JSON_STRING, 0},
+	{"description", JANUS_JSON_STRING, 0},
 	{"is_private", JANUS_JSON_BOOL, 0},
-	{"url", JSON_STRING, 0},
-	{"rtsp_user", JSON_STRING, 0},
-	{"rtsp_pwd", JSON_STRING, 0},
+	{"url", JANUS_JSON_STRING, 0},
+	{"rtsp_user", JANUS_JSON_STRING, 0},
+	{"rtsp_pwd", JANUS_JSON_STRING, 0},
 	{"audio", JANUS_JSON_BOOL, 0},
-	{"audiortpmap", JSON_STRING, 0},
-	{"audiofmtp", JSON_STRING, 0},
+	{"audiortpmap", JANUS_JSON_STRING, 0},
+	{"audiofmtp", JANUS_JSON_STRING, 0},
 	{"audiopt", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"video", JANUS_JSON_BOOL, 0},
-	{"videortpmap", JSON_STRING, 0},
-	{"videofmtp", JSON_STRING, 0},
+	{"videortpmap", JANUS_JSON_STRING, 0},
+	{"videofmtp", JANUS_JSON_STRING, 0},
 	{"videopt", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"rtspiface", JSON_STRING, 0},
+	{"rtspiface", JANUS_JSON_STRING, 0},
 	{"rtsp_failcheck", JANUS_JSON_BOOL, 0}
 };
 #endif
+static struct janus_json_parameter rtp_media_parameters[] = {
+	{"type", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"mid", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"label", JANUS_JSON_STRING, 0},
+	{"mcast", JANUS_JSON_STRING, 0},
+	{"iface", JANUS_JSON_STRING, 0},
+	{"port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"rtcpport", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"pt", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"rtpmap", JANUS_JSON_STRING, 0},
+	{"fmtp", JANUS_JSON_STRING, 0},
+	{"skew", JANUS_JSON_BOOL, 0},
+	/* Video only */
+	{"bufferkf", JANUS_JSON_BOOL, 0},
+	{"simulcast", JANUS_JSON_BOOL, 0},
+	{"port2", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"port3", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"svc", JANUS_JSON_BOOL, 0},
+	/* Data only */
+	{"buffermsg", JANUS_JSON_BOOL, 0},
+};
 static struct janus_json_parameter rtp_audio_parameters[] = {
-	{"audiomcast", JSON_STRING, 0},
-	{"audioport", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"audiortcpport", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"audiopt", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"audiortpmap", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"audiofmtp", JSON_STRING, 0},
-	{"audioiface", JSON_STRING, 0},
+	/* This syntax is deprecated */
+	{"audiomcast", JANUS_JSON_STRING, 0},
+	{"audioport", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"audiortcpport", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audiopt", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"audiortpmap", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"audiofmtp", JANUS_JSON_STRING, 0},
+	{"audioiface", JANUS_JSON_STRING, 0},
 	{"audioskew", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter rtp_video_parameters[] = {
-	{"videomcast", JSON_STRING, 0},
-	{"videoport", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"videortcpport", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"videopt", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"videortpmap", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"videofmtp", JSON_STRING, 0},
+	/* This syntax is deprecated */
+	{"videomcast", JANUS_JSON_STRING, 0},
+	{"videoport", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"videortcpport", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"videopt", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"videortpmap", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"videofmtp", JANUS_JSON_STRING, 0},
 	{"videobufferkf", JANUS_JSON_BOOL, 0},
-	{"videoiface", JSON_STRING, 0},
+	{"videoiface", JANUS_JSON_STRING, 0},
 	{"videosimulcast", JANUS_JSON_BOOL, 0},
-	{"videoport2", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"videoport3", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"videoport2", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"videoport3", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"videoskew", JANUS_JSON_BOOL, 0},
 	{"videosvc", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter rtp_data_parameters[] = {
-	{"dataport", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	/* This syntax is deprecated */
+	{"dataport", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
 	{"databuffermsg", JANUS_JSON_BOOL, 0},
-	{"dataiface", JSON_STRING, 0}
+	{"datamcast", JANUS_JSON_STRING, 0},
+	{"dataiface", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter destroy_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter recording_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"action", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"action", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter recording_start_parameters[] = {
-	{"audio", JSON_STRING, 0},
-	{"video", JSON_STRING, 0},
-	{"data", JSON_STRING, 0}
+	{"audio", JANUS_JSON_STRING, 0},	/* Deprecated */
+	{"video", JANUS_JSON_STRING, 0},	/* Deprecated */
+	{"data", JANUS_JSON_STRING, 0},		/* Deprecated */
+	{"media", JANUS_JSON_ARRAY, 0}
 };
 static struct janus_json_parameter recording_stop_parameters[] = {
-	{"audio", JANUS_JSON_BOOL, 0},
-	{"video", JANUS_JSON_BOOL, 0},
-	{"data", JANUS_JSON_BOOL, 0}
+	{"audio", JANUS_JSON_BOOL, 0},		/* Deprecated */
+	{"video", JANUS_JSON_BOOL, 0},		/* Deprecated */
+	{"data", JANUS_JSON_BOOL, 0},		/* Deprecated */
+	{"media", JANUS_JSON_ARRAY, 0}
 };
 static struct janus_json_parameter simulcast_parameters[] = {
-	{"substream", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"temporal", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"substream", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"temporal", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter svc_parameters[] = {
-	{"spatial_layer", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"temporal_layer", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"spatial_layer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"temporal_layer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter configure_parameters[] = {
-	{"audio", JANUS_JSON_BOOL, 0},
-	{"video", JANUS_JSON_BOOL, 0},
-	{"data", JANUS_JSON_BOOL, 0},
+	{"audio", JANUS_JSON_BOOL, 0},	/* Deprecated */
+	{"video", JANUS_JSON_BOOL, 0},	/* Deprecated */
+	{"data", JANUS_JSON_BOOL, 0},	/* Deprecated */
+	{"mid", JANUS_JSON_STRING, 0},
+	{"send", JANUS_JSON_BOOL, 0},
 	/* For VP8 (or H.264) simulcast */
-	{"substream", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"temporal", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"substream", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"temporal", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	/* For VP9 SVC */
-	{"spatial_layer", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"temporal_layer", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+	{"spatial_layer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"temporal_layer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter disable_parameters[] = {
 	{"stop_recording", JANUS_JSON_BOOL, 0}
@@ -964,6 +1121,7 @@ typedef struct janus_streaming_rtp_keyframe {
 } janus_streaming_rtp_keyframe;
 
 typedef struct janus_streaming_rtp_relay_packet {
+	int mindex;
 	janus_rtp_header *data;
 	gint length;
 	gboolean is_rtp;	/* This may be a data packet and not RTP */
@@ -997,39 +1155,22 @@ typedef struct janus_streaming_buffer {
 } janus_streaming_buffer;
 #endif
 
+typedef struct janus_streaming_codecs {
+	gint pt;
+	char *rtpmap;
+	char *fmtp;
+	janus_videocodec video_codec;
+} janus_streaming_codecs;
+
 typedef struct janus_streaming_rtp_source {
-	gint audio_port, remote_audio_port;
-	gint audio_rtcp_port, remote_audio_rtcp_port;
-	in_addr_t audio_mcast;
-	gint video_port[3], remote_video_port;
-	gint video_rtcp_port, remote_video_rtcp_port;
-	in_addr_t video_mcast;
-	gint data_port;
-	janus_recorder *arc;	/* The Janus recorder instance for this streams's audio, if enabled */
-	janus_recorder *vrc;	/* The Janus recorder instance for this streams's video, if enabled */
-	janus_recorder *drc;	/* The Janus recorder instance for this streams's data, if enabled */
-	janus_mutex rec_mutex;	/* Mutex to protect the recorders from race conditions */
-	janus_rtp_switching_context context[3];
-	int audio_fd;
-	int video_fd[3];
-	int data_fd;
-	int pipefd[2];			/* Just needed to quickly interrupt the poll when it's time to wrap up */
-	int audio_rtcp_fd;
-	int video_rtcp_fd;
-	gboolean simulcast;
-	gboolean svc;
-	gboolean askew, vskew;
-	gint64 last_received_audio;
-	gint64 last_received_video;
-	gint64 last_received_data;
-	uint32_t audio_ssrc;		/* Only needed for fixing outgoing RTCP packets */
-	uint32_t video_ssrc;		/* Only needed for fixing outgoing RTCP packets */
-	volatile gint need_pli;		/* Whether we need to send a PLI later */
-	volatile gint sending_pli;	/* Whether we're currently sending a PLI */
-	gint64 pli_latest;			/* Time of latest sent PLI (to avoid flooding) */
+	GList *media;				/* List of media streams in this RTP source (audio, video and/or data) */
+	GHashTable *media_byid;		/* As above, indexed by mindex */
+	GHashTable *media_byfd;		/* As above, indexed by file descriptor */
+	janus_mutex rec_mutex;		/* Mutex to protect the recorders of all media streams from race conditions */
+	int pipefd[2];				/* Just needed to quickly interrupt the poll when it's time to wrap up */
+	int rtp_collision;			/* Whether we should take care of potential RTP collisions */
 	uint32_t lowest_bitrate;	/* Lowest bitrate received by viewers via REMB since last update */
 	gint64 remb_latest;			/* Time of latest sent REMB (to avoid flooding) */
-	struct sockaddr_storage audio_rtcp_addr, video_rtcp_addr;
 #ifdef HAVE_LIBCURL
 	gboolean rtsp;
 	CURL *curl;
@@ -1038,19 +1179,12 @@ typedef struct janus_streaming_rtp_source {
 	char *rtsp_username, *rtsp_password;
 	int ka_timeout;
 	char *rtsp_ahost, *rtsp_vhost;
+	janus_streaming_codecs rtsp_acodecs, rtsp_vcodecs;
 	gboolean reconnecting;
 	gint64 reconnect_timer;
 	janus_mutex rtsp_mutex;
 #endif
-	janus_streaming_rtp_keyframe keyframe;
-	gboolean buffermsg;
-	int rtp_collision;
-	void *last_msg;
-	janus_mutex buffermsg_mutex;
-	janus_network_address audio_iface;
-	janus_network_address video_iface;
-	janus_network_address data_iface;
-	/* Only needed for SRTP forwarders */
+	/* Only needed for SRTP mountpoints */
 	gboolean is_srtp;
 	int srtpsuite;
 	char *srtpcrypto;
@@ -1058,8 +1192,64 @@ typedef struct janus_streaming_rtp_source {
 	srtp_policy_t srtp_policy;
 } janus_streaming_rtp_source;
 
+typedef enum janus_streaming_media {
+	JANUS_STREAMING_MEDIA_NONE = 0,
+	JANUS_STREAMING_MEDIA_AUDIO,
+	JANUS_STREAMING_MEDIA_VIDEO,
+	JANUS_STREAMING_MEDIA_DATA
+} janus_streaming_media;
+static const char *janus_streaming_media_str(janus_streaming_media type) {
+	switch(type) {
+		case JANUS_STREAMING_MEDIA_AUDIO: return "audio";
+		case JANUS_STREAMING_MEDIA_VIDEO: return "video";
+		case JANUS_STREAMING_MEDIA_DATA: return "data";
+		case JANUS_STREAMING_MEDIA_NONE:
+		default:
+			break;
+	}
+	return NULL;
+}
+
+typedef struct janus_streaming_rtp_source_stream {
+	int mindex;
+	janus_streaming_media type;
+	char *mid, *label;
+	janus_streaming_codecs codecs;
+	gint port[3], remote_port;
+	gint rtcp_port, remote_rtcp_port;
+	in_addr_t mcast;
+	char *mcast_str;
+	janus_network_address iface;
+	char *iface_str;
+	janus_recorder *rc;	/* The Janus recorder instance for this streams, if enabled */
+	janus_rtp_switching_context context[3];
+	int fd[3];
+	int rtcp_fd;
+	gboolean simulcast;
+	gboolean svc;
+	gboolean skew;
+	gint64 last_received;
+	uint32_t ssrc;				/* Only needed for fixing outgoing RTCP packets */
+	uint32_t last_ssrc[3];		/* Only needed for detecting new sources */
+	volatile gint need_pli;		/* Whether we need to send a PLI later */
+	volatile gint sending_pli;	/* Whether we're currently sending a PLI */
+	gint64 pli_latest;			/* Time of latest sent PLI (to avoid flooding) */
+	struct sockaddr_storage rtcp_addr;
+	janus_streaming_rtp_keyframe keyframe;
+	gboolean buffermsg;
+	void *last_msg;
+	janus_mutex buffermsg_mutex;
+	janus_refcount ref;
+} janus_streaming_rtp_source_stream;
+static void janus_streaming_rtp_source_stream_unref(janus_streaming_rtp_source_stream *stream) {
+	/* Decrease the counter */
+	if(stream)
+		janus_refcount_decrease(&stream->ref);
+}
+
 typedef struct janus_streaming_file_source {
 	char *filename;
+	janus_streaming_codecs codecs;
 } janus_streaming_file_source;
 
 /* used for audio/video fd and RTCP fd */
@@ -1068,16 +1258,6 @@ typedef struct multiple_fds {
 	int rtcp_fd;
 } multiple_fds;
 
-typedef struct janus_streaming_codecs {
-	gint audio_pt;
-	char *audio_rtpmap;
-	char *audio_fmtp;
-	janus_videocodec video_codec;
-	gint video_pt;
-	char *video_rtpmap;
-	char *video_fmtp;
-} janus_streaming_codecs;
-
 typedef struct janus_streaming_mountpoint {
 	guint64 id;
 	char *name;
@@ -1087,13 +1267,12 @@ typedef struct janus_streaming_mountpoint {
 	char *pin;
 	gboolean enabled;
 	gboolean active;
+	gboolean audio, video, data;
 	GThread *thread;	/* A mountpoint may or may not have a thread */
 	janus_streaming_type streaming_type;
 	janus_streaming_source streaming_source;
 	void *source;	/* Can differ according to the source type */
 	GDestroyNotify source_destroy;
-	janus_streaming_codecs codecs;
-	gboolean audio, video, data;
 	GList *viewers;
 	int helper_threads;		/* Only relevant for RTP mountpoints */
 	GList *threads;			/* Only relevant for RTP mountpoints */
@@ -1118,26 +1297,27 @@ typedef struct janus_streaming_helper {
 static void *janus_streaming_helper_thread(void *data);
 static void janus_streaming_helper_rtprtcp_packet(gpointer data, gpointer user_data);
 
-/* Helper to create an RTP live source (e.g., from gstreamer/ffmpeg/vlc/etc.) */
+/* Helpers to create an RTP live source (e.g., from gstreamer/ffmpeg/vlc/etc.) */
+janus_streaming_rtp_source_stream *janus_streaming_create_rtp_source_stream(
+	const char *name, int mindex, const char *type, const char *mid, const char *label,
+	char *mcast, char *miface, const janus_network_address *iface, uint16_t port, uint16_t port2, uint16_t port3, uint16_t rtcpport,
+	uint8_t codec, const char *rtpmap, const char *fmtp, gboolean doskew,
+	gboolean bufferkf, gboolean buffermsg, gboolean simulcast, gboolean svc);
 janus_streaming_mountpoint *janus_streaming_create_rtp_source(
-		uint64_t id, char *name, char *desc,
-		int srtpsuite, char *srtpcrypto, int threads,
-		gboolean doaudio, char *amcast, const janus_network_address *aiface, uint16_t aport, uint16_t artcpport, uint8_t acodec, char *artpmap, char *afmtp, gboolean doaskew,
-		gboolean dovideo, char *vmcast, const janus_network_address *viface, uint16_t vport, uint16_t vrtcpport, uint8_t vcodec, char *vrtpmap, char *vfmtp, gboolean bufferkf,
-			gboolean simulcast, uint16_t vport2, uint16_t vport3, gboolean svc, gboolean dovskew, int rtp_collision,
-		gboolean dodata, const janus_network_address *diface, uint16_t dport, gboolean buffermsg);
+	uint64_t id, char *name, char *desc, GList *media,
+	int srtpsuite, char *srtpcrypto, int threads, int rtp_collision);
 /* Helper to create a file/ondemand live source */
 janus_streaming_mountpoint *janus_streaming_create_file_source(
-		uint64_t id, char *name, char *desc, char *filename,
-		gboolean live, gboolean doaudio, gboolean dovideo);
+	uint64_t id, char *name, char *desc, char *filename,
+	gboolean live, gboolean doaudio, gboolean dovideo);
 /* Helper to create a rtsp live source */
 janus_streaming_mountpoint *janus_streaming_create_rtsp_source(
-		uint64_t id, char *name, char *desc,
-		char *url, char *username, char *password,
-		gboolean doaudio, int audiopt, char *artpmap, char *afmtp,
-		gboolean dovideo, int videopt, char *vrtpmap, char *vfmtp,
-		const janus_network_address *iface,
-		gboolean error_on_failure);
+	uint64_t id, char *name, char *desc,
+	char *url, char *username, char *password,
+	gboolean doaudio, int audiopt, char *artpmap, char *afmtp,
+	gboolean dovideo, int videopt, char *vrtpmap, char *vfmtp,
+	const janus_network_address *iface,
+	gboolean error_on_failure);
 
 
 typedef struct janus_streaming_message {
@@ -1149,14 +1329,10 @@ typedef struct janus_streaming_message {
 static GAsyncQueue *messages = NULL;
 static janus_streaming_message exit_message;
 
-typedef struct janus_streaming_session {
-	janus_plugin_session *handle;
-	janus_streaming_mountpoint *mountpoint;
-	gint64 sdp_sessid;
-	gint64 sdp_version;
-	gboolean started;
-	gboolean paused;
-	gboolean audio, video, data;		/* Whether audio, video and/or data must be sent to this listener */
+typedef struct janus_streaming_session_stream {
+	int mindex;				/* The media index of this stream (may not be the same as the mountpoint stream) */
+	janus_streaming_rtp_source_stream *stream;
+	gboolean send;			/* Whether this stream media must be sent to this subscriber */
 	janus_rtp_switching_context context;
 	janus_rtp_simulcasting_context sim_context;
 	janus_vp8_simulcast_context vp8_context;
@@ -1164,6 +1340,22 @@ typedef struct janus_streaming_session {
 	 * simulcast, which has similar info (substream/templayer) but in a completely different context */
 	int spatial_layer, target_spatial_layer;
 	int temporal_layer, target_temporal_layer;
+} janus_streaming_session_stream;
+static void janus_streaming_session_stream_free(janus_streaming_session_stream *s) {
+	if(s && s->stream)
+		janus_streaming_rtp_source_stream_unref(s->stream);
+	g_free(s);
+}
+
+typedef struct janus_streaming_session {
+	janus_plugin_session *handle;
+	janus_streaming_mountpoint *mountpoint;
+	gint64 sdp_sessid;
+	gint64 sdp_version;
+	gboolean started;
+	gboolean paused;
+	GList *streams;				/* List of streams this session is subscribed to */
+	GHashTable *streams_byid;	/* Map of streams this session is subscribed to, indexed by mountpoint mindex */
 	gboolean stopping;
 	volatile gint hangingup;
 	volatile gint destroyed;
@@ -1234,11 +1426,6 @@ static void janus_streaming_mountpoint_free(const janus_refcount *mp_ref) {
 		mp->source_destroy(mp->source);
 	}
 
-	g_free(mp->codecs.audio_rtpmap);
-	g_free(mp->codecs.audio_fmtp);
-	g_free(mp->codecs.video_rtpmap);
-	g_free(mp->codecs.video_fmtp);
-
 	g_free(mp);
 }
 
@@ -1266,41 +1453,41 @@ static void janus_streaming_message_free(janus_streaming_message *msg) {
 
 
 /* Helper method to send an RTCP PLI */
-static void janus_streaming_rtcp_pli_send(janus_streaming_rtp_source *source) {
-	if(source == NULL || source->video_rtcp_fd < 0 || source->video_rtcp_addr.ss_family == 0)
+static void janus_streaming_rtcp_pli_send(janus_streaming_rtp_source_stream *stream) {
+	if(stream == NULL || stream->rtcp_fd < 0 || stream->rtcp_addr.ss_family == 0)
 		return;
-	if(!g_atomic_int_compare_and_exchange(&source->sending_pli, 0, 1))
+	if(!g_atomic_int_compare_and_exchange(&stream->sending_pli, 0, 1))
 		return;
 	gint64 now = janus_get_monotonic_time();
-	if(now - source->pli_latest < G_USEC_PER_SEC) {
+	if(now - stream->pli_latest < G_USEC_PER_SEC) {
 		/* We just sent a PLI less than a second ago, schedule a new delivery later */
-		g_atomic_int_set(&source->need_pli, 1);
-		g_atomic_int_set(&source->sending_pli, 0);
+		g_atomic_int_set(&stream->need_pli, 1);
+		g_atomic_int_set(&stream->sending_pli, 0);
 		return;
 	}
 	/* Update the time of when we last sent a keyframe request */
-	g_atomic_int_set(&source->need_pli, 0);
-	source->pli_latest = janus_get_monotonic_time();
+	g_atomic_int_set(&stream->need_pli, 0);
+	stream->pli_latest = janus_get_monotonic_time();
 	JANUS_LOG(LOG_HUGE, "Sending PLI\n");
 	/* Generate a PLI */
 	char rtcp_buf[12];
 	int rtcp_len = 12;
 	janus_rtcp_pli((char *)&rtcp_buf, rtcp_len);
-	janus_rtcp_fix_ssrc(NULL, rtcp_buf, rtcp_len, 1, 1, source->video_ssrc);
+	janus_rtcp_fix_ssrc(NULL, rtcp_buf, rtcp_len, 1, 1, stream->ssrc);
 	/* Send the packet */
 	int sent = 0;
-	if((sent = sendto(source->video_rtcp_fd, rtcp_buf, rtcp_len, 0,
-			(struct sockaddr *)&source->video_rtcp_addr, sizeof(source->video_rtcp_addr))) < 0) {
+	if((sent = sendto(stream->rtcp_fd, rtcp_buf, rtcp_len, 0,
+			(struct sockaddr *)&stream->rtcp_addr, sizeof(stream->rtcp_addr))) < 0) {
 		JANUS_LOG(LOG_ERR, "Error in sendto... %d (%s)\n", errno, strerror(errno));
 	} else {
 		JANUS_LOG(LOG_HUGE, "Sent %d/%d bytes\n", sent, rtcp_len);
 	}
-	g_atomic_int_set(&source->sending_pli, 0);
+	g_atomic_int_set(&stream->sending_pli, 0);
 }
 
 /* Helper method to send an RTCP REMB */
-static void janus_streaming_rtcp_remb_send(janus_streaming_rtp_source *source) {
-	if(source == NULL || source->video_rtcp_fd < 0 || source->video_rtcp_addr.ss_family == 0)
+static void janus_streaming_rtcp_remb_send(janus_streaming_rtp_source *source, janus_streaming_rtp_source_stream *stream) {
+	if(source == NULL || stream == NULL || stream->rtcp_fd < 0 || stream->rtcp_addr.ss_family == 0)
 		return;
 	/* Update the time of when we last sent REMB feedback */
 	source->remb_latest = janus_get_monotonic_time();
@@ -1308,14 +1495,14 @@ static void janus_streaming_rtcp_remb_send(janus_streaming_rtp_source *source) {
 	char rtcp_buf[24];
 	int rtcp_len = 24;
 	janus_rtcp_remb((char *)(&rtcp_buf), rtcp_len, source->lowest_bitrate);
-	janus_rtcp_fix_ssrc(NULL, rtcp_buf, rtcp_len, 1, 1, source->video_ssrc);
+	janus_rtcp_fix_ssrc(NULL, rtcp_buf, rtcp_len, 1, 1, stream->ssrc);
 	JANUS_LOG(LOG_HUGE, "Sending REMB: %"SCNu32"\n", source->lowest_bitrate);
 	/* Reset the lowest bitrate */
 	source->lowest_bitrate = 0;
 	/* Send the packet */
 	int sent = 0;
-	if((sent = sendto(source->video_rtcp_fd, rtcp_buf, rtcp_len, 0,
-			(struct sockaddr *)&source->video_rtcp_addr, sizeof(source->video_rtcp_addr))) < 0) {
+	if((sent = sendto(stream->rtcp_fd, rtcp_buf, rtcp_len, 0,
+			(struct sockaddr *)&stream->rtcp_addr, sizeof(stream->rtcp_addr))) < 0) {
 		JANUS_LOG(LOG_ERR, "Error in sendto... %d (%s)\n", errno, strerror(errno));
 	} else {
 		JANUS_LOG(LOG_HUGE, "Sent %d/%d bytes\n", sent, rtcp_len);
@@ -1436,143 +1623,332 @@ int janus_streaming_init(janus_callbacks *callback, const char *config_path) {
 				continue;
 			}
 			if(!strcasecmp(type->value, "rtp")) {
-				janus_network_address video_iface, audio_iface, data_iface;
 				/* RTP live source (e.g., from gstreamer/ffmpeg/vlc/etc.) */
+				GList *streams = NULL;
 				janus_config_item *id = janus_config_get(config, cat, janus_config_type_item, "id");
 				janus_config_item *desc = janus_config_get(config, cat, janus_config_type_item, "description");
 				janus_config_item *priv = janus_config_get(config, cat, janus_config_type_item, "is_private");
 				janus_config_item *secret = janus_config_get(config, cat, janus_config_type_item, "secret");
 				janus_config_item *pin = janus_config_get(config, cat, janus_config_type_item, "pin");
-				janus_config_item *audio = janus_config_get(config, cat, janus_config_type_item, "audio");
-				janus_config_item *askew = janus_config_get(config, cat, janus_config_type_item, "audioskew");
-				janus_config_item *video = janus_config_get(config, cat, janus_config_type_item, "video");
-				janus_config_item *vskew = janus_config_get(config, cat, janus_config_type_item, "videoskew");
-				janus_config_item *vsvc = janus_config_get(config, cat, janus_config_type_item, "videosvc");
-				janus_config_item *data = janus_config_get(config, cat, janus_config_type_item, "data");
-				janus_config_item *diface = janus_config_get(config, cat, janus_config_type_item, "dataiface");
-				janus_config_item *amcast = janus_config_get(config, cat, janus_config_type_item, "audiomcast");
-				janus_config_item *aiface = janus_config_get(config, cat, janus_config_type_item, "audioiface");
-				janus_config_item *aport = janus_config_get(config, cat, janus_config_type_item, "audioport");
-				janus_config_item *artcpport = janus_config_get(config, cat, janus_config_type_item, "audiortcpport");
-				janus_config_item *acodec = janus_config_get(config, cat, janus_config_type_item, "audiopt");
-				janus_config_item *artpmap = janus_config_get(config, cat, janus_config_type_item, "audiortpmap");
-				janus_config_item *afmtp = janus_config_get(config, cat, janus_config_type_item, "audiofmtp");
-				janus_config_item *vmcast = janus_config_get(config, cat, janus_config_type_item, "videomcast");
-				janus_config_item *viface = janus_config_get(config, cat, janus_config_type_item, "videoiface");
-				janus_config_item *vport = janus_config_get(config, cat, janus_config_type_item, "videoport");
-				janus_config_item *vrtcpport = janus_config_get(config, cat, janus_config_type_item, "videortcpport");
-				janus_config_item *vcodec = janus_config_get(config, cat, janus_config_type_item, "videopt");
-				janus_config_item *vrtpmap = janus_config_get(config, cat, janus_config_type_item, "videortpmap");
-				janus_config_item *vfmtp = janus_config_get(config, cat, janus_config_type_item, "videofmtp");
-				janus_config_item *vkf = janus_config_get(config, cat, janus_config_type_item, "videobufferkf");
-				janus_config_item *vsc = janus_config_get(config, cat, janus_config_type_item, "videosimulcast");
-				janus_config_item *vport2 = janus_config_get(config, cat, janus_config_type_item, "videoport2");
-				janus_config_item *vport3 = janus_config_get(config, cat, janus_config_type_item, "videoport3");
-				janus_config_item *dport = janus_config_get(config, cat, janus_config_type_item, "dataport");
-				janus_config_item *dbm = janus_config_get(config, cat, janus_config_type_item, "databuffermsg");
+				janus_config_item *media = janus_config_get(config, cat, janus_config_type_array, "media");
 				janus_config_item *rtpcollision = janus_config_get(config, cat, janus_config_type_item, "collision");
 				janus_config_item *threads = janus_config_get(config, cat, janus_config_type_item, "threads");
 				janus_config_item *ssuite = janus_config_get(config, cat, janus_config_type_item, "srtpsuite");
 				janus_config_item *scrypto = janus_config_get(config, cat, janus_config_type_item, "srtpcrypto");
-				gboolean is_private = priv && priv->value && janus_is_true(priv->value);
-				gboolean doaudio = audio && audio->value && janus_is_true(audio->value);
-				gboolean doaskew = audio && askew && askew->value && janus_is_true(askew->value);
-				gboolean dovideo = video && video->value && janus_is_true(video->value);
-				gboolean dovskew = video && vskew && vskew->value && janus_is_true(vskew->value);
-				gboolean dosvc = video && vsvc && vsvc->value && janus_is_true(vsvc->value);
-				gboolean dodata = data && data->value && janus_is_true(data->value);
-				gboolean bufferkf = video && vkf && vkf->value && janus_is_true(vkf->value);
-				gboolean simulcast = video && vsc && vsc->value && janus_is_true(vsc->value);
-				if(simulcast && bufferkf) {
-					/* FIXME We'll need to take care of this */
-					JANUS_LOG(LOG_WARN, "Simulcasting enabled, so disabling buffering of keyframes\n");
-					bufferkf = FALSE;
+				if(ssuite && ssuite->value && atoi(ssuite->value) != 32 && atoi(ssuite->value) != 80) {
+					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid SRTP suite...\n", cat->name);
+					cl = cl->next;
+					continue;
 				}
-				gboolean buffermsg = data && dbm && dbm->value && janus_is_true(dbm->value);
-				if(!doaudio && !dovideo && !dodata) {
-					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', no audio, video or data have to be streamed...\n", cat->name);
+				if(rtpcollision && rtpcollision->value && atoi(rtpcollision->value) < 0) {
+					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid collision configuration...\n", cat->name);
 					cl = cl->next;
 					continue;
 				}
-				if(doaudio &&
-						(aport == NULL || aport->value == NULL || atoi(aport->value) == 0 ||
-						acodec == NULL || acodec->value == NULL ||
-						artpmap == NULL || artpmap->value == NULL)) {
-					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', missing mandatory information for audio...\n", cat->name);
+				if(threads && threads->value && atoi(threads->value) < 0) {
+					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid threads configuration...\n", cat->name);
 					cl = cl->next;
 					continue;
 				}
-				if(doaudio && aiface) {
-					if(!ifas) {
-						JANUS_LOG(LOG_ERR, "Skipping 'rtp' stream '%s', it relies on network configuration but network device information is unavailable...\n", cat->name);
+				gboolean is_private = priv && priv->value && janus_is_true(priv->value);
+				/* How are we adding media? */
+				if(media != NULL) {
+					/* We're using the new media-based configuration, iterate on all media objects */
+					gboolean failed = FALSE;
+					GList *ml = media->list;
+					while(ml) {
+						janus_config_item *m = (janus_config_item *)ml->data;
+						if(m == NULL || m->type != janus_config_type_category) {
+							JANUS_LOG(LOG_WARN, "  -- Invalid media item (not a category?), skipping in '%s'...\n", cat->name);
+							ml = ml->next;
+							continue;
+						}
+						janus_config_item *type = janus_config_get(config, m, janus_config_type_item, "type");
+						if(type == NULL || type->value == NULL) {
+							JANUS_LOG(LOG_WARN, "  -- Invalid media type, skipping in '%s'...\n", cat->name);
+							ml = ml->next;
+							continue;
+						}
+						if(strcasecmp(type->value, "audio") && strcasecmp(type->value, "video") && strcasecmp(type->value, "data")) {
+							JANUS_LOG(LOG_WARN, "  -- Unsupported media type '%s', skipping in '%s'...\n", type->value, cat->name);
+							ml = ml->next;
+							continue;
+						}
+						gboolean audio = !strcasecmp(type->value, "audio");
+						gboolean video = !strcasecmp(type->value, "video");
+						gboolean data = !strcasecmp(type->value, "data");
+						/* We need mid and label for addressing this stream on the client side */
+						janus_config_item *mid = janus_config_get(config, m, janus_config_type_item, "mid");
+						if(mid == NULL || mid->value == NULL) {
+							JANUS_LOG(LOG_WARN, "  -- Missing media mid, skipping in '%s'...\n", cat->name);
+							ml = ml->next;
+							continue;
+						}
+						janus_config_item *label = janus_config_get(config, m, janus_config_type_item, "label");
+						/* These are the attributes we can configure per each media stream */
+						janus_network_address media_iface;
+						janus_config_item *iface = janus_config_get(config, m, janus_config_type_item, "iface");
+						janus_config_item *mcast = janus_config_get(config, m, janus_config_type_item, "mcast");
+						janus_config_item *port = janus_config_get(config, m, janus_config_type_item, "port");
+						janus_config_item *rtcpport = janus_config_get(config, m, janus_config_type_item, "rtcpport");
+						janus_config_item *codec = janus_config_get(config, m, janus_config_type_item, "pt");
+						janus_config_item *rtpmap = janus_config_get(config, m, janus_config_type_item, "rtpmap");
+						janus_config_item *fmtp = janus_config_get(config, m, janus_config_type_item, "fmtp");
+						janus_config_item *vkf = janus_config_get(config, m, janus_config_type_item, "bufferkf");
+						janus_config_item *vsc = janus_config_get(config, m, janus_config_type_item, "simulcast");
+						janus_config_item *dbm = janus_config_get(config, cat, janus_config_type_item, "buffermsg");
+						janus_config_item *vport2 = janus_config_get(config, m, janus_config_type_item, "port2");
+						janus_config_item *vport3 = janus_config_get(config, m, janus_config_type_item, "port3");
+						janus_config_item *vsvc = janus_config_get(config, m, janus_config_type_item, "svc");
+						janus_config_item *skew = janus_config_get(config, m, janus_config_type_item, "skew");
+						gboolean doskew = skew && skew->value && janus_is_true(skew->value);
+						gboolean dosvc = video && vsvc && vsvc->value && janus_is_true(vsvc->value);
+						gboolean bufferkf = video && vkf && vkf->value && janus_is_true(vkf->value);
+						gboolean simulcast = video && vsc && vsc->value && janus_is_true(vsc->value);
+						if(simulcast && bufferkf) {
+							/* FIXME We'll need to take care of this */
+							JANUS_LOG(LOG_WARN, "Simulcasting enabled, so disabling buffering of keyframes\n");
+							bufferkf = FALSE;
+						}
+						gboolean buffermsg = data && dbm && dbm->value && janus_is_true(dbm->value);
+						if((audio || video) &&
+								(port == NULL || port->value == NULL || atoi(port->value) == 0 ||
+								codec == NULL || codec->value == NULL ||
+								rtpmap == NULL || rtpmap->value == NULL)) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', missing mandatory information for audio/video stream...\n", cat->name);
+							failed = TRUE;
+							break;
+						} else if(data && (port == NULL || port->value == NULL || atoi(port->value) == 0)) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', missing mandatory information for data stream...\n", cat->name);
+							failed = TRUE;
+							break;
+						}
+						if(iface) {
+							if(!ifas) {
+								JANUS_LOG(LOG_ERR, "Can't add '%s' stream '%s', it relies on network configuration but network device information is unavailable...\n", type->value, cat->name);
+								failed = TRUE;
+								break;
+							}
+							if(janus_network_lookup_interface(ifas, iface->value, &media_iface) != 0) {
+								JANUS_LOG(LOG_ERR, "Can't add '%s' stream '%s', invalid network interface configuration for media stream...\n", type->value, cat->name);
+								failed = TRUE;
+								break;
+							}
+						}
+						/* Create the source stream */
+						janus_streaming_rtp_source_stream *stream = janus_streaming_create_rtp_source_stream(
+							cat->name, g_list_length(streams),
+							type->value, mid->value, (label && label->value ? label->value : type->value),
+							mcast ? (char *)mcast->value : NULL,
+							iface && iface->value ? (char *)iface->value : NULL,
+							iface && iface->value ? &media_iface : NULL,
+							(port && port->value) ? atoi(port->value) : 0,
+							(vport2 && vport2->value) ? atoi(vport2->value) : 0,
+							(vport3 && vport3->value) ? atoi(vport3->value) : 0,
+							(rtcpport && rtcpport->value) ? atoi(rtcpport->value) : 0,
+							(codec && codec->value) ? atoi(codec->value) : 0,
+							rtpmap ? (char *)rtpmap->value : NULL,
+							fmtp ? (char *)fmtp->value : NULL,
+							doskew, bufferkf, buffermsg, simulcast, dosvc);
+						if(stream == NULL) {
+							JANUS_LOG(LOG_ERR, "Can't add '%s' stream '%s', error creating source stream...\n", type->value, cat->name);
+							failed = TRUE;
+							break;
+						}
+						/* Add to the list of streams */
+						streams = g_list_append(streams, stream);
+						/* Go on */
+						ml = ml->next;
+					}
+					if(failed) {
 						cl = cl->next;
 						continue;
 					}
-					if(janus_network_lookup_interface(ifas, aiface->value, &audio_iface) != 0) {
-						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for audio...\n", cat->name);
+				} else {
+					/* If we got here, we create a mountpoint the "old" way */
+					janus_network_address video_iface, audio_iface, data_iface;
+					janus_config_item *audio = janus_config_get(config, cat, janus_config_type_item, "audio");
+					janus_config_item *askew = janus_config_get(config, cat, janus_config_type_item, "audioskew");
+					janus_config_item *video = janus_config_get(config, cat, janus_config_type_item, "video");
+					janus_config_item *vskew = janus_config_get(config, cat, janus_config_type_item, "videoskew");
+					janus_config_item *vsvc = janus_config_get(config, cat, janus_config_type_item, "videosvc");
+					janus_config_item *data = janus_config_get(config, cat, janus_config_type_item, "data");
+					janus_config_item *dmcast = janus_config_get(config, cat, janus_config_type_item, "datamcast");
+					janus_config_item *diface = janus_config_get(config, cat, janus_config_type_item, "dataiface");
+					janus_config_item *amcast = janus_config_get(config, cat, janus_config_type_item, "audiomcast");
+					janus_config_item *aiface = janus_config_get(config, cat, janus_config_type_item, "audioiface");
+					janus_config_item *aport = janus_config_get(config, cat, janus_config_type_item, "audioport");
+					janus_config_item *artcpport = janus_config_get(config, cat, janus_config_type_item, "audiortcpport");
+					janus_config_item *acodec = janus_config_get(config, cat, janus_config_type_item, "audiopt");
+					janus_config_item *artpmap = janus_config_get(config, cat, janus_config_type_item, "audiortpmap");
+					janus_config_item *afmtp = janus_config_get(config, cat, janus_config_type_item, "audiofmtp");
+					janus_config_item *vmcast = janus_config_get(config, cat, janus_config_type_item, "videomcast");
+					janus_config_item *viface = janus_config_get(config, cat, janus_config_type_item, "videoiface");
+					janus_config_item *vport = janus_config_get(config, cat, janus_config_type_item, "videoport");
+					janus_config_item *vrtcpport = janus_config_get(config, cat, janus_config_type_item, "videortcpport");
+					janus_config_item *vcodec = janus_config_get(config, cat, janus_config_type_item, "videopt");
+					janus_config_item *vrtpmap = janus_config_get(config, cat, janus_config_type_item, "videortpmap");
+					janus_config_item *vfmtp = janus_config_get(config, cat, janus_config_type_item, "videofmtp");
+					janus_config_item *vkf = janus_config_get(config, cat, janus_config_type_item, "videobufferkf");
+					janus_config_item *vsc = janus_config_get(config, cat, janus_config_type_item, "videosimulcast");
+					janus_config_item *vport2 = janus_config_get(config, cat, janus_config_type_item, "videoport2");
+					janus_config_item *vport3 = janus_config_get(config, cat, janus_config_type_item, "videoport3");
+					janus_config_item *dport = janus_config_get(config, cat, janus_config_type_item, "dataport");
+					janus_config_item *dbm = janus_config_get(config, cat, janus_config_type_item, "databuffermsg");
+					gboolean doaudio = audio && audio->value && janus_is_true(audio->value);
+					gboolean doaskew = audio && askew && askew->value && janus_is_true(askew->value);
+					gboolean dovideo = video && video->value && janus_is_true(video->value);
+					gboolean dovskew = video && vskew && vskew->value && janus_is_true(vskew->value);
+					gboolean dosvc = video && vsvc && vsvc->value && janus_is_true(vsvc->value);
+					gboolean dodata = data && data->value && janus_is_true(data->value);
+					gboolean bufferkf = video && vkf && vkf->value && janus_is_true(vkf->value);
+					gboolean simulcast = video && vsc && vsc->value && janus_is_true(vsc->value);
+					if(simulcast && bufferkf) {
+						/* FIXME We'll need to take care of this */
+						JANUS_LOG(LOG_WARN, "Simulcasting enabled, so disabling buffering of keyframes\n");
+						bufferkf = FALSE;
+					}
+					gboolean buffermsg = data && dbm && dbm->value && janus_is_true(dbm->value);
+					if(!doaudio && !dovideo && !dodata) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', no audio, video or data have to be streamed...\n", cat->name);
 						cl = cl->next;
 						continue;
 					}
-				}
-				if(dovideo &&
-						(vport == NULL || vport->value == NULL || atoi(vport->value) == 0 ||
-						vcodec == NULL || vcodec->value == NULL ||
-						vrtpmap == NULL || vrtpmap->value == NULL)) {
-					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', missing mandatory information for video...\n", cat->name);
-					cl = cl->next;
-					continue;
-				}
-				if(dodata && (dport == NULL || dport->value == NULL || atoi(dport->value) == 0)) {
-					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', missing mandatory information for data...\n", cat->name);
-					cl = cl->next;
-					continue;
-				}
-#ifndef HAVE_SCTP
-				if(dodata) {
-					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s': no datachannels support......\n", cat->name);
-					cl = cl->next;
-					continue;
-				}
-#endif
-				if(dodata && diface) {
-					if(!ifas) {
-						JANUS_LOG(LOG_ERR, "Skipping 'rtp' stream '%s', it relies on network configuration but network device information is unavailable...\n", cat->name);
+					if(doaudio &&
+							(aport == NULL || aport->value == NULL || atoi(aport->value) == 0 ||
+							acodec == NULL || acodec->value == NULL ||
+							artpmap == NULL || artpmap->value == NULL)) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', missing mandatory information for audio...\n", cat->name);
 						cl = cl->next;
 						continue;
 					}
-					if(janus_network_lookup_interface(ifas, diface->value, &data_iface) != 0) {
-						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for data...\n", cat->name);
+					if(doaudio && aiface) {
+						if(!ifas) {
+							JANUS_LOG(LOG_ERR, "Skipping 'rtp' stream '%s', it relies on network configuration but network device information is unavailable...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+						if(janus_network_lookup_interface(ifas, aiface->value, &audio_iface) != 0) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for audio...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+					}
+					if(dovideo &&
+							(vport == NULL || vport->value == NULL || atoi(vport->value) == 0 ||
+							vcodec == NULL || vcodec->value == NULL ||
+							vrtpmap == NULL || vrtpmap->value == NULL)) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', missing mandatory information for video...\n", cat->name);
 						cl = cl->next;
 						continue;
 					}
-				}
-				if(dovideo && viface) {
-					if(!ifas) {
-						JANUS_LOG(LOG_ERR, "Skipping 'rtp' stream '%s', it relies on network configuration but network device information is unavailable...\n", cat->name);
+					if(dodata && (dport == NULL || dport->value == NULL || atoi(dport->value) == 0)) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', missing mandatory information for data...\n", cat->name);
 						cl = cl->next;
 						continue;
 					}
-					if(janus_network_lookup_interface(ifas, viface->value, &video_iface) != 0) {
-						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for video...\n", cat->name);
+#ifndef HAVE_SCTP
+					if(dodata) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s': no datachannels support......\n", cat->name);
 						cl = cl->next;
 						continue;
 					}
+#endif
+					if(dodata && diface) {
+						if(!ifas) {
+							JANUS_LOG(LOG_ERR, "Skipping 'rtp' stream '%s', it relies on network configuration but network device information is unavailable...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+						if(janus_network_lookup_interface(ifas, diface->value, &data_iface) != 0) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for data...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+					}
+					if(dovideo && viface) {
+						if(!ifas) {
+							JANUS_LOG(LOG_ERR, "Skipping 'rtp' stream '%s', it relies on network configuration but network device information is unavailable...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+						if(janus_network_lookup_interface(ifas, viface->value, &video_iface) != 0) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for video...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+					}
+					JANUS_LOG(LOG_VERB, "Audio %s, Video %s, Data %s\n",
+						doaudio ? "enabled" : "NOT enabled",
+						dovideo ? "enabled" : "NOT enabled",
+						dodata ? "enabled" : "NOT enabled");
+					/* Create the individual streams */
+					if(doaudio) {
+						/* Create the audio source stream */
+						janus_streaming_rtp_source_stream *stream = janus_streaming_create_rtp_source_stream(
+							cat->name, g_list_length(streams),
+							"audio", "a", "audio",
+							amcast ? (char *)amcast->value : NULL,
+							aiface && aiface->value ? (char *)aiface->value : NULL,
+							aiface && aiface->value ? &audio_iface : NULL,
+							(aport && aport->value) ? atoi(aport->value) : 0, 0, 0,
+							(artcpport && artcpport->value) ? atoi(artcpport->value) : 0,
+							(acodec && acodec->value) ? atoi(acodec->value) : 0,
+							artpmap ? (char *)artpmap->value : NULL,
+							afmtp ? (char *)afmtp->value : NULL,
+							doaskew, FALSE, FALSE, FALSE, FALSE);
+						if(stream == NULL) {
+							JANUS_LOG(LOG_ERR, "Skipping 'audio' stream '%s', error creating source stream...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+						/* Add to the list of streams */
+						streams = g_list_append(streams, stream);
+					}
+					if(dovideo) {
+						/* Create the video source stream */
+						janus_streaming_rtp_source_stream *stream = janus_streaming_create_rtp_source_stream(
+							cat->name, g_list_length(streams),
+							"video", "v", "video",
+							vmcast ? (char *)vmcast->value : NULL,
+							viface && viface->value ? (char *)viface->value : NULL,
+							viface && viface->value ? &video_iface : NULL,
+							(vport && vport->value) ? atoi(vport->value) : 0,
+							(vport2 && vport2->value) ? atoi(vport2->value) : 0,
+							(vport3 && vport3->value) ? atoi(vport3->value) : 0,
+							(vrtcpport && vrtcpport->value) ? atoi(vrtcpport->value) : 0,
+							(vcodec && vcodec->value) ? atoi(vcodec->value) : 0,
+							vrtpmap ? (char *)vrtpmap->value : NULL,
+							vfmtp ? (char *)vfmtp->value : NULL,
+							dovskew, bufferkf, FALSE, simulcast, dosvc);
+						if(stream == NULL) {
+							JANUS_LOG(LOG_ERR, "Skipping 'video' stream '%s', error creating source stream...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+						/* Add to the list of streams */
+						streams = g_list_append(streams, stream);
+					}
+					if(dodata) {
+						/* Create the data source stream */
+						janus_streaming_rtp_source_stream *stream = janus_streaming_create_rtp_source_stream(
+							cat->name, g_list_length(streams),
+							"data", "d", "data",
+							dmcast ? (char *)dmcast->value : NULL,
+							diface && diface->value ? (char *)diface->value : NULL,
+							diface && diface->value ? &data_iface : NULL,
+							(dport && dport->value) ? atoi(dport->value) : 0, 0, 0, 0,
+							0, NULL, NULL, FALSE, FALSE, buffermsg, FALSE, FALSE);
+						if(stream == NULL) {
+							JANUS_LOG(LOG_ERR, "Skipping 'data' stream '%s', error creating source stream...\n", cat->name);
+							cl = cl->next;
+							continue;
+						}
+						/* Add to the list of streams */
+						streams = g_list_append(streams, stream);
+					}
 				}
-				if(ssuite && ssuite->value && atoi(ssuite->value) != 32 && atoi(ssuite->value) != 80) {
-					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid SRTP suite...\n", cat->name);
-					cl = cl->next;
-					continue;
-				}
-				if(rtpcollision && rtpcollision->value && atoi(rtpcollision->value) < 0) {
-					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid collision configuration...\n", cat->name);
-					cl = cl->next;
-					continue;
-				}
-				if(threads && threads->value && atoi(threads->value) < 0) {
-					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid threads configuration...\n", cat->name);
-					cl = cl->next;
-					continue;
-				}
+				/* Back to the common part */
 				if(id == NULL || id->value == NULL) {
 					JANUS_LOG(LOG_VERB, "Missing id for stream '%s', will generate a random one...\n", cat->name);
 				} else {
@@ -1586,46 +1962,16 @@ int janus_streaming_init(janus_callbacks *callback, const char *config_path) {
 						continue;
 					}
 				}
-				JANUS_LOG(LOG_VERB, "Audio %s, Video %s, Data %s\n",
-					doaudio ? "enabled" : "NOT enabled",
-					dovideo ? "enabled" : "NOT enabled",
-					dodata ? "enabled" : "NOT enabled");
+				/* Streams created, create the actual mountpoint now */
 				janus_streaming_mountpoint *mp = NULL;
 				if((mp = janus_streaming_create_rtp_source(
 						(id && id->value) ? g_ascii_strtoull(id->value, 0, 10) : 0,
-						(char *)cat->name,
-						desc ? (char *)desc->value : NULL,
+						(char *)cat->name, desc ? (char *)desc->value : NULL,
+						streams,
 						ssuite && ssuite->value ? atoi(ssuite->value) : 0,
 						scrypto && scrypto->value ? (char *)scrypto->value : NULL,
 						(threads && threads->value) ?  atoi(threads->value) : 0,
-						doaudio,
-						amcast ? (char *)amcast->value : NULL,
-						doaudio && aiface && aiface->value ? &audio_iface : NULL,
-						(aport && aport->value) ? atoi(aport->value) : 0,
-						(artcpport && artcpport->value) ? atoi(artcpport->value) : 0,
-						(acodec && acodec->value) ? atoi(acodec->value) : 0,
-						artpmap ? (char *)artpmap->value : NULL,
-						afmtp ? (char *)afmtp->value : NULL,
-						doaskew,
-						dovideo,
-						vmcast ? (char *)vmcast->value : NULL,
-						dovideo && viface && viface->value ? &video_iface : NULL,
-						(vport && vport->value) ? atoi(vport->value) : 0,
-						(vrtcpport && vrtcpport->value) ? atoi(vrtcpport->value) : 0,
-						(vcodec && vcodec->value) ? atoi(vcodec->value) : 0,
-						vrtpmap ? (char *)vrtpmap->value : NULL,
-						vfmtp ? (char *)vfmtp->value : NULL,
-						bufferkf,
-						simulcast,
-						(vport2 && vport2->value) ? atoi(vport2->value) : 0,
-						(vport3 && vport3->value) ? atoi(vport3->value) : 0,
-						dosvc,
-						dovskew,
-						(rtpcollision && rtpcollision->value) ?  atoi(rtpcollision->value) : 0,
-						dodata,
-						dodata && diface && diface->value ? &data_iface : NULL,
-						(dport && dport->value) ? atoi(dport->value) : 0,
-						buffermsg)) == NULL) {
+						(rtpcollision && rtpcollision->value) ?  atoi(rtpcollision->value) : 0)) == NULL) {
 					JANUS_LOG(LOG_ERR, "Error creating 'rtp' stream '%s'...\n", cat->name);
 					cl = cl->next;
 					continue;
@@ -1884,7 +2230,7 @@ int janus_streaming_init(janus_callbacks *callback, const char *config_path) {
 	janus_mutex_unlock(&mountpoints_mutex);
 
 	sessions = g_hash_table_new_full(NULL, NULL, NULL, (GDestroyNotify)janus_streaming_session_destroy);
-	messages = g_async_queue_new_full((GDestroyNotify) janus_streaming_message_free);
+	messages = g_async_queue_new_full((GDestroyNotify)janus_streaming_message_free);
 	/* This is the callback we'll need to invoke to contact the Janus core */
 	gateway = callback;
 
@@ -1895,6 +2241,7 @@ int janus_streaming_init(janus_callbacks *callback, const char *config_path) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Streaming handler thread...\n", error->code, error->message ? error->message : "??");
 		janus_config_destroy(config);
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_STREAMING_NAME);
@@ -2036,34 +2383,47 @@ json_t *janus_streaming_query_session(janus_plugin_session *handle) {
 		janus_mutex_lock(&mp->mutex);
 		json_object_set_new(info, "mountpoint_viewers", json_integer(mp->viewers ? g_list_length(mp->viewers) : 0));
 		janus_mutex_unlock(&mp->mutex);
-		json_t *media = json_object();
-		json_object_set_new(media, "audio", session->audio ? json_true() : json_false());
-		json_object_set_new(media, "video", session->video ? json_true() : json_false());
-		json_object_set_new(media, "data", session->data ? json_true() : json_false());
-		json_object_set_new(info, "media", media);
-		if(mp->streaming_source == janus_streaming_source_rtp) {
-			janus_streaming_rtp_source *source = mp->source;
-			if(source->simulcast) {
-				json_t *simulcast = json_object();
-				json_object_set_new(simulcast, "substream", json_integer(session->sim_context.substream));
-				json_object_set_new(simulcast, "substream-target", json_integer(session->sim_context.substream_target));
-				json_object_set_new(simulcast, "temporal-layer", json_integer(session->sim_context.templayer));
-				json_object_set_new(simulcast, "temporal-layer-target", json_integer(session->sim_context.templayer_target));
-				json_object_set_new(info, "simulcast", simulcast);
-			}
-			if(source->svc) {
-				json_t *svc = json_object();
-				json_object_set_new(svc, "spatial-layer", json_integer(session->spatial_layer));
-				json_object_set_new(svc, "target-spatial-layer", json_integer(session->target_spatial_layer));
-				json_object_set_new(svc, "temporal-layer", json_integer(session->temporal_layer));
-				json_object_set_new(svc, "target-temporal-layer", json_integer(session->target_temporal_layer));
-				json_object_set_new(info, "svc", svc);
+		if(mp->streaming_source == janus_streaming_source_file) {
+			janus_streaming_file_source *source = mp->source;
+			json_t *media = json_object();
+			json_object_set_new(media, "type", json_string("audio"));
+			json_object_set_new(media, "filename", json_string(source->filename));
+			json_object_set_new(info, "media", media);
+		} else if(mp->streaming_source == janus_streaming_source_rtp) {
+			json_t *media = json_array();
+			GList *temp = session->streams;
+			while(temp) {
+				janus_streaming_session_stream *s = (janus_streaming_session_stream *)temp->data;
+				janus_streaming_rtp_source_stream *stream = s->stream;
+				json_t *info = json_object();
+				json_object_set_new(info, "type", json_string(janus_streaming_media_str(stream->type)));
+				json_object_set_new(info, "mindex", json_integer(stream->mindex));
+				json_object_set_new(info, "mid", json_string(stream->mid));
+				if(stream->simulcast) {
+					json_t *simulcast = json_object();
+					json_object_set_new(simulcast, "substream", json_integer(s->sim_context.substream));
+					json_object_set_new(simulcast, "substream-target", json_integer(s->sim_context.substream_target));
+					json_object_set_new(simulcast, "temporal-layer", json_integer(s->sim_context.templayer));
+					json_object_set_new(simulcast, "temporal-layer-target", json_integer(s->sim_context.templayer_target));
+					json_object_set_new(info, "simulcast", simulcast);
+				}
+				if(stream->svc) {
+					json_t *svc = json_object();
+					json_object_set_new(svc, "spatial-layer", json_integer(s->spatial_layer));
+					json_object_set_new(svc, "target-spatial-layer", json_integer(s->target_spatial_layer));
+					json_object_set_new(svc, "temporal-layer", json_integer(s->temporal_layer));
+					json_object_set_new(svc, "target-temporal-layer", json_integer(s->target_temporal_layer));
+					json_object_set_new(info, "svc", svc);
+				}
+				json_array_append_new(media, info);
+				temp = temp->next;
 			}
+			json_object_set_new(info, "media", media);
 		}
 		janus_refcount_decrease(&mp->ref);
 	}
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -2101,12 +2461,22 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 			json_object_set_new(ml, "description", json_string(mp->description));
 			json_object_set_new(ml, "type", json_string(mp->streaming_type == janus_streaming_type_live ? "live" : "on demand"));
 			if(mp->streaming_source == janus_streaming_source_rtp) {
+				json_t *media = json_array();
 				janus_streaming_rtp_source *source = mp->source;
 				gint64 now = janus_get_monotonic_time();
-				if(source->audio_fd != -1)
-					json_object_set_new(ml, "audio_age_ms", json_integer((now - source->last_received_audio) / 1000));
-				if(source->video_fd[0] != -1 || source->video_fd[1] != -1 || source->video_fd[2] != -1)
-					json_object_set_new(ml, "video_age_ms", json_integer((now - source->last_received_video) / 1000));
+				GList *temp = source->media;
+				while(temp) {
+					json_t *info = json_object();
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					json_object_set_new(info, "mid", json_string(stream->mid));
+					json_object_set_new(info, "type", json_string(janus_streaming_media_str(stream->type)));
+					json_object_set_new(info, "label", json_string(stream->label));
+					if(stream->fd[0] != -1 || stream->fd[1] != -1 || stream->fd[2] != -1)
+						json_object_set_new(info, "age_ms", json_integer((now - stream->last_received) / 1000));
+					json_array_append_new(media, info);
+					temp = temp->next;
+				}
+				json_object_set_new(ml, "media", media);
 			}
 			json_array_append_new(list, ml);
 			janus_refcount_decrease(&mp->ref);
@@ -2159,37 +2529,25 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 		if(admin && mp->is_private)
 			json_object_set_new(ml, "is_private", json_true());
 		json_object_set_new(ml, "enabled", mp->enabled ? json_true() : json_false());
-		if(mp->audio) {
-			json_object_set_new(ml, "audio", json_true());
-			if(mp->codecs.audio_pt != -1)
-				json_object_set_new(ml, "audiopt", json_integer(mp->codecs.audio_pt));
-			if(mp->codecs.audio_rtpmap)
-				json_object_set_new(ml, "audiortpmap", json_string(mp->codecs.audio_rtpmap));
-			if(mp->codecs.audio_fmtp)
-				json_object_set_new(ml, "audiofmtp", json_string(mp->codecs.audio_fmtp));
-		}
-		if(mp->video) {
-			json_object_set_new(ml, "video", json_true());
-			if(mp->codecs.video_pt != -1)
-				json_object_set_new(ml, "videopt", json_integer(mp->codecs.video_pt));
-			if(mp->codecs.video_rtpmap)
-				json_object_set_new(ml, "videortpmap", json_string(mp->codecs.video_rtpmap));
-			if(mp->codecs.video_fmtp)
-				json_object_set_new(ml, "videofmtp", json_string(mp->codecs.video_fmtp));
-		}
-		if(mp->data) {
-			json_object_set_new(ml, "data", json_true());
-		}
 		json_object_set_new(ml, "type", json_string(mp->streaming_type == janus_streaming_type_live ? "live" : "on demand"));
+		json_t *media = json_array();
+		json_object_set_new(ml, "media", media);
 		if(mp->streaming_source == janus_streaming_source_file) {
 			janus_streaming_file_source *source = mp->source;
 			if(admin && source->filename)
 				json_object_set_new(ml, "filename", json_string(source->filename));
+			json_t *info = json_object();
+			json_object_set_new(info, "type", json_string("audio"));
+			if(source->codecs.pt != -1)
+				json_object_set_new(info, "pt", json_integer(source->codecs.pt));
+			if(source->codecs.rtpmap)
+				json_object_set_new(info, "rtpmap", json_string(source->codecs.rtpmap));
+			if(source->codecs.fmtp)
+				json_object_set_new(info, "fmtp", json_string(source->codecs.fmtp));
+			json_array_append_new(media, info);
 		} else if(mp->streaming_source == janus_streaming_source_rtp) {
 			janus_streaming_rtp_source *source = mp->source;
-			if(source->is_srtp) {
-				json_object_set_new(ml, "srtp", json_true());
-			}
+			/* Global stuff first */
 			gint64 now = janus_get_monotonic_time();
 #ifdef HAVE_LIBCURL
 			if(source->rtsp) {
@@ -2204,59 +2562,57 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 				}
 			}
 #endif
-			if(source->keyframe.enabled) {
-				json_object_set_new(ml, "videobufferkf", json_true());
-			}
-			if(source->simulcast) {
-				json_object_set_new(ml, "videosimulcast", json_true());
-			}
-			if(source->svc) {
-				json_object_set_new(ml, "videosvc", json_true());
+			if(source->is_srtp) {
+				json_object_set_new(ml, "srtp", json_true());
 			}
-			if(source->askew)
-				json_object_set_new(ml, "audioskew", json_true());
-			if(source->vskew)
-				json_object_set_new(ml, "videoskew", json_true());
 			if(source->rtp_collision > 0)
 				json_object_set_new(ml, "collision", json_integer(source->rtp_collision));
 			if(mp->helper_threads > 0)
 				json_object_set_new(ml, "threads", json_integer(mp->helper_threads));
-			if(admin) {
-				if(mp->audio) {
-					json_object_set_new(ml, "audioport", json_integer(source->audio_port));
-					if(source->audio_rtcp_port > -1)
-						json_object_set_new(ml, "audiortcpport", json_integer(source->audio_rtcp_port));
+			/* Iterate on media now */
+			GList *temp = source->media;
+			while(temp) {
+				janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+				json_t *info = json_object();
+				json_object_set_new(info, "mindex", json_integer(stream->mindex));
+				json_object_set_new(info, "type", json_string(janus_streaming_media_str(stream->type)));
+				json_object_set_new(info, "mid", json_string(stream->mid));
+				json_object_set_new(info, "label", json_string(stream->label));
+				if(stream->codecs.pt != -1)
+					json_object_set_new(info, "pt", json_integer(stream->codecs.pt));
+				if(stream->codecs.rtpmap)
+					json_object_set_new(info, "rtpmap", json_string(stream->codecs.rtpmap));
+				if(stream->codecs.fmtp)
+					json_object_set_new(info, "fmtp", json_string(stream->codecs.fmtp));
+				if(stream->keyframe.enabled) {
+					json_object_set_new(info, "videobufferkf", json_true());
 				}
-				if(mp->video) {
-					json_object_set_new(ml, "videoport", json_integer(source->video_port[0]));
-					if(source->video_rtcp_port > -1)
-						json_object_set_new(ml, "videortcpport", json_integer(source->video_rtcp_port));
-					if(source->video_port[1] > -1)
-						json_object_set_new(ml, "videoport2", json_integer(source->video_port[1]));
-					if(source->video_port[2] > -1)
-						json_object_set_new(ml, "videoport3", json_integer(source->video_port[2]));
+				if(stream->simulcast) {
+					json_object_set_new(info, "videosimulcast", json_true());
 				}
-				if(mp->data)
-					json_object_set_new(ml, "dataport", json_integer(source->data_port));
-			}
-			if(source->audio_fd != -1)
-				json_object_set_new(ml, "audio_age_ms", json_integer((now - source->last_received_audio) / 1000));
-			if(source->video_fd[0] != -1 || source->video_fd[1] != -1 || source->video_fd[2] != -1)
-				json_object_set_new(ml, "video_age_ms", json_integer((now - source->last_received_video) / 1000));
-			if(source->data_fd != -1)
-				json_object_set_new(ml, "data_age_ms", json_integer((now - source->last_received_data) / 1000));
-			janus_mutex_lock(&source->rec_mutex);
-			if(admin && (source->arc || source->vrc || source->drc)) {
-				json_t *recording = json_object();
-				if(source->arc && source->arc->filename)
-					json_object_set_new(recording, "audio", json_string(source->arc->filename));
-				if(source->vrc && source->vrc->filename)
-					json_object_set_new(recording, "video", json_string(source->vrc->filename));
-				if(source->drc && source->drc->filename)
-					json_object_set_new(recording, "data", json_string(source->drc->filename));
-				json_object_set_new(ml, "recording", recording);
+				if(stream->svc) {
+					json_object_set_new(info, "videosvc", json_true());
+				}
+				if(stream->skew)
+					json_object_set_new(info, "skew_compensation", json_true());
+				if(admin) {
+					json_object_set_new(info, "port", json_integer(stream->port[0]));
+					if(stream->rtcp_port > -1)
+						json_object_set_new(info, "rtcpport", json_integer(stream->rtcp_port));
+					if(stream->port[1] > -1)
+						json_object_set_new(info, "port2", json_integer(stream->port[1]));
+					if(stream->port[2] > -1)
+						json_object_set_new(info, "port3", json_integer(stream->port[2]));
+				}
+				if(stream->fd[0] != -1 || stream->fd[1] != -1 || stream->fd[2] != -1)
+					json_object_set_new(info, "age_ms", json_integer((now - stream->last_received / 1000)));
+				janus_mutex_lock(&source->rec_mutex);
+				if(admin && stream->rc && stream->rc->filename)
+					json_object_set_new(info, "recording", json_string(stream->rc->filename));
+				janus_mutex_unlock(&source->rec_mutex);
+				json_array_append_new(media, info);
+				temp = temp->next;
 			}
-			janus_mutex_unlock(&source->rec_mutex);
 		}
 		janus_refcount_decrease(&mp->ref);
 		janus_mutex_unlock(&mountpoints_mutex);
@@ -2303,7 +2659,6 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 		}
 		janus_streaming_mountpoint *mp = NULL;
 		if(!strcasecmp(type_text, "rtp")) {
-			janus_network_address audio_iface, video_iface, data_iface;
 			/* RTP live source (e.g., from gstreamer/ffmpeg/vlc/etc.) */
 			JANUS_VALIDATE_JSON_OBJECT(root, rtp_parameters,
 				error_code, error_cause, TRUE,
@@ -2314,152 +2669,306 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 			json_t *name = json_object_get(root, "name");
 			json_t *desc = json_object_get(root, "description");
 			json_t *is_private = json_object_get(root, "is_private");
-			json_t *audio = json_object_get(root, "audio");
-			json_t *video = json_object_get(root, "video");
-			json_t *data = json_object_get(root, "data");
 			json_t *rtpcollision = json_object_get(root, "collision");
 			json_t *threads = json_object_get(root, "threads");
 			json_t *ssuite = json_object_get(root, "srtpsuite");
 			json_t *scrypto = json_object_get(root, "srtpcrypto");
-			gboolean doaudio = audio ? json_is_true(audio) : FALSE;
-			gboolean dovideo = video ? json_is_true(video) : FALSE;
-			gboolean dodata = data ? json_is_true(data) : FALSE;
-			gboolean doaskew = FALSE, dovskew = FALSE, dosvc = FALSE;
-			if(!doaudio && !dovideo && !dodata) {
-				JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream, no audio, video or data have to be streamed...\n");
-				error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
-				g_snprintf(error_cause, 512, "Can't add 'rtp' stream, no audio or video have to be streamed...");
-				goto prepare_response;
-			}
 			if(ssuite && json_integer_value(ssuite) != 32 && json_integer_value(ssuite) != 80) {
 				JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream, invalid SRTP suite...\n");
 				error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
 				g_snprintf(error_cause, 512, "Can't add 'rtp' stream, invalid SRTP suite...");
 				goto prepare_response;
 			}
-			uint16_t aport = 0;
-			uint16_t artcpport = 0;
-			uint8_t acodec = 0;
-			char *artpmap = NULL, *afmtp = NULL, *amcast = NULL;
-			if(doaudio) {
-				JANUS_VALIDATE_JSON_OBJECT(root, rtp_audio_parameters,
-					error_code, error_cause, TRUE,
-					JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
-				if(error_code != 0)
+			json_t *media = json_object_get(root, "media");
+			GList *streams = NULL;
+			/* How are we adding media? */
+			if(media != NULL) {
+				/* We're using the new media-based configuration, iterate on all media objects */
+				if(json_array_size(media) == 0) {
+					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream, no audio, video or data have to be streamed...\n");
+					error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+					g_snprintf(error_cause, 512, "Can't add 'rtp' stream, no audio or video have to be streamed...");
 					goto prepare_response;
-				json_t *audiomcast = json_object_get(root, "audiomcast");
-				amcast = (char *)json_string_value(audiomcast);
-				json_t *audioport = json_object_get(root, "audioport");
-				aport = json_integer_value(audioport);
-				json_t *audiortcpport = json_object_get(root, "audiortcpport");
-				if(audiortcpport)
-					artcpport = json_integer_value(audiortcpport);
-				json_t *audiopt = json_object_get(root, "audiopt");
-				acodec = json_integer_value(audiopt);
-				json_t *audiortpmap = json_object_get(root, "audiortpmap");
-				artpmap = (char *)json_string_value(audiortpmap);
-				json_t *audiofmtp = json_object_get(root, "audiofmtp");
-				afmtp = (char *)json_string_value(audiofmtp);
-				json_t *aiface = json_object_get(root, "audioiface");
-				if(aiface) {
-					const char *miface = (const char *)json_string_value(aiface);
-					if(janus_network_lookup_interface(ifas, miface, &audio_iface) != 0) {
-						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for audio...\n", (const char *)json_string_value(name));
+				}
+				size_t i = 0;
+				for(i=0; i<json_array_size(media); i++) {
+					json_t *m = json_array_get(media, i);
+					JANUS_VALIDATE_JSON_OBJECT(m, rtp_media_parameters,
+						error_code, error_cause, TRUE,
+						JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
+					if(error_code != 0)
+						goto prepare_response;
+					janus_network_address iface;
+					uint16_t port = 0, port2 = 0, port3 = 0;
+					uint16_t rtcpport = 0;
+					uint8_t codec = 0;
+					char *mtype = NULL, *mid = NULL, *label = NULL, *rtpmap = NULL, *fmtp = NULL, *mcast = NULL, *miface = NULL;
+					gboolean doskew = FALSE, bufferkf = FALSE, simulcast = FALSE, dosvc = FALSE, buffermsg = FALSE;
+					json_t *jmtype = json_object_get(m, "type");
+					mtype = (char *)json_string_value(jmtype);
+					if(strcasecmp(mtype, "audio") && strcasecmp(mtype, "video") && strcasecmp(mtype, "data")) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', unsupported media type...\n", (const char *)json_string_value(name));
 						error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
-						g_snprintf(error_cause, 512, ifas ? "Invalid network interface configuration for audio" : "Unable to query network device information");
+						g_snprintf(error_cause, 512, "Unsupported media type in media stream");
 						goto prepare_response;
 					}
-				} else {
-					janus_network_address_nullify(&audio_iface);
-				}
-				json_t *askew = json_object_get(root, "audioskew");
-				doaskew = askew ? json_is_true(askew) : FALSE;
-			}
-			uint16_t vport = 0, vport2 = 0, vport3 = 0;
-			uint16_t vrtcpport = 0;
-			uint8_t vcodec = 0;
-			char *vrtpmap = NULL, *vfmtp = NULL, *vmcast = NULL;
-			gboolean bufferkf = FALSE, simulcast = FALSE;
-			if(dovideo) {
-				JANUS_VALIDATE_JSON_OBJECT(root, rtp_video_parameters,
-					error_code, error_cause, TRUE,
-					JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
-				if(error_code != 0)
+					/* We need mid and label for addressing this stream on the client side */
+					json_t *jmid = json_object_get(m, "mid");
+					mid = (char *)json_string_value(jmid);
+					json_t *jlabel = json_object_get(m, "label");
+					label = (char *)json_string_value(jlabel);
+					json_t *jmcast = json_object_get(m, "mcast");
+					mcast = (char *)json_string_value(jmcast);
+					json_t *jport = json_object_get(m, "port");
+					port = json_integer_value(jport);
+					json_t *jrtcpport = json_object_get(m, "rtcpport");
+					if(jrtcpport)
+						rtcpport = json_integer_value(jrtcpport);
+					json_t *jpt = json_object_get(m, "pt");
+					codec = json_integer_value(jpt);
+					json_t *jrtpmap = json_object_get(m, "rtpmap");
+					rtpmap = (char *)json_string_value(jrtpmap);
+					json_t *jfmtp = json_object_get(m, "fmtp");
+					fmtp = (char *)json_string_value(jfmtp);
+					json_t *jiface = json_object_get(m, "iface");
+					miface = (char *)json_string_value(jiface);
+					if(jiface) {
+						if(janus_network_lookup_interface(ifas, miface, &iface) != 0) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for media stream...\n", (const char *)json_string_value(name));
+							error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+							g_snprintf(error_cause, 512, ifas ? "Invalid network interface configuration for media stream" : "Unable to query network device information");
+							goto prepare_response;
+						}
+					} else {
+						janus_network_address_nullify(&iface);
+					}
+					json_t *jskew = json_object_get(m, "skew");
+					doskew = jskew ? json_is_true(jskew) : FALSE;
+					if(!strcasecmp(mtype, "video")) {
+						json_t *vkf = json_object_get(m, "bufferkf");
+						bufferkf = vkf ? json_is_true(vkf) : FALSE;
+						json_t *vsc = json_object_get(m, "simulcast");
+						simulcast = vsc ? json_is_true(vsc) : FALSE;
+						if(simulcast && bufferkf) {
+							/* FIXME We'll need to take care of this */
+							JANUS_LOG(LOG_WARN, "Simulcasting enabled, so disabling buffering of keyframes\n");
+							bufferkf = FALSE;
+						}
+						json_t *videoport2 = json_object_get(m, "port2");
+						port2 = json_integer_value(videoport2);
+						json_t *videoport3 = json_object_get(m, "port3");
+						port3 = json_integer_value(videoport3);
+						json_t *vsvc = json_object_get(m, "svc");
+						dosvc = vsvc ? json_is_true(vsvc) : FALSE;
+					} else if(!strcasecmp(mtype, "data")) {
+						json_t *dbm = json_object_get(root, "buffermsg");
+						buffermsg = dbm ? json_is_true(dbm) : FALSE;
+					}
+					/* Create the data source stream */
+					janus_streaming_rtp_source_stream *stream = janus_streaming_create_rtp_source_stream(
+						name ? (char *)json_string_value(name) : NULL,
+						g_list_length(streams),
+						mtype, mid, label ? label : mtype,
+						mcast, miface, &iface, port, port2, port3, rtcpport,
+						codec, rtpmap, fmtp, doskew, bufferkf, buffermsg, simulcast, dosvc);
+					if(stream == NULL) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', error creating data source stream...\n", (const char *)json_string_value(name));
+						error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+						g_snprintf(error_cause, 512, "Can't add 'rtp' stream, error creating data source stream");
+						goto prepare_response;
+					}
+					/* Add to the list of streams */
+					streams = g_list_append(streams, stream);
+				}
+			} else {
+				/* If we got here, we create a mountpoint the "old" way */
+				janus_network_address audio_iface, video_iface, data_iface;
+				json_t *audio = json_object_get(root, "audio");
+				json_t *video = json_object_get(root, "video");
+				json_t *data = json_object_get(root, "data");
+				gboolean doaudio = audio ? json_is_true(audio) : FALSE;
+				gboolean dovideo = video ? json_is_true(video) : FALSE;
+				gboolean dodata = data ? json_is_true(data) : FALSE;
+				gboolean doaskew = FALSE, dovskew = FALSE, dosvc = FALSE;
+				if(!doaudio && !dovideo && !dodata) {
+					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream, no audio, video or data have to be streamed...\n");
+					error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+					g_snprintf(error_cause, 512, "Can't add 'rtp' stream, no audio or video have to be streamed...");
 					goto prepare_response;
-				json_t *videomcast = json_object_get(root, "videomcast");
-				vmcast = (char *)json_string_value(videomcast);
-				json_t *videoport = json_object_get(root, "videoport");
-				vport = json_integer_value(videoport);
-				json_t *videortcpport = json_object_get(root, "videortcpport");
-				if(videortcpport)
-					vrtcpport = json_integer_value(videortcpport);
-				json_t *videopt = json_object_get(root, "videopt");
-				vcodec = json_integer_value(videopt);
-				json_t *videortpmap = json_object_get(root, "videortpmap");
-				vrtpmap = (char *)json_string_value(videortpmap);
-				json_t *videofmtp = json_object_get(root, "videofmtp");
-				vfmtp = (char *)json_string_value(videofmtp);
-				json_t *vkf = json_object_get(root, "videobufferkf");
-				bufferkf = vkf ? json_is_true(vkf) : FALSE;
-				json_t *vsc = json_object_get(root, "videosimulcast");
-				simulcast = vsc ? json_is_true(vsc) : FALSE;
-				if(simulcast && bufferkf) {
-					/* FIXME We'll need to take care of this */
-					JANUS_LOG(LOG_WARN, "Simulcasting enabled, so disabling buffering of keyframes\n");
-					bufferkf = FALSE;
 				}
-				json_t *videoport2 = json_object_get(root, "videoport2");
-				vport2 = json_integer_value(videoport2);
-				json_t *videoport3 = json_object_get(root, "videoport3");
-				vport3 = json_integer_value(videoport3);
-				json_t *viface = json_object_get(root, "videoiface");
-				if(viface) {
-					const char *miface = (const char *)json_string_value(viface);
-					if(janus_network_lookup_interface(ifas, miface, &video_iface) != 0) {
-						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for video...\n", (const char *)json_string_value(name));
+				uint16_t aport = 0;
+				uint16_t artcpport = 0;
+				uint8_t acodec = 0;
+				char *artpmap = NULL, *afmtp = NULL, *amcast = NULL, *amiface = NULL;
+				if(doaudio) {
+					JANUS_VALIDATE_JSON_OBJECT(root, rtp_audio_parameters,
+						error_code, error_cause, TRUE,
+						JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
+					if(error_code != 0)
+						goto prepare_response;
+					json_t *audiomcast = json_object_get(root, "audiomcast");
+					amcast = (char *)json_string_value(audiomcast);
+					json_t *audioport = json_object_get(root, "audioport");
+					aport = json_integer_value(audioport);
+					json_t *audiortcpport = json_object_get(root, "audiortcpport");
+					if(audiortcpport)
+						artcpport = json_integer_value(audiortcpport);
+					json_t *audiopt = json_object_get(root, "audiopt");
+					acodec = json_integer_value(audiopt);
+					json_t *audiortpmap = json_object_get(root, "audiortpmap");
+					artpmap = (char *)json_string_value(audiortpmap);
+					json_t *audiofmtp = json_object_get(root, "audiofmtp");
+					afmtp = (char *)json_string_value(audiofmtp);
+					json_t *aiface = json_object_get(root, "audioiface");
+					amiface = (char *)json_string_value(aiface);
+					if(aiface) {
+						if(janus_network_lookup_interface(ifas, amiface, &audio_iface) != 0) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for audio...\n", (const char *)json_string_value(name));
+							error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+							g_snprintf(error_cause, 512, ifas ? "Invalid network interface configuration for audio" : "Unable to query network device information");
+							goto prepare_response;
+						}
+					} else {
+						janus_network_address_nullify(&audio_iface);
+					}
+					json_t *askew = json_object_get(root, "audioskew");
+					doaskew = askew ? json_is_true(askew) : FALSE;
+					/* Create the audio source stream */
+					janus_streaming_rtp_source_stream *stream = janus_streaming_create_rtp_source_stream(
+						name ? (char *)json_string_value(name) : NULL,
+						g_list_length(streams),
+						"audio", "a", "audio",
+						amcast, amiface, &audio_iface, aport, 0, 0, artcpport,
+						acodec, artpmap, afmtp, doaskew, FALSE, FALSE, FALSE, FALSE);
+					if(stream == NULL) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', error creating audio source stream...\n", (const char *)json_string_value(name));
 						error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
-						g_snprintf(error_cause, 512, ifas ? "Invalid network interface configuration for video" : "Unable to query network device information");
+						g_snprintf(error_cause, 512, "Can't add 'rtp' stream, error creating audio source stream");
 						goto prepare_response;
 					}
-				} else {
-					janus_network_address_nullify(&video_iface);
+					/* Add to the list of streams */
+					streams = g_list_append(streams, stream);
 				}
-				json_t *vskew = json_object_get(root, "videoskew");
-				dovskew = vskew ? json_is_true(vskew) : FALSE;
-				json_t *vsvc = json_object_get(root, "videosvc");
-				dosvc = vsvc ? json_is_true(vsvc) : FALSE;
-			}
-			uint16_t dport = 0;
-			gboolean buffermsg = FALSE;
-			if(dodata) {
-				JANUS_VALIDATE_JSON_OBJECT(root, rtp_data_parameters,
-					error_code, error_cause, TRUE,
-					JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
-				if(error_code != 0)
-					goto prepare_response;
-#ifdef HAVE_SCTP
-				json_t *dataport = json_object_get(root, "dataport");
-				dport = json_integer_value(dataport);
-				json_t *dbm = json_object_get(root, "databuffermsg");
-				buffermsg = dbm ? json_is_true(dbm) : FALSE;
-				json_t *diface = json_object_get(root, "dataiface");
-				if(diface) {
-					const char *miface = (const char *)json_string_value(diface);
-					if(janus_network_lookup_interface(ifas, miface, &data_iface) != 0) {
-						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for data...\n", (const char *)json_string_value(name));
+				uint16_t vport = 0, vport2 = 0, vport3 = 0;
+				uint16_t vrtcpport = 0;
+				uint8_t vcodec = 0;
+				char *vrtpmap = NULL, *vfmtp = NULL, *vmcast = NULL, *vmiface = NULL;
+				gboolean bufferkf = FALSE, simulcast = FALSE;
+				if(dovideo) {
+					JANUS_VALIDATE_JSON_OBJECT(root, rtp_video_parameters,
+						error_code, error_cause, TRUE,
+						JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
+					if(error_code != 0)
+						goto prepare_response;
+					json_t *videomcast = json_object_get(root, "videomcast");
+					vmcast = (char *)json_string_value(videomcast);
+					json_t *videoport = json_object_get(root, "videoport");
+					vport = json_integer_value(videoport);
+					json_t *videortcpport = json_object_get(root, "videortcpport");
+					if(videortcpport)
+						vrtcpport = json_integer_value(videortcpport);
+					json_t *videopt = json_object_get(root, "videopt");
+					vcodec = json_integer_value(videopt);
+					json_t *videortpmap = json_object_get(root, "videortpmap");
+					vrtpmap = (char *)json_string_value(videortpmap);
+					json_t *videofmtp = json_object_get(root, "videofmtp");
+					vfmtp = (char *)json_string_value(videofmtp);
+					json_t *vkf = json_object_get(root, "videobufferkf");
+					bufferkf = vkf ? json_is_true(vkf) : FALSE;
+					json_t *vsc = json_object_get(root, "videosimulcast");
+					simulcast = vsc ? json_is_true(vsc) : FALSE;
+					if(simulcast && bufferkf) {
+						/* FIXME We'll need to take care of this */
+						JANUS_LOG(LOG_WARN, "Simulcasting enabled, so disabling buffering of keyframes\n");
+						bufferkf = FALSE;
+					}
+					json_t *videoport2 = json_object_get(root, "videoport2");
+					vport2 = json_integer_value(videoport2);
+					json_t *videoport3 = json_object_get(root, "videoport3");
+					vport3 = json_integer_value(videoport3);
+					json_t *viface = json_object_get(root, "videoiface");
+					vmiface = (char *)json_string_value(viface);
+					if(viface) {
+						if(janus_network_lookup_interface(ifas, vmiface, &video_iface) != 0) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for video...\n", (const char *)json_string_value(name));
+							error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+							g_snprintf(error_cause, 512, ifas ? "Invalid network interface configuration for video" : "Unable to query network device information");
+							goto prepare_response;
+						}
+					} else {
+						janus_network_address_nullify(&video_iface);
+					}
+					json_t *vskew = json_object_get(root, "videoskew");
+					dovskew = vskew ? json_is_true(vskew) : FALSE;
+					json_t *vsvc = json_object_get(root, "videosvc");
+					dosvc = vsvc ? json_is_true(vsvc) : FALSE;
+					/* Create the video source stream */
+					janus_streaming_rtp_source_stream *stream = janus_streaming_create_rtp_source_stream(
+						name ? (char *)json_string_value(name) : NULL,
+						g_list_length(streams),
+						"video", "v", "video",
+						vmcast, vmiface, &video_iface, vport, vport2, vport3, vrtcpport,
+						vcodec, vrtpmap, vfmtp, dovskew, bufferkf, FALSE, simulcast, dosvc);
+					if(stream == NULL) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', error creating video source stream...\n", (const char *)json_string_value(name));
 						error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
-						g_snprintf(error_cause, 512, ifas ? "Invalid network interface configuration for data" : "Unable to query network device information");
+						g_snprintf(error_cause, 512, "Can't add 'rtp' stream, error creating video source stream");
 						goto prepare_response;
 					}
-				} else {
-					janus_network_address_nullify(&data_iface);
+					/* Add to the list of streams */
+					streams = g_list_append(streams, stream);
 				}
+				uint16_t dport = 0;
+				gboolean buffermsg = FALSE;
+				char *dmcast = NULL, *dmiface = NULL;
+				if(dodata) {
+					JANUS_VALIDATE_JSON_OBJECT(root, rtp_data_parameters,
+						error_code, error_cause, TRUE,
+						JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
+					if(error_code != 0)
+						goto prepare_response;
+#ifdef HAVE_SCTP
+					json_t *datamcast = json_object_get(root, "datamcast");
+					dmcast = (char *)json_string_value(datamcast);
+					json_t *dataport = json_object_get(root, "dataport");
+					dport = json_integer_value(dataport);
+					json_t *dbm = json_object_get(root, "databuffermsg");
+					buffermsg = dbm ? json_is_true(dbm) : FALSE;
+					json_t *diface = json_object_get(root, "dataiface");
+					dmiface = (char *)json_string_value(diface);
+					if(diface) {
+						if(janus_network_lookup_interface(ifas, dmiface, &data_iface) != 0) {
+							JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', invalid network interface configuration for data...\n", (const char *)json_string_value(name));
+							error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+							g_snprintf(error_cause, 512, ifas ? "Invalid network interface configuration for data" : "Unable to query network device information");
+							goto prepare_response;
+						}
+					} else {
+						janus_network_address_nullify(&data_iface);
+					}
+					/* Create the data source stream */
+					janus_streaming_rtp_source_stream *stream = janus_streaming_create_rtp_source_stream(
+						name ? (char *)json_string_value(name) : NULL,
+						g_list_length(streams),
+						"data", "d", "data",
+						dmcast, dmiface, &data_iface, dport, 0, 0, 0, 0,
+						NULL, NULL, FALSE, FALSE, buffermsg, FALSE, FALSE);
+					if(stream == NULL) {
+						JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream '%s', error creating data source stream...\n", (const char *)json_string_value(name));
+						error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+						g_snprintf(error_cause, 512, "Can't add 'rtp' stream, error creating data source stream");
+						goto prepare_response;
+					}
+					/* Add to the list of streams */
+					streams = g_list_append(streams, stream);
 #else
-				JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream: no datachannels support...\n");
-				error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
-				g_snprintf(error_cause, 512, "Can't add 'rtp' stream: no datachannels support...");
-				goto prepare_response;
+					JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream: no datachannels support...\n");
+					error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
+					g_snprintf(error_cause, 512, "Can't add 'rtp' stream: no datachannels support...");
+					goto prepare_response;
 #endif
+				}
+				JANUS_LOG(LOG_VERB, "Audio %s, Video %s\n", doaudio ? "enabled" : "NOT enabled", dovideo ? "enabled" : "NOT enabled");
 			}
 			/* Check if an ID has been provided, or if we need to generate one ourselves */
 			janus_mutex_lock(&mountpoints_mutex);
@@ -2485,24 +2994,15 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 					goto prepare_response;
 				}
 			}
-			g_hash_table_insert(mountpoints_temp, &mpid, GUINT_TO_POINTER(TRUE));
-			janus_mutex_unlock(&mountpoints_mutex);
-			JANUS_LOG(LOG_VERB, "Audio %s, Video %s\n", doaudio ? "enabled" : "NOT enabled", dovideo ? "enabled" : "NOT enabled");
 			mp = janus_streaming_create_rtp_source(
-					mpid,
-					name ? (char *)json_string_value(name) : NULL,
-					desc ? (char *)json_string_value(desc) : NULL,
-					ssuite ? json_integer_value(ssuite) : 0,
-					scrypto ? (char *)json_string_value(scrypto) : NULL,
-					threads ? json_integer_value(threads) : 0,
-					doaudio, amcast, &audio_iface, aport, artcpport, acodec, artpmap, afmtp, doaskew,
-					dovideo, vmcast, &video_iface, vport, vrtcpport, vcodec, vrtpmap, vfmtp, bufferkf,
-					simulcast, vport2, vport3, dosvc, dovskew,
-					rtpcollision ? json_integer_value(rtpcollision) : 0,
-					dodata, &data_iface, dport, buffermsg);
-			janus_mutex_lock(&mountpoints_mutex);
-			g_hash_table_remove(mountpoints_temp, &mpid);
-			janus_mutex_unlock(&mountpoints_mutex);
+				mpid,
+				name ? (char *)json_string_value(name) : NULL,
+				desc ? (char *)json_string_value(desc) : NULL,
+				streams,
+				ssuite ? json_integer_value(ssuite) : 0,
+				scrypto ? (char *)json_string_value(scrypto) : NULL,
+				threads ? json_integer_value(threads) : 0,
+				rtpcollision ? json_integer_value(rtpcollision) : 0);
 			if(mp == NULL) {
 				JANUS_LOG(LOG_ERR, "Error creating 'rtp' stream...\n");
 				error_code = JANUS_STREAMING_ERROR_CANT_CREATE;
@@ -2801,82 +3301,18 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 			janus_config_add(config, c, janus_config_item_create("description", mp->description));
 			if(mp->is_private)
 				janus_config_add(config, c, janus_config_item_create("is_private", "yes"));
+			if(mp->secret)
+				janus_config_add(config, c, janus_config_item_create("secret", mp->secret));
+			if(mp->pin)
+				janus_config_add(config, c, janus_config_item_create("pin", mp->pin));
 			/* Per type values */
 			if(!strcasecmp(type_text, "rtp")) {
-				janus_config_add(config, c, janus_config_item_create("audio", mp->codecs.audio_pt >= 0 ? "yes" : "no"));
+				/* We save using the new format, not the old deprecated one */
 				janus_streaming_rtp_source *source = mp->source;
-				if(mp->codecs.audio_pt >= 0) {
-					g_snprintf(value, BUFSIZ, "%d", source->audio_port);
-					janus_config_add(config, c, janus_config_item_create("audioport", value));
-					if(source->audio_rtcp_port > 0) {
-						g_snprintf(value, BUFSIZ, "%d", source->audio_rtcp_port);
-						janus_config_add(config, c, janus_config_item_create("audiortcpport", value));
-					}
-					json_t *audiomcast = json_object_get(root, "audiomcast");
-					if(audiomcast)
-						janus_config_add(config, c, janus_config_item_create("audiomcast", json_string_value(audiomcast)));
-					g_snprintf(value, BUFSIZ, "%d", mp->codecs.audio_pt);
-					janus_config_add(config, c, janus_config_item_create("audiopt", value));
-					janus_config_add(config, c, janus_config_item_create("audiortpmap", mp->codecs.audio_rtpmap));
-					if(mp->codecs.audio_fmtp)
-						janus_config_add(config, c, janus_config_item_create("audiofmtp", mp->codecs.audio_fmtp));
-					json_t *aiface = json_object_get(root, "audioiface");
-					if(aiface)
-						janus_config_add(config, c, janus_config_item_create("audioiface", json_string_value(aiface)));
-					if(source->askew)
-						janus_config_add(config, c, janus_config_item_create("askew", "yes"));
-				}
-				janus_config_add(config, c, janus_config_item_create("video", mp->codecs.video_pt > 0 ? "yes" : "no"));
-				if(mp->codecs.video_pt > 0) {
-					g_snprintf(value, BUFSIZ, "%d", source->video_port[0]);
-					janus_config_add(config, c, janus_config_item_create("videoport", value));
-					if(source->video_rtcp_port > 0) {
-						g_snprintf(value, BUFSIZ, "%d", source->video_rtcp_port);
-						janus_config_add(config, c, janus_config_item_create("videortcpport", value));
-					}
-					json_t *videomcast = json_object_get(root, "videomcast");
-					if(videomcast)
-						janus_config_add(config, c, janus_config_item_create("videomcast", json_string_value(videomcast)));
-					g_snprintf(value, BUFSIZ, "%d", mp->codecs.video_pt);
-					janus_config_add(config, c, janus_config_item_create("videopt", value));
-					janus_config_add(config, c, janus_config_item_create("videortpmap", mp->codecs.video_rtpmap));
-					if(mp->codecs.video_fmtp)
-						janus_config_add(config, c, janus_config_item_create("videofmtp", mp->codecs.video_fmtp));
-					if(source->keyframe.enabled)
-						janus_config_add(config, c, janus_config_item_create("videobufferkf", "yes"));
-					if(source->simulcast) {
-						janus_config_add(config, c, janus_config_item_create("videosimulcast", "yes"));
-						if(source->video_port[1]) {
-							g_snprintf(value, BUFSIZ, "%d", source->video_port[1]);
-							janus_config_add(config, c, janus_config_item_create("videoport2", value));
-						}
-						if(source->video_port[2]) {
-							g_snprintf(value, BUFSIZ, "%d", source->video_port[2]);
-							janus_config_add(config, c, janus_config_item_create("videoport3", value));
-						}
-					}
-					if(source->svc)
-						janus_config_add(config, c, janus_config_item_create("videosvc", "yes"));
-					json_t *viface = json_object_get(root, "videoiface");
-					if(viface)
-						janus_config_add(config, c, janus_config_item_create("videoiface", json_string_value(viface)));
-					if(source->vskew)
-						janus_config_add(config, c, janus_config_item_create("videoskew", "yes"));
-				}
 				if(source->rtp_collision > 0) {
 					g_snprintf(value, BUFSIZ, "%d", source->rtp_collision);
 					janus_config_add(config, c, janus_config_item_create("collision", value));
 				}
-				janus_config_add(config, c, janus_config_item_create("data", mp->data ? "yes" : "no"));
-				if(source->data_port > -1) {
-					g_snprintf(value, BUFSIZ, "%d", source->data_port);
-					janus_config_add(config, c, janus_config_item_create("dataport", value));
-					if(source->buffermsg)
-						janus_config_add(config, c, janus_config_item_create("databuffermsg", "yes"));
-					json_t *diface = json_object_get(root, "dataiface");
-					if(diface)
-						janus_config_add(config, c, janus_config_item_create("dataiface", json_string_value(diface)));
-				}
 				if(source->srtpsuite > 0 && source->srtpcrypto) {
 					g_snprintf(value, BUFSIZ, "%d", source->srtpsuite);
 					janus_config_add(config, c, janus_config_item_create("srtpsuite", value));
@@ -2886,14 +3322,68 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 					g_snprintf(value, BUFSIZ, "%d", mp->helper_threads);
 					janus_config_add(config, c, janus_config_item_create("threads", value));
 				}
+				/* Iterate on all media streams */
+				janus_config_array *media = janus_config_array_create("media");
+				janus_config_add(config, c, media);
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					janus_config_category *m = janus_config_category_create(NULL);
+					janus_config_add(config, media, m);
+					janus_config_add(config, m, janus_config_item_create("type", janus_streaming_media_str(stream->type)));
+					janus_config_add(config, m, janus_config_item_create("mid", stream->mid));
+					janus_config_add(config, m, janus_config_item_create("label", stream->label));
+					if(stream->port[0] > 0) {
+						g_snprintf(value, BUFSIZ, "%d", stream->port[0]);
+						janus_config_add(config, m, janus_config_item_create("port", value));
+					}
+					if(stream->rtcp_port > 0) {
+						g_snprintf(value, BUFSIZ, "%d", stream->rtcp_port);
+						janus_config_add(config, m, janus_config_item_create("rtcpport", value));
+					}
+					if(stream->codecs.pt >= 0) {
+						g_snprintf(value, BUFSIZ, "%d", stream->codecs.pt);
+						janus_config_add(config, m, janus_config_item_create("pt", value));
+					}
+					if(stream->codecs.rtpmap) {
+						janus_config_add(config, m, janus_config_item_create("rtpmap", stream->codecs.rtpmap));
+						if(stream->codecs.fmtp)
+							janus_config_add(config, m, janus_config_item_create("fmtp", stream->codecs.fmtp));
+						if(stream->skew)
+							janus_config_add(config, m, janus_config_item_create("skew", "yes"));
+					}
+					if(stream->keyframe.enabled)
+						janus_config_add(config, m, janus_config_item_create("videobufferkf", "yes"));
+					if(stream->simulcast) {
+						janus_config_add(config, m, janus_config_item_create("videosimulcast", "yes"));
+						if(stream->port[1]) {
+							g_snprintf(value, BUFSIZ, "%d", stream->port[1]);
+							janus_config_add(config, m, janus_config_item_create("port2", value));
+						}
+						if(stream->port[2]) {
+							g_snprintf(value, BUFSIZ, "%d", stream->port[2]);
+							janus_config_add(config, m, janus_config_item_create("port3", value));
+						}
+					}
+					if(stream->svc)
+						janus_config_add(config, m, janus_config_item_create("videosvc", "yes"));
+					if(stream->skew)
+						janus_config_add(config, m, janus_config_item_create("skew", "yes"));
+					if(stream->mcast_str)
+						janus_config_add(config, m, janus_config_item_create("mcast", stream->mcast_str));
+					if(stream->iface_str)
+						janus_config_add(config, m, janus_config_item_create("iface", stream->iface_str));
+					if(stream->buffermsg)
+						janus_config_add(config, m, janus_config_item_create("databuffermsg", "yes"));
+					temp = temp->next;
+				}
 			} else if(!strcasecmp(type_text, "live") || !strcasecmp(type_text, "ondemand")) {
 				janus_streaming_file_source *source = mp->source;
 				janus_config_add(config, c, janus_config_item_create("filename", source->filename));
-				janus_config_add(config, c, janus_config_item_create("audio", mp->codecs.audio_pt >= 0 ? "yes" : "no"));
-				janus_config_add(config, c, janus_config_item_create("video", mp->codecs.video_pt > 0 ? "yes" : "no"));
+				janus_config_add(config, c, janus_config_item_create("audio", "yes"));
 			} else if(!strcasecmp(type_text, "rtsp")) {
-#ifdef HAVE_LIBCURL
 				janus_streaming_rtp_source *source = mp->source;
+#ifdef HAVE_LIBCURL
 				if(source->rtsp_url)
 					janus_config_add(config, c, janus_config_item_create("url", source->rtsp_url));
 				if(source->rtsp_username)
@@ -2901,29 +3391,29 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 				if(source->rtsp_password)
 					janus_config_add(config, c, janus_config_item_create("rtsp_pwd", source->rtsp_password));
 #endif
-				janus_config_add(config, c, janus_config_item_create("audio", mp->codecs.audio_pt >= 0 ? "yes" : "no"));
-				if(mp->codecs.audio_pt >= 0) {
-					if(mp->codecs.audio_rtpmap)
-						janus_config_add(config, c, janus_config_item_create("audiortpmap", mp->codecs.audio_rtpmap));
-					if(mp->codecs.audio_fmtp)
-						janus_config_add(config, c, janus_config_item_create("audiofmtp", mp->codecs.audio_fmtp));
-				}
-				janus_config_add(config, c, janus_config_item_create("video", mp->codecs.video_pt > 0 ? "yes" : "no"));
-				if(mp->codecs.video_pt > 0) {
-					if(mp->codecs.video_rtpmap)
-						janus_config_add(config, c, janus_config_item_create("videortpmap", mp->codecs.video_rtpmap));
-					if(mp->codecs.video_fmtp)
-						janus_config_add(config, c, janus_config_item_create("videofmtp", mp->codecs.video_fmtp));
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					/* FIXME Should we support RTSP streams with multiple media? */
+					if(stream->type == JANUS_STREAMING_MEDIA_AUDIO) {
+						janus_config_add(config, c, janus_config_item_create("audio", "yes"));
+						if(stream->codecs.rtpmap)
+							janus_config_add(config, c, janus_config_item_create("audiortpmap", stream->codecs.rtpmap));
+						if(stream->codecs.fmtp)
+							janus_config_add(config, c, janus_config_item_create("audiofmtp", stream->codecs.fmtp));
+					} else if(stream->type == JANUS_STREAMING_MEDIA_VIDEO) {
+						janus_config_add(config, c, janus_config_item_create("video", "yes"));
+						if(stream->codecs.rtpmap)
+							janus_config_add(config, c, janus_config_item_create("videortpmap", stream->codecs.rtpmap));
+						if(stream->codecs.fmtp)
+							janus_config_add(config, c, janus_config_item_create("videofmtp", stream->codecs.fmtp));
+					}
+					temp = temp->next;
 				}
 				json_t *iface = json_object_get(root, "rtspiface");
 				if(iface)
 					janus_config_add(config, c, janus_config_item_create("rtspiface", json_string_value(iface)));
 			}
-			/* Some more common values */
-			if(mp->secret)
-				janus_config_add(config, c, janus_config_item_create("secret", mp->secret));
-			if(mp->pin)
-				janus_config_add(config, c, janus_config_item_create("pin", mp->pin));
 			/* Save modified configuration */
 			if(janus_config_save(config, config_folder, JANUS_STREAMING_PACKAGE) < 0)
 				save = FALSE;	/* This will notify the user the mountpoint is not permanent */
@@ -2941,27 +3431,29 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 		json_object_set_new(ml, "is_private", mp->is_private ? json_true() : json_false());
 		if(!strcasecmp(type_text, "rtp")) {
 			janus_streaming_rtp_source *source = mp->source;
-			if(source->audio_fd != -1) {
-				json_object_set_new(ml, "audio_port", json_integer(source->audio_port));
-			}
-			if(source->audio_rtcp_fd != -1) {
-				json_object_set_new(ml, "audio_rtcp_port", json_integer(source->audio_rtcp_port));
-			}
-			if(source->video_fd[0] != -1) {
-				json_object_set_new(ml, "video_port", json_integer(source->video_port[0]));
-			}
-			if(source->video_rtcp_fd != -1) {
-				json_object_set_new(ml, "video_rtcp_port", json_integer(source->video_rtcp_port));
-			}
-			if(source->video_fd[1] != -1) {
-				json_object_set_new(ml, "video_port_2", json_integer(source->video_port[1]));
-			}
-			if(source->video_fd[2] != -1) {
-				json_object_set_new(ml, "video_port_3", json_integer(source->video_port[2]));
-			}
-			if(source->data_fd != -1) {
-				json_object_set_new(ml, "data_port", json_integer(source->data_port));
+			json_t *media = json_array();
+			GList *temp = source->media;
+			while(temp) {
+				janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+				json_t *info = json_object();
+				json_object_set_new(info, "type", json_string(janus_streaming_media_str(stream->type)));
+				json_object_set_new(info, "mid", json_string(stream->mid));
+				if(stream->fd[0] != -1) {
+					json_object_set_new(info, "port", json_integer(stream->port[0]));
+				}
+				if(stream->rtcp_fd != -1) {
+					json_object_set_new(info, "rtcp_port", json_integer(stream->rtcp_port));
+				}
+				if(stream->fd[1] != -1) {
+					json_object_set_new(info, "port_2", json_integer(stream->port[1]));
+				}
+				if(stream->fd[2] != -1) {
+					json_object_set_new(info, "port_3", json_integer(stream->port[2]));
+				}
+				json_array_append_new(media, info);
+				temp = temp->next;
 			}
+			json_object_set_new(ml, "ports", media);
 		}
 		json_object_set_new(response, "stream", ml);
 		/* Also notify event handlers */
@@ -3046,19 +3538,23 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 			g_free(old_pin);
 		}
 		if(save) {
-			JANUS_LOG(LOG_VERB, "Saving editted mountpoint %"SCNu64" permanently in config file\n", mp->id);
+			JANUS_LOG(LOG_VERB, "Saving edited mountpoint %"SCNu64" permanently in config file\n", mp->id);
 			janus_mutex_lock(&config_mutex);
 			char value[BUFSIZ];
 			/* Remove the old category first */
 			janus_config_remove(config, NULL, mp->name);
 			/* Now write the room details again */
 			janus_config_category *c = janus_config_get_create(config, NULL, janus_config_type_category, mp->name);
-			/* Now for the common values at top */
+			/* Now for the common values */
 			g_snprintf(value, BUFSIZ, "%"SCNu64, mp->id);
 			janus_config_add(config, c, janus_config_item_create("id", value));
 			janus_config_add(config, c, janus_config_item_create("description", mp->description));
 			if(mp->is_private)
 				janus_config_add(config, c, janus_config_item_create("is_private", "yes"));
+			if(mp->secret)
+				janus_config_add(config, c, janus_config_item_create("secret", mp->secret));
+			if(mp->pin)
+				janus_config_add(config, c, janus_config_item_create("pin", mp->pin));
 			/* Per type values */
 			if(mp->streaming_source == janus_streaming_source_rtp) {
 				gboolean rtsp = FALSE;
@@ -3068,6 +3564,7 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 						rtsp = TRUE;
 #endif
 				if(rtsp) {
+					janus_streaming_rtp_source *source = mp->source;
 #ifdef HAVE_LIBCURL
 					janus_config_add(config, c, janus_config_item_create("type", "rtsp"));
 					if(source->rtsp_url)
@@ -3077,99 +3574,36 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 					if(source->rtsp_password)
 						janus_config_add(config, c, janus_config_item_create("rtsp_pwd", source->rtsp_password));
 #endif
-					janus_config_add(config, c, janus_config_item_create("audio", mp->codecs.audio_pt >= 0 ? "yes" : "no"));
-					if(mp->codecs.audio_pt >= 0) {
-						if(mp->codecs.audio_rtpmap)
-							janus_config_add(config, c, janus_config_item_create("audiortpmap", mp->codecs.audio_rtpmap));
-						if(mp->codecs.audio_fmtp)
-							janus_config_add(config, c, janus_config_item_create("audiofmtp", mp->codecs.audio_fmtp));
-					}
-					janus_config_add(config, c, janus_config_item_create("video", mp->codecs.video_pt > 0 ? "yes" : "no"));
-					if(mp->codecs.video_pt > 0) {
-						if(mp->codecs.video_rtpmap)
-							janus_config_add(config, c, janus_config_item_create("videortpmap", mp->codecs.video_rtpmap));
-						if(mp->codecs.video_fmtp)
-							janus_config_add(config, c, janus_config_item_create("videofmtp", mp->codecs.video_fmtp));
+					GList *temp = source->media;
+					while(temp) {
+						janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+						/* FIXME Should we support RTSP streams with multiple media? */
+						if(stream->type == JANUS_STREAMING_MEDIA_AUDIO) {
+							janus_config_add(config, c, janus_config_item_create("audio", "yes"));
+							if(stream->codecs.rtpmap)
+								janus_config_add(config, c, janus_config_item_create("audiortpmap", stream->codecs.rtpmap));
+							if(stream->codecs.fmtp)
+								janus_config_add(config, c, janus_config_item_create("audiofmtp", stream->codecs.fmtp));
+						} else if(stream->type == JANUS_STREAMING_MEDIA_VIDEO) {
+							janus_config_add(config, c, janus_config_item_create("video", "yes"));
+							if(stream->codecs.rtpmap)
+								janus_config_add(config, c, janus_config_item_create("videortpmap", stream->codecs.rtpmap));
+							if(stream->codecs.fmtp)
+								janus_config_add(config, c, janus_config_item_create("videofmtp", stream->codecs.fmtp));
+						}
+						temp = temp->next;
 					}
 					json_t *iface = json_object_get(root, "rtspiface");
 					if(iface)
 						janus_config_add(config, c, janus_config_item_create("rtspiface", json_string_value(iface)));
 				} else {
 					janus_config_add(config, c, janus_config_item_create("type", "rtp"));
-					janus_config_add(config, c, janus_config_item_create("audio", mp->codecs.audio_pt >= 0 ? "yes" : "no"));
+					/* We save using the new format, not the old deprecated one */
 					janus_streaming_rtp_source *source = mp->source;
-					if(mp->codecs.audio_pt >= 0) {
-						g_snprintf(value, BUFSIZ, "%d", source->audio_port);
-						janus_config_add(config, c, janus_config_item_create("audioport", value));
-						if(source->audio_rtcp_port > 0) {
-							g_snprintf(value, BUFSIZ, "%d", source->audio_rtcp_port);
-							janus_config_add(config, c, janus_config_item_create("audiortcpport", value));
-						}
-						json_t *audiomcast = json_object_get(root, "audiomcast");
-						if(audiomcast)
-							janus_config_add(config, c, janus_config_item_create("audiomcast", json_string_value(audiomcast)));
-						g_snprintf(value, BUFSIZ, "%d", mp->codecs.audio_pt);
-						janus_config_add(config, c, janus_config_item_create("audiopt", value));
-						janus_config_add(config, c, janus_config_item_create("audiortpmap", mp->codecs.audio_rtpmap));
-						if(mp->codecs.audio_fmtp)
-							janus_config_add(config, c, janus_config_item_create("audiofmtp", mp->codecs.audio_fmtp));
-						json_t *aiface = json_object_get(root, "audioiface");
-						if(aiface)
-							janus_config_add(config, c, janus_config_item_create("audioiface", json_string_value(aiface)));
-						if(source->askew)
-							janus_config_add(config, c, janus_config_item_create("askew", "yes"));
-					}
-					janus_config_add(config, c, janus_config_item_create("video", mp->codecs.video_pt > 0 ? "yes" : "no"));
-					if(mp->codecs.video_pt > 0) {
-						g_snprintf(value, BUFSIZ, "%d", source->video_port[0]);
-						janus_config_add(config, c, janus_config_item_create("videoport", value));
-						if(source->video_rtcp_port > 0) {
-							g_snprintf(value, BUFSIZ, "%d", source->video_rtcp_port);
-							janus_config_add(config, c, janus_config_item_create("videortcpport", value));
-						}
-						json_t *videomcast = json_object_get(root, "videomcast");
-						if(videomcast)
-							janus_config_add(config, c, janus_config_item_create("videomcast", json_string_value(videomcast)));
-						g_snprintf(value, BUFSIZ, "%d", mp->codecs.video_pt);
-						janus_config_add(config, c, janus_config_item_create("videopt", value));
-						janus_config_add(config, c, janus_config_item_create("videortpmap", mp->codecs.video_rtpmap));
-						if(mp->codecs.video_fmtp)
-							janus_config_add(config, c, janus_config_item_create("videofmtp", mp->codecs.video_fmtp));
-						if(source->keyframe.enabled)
-							janus_config_add(config, c, janus_config_item_create("videobufferkf", "yes"));
-						if(source->simulcast) {
-							janus_config_add(config, c, janus_config_item_create("videosimulcast", "yes"));
-							if(source->video_port[1]) {
-								g_snprintf(value, BUFSIZ, "%d", source->video_port[1]);
-								janus_config_add(config, c, janus_config_item_create("videoport2", value));
-							}
-							if(source->video_port[2]) {
-								g_snprintf(value, BUFSIZ, "%d", source->video_port[2]);
-								janus_config_add(config, c, janus_config_item_create("videoport3", value));
-							}
-						}
-						if(source->svc)
-							janus_config_add(config, c, janus_config_item_create("videosvc", "yes"));
-						json_t *viface = json_object_get(root, "videoiface");
-						if(viface)
-							janus_config_add(config, c, janus_config_item_create("videoiface", json_string_value(viface)));
-						if(source->vskew)
-							janus_config_add(config, c, janus_config_item_create("videoskew", "yes"));
-					}
 					if(source->rtp_collision > 0) {
 						g_snprintf(value, BUFSIZ, "%d", source->rtp_collision);
 						janus_config_add(config, c, janus_config_item_create("collision", value));
 					}
-					janus_config_add(config, c, janus_config_item_create("data", mp->data ? "yes" : "no"));
-					if(source->data_port > -1) {
-						g_snprintf(value, BUFSIZ, "%d", source->data_port);
-						janus_config_add(config, c, janus_config_item_create("dataport", value));
-						if(source->buffermsg)
-							janus_config_add(config, c, janus_config_item_create("databuffermsg", "yes"));
-						json_t *diface = json_object_get(root, "dataiface");
-						if(diface)
-							janus_config_add(config, c, janus_config_item_create("dataiface", json_string_value(diface)));
-					}
 					if(source->srtpsuite > 0 && source->srtpcrypto) {
 						g_snprintf(value, BUFSIZ, "%d", source->srtpsuite);
 						janus_config_add(config, c, janus_config_item_create("srtpsuite", value));
@@ -3179,19 +3613,68 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 						g_snprintf(value, BUFSIZ, "%d", mp->helper_threads);
 						janus_config_add(config, c, janus_config_item_create("threads", value));
 					}
+					/* Iterate on all media streams */
+					janus_config_array *media = janus_config_array_create("media");
+					janus_config_add(config, c, media);
+					GList *temp = source->media;
+					while(temp) {
+						janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+						janus_config_category *m = janus_config_category_create(NULL);
+						janus_config_add(config, media, m);
+						janus_config_add(config, m, janus_config_item_create("type", janus_streaming_media_str(stream->type)));
+						janus_config_add(config, m, janus_config_item_create("mid", stream->mid));
+						janus_config_add(config, m, janus_config_item_create("label", stream->label));
+						if(stream->port[0] > 0) {
+							g_snprintf(value, BUFSIZ, "%d", stream->port[0]);
+							janus_config_add(config, m, janus_config_item_create("port", value));
+						}
+						if(stream->rtcp_port > 0) {
+							g_snprintf(value, BUFSIZ, "%d", stream->rtcp_port);
+							janus_config_add(config, m, janus_config_item_create("rtcpport", value));
+						}
+						if(stream->codecs.pt >= 0) {
+							g_snprintf(value, BUFSIZ, "%d", stream->codecs.pt);
+							janus_config_add(config, m, janus_config_item_create("pt", value));
+						}
+						if(stream->codecs.rtpmap) {
+							janus_config_add(config, m, janus_config_item_create("rtpmap", stream->codecs.rtpmap));
+							if(stream->codecs.fmtp)
+								janus_config_add(config, m, janus_config_item_create("fmtp", stream->codecs.fmtp));
+							if(stream->skew)
+								janus_config_add(config, m, janus_config_item_create("skew", "yes"));
+						}
+						if(stream->keyframe.enabled)
+							janus_config_add(config, m, janus_config_item_create("videobufferkf", "yes"));
+						if(stream->simulcast) {
+							janus_config_add(config, m, janus_config_item_create("videosimulcast", "yes"));
+							if(stream->port[1]) {
+								g_snprintf(value, BUFSIZ, "%d", stream->port[1]);
+								janus_config_add(config, m, janus_config_item_create("port2", value));
+							}
+							if(stream->port[2]) {
+								g_snprintf(value, BUFSIZ, "%d", stream->port[2]);
+								janus_config_add(config, m, janus_config_item_create("port3", value));
+							}
+						}
+						if(stream->svc)
+							janus_config_add(config, m, janus_config_item_create("videosvc", "yes"));
+						if(stream->skew)
+							janus_config_add(config, m, janus_config_item_create("skew", "yes"));
+						if(stream->mcast_str)
+							janus_config_add(config, m, janus_config_item_create("mcast", stream->mcast_str));
+						if(stream->iface_str)
+							janus_config_add(config, m, janus_config_item_create("iface", stream->iface_str));
+						if(stream->buffermsg)
+							janus_config_add(config, m, janus_config_item_create("databuffermsg", "yes"));
+						temp = temp->next;
+					}
 				}
 			} else {
 				janus_config_add(config, c, janus_config_item_create("type", (mp->streaming_type == janus_streaming_type_live) ? "live" : "ondemand"));
 				janus_streaming_file_source *source = mp->source;
 				janus_config_add(config, c, janus_config_item_create("filename", source->filename));
-				janus_config_add(config, c, janus_config_item_create("audio", mp->codecs.audio_pt >= 0 ? "yes" : "no"));
-				janus_config_add(config, c, janus_config_item_create("video", mp->codecs.video_pt > 0 ? "yes" : "no"));
+				janus_config_add(config, c, janus_config_item_create("audio", "yes"));
 			}
-			/* Some more common values */
-			if(mp->secret)
-				janus_config_add(config, c, janus_config_item_create("secret", mp->secret));
-			if(mp->pin)
-				janus_config_add(config, c, janus_config_item_create("pin", mp->pin));
 			/* Save modified configuration */
 			if(janus_config_save(config, config_folder, JANUS_STREAMING_PACKAGE) < 0)
 				save = FALSE;	/* This will notify the user the mountpoint is not permanent */
@@ -3379,18 +3862,85 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 				janus_mutex_unlock(&mountpoints_mutex);
 				goto prepare_response;
 			}
+			json_t *media = json_object_get(root, "media");
 			json_t *audio = json_object_get(root, "audio");
 			json_t *video = json_object_get(root, "video");
 			json_t *data = json_object_get(root, "data");
-			janus_recorder *arc = NULL, *vrc = NULL, *drc = NULL;
-			if((audio && source->arc) || (video && source->vrc) || (data && source->drc)) {
-				janus_refcount_decrease(&mp->ref);
-				janus_mutex_unlock(&mountpoints_mutex);
-				JANUS_LOG(LOG_ERR, "Recording for audio, video and/or data already started for this stream\n");
-				error_code = JANUS_STREAMING_ERROR_INVALID_REQUEST;
-				g_snprintf(error_cause, 512, "Recording for audio, video and/or data already started for this stream");
+			if(media) {
+				/* Iterate on all media to stop */
+				janus_mutex_lock(&source->rec_mutex);
+				if(json_array_size(media) > 0) {
+					size_t i = 0;
+					for(i=0; i<json_array_size(media); i++) {
+						json_t *m = json_array_get(media, i);
+						if(!json_is_object(m))
+							continue;
+						const char *mid = json_string_value(json_object_get(m, "mid"));
+						const char *filename = json_string_value(json_object_get(m, "filename"));
+						if(mid == NULL || filename == NULL) {
+							JANUS_LOG(LOG_WARN, "No mid/filename, can't start recording for this stream...\n");
+							continue;
+						}
+						GList *temp = source->media;
+						while(temp) {
+							janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+							if(strcasecmp(stream->mid, mid)) {
+								temp = temp->next;
+								continue;
+							}
+							if(stream->rc) {
+								janus_mutex_unlock(&source->rec_mutex);
+								janus_refcount_decrease(&mp->ref);
+								janus_mutex_unlock(&mountpoints_mutex);
+								JANUS_LOG(LOG_ERR, "Recording already started for this stream\n");
+								error_code = JANUS_STREAMING_ERROR_INVALID_REQUEST;
+								g_snprintf(error_cause, 512, "Recording for audio, video and/or data already started for this stream");
+								goto prepare_response;
+							}
+							const char *codec = NULL;
+							if(stream->type == JANUS_STREAMING_MEDIA_AUDIO) {
+								if(strstr(stream->codecs.rtpmap, "opus") || strstr(stream->codecs.rtpmap, "OPUS"))
+									codec = "opus";
+								else if(strstr(stream->codecs.rtpmap, "pcma") || strstr(stream->codecs.rtpmap, "PCMA"))
+									codec = "pcma";
+								else if(strstr(stream->codecs.rtpmap, "pcmu") || strstr(stream->codecs.rtpmap, "PCMU"))
+									codec = "pcmu";
+								else if(strstr(stream->codecs.rtpmap, "g722") || strstr(stream->codecs.rtpmap, "G722"))
+									codec = "g722";
+							} else if(stream->type == JANUS_STREAMING_MEDIA_VIDEO) {
+								if(strstr(stream->codecs.rtpmap, "vp8") || strstr(stream->codecs.rtpmap, "VP8"))
+									codec = "vp8";
+								else if(strstr(stream->codecs.rtpmap, "vp9") || strstr(stream->codecs.rtpmap, "VP9"))
+									codec = "vp9";
+								else if(strstr(stream->codecs.rtpmap, "h264") || strstr(stream->codecs.rtpmap, "H264"))
+									codec = "h264";
+							} else {
+								codec = "text";
+							}
+							stream->rc = janus_recorder_create(NULL, codec, (char *)filename);
+							if(stream->rc == NULL) {
+								JANUS_LOG(LOG_ERR, "[%s] Error starting recorder for %s\n",
+									mp->name, janus_streaming_media_str(stream->type));
+								janus_mutex_unlock(&source->rec_mutex);
+								janus_refcount_decrease(&mp->ref);
+								janus_mutex_unlock(&mountpoints_mutex);
+								error_code = JANUS_STREAMING_ERROR_CANT_RECORD;
+								g_snprintf(error_cause, 512, "Error starting recorder for %s", janus_streaming_media_str(stream->type));
+								goto prepare_response;
+							}
+							JANUS_LOG(LOG_INFO, "[%s] %s recording started (%s)\n",
+								mp->name, janus_streaming_media_str(stream->type), stream->mid);
+							break;
+						}
+					}
+				}
+				janus_mutex_unlock(&source->rec_mutex);
+				/* Send a success response back */
+				response = json_object();
+				json_object_set_new(response, "streaming", json_string("ok"));
 				goto prepare_response;
 			}
+			/* If we're here, we're using the old method to record */
 			if(!audio && !video && !data) {
 				janus_refcount_decrease(&mp->ref);
 				janus_mutex_unlock(&mountpoints_mutex);
@@ -3399,86 +3949,124 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 				g_snprintf(error_cause, 512, "Missing audio, video and/or data");
 				goto prepare_response;
 			}
+			janus_mutex_lock(&source->rec_mutex);
 			if(audio) {
-				const char *codec = NULL;
-				if (!mp->codecs.audio_rtpmap)
-					JANUS_LOG(LOG_ERR, "[%s] Audio RTP map is uninitialized\n", mp->name);
-				else if(strstr(mp->codecs.audio_rtpmap, "opus") || strstr(mp->codecs.audio_rtpmap, "OPUS"))
-					codec = "opus";
-				else if(strstr(mp->codecs.audio_rtpmap, "pcma") || strstr(mp->codecs.audio_rtpmap, "PCMA"))
-					codec = "pcma";
-				else if(strstr(mp->codecs.audio_rtpmap, "pcmu") || strstr(mp->codecs.audio_rtpmap, "PCMU"))
-					codec = "pcmu";
-				else if(strstr(mp->codecs.audio_rtpmap, "g722") || strstr(mp->codecs.audio_rtpmap, "G722"))
-					codec = "g722";
 				const char *audiofile = json_string_value(audio);
-				arc = janus_recorder_create(NULL, codec, (char *)audiofile);
-				if(arc == NULL) {
-					JANUS_LOG(LOG_ERR, "[%s] Error starting recorder for audio\n", mp->name);
-					janus_refcount_decrease(&mp->ref);
-					janus_mutex_unlock(&mountpoints_mutex);
-					error_code = JANUS_STREAMING_ERROR_CANT_RECORD;
-					g_snprintf(error_cause, 512, "Error starting recorder for audio");
-					goto prepare_response;
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					if(stream->type != JANUS_STREAMING_MEDIA_AUDIO) {
+						temp = temp->next;
+						continue;
+					}
+					if(stream->rc) {
+						janus_mutex_unlock(&source->rec_mutex);
+						janus_refcount_decrease(&mp->ref);
+						janus_mutex_unlock(&mountpoints_mutex);
+						JANUS_LOG(LOG_ERR, "Recording already started for this audio stream\n");
+						error_code = JANUS_STREAMING_ERROR_INVALID_REQUEST;
+						g_snprintf(error_cause, 512, "Recording for audio started for this stream");
+						goto prepare_response;
+					}
+					const char *codec = NULL;
+					if(!stream->codecs.rtpmap)
+						JANUS_LOG(LOG_ERR, "[%s] Audio RTP map is uninitialized\n", mp->name);
+					else if(strstr(stream->codecs.rtpmap, "opus") || strstr(stream->codecs.rtpmap, "OPUS"))
+						codec = "opus";
+					else if(strstr(stream->codecs.rtpmap, "pcma") || strstr(stream->codecs.rtpmap, "PCMA"))
+						codec = "pcma";
+					else if(strstr(stream->codecs.rtpmap, "pcmu") || strstr(stream->codecs.rtpmap, "PCMU"))
+						codec = "pcmu";
+					else if(strstr(stream->codecs.rtpmap, "g722") || strstr(stream->codecs.rtpmap, "G722"))
+						codec = "g722";
+					stream->rc = janus_recorder_create(NULL, codec, (char *)audiofile);
+					if(stream->rc == NULL) {
+						JANUS_LOG(LOG_ERR, "[%s] Error starting recorder for audio\n", mp->name);
+						janus_mutex_unlock(&source->rec_mutex);
+						janus_refcount_decrease(&mp->ref);
+						janus_mutex_unlock(&mountpoints_mutex);
+						error_code = JANUS_STREAMING_ERROR_CANT_RECORD;
+						g_snprintf(error_cause, 512, "Error starting recorder for audio");
+						goto prepare_response;
+					}
+					JANUS_LOG(LOG_INFO, "[%s] Audio recording started (%s)\n", mp->name, stream->mid);
+					break;
 				}
-				JANUS_LOG(LOG_INFO, "[%s] Audio recording started\n", mp->name);
 			}
 			if(video) {
-				const char *codec = NULL;
-				if (!mp->codecs.video_rtpmap)
-					JANUS_LOG(LOG_ERR, "[%s] Video RTP map is uninitialized\n", mp->name);
-				else if(strstr(mp->codecs.video_rtpmap, "vp8") || strstr(mp->codecs.video_rtpmap, "VP8"))
-					codec = "vp8";
-				else if(strstr(mp->codecs.video_rtpmap, "vp9") || strstr(mp->codecs.video_rtpmap, "VP9"))
-					codec = "vp9";
-				else if(strstr(mp->codecs.video_rtpmap, "h264") || strstr(mp->codecs.video_rtpmap, "H264"))
-					codec = "h264";
 				const char *videofile = json_string_value(video);
-				vrc = janus_recorder_create(NULL, codec, (char *)videofile);
-				if(vrc == NULL) {
-					if(arc != NULL) {
-						janus_recorder_close(arc);
-						janus_recorder_destroy(arc);
-						arc = NULL;
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					if(stream->type != JANUS_STREAMING_MEDIA_VIDEO) {
+						temp = temp->next;
+						continue;
 					}
-					JANUS_LOG(LOG_ERR, "[%s] Error starting recorder for video\n", mp->name);
-					janus_refcount_decrease(&mp->ref);
-					janus_mutex_unlock(&mountpoints_mutex);
-					error_code = JANUS_STREAMING_ERROR_CANT_RECORD;
-					g_snprintf(error_cause, 512, "Error starting recorder for video");
-					goto prepare_response;
+					if(stream->rc) {
+						janus_mutex_unlock(&source->rec_mutex);
+						janus_refcount_decrease(&mp->ref);
+						janus_mutex_unlock(&mountpoints_mutex);
+						JANUS_LOG(LOG_ERR, "Recording already started for this video stream\n");
+						error_code = JANUS_STREAMING_ERROR_INVALID_REQUEST;
+						g_snprintf(error_cause, 512, "Recording for video started for this stream");
+						goto prepare_response;
+					}
+					const char *codec = NULL;
+					if(!stream->codecs.rtpmap)
+						JANUS_LOG(LOG_ERR, "[%s] Video RTP map is uninitialized\n", mp->name);
+					else if(strstr(stream->codecs.rtpmap, "vp8") || strstr(stream->codecs.rtpmap, "VP8"))
+						codec = "vp8";
+					else if(strstr(stream->codecs.rtpmap, "vp9") || strstr(stream->codecs.rtpmap, "VP9"))
+						codec = "vp9";
+					else if(strstr(stream->codecs.rtpmap, "h264") || strstr(stream->codecs.rtpmap, "H264"))
+						codec = "h264";
+					stream->rc = janus_recorder_create(NULL, codec, (char *)videofile);
+					if(stream->rc == NULL) {
+						janus_mutex_unlock(&source->rec_mutex);
+						JANUS_LOG(LOG_ERR, "[%s] Error starting recorder for video\n", mp->name);
+						janus_refcount_decrease(&mp->ref);
+						janus_mutex_unlock(&mountpoints_mutex);
+						error_code = JANUS_STREAMING_ERROR_CANT_RECORD;
+						g_snprintf(error_cause, 512, "Error starting recorder for video");
+						goto prepare_response;
+					}
+					JANUS_LOG(LOG_INFO, "[%s] Video recording started (%s)\n", mp->name, stream->mid);
+					break;
 				}
-				JANUS_LOG(LOG_INFO, "[%s] Video recording started\n", mp->name);
 			}
 			if(data) {
 				const char *datafile = json_string_value(data);
-				drc = janus_recorder_create(NULL, "text", (char *)datafile);
-				if(drc == NULL) {
-					if(arc != NULL) {
-						janus_recorder_close(arc);
-						janus_recorder_destroy(arc);
-						arc = NULL;
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					if(stream->type != JANUS_STREAMING_MEDIA_DATA) {
+						temp = temp->next;
+						continue;
 					}
-					if(vrc != NULL) {
-						janus_recorder_close(vrc);
-						janus_recorder_destroy(vrc);
-						vrc = NULL;
+					if(stream->rc) {
+						janus_mutex_unlock(&source->rec_mutex);
+						janus_refcount_decrease(&mp->ref);
+						janus_mutex_unlock(&mountpoints_mutex);
+						JANUS_LOG(LOG_ERR, "Recording already started for this data stream\n");
+						error_code = JANUS_STREAMING_ERROR_INVALID_REQUEST;
+						g_snprintf(error_cause, 512, "Recording for data started for this stream");
+						goto prepare_response;
 					}
-					JANUS_LOG(LOG_ERR, "[%s] Error starting recorder for data\n", mp->name);
-					janus_refcount_decrease(&mp->ref);
-					janus_mutex_unlock(&mountpoints_mutex);
-					error_code = JANUS_STREAMING_ERROR_CANT_RECORD;
-					g_snprintf(error_cause, 512, "Error starting recorder for data");
-					goto prepare_response;
+					stream->rc = janus_recorder_create(NULL, "text", (char *)datafile);
+					if(stream->rc == NULL) {
+						JANUS_LOG(LOG_ERR, "[%s] Error starting recorder for data\n", mp->name);
+						janus_mutex_unlock(&source->rec_mutex);
+						janus_refcount_decrease(&mp->ref);
+						janus_mutex_unlock(&mountpoints_mutex);
+						error_code = JANUS_STREAMING_ERROR_CANT_RECORD;
+						g_snprintf(error_cause, 512, "Error starting recorder for data");
+						goto prepare_response;
+					}
+					JANUS_LOG(LOG_INFO, "[%s] Data recording started (%s)\n", mp->name, stream->mid);
+					break;
 				}
-				JANUS_LOG(LOG_INFO, "[%s] Data recording started\n", mp->name);
 			}
-			if(arc != NULL)
-				source->arc = arc;
-			if(vrc != NULL)
-				source->vrc = vrc;
-			if(drc != NULL)
-				source->drc = drc;
+			janus_mutex_unlock(&source->rec_mutex);
 			janus_refcount_decrease(&mp->ref);
 			janus_mutex_unlock(&mountpoints_mutex);
 			/* Send a success response back */
@@ -3494,40 +4082,108 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 				janus_mutex_unlock(&mountpoints_mutex);
 				goto prepare_response;
 			}
+			json_t *media = json_object_get(root, "media");
 			json_t *audio = json_object_get(root, "audio");
 			json_t *video = json_object_get(root, "video");
 			json_t *data = json_object_get(root, "data");
-			if(!audio && !video) {
+			if(media) {
+				/* Iterate on all media to stop */
+				janus_mutex_lock(&source->rec_mutex);
+				if(json_array_size(media) > 0) {
+					size_t i = 0;
+					for(i=0; i<json_array_size(media); i++) {
+						json_t *m = json_array_get(media, i);
+						if(!json_is_string(m))
+							continue;
+						const char *mid = json_string_value(m);
+						if(mid == NULL)
+							continue;
+						GList *temp = source->media;
+						while(temp) {
+							janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+							if(strcasecmp(stream->mid, mid)) {
+								temp = temp->next;
+								continue;
+							}
+							janus_recorder_close(stream->rc);
+							JANUS_LOG(LOG_INFO, "[%s] Closed %s recording %s (%s)\n",
+								mp->name, janus_streaming_media_str(stream->type),
+								stream->rc->filename ? stream->rc->filename : "??", stream->mid);
+							janus_recorder *tmp = stream->rc;
+							stream->rc = NULL;
+							janus_recorder_destroy(tmp);
+							break;
+						}
+					}
+				}
+				janus_mutex_unlock(&source->rec_mutex);
+				/* Send a success response back */
+				response = json_object();
+				json_object_set_new(response, "streaming", json_string("ok"));
+				goto prepare_response;
+			}
+			/* If we're here, we're using the old method to stop recording */
+			if(!audio && !video && !data) {
 				janus_mutex_unlock(&mountpoints_mutex);
-				JANUS_LOG(LOG_ERR, "Missing audio and/or video\n");
+				JANUS_LOG(LOG_ERR, "Missing audio, video and or data\n");
 				error_code = JANUS_STREAMING_ERROR_INVALID_REQUEST;
-				g_snprintf(error_cause, 512, "Missing audio and/or video");
+				g_snprintf(error_cause, 512, "Missing audio, video and or data");
 				goto prepare_response;
 			}
 			janus_mutex_lock(&source->rec_mutex);
-			if(audio && json_is_true(audio) && source->arc) {
+			if(audio && json_is_true(audio)) {
 				/* Close the audio recording */
-				janus_recorder_close(source->arc);
-				JANUS_LOG(LOG_INFO, "[%s] Closed audio recording %s\n", mp->name, source->arc->filename ? source->arc->filename : "??");
-				janus_recorder *tmp = source->arc;
-				source->arc = NULL;
-				janus_recorder_destroy(tmp);
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					if(stream->type != JANUS_STREAMING_MEDIA_AUDIO) {
+						temp = temp->next;
+						continue;
+					}
+					janus_recorder_close(stream->rc);
+					JANUS_LOG(LOG_INFO, "[%s] Closed audio recording %s (%s)\n",
+						mp->name, stream->rc->filename ? stream->rc->filename : "??", stream->mid);
+					janus_recorder *tmp = stream->rc;
+					stream->rc = NULL;
+					janus_recorder_destroy(tmp);
+					break;
+				}
 			}
-			if(video && json_is_true(video) && source->vrc) {
+			if(video && json_is_true(video)) {
 				/* Close the video recording */
-				janus_recorder_close(source->vrc);
-				JANUS_LOG(LOG_INFO, "[%s] Closed video recording %s\n", mp->name, source->vrc->filename ? source->vrc->filename : "??");
-				janus_recorder *tmp = source->vrc;
-				source->vrc = NULL;
-				janus_recorder_destroy(tmp);
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					if(stream->type != JANUS_STREAMING_MEDIA_VIDEO) {
+						temp = temp->next;
+						continue;
+					}
+					janus_recorder_close(stream->rc);
+					JANUS_LOG(LOG_INFO, "[%s] Closed video recording %s (%s)\n",
+						mp->name, stream->rc->filename ? stream->rc->filename : "??", stream->mid);
+					janus_recorder *tmp = stream->rc;
+					stream->rc = NULL;
+					janus_recorder_destroy(tmp);
+					break;
+				}
 			}
-			if(data && json_is_true(data) && source->drc) {
+			if(data && json_is_true(data)) {
 				/* Close the data recording */
-				janus_recorder_close(source->drc);
-				JANUS_LOG(LOG_INFO, "[%s] Closed data recording %s\n", mp->name, source->drc->filename ? source->drc->filename : "??");
-				janus_recorder *tmp = source->drc;
-				source->drc = NULL;
-				janus_recorder_destroy(tmp);
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					if(stream->type != JANUS_STREAMING_MEDIA_DATA) {
+						temp = temp->next;
+						continue;
+					}
+					janus_recorder_close(stream->rc);
+					JANUS_LOG(LOG_INFO, "[%s] Closed data recording %s (%s)\n",
+						mp->name, stream->rc->filename ? stream->rc->filename : "??", stream->mid);
+					janus_recorder *tmp = stream->rc;
+					stream->rc = NULL;
+					janus_recorder_destroy(tmp);
+					break;
+				}
 			}
 			janus_mutex_unlock(&source->rec_mutex);
 			janus_refcount_decrease(&mp->ref);
@@ -3587,26 +4243,17 @@ static json_t *janus_streaming_process_synchronous_request(janus_streaming_sessi
 			if(mp->streaming_source == janus_streaming_source_rtp && stop_recording) {
 				janus_streaming_rtp_source *source = mp->source;
 				janus_mutex_lock(&source->rec_mutex);
-				if(source->arc) {
-					janus_recorder_close(source->arc);
-					JANUS_LOG(LOG_INFO, "[%s] Closed audio recording %s\n", mp->name, source->arc->filename ? source->arc->filename : "??");
-					janus_recorder *tmp = source->arc;
-					source->arc = NULL;
-					janus_recorder_destroy(tmp);
-				}
-				if(source->vrc) {
-					janus_recorder_close(source->vrc);
-					JANUS_LOG(LOG_INFO, "[%s] Closed video recording %s\n", mp->name, source->vrc->filename ? source->vrc->filename : "??");
-					janus_recorder *tmp = source->vrc;
-					source->vrc = NULL;
-					janus_recorder_destroy(tmp);
-				}
-				if(source->drc) {
-					janus_recorder_close(source->drc);
-					JANUS_LOG(LOG_INFO, "[%s] Closed data recording %s\n", mp->name, source->drc->filename ? source->drc->filename : "??");
-					janus_recorder *tmp = source->drc;
-					source->drc = NULL;
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					janus_recorder_close(stream->rc);
+					JANUS_LOG(LOG_INFO, "[%s] Closed %s recording %s (%s)\n",
+						mp->name, janus_streaming_media_str(stream->type),
+						stream->rc->filename ? stream->rc->filename : "??", stream->mid);
+					janus_recorder *tmp = stream->rc;
+					stream->rc = NULL;
 					janus_recorder_destroy(tmp);
+					temp = temp->next;
 				}
 				janus_mutex_unlock(&source->rec_mutex);
 			}
@@ -3799,34 +4446,38 @@ void janus_streaming_setup_media(janus_plugin_session *handle) {
 	}
 	janus_refcount_increase(&session->ref);
 	janus_mutex_unlock(&sessions_mutex);
-	g_atomic_int_set(&session->hangingup, 0);
 	/* We only start streaming towards this user when we get this event */
-	janus_rtp_switching_context_reset(&session->context);
+	g_atomic_int_set(&session->hangingup, 0);
 	/* If this is related to a live RTP mountpoint, any keyframe we can shoot already? */
 	janus_streaming_mountpoint *mountpoint = session->mountpoint;
 	if(mountpoint->streaming_source == janus_streaming_source_rtp) {
 		janus_streaming_rtp_source *source = mountpoint->source;
-		if(source->keyframe.enabled) {
-			JANUS_LOG(LOG_HUGE, "Any keyframe to send?\n");
-			janus_mutex_lock(&source->keyframe.mutex);
-			if(source->keyframe.latest_keyframe != NULL) {
-				JANUS_LOG(LOG_HUGE, "Yep! %d packets\n", g_list_length(source->keyframe.latest_keyframe));
-				GList *temp = source->keyframe.latest_keyframe;
-				while(temp) {
-					janus_streaming_relay_rtp_packet(session, temp->data);
-					temp = temp->next;
+		GList *temp = source->media;
+		while(temp) {
+			janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+			if(stream->keyframe.enabled) {
+				JANUS_LOG(LOG_HUGE, "Any keyframe to send? (%s)\n", stream->mid);
+				janus_mutex_lock(&stream->keyframe.mutex);
+				if(stream->keyframe.latest_keyframe != NULL) {
+					JANUS_LOG(LOG_HUGE, "Yep! %d packets\n", g_list_length(stream->keyframe.latest_keyframe));
+					GList *temp = stream->keyframe.latest_keyframe;
+					while(temp) {
+						janus_streaming_relay_rtp_packet(session, temp->data);
+						temp = temp->next;
+					}
 				}
+				janus_mutex_unlock(&stream->keyframe.mutex);
 			}
-			janus_mutex_unlock(&source->keyframe.mutex);
-		}
-		if(source->buffermsg) {
-			JANUS_LOG(LOG_HUGE, "Any recent datachannel message to send?\n");
-			janus_mutex_lock(&source->buffermsg_mutex);
-			if(source->last_msg != NULL) {
-				JANUS_LOG(LOG_HUGE, "Yep!\n");
-				janus_streaming_relay_rtp_packet(session, source->last_msg);
+			if(stream->buffermsg) {
+				JANUS_LOG(LOG_HUGE, "Any recent datachannel message to send? (%s)\n", stream->mid);
+				janus_mutex_lock(&stream->buffermsg_mutex);
+				if(stream->last_msg != NULL) {
+					JANUS_LOG(LOG_HUGE, "Yep!\n");
+					janus_streaming_relay_rtp_packet(session, stream->last_msg);
+				}
+				janus_mutex_unlock(&stream->buffermsg_mutex);
 			}
-			janus_mutex_unlock(&source->buffermsg_mutex);
+			temp = temp->next;
 		}
 	}
 	session->started = TRUE;
@@ -3842,13 +4493,13 @@ void janus_streaming_setup_media(janus_plugin_session *handle) {
 	janus_refcount_decrease(&session->ref);
 }
 
-void janus_streaming_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_streaming_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	/* FIXME We don't care about what the browser sends us, we're sendonly */
 }
 
-void janus_streaming_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_streaming_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	janus_streaming_session *session = (janus_streaming_session *)handle->plugin_handle;
@@ -3858,18 +4509,22 @@ void janus_streaming_incoming_rtcp(janus_plugin_session *handle, int video, char
 	if(mp->streaming_source != janus_streaming_source_rtp)
 		return;
 	janus_streaming_rtp_source *source = (janus_streaming_rtp_source *)mp->source;
-	if(!video && (source->audio_rtcp_fd > -1) && (source->audio_rtcp_addr.ss_family != 0)) {
-		JANUS_LOG(LOG_HUGE, "Got audio RTCP feedback from a viewer: SSRC %"SCNu32"\n",
-			janus_rtcp_get_sender_ssrc(buf, len));
+	/* Check which stream this feedback refers to */
+	janus_streaming_rtp_source_stream *stream = g_hash_table_lookup(source->media_byid, GINT_TO_POINTER(mindex));
+	if(stream == NULL)
+		return;
+	if(stream->type != JANUS_STREAMING_MEDIA_VIDEO && (stream->rtcp_fd > -1) && (stream->rtcp_addr.ss_family != 0)) {
+		JANUS_LOG(LOG_HUGE, "Got audio RTCP feedback from a viewer: #%d, SSRC %"SCNu32"\n",
+			mindex, janus_rtcp_get_sender_ssrc(buf, len));
 		/* FIXME We don't forward RR packets, so what should we check here? */
-	} else if(video && (source->video_rtcp_fd > -1) && (source->video_rtcp_addr.ss_family != 0)) {
-		JANUS_LOG(LOG_HUGE, "Got video RTCP feedback from a viewer: SSRC %"SCNu32"\n",
-			janus_rtcp_get_sender_ssrc(buf, len));
+	} else if(stream->type == JANUS_STREAMING_MEDIA_VIDEO && (stream->rtcp_fd > -1) && (stream->rtcp_addr.ss_family != 0)) {
+		JANUS_LOG(LOG_HUGE, "Got video RTCP feedback from a viewer: #%d, SSRC %"SCNu32"\n",
+			mindex, janus_rtcp_get_sender_ssrc(buf, len));
 		/* We only relay PLI/FIR and REMB packets, but in a selective way */
 		if(janus_rtcp_has_fir(buf, len) || janus_rtcp_has_pli(buf, len)) {
 			/* We got a PLI/FIR, pass it along unless we just sent one */
 			JANUS_LOG(LOG_HUGE, "  -- Keyframe request\n");
-			janus_streaming_rtcp_pli_send(source);
+			janus_streaming_rtcp_pli_send(stream);
 		}
 		uint64_t bw = janus_rtcp_get_remb(buf, len);
 		if(bw > 0) {
@@ -3900,13 +4555,6 @@ static void janus_streaming_hangup_media_internal(janus_plugin_session *handle)
 		return;
 	if(!g_atomic_int_compare_and_exchange(&session->hangingup, 0, 1))
 		return;
-	janus_rtp_switching_context_reset(&session->context);
-	janus_rtp_simulcasting_context_reset(&session->sim_context);
-	janus_vp8_simulcast_context_reset(&session->vp8_context);
-	session->spatial_layer = -1;
-	session->target_spatial_layer = 1;	/* FIXME Chrome sends 0 and 1 */
-	session->temporal_layer = -1;
-	session->target_temporal_layer = 2;	/* FIXME Chrome sends 0, 1 and 2 */
 	session->stopping = TRUE;
 	session->started = FALSE;
 	session->paused = FALSE;
@@ -3942,6 +4590,13 @@ static void janus_streaming_hangup_media_internal(janus_plugin_session *handle)
 		janus_mutex_unlock(&mp->mutex);
 	}
 	session->mountpoint = NULL;
+	/* Get rid of streams and streams_byid */
+	g_list_free_full(session->streams, (GDestroyNotify)(janus_streaming_session_stream_free));
+	session->streams = NULL;
+	if(session->streams_byid != NULL)
+		g_hash_table_unref(session->streams_byid);
+	session->streams_byid = NULL;
+
 	g_atomic_int_set(&session->hangingup, 0);
 }
 
@@ -4004,12 +4659,30 @@ static void *janus_streaming_handler(void *data) {
 			if(error_code != 0)
 				goto error;
 			json_t *id = json_object_get(root, "id");
+			guint64 id_value = json_integer_value(id);
+			/* The proper way is listing the mids of the streams we want to
+			 * receive: a missing or empty list will subscribe to them all */
+			json_t *mids = json_object_get(root, "media");
+			if(mids != NULL && json_array_size(mids) > 0) {
+				size_t i = 0;
+				for(i=0; i<json_array_size(mids); i++) {
+					json_t *s = json_array_get(mids, i);
+					if(!json_is_string(s)) {
+						error_code = JANUS_STREAMING_ERROR_INVALID_ELEMENT;
+						g_snprintf(error_cause, 512, "The media array must only contain strings (mid values)");
+						goto error;
+					}
+				}
+			}
+			/* The offer_audio/video/data attribute are deprecated,
+			 * but we keep them there for backwards compatibility */
 			json_t *offer_audio = json_object_get(root, "offer_audio");
 			json_t *offer_video = json_object_get(root, "offer_video");
 			json_t *offer_data = json_object_get(root, "offer_data");
-			guint64 id_value = json_integer_value(id);
+			/* There may be an ICE restart request involved */
 			json_t *restart = json_object_get(root, "restart");
 			do_restart = restart ? json_is_true(restart) : FALSE;
+			/* Find the mountpoint and go on */
 			janus_mutex_lock(&mountpoints_mutex);
 			janus_streaming_mountpoint *mp = g_hash_table_lookup(mountpoints, &id_value);
 			if(mp == NULL) {
@@ -4066,19 +4739,25 @@ static void *janus_streaming_handler(void *data) {
 			session->mountpoint = mp;
 			session->sdp_version = 1;	/* This needs to be increased when it changes */
 			session->sdp_sessid = janus_get_real_time();
-			/* Check what we should offer */
-			session->audio = offer_audio ? json_is_true(offer_audio) : TRUE;	/* True by default */
+			/* Check what we should offer: the new way of subscribing is
+			 * specifying the streams we're interested in by mid */
+			gboolean legacy = FALSE;
+			if(mids == NULL || json_array_size(mids) == 0) {
+				/* We'll treat this as a legacy request, even though
+				 * it may simply be a "subscribe to all streams" */
+				legacy = TRUE;
+			}
+			/* These are deprecated attributes, that we handle for backwards compatibility */
+			gboolean audio = offer_audio ? json_is_true(offer_audio) : TRUE;	/* True by default */
 			if(!mp->audio)
-				session->audio = FALSE;	/* ... unless the mountpoint isn't sending any audio */
-			session->video = offer_video ? json_is_true(offer_video) : TRUE;	/* True by default */
+				audio = FALSE;	/* ... unless the mountpoint isn't sending any audio */
+			gboolean video = offer_video ? json_is_true(offer_video) : TRUE;	/* True by default */
 			if(!mp->video)
-				session->video = FALSE;	/* ... unless the mountpoint isn't sending any video */
-			session->data = offer_data ? json_is_true(offer_data) : TRUE;	/* True by default */
+				video = FALSE;	/* ... unless the mountpoint isn't sending any video */
+			gboolean data = offer_data ? json_is_true(offer_data) : TRUE;	/* True by default */
 			if(!mp->data)
-				session->data = FALSE;	/* ... unless the mountpoint isn't sending any data */
-			if((!mp->audio || !session->audio) &&
-					(!mp->video || !session->video) &&
-					(!mp->data || !session->data)) {
+				data = FALSE;	/* ... unless the mountpoint isn't sending any data */
+			if((!mp->audio || !audio) && (!mp->video || !video) && (!mp->data || !data)) {
 				session->mountpoint = NULL;
 				janus_mutex_unlock(&mp->mutex);
 				janus_refcount_decrease(&mp->ref);
@@ -4087,7 +4766,19 @@ static void *janus_streaming_handler(void *data) {
 				g_snprintf(error_cause, 512, "Can't offer an SDP with no audio, video or data for this mountpoint");
 				goto error;
 			}
+			if(mp->streaming_source == janus_streaming_source_file) {
+				/* Create a session stream */
+				janus_streaming_session_stream *s = g_malloc0(sizeof(janus_streaming_session_stream));
+				s->mindex = -1;
+				s->send = TRUE;
+				janus_rtp_switching_context_reset(&s->context);
+				session->streams = g_list_append(session->streams, s);
+				if(session->streams_byid == NULL)
+					session->streams_byid = g_hash_table_new(NULL, NULL);
+				g_hash_table_insert(session->streams_byid, GINT_TO_POINTER(s->mindex), s);
+			}
 			if(mp->streaming_type == janus_streaming_type_on_demand) {
+				/* Spawn a thread */
 				GError *error = NULL;
 				char tname[16];
 				g_snprintf(tname, sizeof(tname), "mp %"SCNu64, id_value);
@@ -4103,67 +4794,112 @@ static void *janus_streaming_handler(void *data) {
 					JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the on-demand thread...\n", error->code, error->message ? error->message : "??");
 					error_code = JANUS_STREAMING_ERROR_UNKNOWN_ERROR;
 					g_snprintf(error_cause, 512, "Got error %d (%s) trying to launch the on-demand thread", error->code, error->message ? error->message : "??");
+					g_error_free(error);
 					goto error;
 				}
 			} else if(mp->streaming_source == janus_streaming_source_rtp) {
+				/* Create a session stream for each source stream we're subscribing to */
+				janus_streaming_session_stream *s = NULL;
 				janus_streaming_rtp_source *source = (janus_streaming_rtp_source *)mp->source;
-				if(source && source->simulcast) {
-					JANUS_VALIDATE_JSON_OBJECT(root, simulcast_parameters,
-						error_code, error_cause, TRUE,
-						JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
-					if(error_code != 0) {
-						session->mountpoint = NULL;
-						janus_mutex_unlock(&mp->mutex);
-						janus_refcount_decrease(&mp->ref);
-						goto error;
-					}
-					/* In case this mountpoint is simulcasting, let's aim high by default */
-					janus_rtp_switching_context_reset(&session->context);
-					janus_rtp_simulcasting_context_reset(&session->sim_context);
-					session->sim_context.substream_target = 2;
-					session->sim_context.templayer_target = 2;
-					janus_vp8_simulcast_context_reset(&session->vp8_context);
-					/* Unless the request contains a target for either layer */
-					json_t *substream = json_object_get(root, "substream");
-					if(substream) {
-						session->sim_context.substream_target = json_integer_value(substream);
-						JANUS_LOG(LOG_VERB, "Setting video substream to let through (simulcast): %d (was %d)\n",
-							session->sim_context.substream_target, session->sim_context.substream);
-					}
-					json_t *temporal = json_object_get(root, "temporal");
-					if(temporal) {
-						session->sim_context.templayer_target = json_integer_value(temporal);
-						JANUS_LOG(LOG_VERB, "Setting video temporal layer to let through (simulcast): %d (was %d)\n",
-							session->sim_context.templayer_target, session->sim_context.templayer);
-					}
-				} else if(source && source->svc) {
-					JANUS_VALIDATE_JSON_OBJECT(root, svc_parameters,
-						error_code, error_cause, TRUE,
-						JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
-					if(error_code != 0) {
-						session->mountpoint = NULL;
-						janus_mutex_unlock(&mp->mutex);
-						janus_refcount_decrease(&mp->ref);
-						goto error;
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					/* If we're using the legacy API, make sure we honor the deprecated attributes */
+					if(legacy && ((stream->type == JANUS_STREAMING_MEDIA_AUDIO && !audio) ||
+							(stream->type == JANUS_STREAMING_MEDIA_VIDEO && !video) ||
+							(stream->type == JANUS_STREAMING_MEDIA_DATA && !data))) {
+						/* FIXME Don't subscribe to this stream */
+						temp = temp->next;
+						continue;
 					}
-					/* In case this mountpoint is doing VP9-SVC, let's aim high by default */
-					session->spatial_layer = -1;
-					session->target_spatial_layer = 1;	/* FIXME Chrome sends 0 and 1 */
-					session->temporal_layer = -1;
-					session->target_temporal_layer = 2;	/* FIXME Chrome sends 0, 1 and 2 */
-					/* Unless the request contains a target for either layer */
-					json_t *spatial = json_object_get(root, "spatial_layer");
-					if(spatial) {
-						session->target_spatial_layer = json_integer_value(spatial);
-						JANUS_LOG(LOG_VERB, "Setting video spatial layer to let through (SVC): %d (was %d)\n",
-							session->target_spatial_layer, session->spatial_layer);
+					/* Check if this stream is among the mids the user subscribed to */
+					if(mids != NULL && json_array_size(mids) > 0) {
+						gboolean found = FALSE;
+						size_t i = 0;
+						for(i=0; i<json_array_size(mids); i++) {
+							json_t *s = json_array_get(mids, i);
+							const char *mid = json_string_value(s);
+							if(mid && stream->mid && !strcasecmp(stream->mid, mid)) {
+								found = TRUE;
+								break;
+							}
+						}
+						if(!found) {
+							/* Not in the mids list, don't subscribe to this stream */
+							temp = temp->next;
+							continue;
+						}
 					}
-					json_t *temporal = json_object_get(root, "temporal_layer");
-					if(temporal) {
-						session->target_temporal_layer = json_integer_value(temporal);
-						JANUS_LOG(LOG_VERB, "Setting video temporal layer to let through (SVC): %d (was %d)\n",
-							session->target_temporal_layer, session->temporal_layer);
+					/* Create a new session stream and add a reference to the source stream */
+					s = g_malloc0(sizeof(janus_streaming_session_stream));
+					s->send = TRUE;
+					janus_rtp_switching_context_reset(&s->context);
+					if(stream && stream->simulcast) {
+						/* FIXME What if we're simulcasting on two different video streams? */
+						JANUS_VALIDATE_JSON_OBJECT(root, simulcast_parameters,
+							error_code, error_cause, TRUE,
+							JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
+						if(error_code != 0) {
+							session->mountpoint = NULL;
+							janus_mutex_unlock(&mp->mutex);
+							janus_refcount_decrease(&mp->ref);
+							goto error;
+						}
+						/* In case this mountpoint is simulcasting, let's aim high by default */
+						s->sim_context.substream_target = 2;
+						s->sim_context.templayer_target = 2;
+						janus_vp8_simulcast_context_reset(&s->vp8_context);
+						/* Unless the request contains a target for either layer */
+						json_t *substream = json_object_get(root, "substream");
+						if(substream) {
+							s->sim_context.substream_target = json_integer_value(substream);
+							JANUS_LOG(LOG_VERB, "Setting video substream to let through (simulcast): %d (was %d)\n",
+								s->sim_context.substream_target, s->sim_context.substream);
+						}
+						json_t *temporal = json_object_get(root, "temporal");
+						if(temporal) {
+							s->sim_context.templayer_target = json_integer_value(temporal);
+							JANUS_LOG(LOG_VERB, "Setting video temporal layer to let through (simulcast): %d (was %d)\n",
+								s->sim_context.templayer_target, s->sim_context.templayer);
+						}
+					} else if(stream && stream->svc) {
+						/* FIXME What if we're doing SVC on two different video streams? */
+						JANUS_VALIDATE_JSON_OBJECT(root, svc_parameters,
+							error_code, error_cause, TRUE,
+							JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
+						if(error_code != 0) {
+							session->mountpoint = NULL;
+							janus_mutex_unlock(&mp->mutex);
+							janus_refcount_decrease(&mp->ref);
+							goto error;
+						}
+						/* In case this mountpoint is doing VP9-SVC, let's aim high by default */
+						s->spatial_layer = -1;
+						s->target_spatial_layer = 1;	/* FIXME Chrome sends 0 and 1 */
+						s->temporal_layer = -1;
+						s->target_temporal_layer = 2;	/* FIXME Chrome sends 0, 1 and 2 */
+						/* Unless the request contains a target for either layer */
+						json_t *spatial = json_object_get(root, "spatial_layer");
+						if(spatial) {
+							s->target_spatial_layer = json_integer_value(spatial);
+							JANUS_LOG(LOG_VERB, "Setting video spatial layer to let through (SVC): %d (was %d)\n",
+								s->target_spatial_layer, s->spatial_layer);
+						}
+						json_t *temporal = json_object_get(root, "temporal_layer");
+						if(temporal) {
+							s->target_temporal_layer = json_integer_value(temporal);
+							JANUS_LOG(LOG_VERB, "Setting video temporal layer to let through (SVC): %d (was %d)\n",
+								s->target_temporal_layer, s->temporal_layer);
+						}
 					}
+					s->mindex = g_list_length(session->streams);
+					s->stream = stream;
+					janus_refcount_increase(&stream->ref);
+					session->streams = g_list_append(session->streams, s);
+					if(session->streams_byid == NULL)
+						session->streams_byid = g_hash_table_new(NULL, NULL);
+					g_hash_table_insert(session->streams_byid, GINT_TO_POINTER(stream->mindex), s);
+					temp = temp->next;
 				}
 			}
 			janus_refcount_increase(&session->ref);
@@ -4182,70 +4918,104 @@ static void *janus_streaming_handler(void *data) {
 				"s=Mountpoint %"SCNu64"\r\n", mp->id);
 			g_strlcat(sdptemp, buffer, 2048);
 			g_strlcat(sdptemp, "t=0 0\r\n", 2048);
-			if(mp->codecs.audio_pt >= 0 && session->audio) {
-				/* Add audio line */
-				g_snprintf(buffer, 512,
-					"m=audio 1 RTP/SAVPF %d\r\n"
-					"c=IN IP4 1.1.1.1\r\n",
-					mp->codecs.audio_pt);
-				g_strlcat(sdptemp, buffer, 2048);
-				if(mp->codecs.audio_rtpmap) {
-					g_snprintf(buffer, 512,
-						"a=rtpmap:%d %s\r\n",
-						mp->codecs.audio_pt, mp->codecs.audio_rtpmap);
-					g_strlcat(sdptemp, buffer, 2048);
-				}
-				if(mp->codecs.audio_fmtp) {
-					g_snprintf(buffer, 512,
-						"a=fmtp:%d %s\r\n",
-						mp->codecs.audio_pt, mp->codecs.audio_fmtp);
-					g_strlcat(sdptemp, buffer, 2048);
-				}
-				g_strlcat(sdptemp, "a=sendonly\r\n", 2048);
-			}
-			if(mp->codecs.video_pt > 0 && session->video) {
-				/* Add video line */
+			if(mp->streaming_source == janus_streaming_source_file) {
+				janus_streaming_file_source *source = mp->source;
+				/* Add audio line */
 				g_snprintf(buffer, 512,
-					"m=video 1 RTP/SAVPF %d\r\n"
+					"m=audio 1 RTP/SAVPF %d\r\n"
 					"c=IN IP4 1.1.1.1\r\n",
-					mp->codecs.video_pt);
+					source->codecs.pt);
 				g_strlcat(sdptemp, buffer, 2048);
-				if(mp->codecs.video_rtpmap) {
+				if(source->codecs.rtpmap) {
 					g_snprintf(buffer, 512,
 						"a=rtpmap:%d %s\r\n",
-						mp->codecs.video_pt, mp->codecs.video_rtpmap);
+						source->codecs.pt, source->codecs.rtpmap);
 					g_strlcat(sdptemp, buffer, 2048);
 				}
-				if(mp->codecs.video_fmtp) {
+				if(source->codecs.fmtp) {
 					g_snprintf(buffer, 512,
 						"a=fmtp:%d %s\r\n",
-						mp->codecs.video_pt, mp->codecs.video_fmtp);
+						source->codecs.pt, source->codecs.fmtp);
 					g_strlcat(sdptemp, buffer, 2048);
 				}
-				g_snprintf(buffer, 512,
-					"a=rtcp-fb:%d nack\r\n",
-					mp->codecs.video_pt);
-				g_strlcat(sdptemp, buffer, 2048);
-				g_snprintf(buffer, 512,
-					"a=rtcp-fb:%d nack pli\r\n",
-					mp->codecs.video_pt);
-				g_strlcat(sdptemp, buffer, 2048);
-				g_snprintf(buffer, 512,
-					"a=rtcp-fb:%d goog-remb\r\n",
-					mp->codecs.video_pt);
-				g_strlcat(sdptemp, buffer, 2048);
 				g_strlcat(sdptemp, "a=sendonly\r\n", 2048);
-			}
+			} else {
+				/* Iterate on all media streams */
+				GList *temp = session->streams;
+				while(temp) {
+					janus_streaming_session_stream *s = (janus_streaming_session_stream *)temp->data;
+					janus_streaming_rtp_source_stream *stream = s->stream;
+					if(stream->type == JANUS_STREAMING_MEDIA_AUDIO && audio) {
+						/* Add audio line */
+						g_snprintf(buffer, 512,
+							"m=audio 1 RTP/SAVPF %d\r\n"
+							"c=IN IP4 1.1.1.1\r\n",
+							stream->codecs.pt);
+						g_strlcat(sdptemp, buffer, 2048);
+						g_snprintf(buffer, 512, "a=mid:%s\r\n", stream->mid);
+						g_strlcat(sdptemp, buffer, 2048);
+						if(stream->codecs.rtpmap) {
+							g_snprintf(buffer, 512,
+								"a=rtpmap:%d %s\r\n",
+								stream->codecs.pt, stream->codecs.rtpmap);
+							g_strlcat(sdptemp, buffer, 2048);
+						}
+						if(stream->codecs.fmtp) {
+							g_snprintf(buffer, 512,
+								"a=fmtp:%d %s\r\n",
+								stream->codecs.pt, stream->codecs.fmtp);
+							g_strlcat(sdptemp, buffer, 2048);
+						}
+						g_strlcat(sdptemp, "a=sendonly\r\n", 2048);
+					} else if(stream->type == JANUS_STREAMING_MEDIA_VIDEO && video) {
+						/* Add video line */
+						g_snprintf(buffer, 512,
+							"m=video 1 RTP/SAVPF %d\r\n"
+							"c=IN IP4 1.1.1.1\r\n",
+							stream->codecs.pt);
+						g_strlcat(sdptemp, buffer, 2048);
+						g_snprintf(buffer, 512, "a=mid:%s\r\n", stream->mid);
+						g_strlcat(sdptemp, buffer, 2048);
+						if(stream->codecs.rtpmap) {
+							g_snprintf(buffer, 512,
+								"a=rtpmap:%d %s\r\n",
+								stream->codecs.pt, stream->codecs.rtpmap);
+							g_strlcat(sdptemp, buffer, 2048);
+						}
+						if(stream->codecs.fmtp) {
+							g_snprintf(buffer, 512,
+								"a=fmtp:%d %s\r\n",
+								stream->codecs.pt, stream->codecs.fmtp);
+							g_strlcat(sdptemp, buffer, 2048);
+						}
+						g_snprintf(buffer, 512,
+							"a=rtcp-fb:%d nack\r\n",
+							stream->codecs.pt);
+						g_strlcat(sdptemp, buffer, 2048);
+						g_snprintf(buffer, 512,
+							"a=rtcp-fb:%d nack pli\r\n",
+							stream->codecs.pt);
+						g_strlcat(sdptemp, buffer, 2048);
+						g_snprintf(buffer, 512,
+							"a=rtcp-fb:%d goog-remb\r\n",
+							stream->codecs.pt);
+						g_strlcat(sdptemp, buffer, 2048);
+						g_strlcat(sdptemp, "a=sendonly\r\n", 2048);
+					}
 #ifdef HAVE_SCTP
-			if(mp->data && session->data) {
-				/* Add data line */
-				g_snprintf(buffer, 512,
-					"m=application 1 DTLS/SCTP 5000\r\n"
-					"c=IN IP4 1.1.1.1\r\n"
-					"a=sctpmap:5000 webrtc-datachannel 16\r\n");
-				g_strlcat(sdptemp, buffer, 2048);
-			}
+					else if(stream->type == JANUS_STREAMING_MEDIA_DATA && data) {
+						/* Add data line */
+						g_snprintf(buffer, 512,
+							"m=application 1 DTLS/SCTP 5000\r\n"
+							"c=IN IP4 1.1.1.1\r\n"
+							"a=mid:%s\r\n"
+							"a=sctpmap:5000 webrtc-datachannel 16\r\n", stream->mid);
+						g_strlcat(sdptemp, buffer, 2048);
+					}
 #endif
+					temp = temp->next;
+				}
+			}
 			sdp = g_strdup(sdptemp);
 			JANUS_LOG(LOG_VERB, "Going to %s this SDP:\n%s\n", sdp_type, sdp);
 			result = json_object();
@@ -4325,97 +5095,116 @@ static void *janus_streaming_handler(void *data) {
 			JANUS_VALIDATE_JSON_OBJECT(root, configure_parameters,
 				error_code, error_cause, TRUE,
 				JANUS_STREAMING_ERROR_MISSING_ELEMENT, JANUS_STREAMING_ERROR_INVALID_ELEMENT);
+			/* Audio, video and data are deprecated properties */
 			json_t *audio = json_object_get(root, "audio");
-			if(audio)
-				session->audio = json_is_true(audio);
 			json_t *video = json_object_get(root, "video");
-			if(video)
-				session->video = json_is_true(video);
 			json_t *data = json_object_get(root, "data");
-			if(data)
-				session->data = json_is_true(data);
+			/* Better to specify the 'send' property of a specific 'mid' */
+			const char *mid = json_string_value(json_object_get(root, "mid"));
+			json_t *send = json_object_get(root, "send");
 			if(mp->streaming_source == janus_streaming_source_rtp) {
-				janus_streaming_rtp_source *source = (janus_streaming_rtp_source *)mp->source;
-				if(source && source->simulcast) {
-					/* Check if the viewer is requesting a different substream/temporal layer */
-					json_t *substream = json_object_get(root, "substream");
-					if(substream) {
-						session->sim_context.substream_target = json_integer_value(substream);
-						JANUS_LOG(LOG_VERB, "Setting video substream to let through (simulcast): %d (was %d)\n",
-							session->sim_context.substream_target, session->sim_context.substream);
-						if(session->sim_context.substream_target == session->sim_context.substream) {
-							/* No need to do anything, we're already getting the right substream, so notify the viewer */
-							json_t *event = json_object();
-							json_object_set_new(event, "streaming", json_string("event"));
-							json_t *result = json_object();
-							json_object_set_new(result, "substream", json_integer(session->sim_context.substream));
-							json_object_set_new(event, "result", result);
-							gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
-							json_decref(event);
-						} else {
-							/* Schedule a PLI */
-							JANUS_LOG(LOG_VERB, "We need a PLI for the simulcast context\n");
-							g_atomic_int_set(&source->need_pli, 1);
-						}
-					}
-					json_t *temporal = json_object_get(root, "temporal");
-					if(temporal) {
-						session->sim_context.templayer_target = json_integer_value(temporal);
-						JANUS_LOG(LOG_VERB, "Setting video temporal layer to let through (simulcast): %d (was %d)\n",
-							session->sim_context.templayer_target, session->sim_context.templayer);
-						if(mp->codecs.video_codec == JANUS_VIDEOCODEC_VP8 && session->sim_context.templayer_target == session->sim_context.templayer) {
-							/* No need to do anything, we're already getting the right temporal layer, so notify the viewer */
-							json_t *event = json_object();
-							json_object_set_new(event, "streaming", json_string("event"));
-							json_t *result = json_object();
-							json_object_set_new(result, "temporal", json_integer(session->sim_context.templayer));
-							json_object_set_new(event, "result", result);
-							gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
-							json_decref(event);
-						}
+				GList *temp = session->streams;
+				while(temp) {
+					janus_streaming_session_stream *s = (janus_streaming_session_stream *)temp->data;
+					janus_streaming_rtp_source_stream *stream = s->stream;
+					/* Check the old and deprecated approach first */
+					if(audio && stream->type == JANUS_STREAMING_MEDIA_AUDIO)
+						s->send = json_is_true(audio);
+					else if(video && stream->type == JANUS_STREAMING_MEDIA_VIDEO)
+						s->send = json_is_true(video);
+					else if(data && stream->type == JANUS_STREAMING_MEDIA_DATA)
+						s->send = json_is_true(data);
+					/* Now let's see if this is the right mid */
+					if(mid && strcasecmp(stream->mid, mid)) {
+						temp = temp->next;
+						continue;
 					}
-				}
-				if(source && source->svc) {
-					/* Check if the viewer is requesting a different SVC spatial/temporal layer */
-					json_t *spatial = json_object_get(root, "spatial_layer");
-					if(spatial) {
-						int spatial_layer = json_integer_value(spatial);
-						if(spatial_layer > 1) {
-							JANUS_LOG(LOG_WARN, "Spatial layer higher than 1, will probably be ignored\n");
+					if(send)
+						s->send = json_is_true(send);
+					/* FIXME What if we're simulcasting or doing SVC on two different video streams? */
+					if(stream->simulcast) {
+						/* Check if the viewer is requesting a different substream/temporal layer */
+						json_t *substream = json_object_get(root, "substream");
+						if(substream) {
+							s->sim_context.substream_target = json_integer_value(substream);
+							JANUS_LOG(LOG_VERB, "Setting video substream to let through (simulcast): %d (was %d)\n",
+								s->sim_context.substream_target, s->sim_context.substream);
+							if(s->sim_context.substream_target == s->sim_context.substream) {
+								/* No need to do anything, we're already getting the right substream, so notify the viewer */
+								json_t *event = json_object();
+								json_object_set_new(event, "streaming", json_string("event"));
+								json_t *result = json_object();
+								json_object_set_new(result, "mid", json_string(stream->mid));
+								json_object_set_new(result, "substream", json_integer(s->sim_context.substream));
+								json_object_set_new(event, "result", result);
+								gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
+								json_decref(event);
+							}
 						}
-						if(spatial_layer == session->spatial_layer) {
-							/* No need to do anything, we're already getting the right spatial layer, so notify the user */
-							json_t *event = json_object();
-							json_object_set_new(event, "streaming", json_string("event"));
-							json_t *result = json_object();
-							json_object_set_new(result, "spatial_layer", json_integer(session->spatial_layer));
-							json_object_set_new(event, "result", result);
-							gateway->push_event(msg->handle, &janus_streaming_plugin, NULL, event, NULL);
-							json_decref(event);
-						} else if(spatial_layer != session->target_spatial_layer) {
-							/* Send a FIR to the source, if RTCP is enabled */
-							g_atomic_int_set(&source->need_pli, 1);
+						json_t *temporal = json_object_get(root, "temporal");
+						if(temporal) {
+							s->sim_context.templayer_target = json_integer_value(temporal);
+							JANUS_LOG(LOG_VERB, "Setting video temporal layer to let through (simulcast): %d (was %d)\n",
+								s->sim_context.templayer_target, s->sim_context.templayer);
+							if(stream->codecs.video_codec == JANUS_VIDEOCODEC_VP8 &&
+									s->sim_context.templayer_target == s->sim_context.templayer) {
+								/* No need to do anything, we're already getting the right temporal layer, so notify the viewer */
+								json_t *event = json_object();
+								json_object_set_new(event, "streaming", json_string("event"));
+								json_t *result = json_object();
+								json_object_set_new(result, "mid", json_string(stream->mid));
+								json_object_set_new(result, "temporal", json_integer(s->sim_context.templayer));
+								json_object_set_new(event, "result", result);
+								gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
+								json_decref(event);
+							}
 						}
-						session->target_spatial_layer = spatial_layer;
 					}
-					json_t *temporal = json_object_get(root, "temporal_layer");
-					if(temporal) {
-						int temporal_layer = json_integer_value(temporal);
-						if(temporal_layer > 2) {
-							JANUS_LOG(LOG_WARN, "Temporal layer higher than 2, will probably be ignored\n");
+					if(stream->svc) {
+						/* Check if the viewer is requesting a different SVC spatial/temporal layer */
+						json_t *spatial = json_object_get(root, "spatial_layer");
+						if(spatial) {
+							int spatial_layer = json_integer_value(spatial);
+							if(spatial_layer > 1) {
+								JANUS_LOG(LOG_WARN, "Spatial layer higher than 1, will probably be ignored\n");
+							}
+							if(spatial_layer == s->spatial_layer) {
+								/* No need to do anything, we're already getting the right spatial layer, so notify the user */
+								json_t *event = json_object();
+								json_object_set_new(event, "streaming", json_string("event"));
+								json_t *result = json_object();
+								json_object_set_new(result, "mid", json_string(stream->mid));
+								json_object_set_new(result, "spatial_layer", json_integer(s->spatial_layer));
+								json_object_set_new(event, "result", result);
+								gateway->push_event(msg->handle, &janus_streaming_plugin, NULL, event, NULL);
+								json_decref(event);
+							} else if(spatial_layer != s->target_spatial_layer) {
+								/* Send a FIR to the source, if RTCP is enabled */
+								g_atomic_int_set(&stream->need_pli, 1);
+							}
+							s->target_spatial_layer = spatial_layer;
 						}
-						if(temporal_layer == session->temporal_layer) {
-							/* No need to do anything, we're already getting the right temporal layer, so notify the user */
-							json_t *event = json_object();
-							json_object_set_new(event, "streaming", json_string("event"));
-							json_t *result = json_object();
-							json_object_set_new(result, "temporal_layer", json_integer(session->temporal_layer));
-							json_object_set_new(event, "result", result);
-							gateway->push_event(msg->handle, &janus_streaming_plugin, NULL, event, NULL);
-							json_decref(event);
+						json_t *temporal = json_object_get(root, "temporal_layer");
+						if(temporal) {
+							int temporal_layer = json_integer_value(temporal);
+							if(temporal_layer > 2) {
+								JANUS_LOG(LOG_WARN, "Temporal layer higher than 2, will probably be ignored\n");
+							}
+							if(temporal_layer == s->temporal_layer) {
+								/* No need to do anything, we're already getting the right temporal layer, so notify the user */
+								json_t *event = json_object();
+								json_object_set_new(event, "streaming", json_string("event"));
+								json_t *result = json_object();
+								json_object_set_new(result, "mid", json_string(stream->mid));
+								json_object_set_new(result, "temporal_layer", json_integer(s->temporal_layer));
+								json_object_set_new(event, "result", result);
+								gateway->push_event(msg->handle, &janus_streaming_plugin, NULL, event, NULL);
+								json_decref(event);
+							}
+							s->target_temporal_layer = temporal_layer;
 						}
-						session->target_temporal_layer = temporal_layer;
 					}
+					temp = temp->next;
 				}
 			}
 			/* Done */
@@ -4425,7 +5214,8 @@ static void *janus_streaming_handler(void *data) {
 			/* This listener wants to switch to a different mountpoint
 			 * NOTE: this only works for live RTP streams as of now: you
 			 * cannot, for instance, switch from a live RTP mountpoint to
-			 * an on demand one or viceversa (TBD.) */
+			 * an on demand one or viceversa (TBD.). Besides, it needs
+			 * to be mountpoints with the same media in the same order. */
 			janus_streaming_mountpoint *oldmp = session->mountpoint;
 			if(oldmp == NULL) {
 				JANUS_LOG(LOG_VERB, "Can't switch: not on a mountpoint\n");
@@ -4470,6 +5260,7 @@ static void *janus_streaming_handler(void *data) {
 				g_snprintf(error_cause, 512, "Can't switch: target is not a live RTP mountpoint");
 				goto error;
 			}
+			/* TODO: compare the streams of the two mountpoints */
 			janus_mutex_unlock(&mountpoints_mutex);
 			JANUS_LOG(LOG_VERB, "Request to switch to mountpoint/stream %"SCNu64" (old: %"SCNu64")\n", id_value, oldmp->id);
 			session->paused = TRUE;
@@ -4602,7 +5393,7 @@ static void *janus_streaming_handler(void *data) {
 
 /* Helpers to create a listener filedescriptor */
 static int janus_streaming_create_fd(int port, in_addr_t mcast, const janus_network_address *iface,
-		const char *listenername, const char *medianame, const char *mountpointname, gboolean quiet) {
+		const char *medianame, const char *mountpointname, gboolean quiet) {
 	janus_mutex_lock(&fd_mutex);
 	struct sockaddr_in address;
 	struct sockaddr_in6 address6;
@@ -4632,7 +5423,7 @@ static int janus_streaming_create_fd(int port, in_addr_t mcast, const janus_netw
 #ifdef IP_MULTICAST_ALL
 				int mc_all = 0;
 				if((setsockopt(fd, IPPROTO_IP, IP_MULTICAST_ALL, (void*) &mc_all, sizeof(mc_all))) < 0) {
-					JANUS_LOG(LOG_ERR, "[%s] %s listener setsockopt IP_MULTICAST_ALL failed\n", mountpointname, listenername);
+					JANUS_LOG(LOG_ERR, "[%s] %s listener setsockopt IP_MULTICAST_ALL failed\n", mountpointname, medianame);
 					close(fd);
 					janus_mutex_unlock(&fd_mutex);
 					return -1;
@@ -4646,23 +5437,23 @@ static int janus_streaming_create_fd(int port, in_addr_t mcast, const janus_netw
 					if(iface->family == AF_INET) {
 						mreq.imr_interface = iface->ipv4;
 						(void) janus_network_address_to_string_buffer(iface, &address_representation); /* This is OK: if we get here iface must be non-NULL */
-						JANUS_LOG(LOG_INFO, "[%s] %s listener using interface address: %s\n", mountpointname, listenername, janus_network_address_string_from_buffer(&address_representation));
+						JANUS_LOG(LOG_INFO, "[%s] %s listener using interface address: %s\n", mountpointname, medianame, janus_network_address_string_from_buffer(&address_representation));
 					} else {
-						JANUS_LOG(LOG_ERR, "[%s] %s listener: invalid multicast address type (only IPv4 multicast is currently supported by this plugin)\n", mountpointname, listenername);
+						JANUS_LOG(LOG_ERR, "[%s] %s listener: invalid multicast address type (only IPv4 multicast is currently supported by this plugin)\n", mountpointname, medianame);
 						close(fd);
 						janus_mutex_unlock(&fd_mutex);
 						return -1;
 					}
 				} else {
-					JANUS_LOG(LOG_WARN, "[%s] No multicast interface for: %s. This may not work as expected if you have multiple network devices (NICs)\n", mountpointname, listenername);
+					JANUS_LOG(LOG_WARN, "[%s] No multicast interface for: %s. This may not work as expected if you have multiple network devices (NICs)\n", mountpointname, medianame);
 				}
 				if(setsockopt(fd, IPPROTO_IP, IP_ADD_MEMBERSHIP, &mreq, sizeof(mreq)) == -1) {
-					JANUS_LOG(LOG_ERR, "[%s] %s listener IP_ADD_MEMBERSHIP failed\n", mountpointname, listenername);
+					JANUS_LOG(LOG_ERR, "[%s] %s listener IP_ADD_MEMBERSHIP failed\n", mountpointname, medianame);
 					close(fd);
 					janus_mutex_unlock(&fd_mutex);
 					return -1;
 				}
-				JANUS_LOG(LOG_INFO, "[%s] %s listener IP_ADD_MEMBERSHIP ok\n", mountpointname, listenername);
+				JANUS_LOG(LOG_INFO, "[%s] %s listener IP_ADD_MEMBERSHIP ok\n", mountpointname, medianame);
 			}
 		} else {
 			/* Pick a port in the configured range */
@@ -4684,7 +5475,7 @@ static int janus_streaming_create_fd(int port, in_addr_t mcast, const janus_netw
 		if(!use_range && IN_MULTICAST(ntohl(mcast))) {
 			int reuse = 1;
 			if(setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, &reuse, sizeof(reuse)) == -1) {
-				JANUS_LOG(LOG_ERR, "[%s] %s listener setsockopt SO_REUSEADDR failed\n", mountpointname, listenername);
+				JANUS_LOG(LOG_ERR, "[%s] %s listener setsockopt SO_REUSEADDR failed\n", mountpointname, medianame);
 				close(fd);
 				janus_mutex_unlock(&fd_mutex);
 				return -1;
@@ -4698,14 +5489,14 @@ static int janus_streaming_create_fd(int port, in_addr_t mcast, const janus_netw
 					address.sin_addr = iface->ipv4;
 					(void) janus_network_address_to_string_buffer(iface, &address_representation); /* This is OK: if we get here iface must be non-NULL */
 					JANUS_LOG(LOG_INFO, "[%s] %s listener restricted to interface address: %s\n",
-						mountpointname, listenername, janus_network_address_string_from_buffer(&address_representation));
+						mountpointname, medianame, janus_network_address_string_from_buffer(&address_representation));
 				} else if(iface->family == AF_INET6) {
 					memcpy(&address6.sin6_addr, &iface->ipv6, sizeof(iface->ipv6));
 					(void) janus_network_address_to_string_buffer(iface, &address_representation); /* This is OK: if we get here iface must be non-NULL */
 					JANUS_LOG(LOG_INFO, "[%s] %s listener restricted to interface address: %s\n",
-						mountpointname, listenername, janus_network_address_string_from_buffer(&address_representation));
+						mountpointname, medianame, janus_network_address_string_from_buffer(&address_representation));
 				} else {
-					JANUS_LOG(LOG_ERR, "[%s] %s listener: invalid address/restriction type\n", mountpointname, listenername);
+					JANUS_LOG(LOG_ERR, "[%s] %s listener: invalid address/restriction type\n", mountpointname, medianame);
 					continue;
 				}
 			}
@@ -4765,9 +5556,9 @@ static int janus_streaming_allocate_port_pair(const char *name, const char *medi
 			rtp_port_next = rtp_range_min;
 			rtp_port_wrap = TRUE;
 		}
-		rtp_fd = janus_streaming_create_fd(rtp_port, mcast, iface, media, media, name, TRUE);
+		rtp_fd = janus_streaming_create_fd(rtp_port, mcast, iface, media, name, TRUE);
 		if(rtp_fd != -1) {
-			rtcp_fd = janus_streaming_create_fd(rtcp_port, mcast, iface, media, media, name, TRUE);
+			rtcp_fd = janus_streaming_create_fd(rtcp_port, mcast, iface, media, name, TRUE);
 			if(rtcp_fd != -1) {
 				/* Done */
 				fds->fd = rtp_fd;
@@ -4797,45 +5588,48 @@ static int janus_streaming_get_fd_port(int fd) {
 	return ntohs(server.sin6_port);
 }
 
-/* Helpers to destroy a streaming mountpoint. */
-static void janus_streaming_rtp_source_free(janus_streaming_rtp_source *source) {
-	if(source->audio_fd > -1) {
-		close(source->audio_fd);
+/* Helpers to destroy a streaming mountpoint source */
+static void janus_streaming_rtp_source_stream_free(const janus_refcount *st_ref) {
+	janus_streaming_rtp_source_stream *stream = janus_refcount_containerof(st_ref, janus_streaming_rtp_source_stream, ref);
+	/* This stream can be destroyed, free all the resources */
+	if(stream->fd[0] > -1) {
+		close(stream->fd[0]);
 	}
-	if(source->video_fd[0] > -1) {
-		close(source->video_fd[0]);
+	if(stream->fd[1] > -1) {
+		close(stream->fd[1]);
 	}
-	if(source->video_fd[1] > -1) {
-		close(source->video_fd[1]);
+	if(stream->fd[2] > -1) {
+		close(stream->fd[2]);
 	}
-	if(source->video_fd[2] > -1) {
-		close(source->video_fd[2]);
-	}
-	if(source->data_fd > -1) {
-		close(source->data_fd);
-	}
-	if(source->audio_rtcp_fd > -1) {
-		close(source->audio_rtcp_fd);
-	}
-	if(source->video_rtcp_fd > -1) {
-		close(source->video_rtcp_fd);
+	if(stream->rtcp_fd > -1) {
+		close(stream->rtcp_fd);
 	}
+	janus_mutex_lock(&stream->keyframe.mutex);
+	if(stream->keyframe.latest_keyframe != NULL)
+		g_list_free_full(stream->keyframe.latest_keyframe, (GDestroyNotify)janus_streaming_rtp_relay_packet_free);
+	stream->keyframe.latest_keyframe = NULL;
+	janus_mutex_unlock(&stream->keyframe.mutex);
+	janus_mutex_lock(&stream->buffermsg_mutex);
+	if(stream->last_msg != NULL)
+		janus_streaming_rtp_relay_packet_free((janus_streaming_rtp_relay_packet *)stream->last_msg);
+	stream->last_msg = NULL;
+	janus_mutex_unlock(&stream->buffermsg_mutex);
+	g_free(stream->codecs.rtpmap);
+	g_free(stream->codecs.fmtp);
+	g_free(stream->mid);
+	g_free(stream->label);
+	g_free(stream->mcast_str);
+	g_free(stream->iface_str);
+	g_free(stream);
+}
+
+static void janus_streaming_rtp_source_free(janus_streaming_rtp_source *source) {
 	if(source->pipefd[0] > -1) {
 		close(source->pipefd[0]);
 	}
 	if(source->pipefd[1] > -1) {
 		close(source->pipefd[1]);
 	}
-	janus_mutex_lock(&source->keyframe.mutex);
-	if(source->keyframe.latest_keyframe != NULL)
-		g_list_free_full(source->keyframe.latest_keyframe, (GDestroyNotify)janus_streaming_rtp_relay_packet_free);
-	source->keyframe.latest_keyframe = NULL;
-	janus_mutex_unlock(&source->keyframe.mutex);
-	janus_mutex_lock(&source->buffermsg_mutex);
-	if(source->last_msg != NULL)
-		janus_streaming_rtp_relay_packet_free((janus_streaming_rtp_relay_packet *)source->last_msg);
-	source->last_msg = NULL;
-	janus_mutex_unlock(&source->buffermsg_mutex);
 	if(source->is_srtp) {
 		g_free(source->srtpcrypto);
 		srtp_dealloc(source->srtp_ctx);
@@ -4862,201 +5656,182 @@ static void janus_streaming_rtp_source_free(janus_streaming_rtp_source *source)
 	g_free(source->rtsp_password);
 	g_free(source->rtsp_ahost);
 	g_free(source->rtsp_vhost);
+	g_free(source->rtsp_acodecs.rtpmap);
+	g_free(source->rtsp_vcodecs.fmtp);
 	janus_mutex_unlock(&source->rtsp_mutex);
 #endif
+	g_list_free_full(source->media, (GDestroyNotify)(janus_streaming_rtp_source_stream_unref));
+	g_hash_table_unref(source->media_byid);
+	g_hash_table_unref(source->media_byfd);
 	g_free(source);
 }
 
 static void janus_streaming_file_source_free(janus_streaming_file_source *source) {
+	g_free(source->codecs.rtpmap);
+	g_free(source->codecs.fmtp);
 	g_free(source->filename);
 	g_free(source);
 }
 
 /* Helper to create an RTP live source (e.g., from gstreamer/ffmpeg/vlc/etc.) */
-janus_streaming_mountpoint *janus_streaming_create_rtp_source(
-		uint64_t id, char *name, char *desc,
-		int srtpsuite, char *srtpcrypto, int threads,
-		gboolean doaudio, char *amcast, const janus_network_address *aiface, uint16_t aport, uint16_t artcpport, uint8_t acodec, char *artpmap, char *afmtp, gboolean doaskew,
-		gboolean dovideo, char *vmcast, const janus_network_address *viface, uint16_t vport, uint16_t vrtcpport, uint8_t vcodec, char *vrtpmap, char *vfmtp, gboolean bufferkf,
-			gboolean simulcast, uint16_t vport2, uint16_t vport3, gboolean svc, gboolean dovskew, int rtp_collision,
-		gboolean dodata, const janus_network_address *diface, uint16_t dport, gboolean buffermsg) {
-	char tempname[255];
-	if(name == NULL) {
-		JANUS_LOG(LOG_VERB, "Missing name, will generate a random one...\n");
-		memset(tempname, 0, 255);
-		g_snprintf(tempname, 255, "mp-%"SCNu64, id);
-	} else if(atoi(name) != 0) {
-		JANUS_LOG(LOG_VERB, "Names can't start with a number, prefixing it...\n");
-		memset(tempname, 0, 255);
-		g_snprintf(tempname, 255, "mp-%s", name);
-		name = NULL;
+janus_streaming_rtp_source_stream *janus_streaming_create_rtp_source_stream(
+		const char *name, int mindex, const char *type, const char *mid, const char *label,
+		char *mcast, char *miface, const janus_network_address *iface, uint16_t port, uint16_t port2, uint16_t port3, uint16_t rtcpport,
+		uint8_t codec, const char *rtpmap, const char *fmtp, gboolean doskew,
+		gboolean bufferkf, gboolean buffermsg, gboolean simulcast, gboolean svc) {
+	if(type == NULL || mid == NULL || label == NULL) {
+		JANUS_LOG(LOG_ERR, "[%s] Can't add 'rtp' stream, missing media type, mid or label...\n", name);
+		return NULL;
 	}
-	if(!doaudio && !dovideo && !dodata) {
-		JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream, no audio, video or data have to be streamed...\n");
-		janus_mutex_lock(&mountpoints_mutex);
-		g_hash_table_remove(mountpoints_temp, &id);
-		janus_mutex_unlock(&mountpoints_mutex);
+	janus_streaming_media mtype = JANUS_STREAMING_MEDIA_NONE;
+	if(!strcasecmp(type, "audio")) {
+		mtype = JANUS_STREAMING_MEDIA_AUDIO;
+	} else if(!strcasecmp(type, "video")) {
+		mtype = JANUS_STREAMING_MEDIA_VIDEO;
+	} else if(!strcasecmp(type, "data")) {
+		mtype = JANUS_STREAMING_MEDIA_DATA;
+	} else {
+		JANUS_LOG(LOG_ERR, "[%s] Can't add 'rtp' stream, invalid type '%s'...\n", name, type);
 		return NULL;
 	}
-	if(doaudio && (artpmap == NULL)) {
-		JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream, missing mandatory information for audio...\n");
-		janus_mutex_lock(&mountpoints_mutex);
-		g_hash_table_remove(mountpoints_temp, &id);
-		janus_mutex_unlock(&mountpoints_mutex);
+#ifndef HAVE_SCTP
+	if(mtype == JANUS_STREAMING_MEDIA_DATA) {
+		JANUS_LOG(LOG_WARN, "[%s] Mountpoint wants to do datachannel relaying, but datachannels support was not compiled...\n", name);
 		return NULL;
 	}
-	if(dovideo && (vcodec == 0 || vrtpmap == NULL)) {
-		JANUS_LOG(LOG_ERR, "Can't add 'rtp' stream, missing mandatory information for video...\n");
-		janus_mutex_lock(&mountpoints_mutex);
-		g_hash_table_remove(mountpoints_temp, &id);
-		janus_mutex_unlock(&mountpoints_mutex);
+#endif
+	if(mtype != JANUS_STREAMING_MEDIA_DATA && rtpmap == NULL) {
+		JANUS_LOG(LOG_ERR, "[%s] Can't add 'rtp' stream, missing mandatory information for audio/video...\n", name);
 		return NULL;
 	}
-	JANUS_LOG(LOG_VERB, "Audio %s, Video %s, Data %s\n",
-		doaudio ? "enabled" : "NOT enabled",
-		dovideo ? "enabled" : "NOT enabled",
-		dodata ? "enabled" : "NOT enabled");
-	/* First of all, let's check if the requested ports are free */
-	int audio_fd = -1;
-	int audio_rtcp_fd = -1;
-	if(doaudio) {
-		audio_fd = janus_streaming_create_fd(aport, amcast ? inet_addr(amcast) : INADDR_ANY, aiface,
-			"Audio", "audio", name ? name : tempname, aport == 0);
-		if(audio_fd < 0) {
-			JANUS_LOG(LOG_ERR, "Can't bind to port %d for audio...\n", aport);
-			janus_mutex_lock(&mountpoints_mutex);
-			g_hash_table_remove(mountpoints_temp, &id);
-			janus_mutex_unlock(&mountpoints_mutex);
+	/* Allocate file descriptors (if video, we may be simulcasting) */
+	int fd[3] = {-1, -1, -1};
+	int rtcp_fd = -1;
+	fd[0] = janus_streaming_create_fd(port, mcast ? inet_addr(mcast) : INADDR_ANY, iface, type, name, port == 0);
+	if(fd[0] < 0) {
+		JANUS_LOG(LOG_ERR, "[%s] Can't bind to port %d for video...\n", name, port);
+		return NULL;
+	}
+	port = janus_streaming_get_fd_port(fd[0]);
+	if(rtcpport > 0) {
+		rtcp_fd = janus_streaming_create_fd(rtcpport, mcast ? inet_addr(mcast) : INADDR_ANY, iface, type, name, FALSE);
+		if(rtcp_fd < 0) {
+			JANUS_LOG(LOG_ERR, "[%s] Can't bind to port %d for video RTCP...\n", name, rtcpport);
+			if(fd[0] > -1)
+				close(fd[0]);
 			return NULL;
 		}
-		aport = janus_streaming_get_fd_port(audio_fd);
-		if(artcpport > 0) {
-			audio_rtcp_fd = janus_streaming_create_fd(artcpport, amcast ? inet_addr(amcast) : INADDR_ANY, aiface,
-				"Audio", "audio", name ? name : tempname, FALSE);
-			if(audio_rtcp_fd < 0) {
-				JANUS_LOG(LOG_ERR, "Can't bind to port %d for audio RTCP...\n", aport+1);
-				if(audio_fd > -1)
-					close(audio_fd);
-				janus_mutex_lock(&mountpoints_mutex);
-				g_hash_table_remove(mountpoints_temp, &id);
-				janus_mutex_unlock(&mountpoints_mutex);
+	}
+	if(mtype == JANUS_STREAMING_MEDIA_VIDEO && simulcast) {
+		if(port2 > 0) {
+			fd[1] = janus_streaming_create_fd(port2, mcast ? inet_addr(mcast) : INADDR_ANY, iface, type, name, FALSE);
+			if(fd[1] < 0) {
+				JANUS_LOG(LOG_ERR, "[%s] Can't bind to port %d for video (2nd port)...\n", name, port2);
+				if(fd[0] > -1)
+					close(fd[0]);
+				if(rtcp_fd > -1)
+					close(rtcp_fd);
 				return NULL;
 			}
+			port2 = janus_streaming_get_fd_port(fd[1]);
 		}
-	}
-	int video_fd[3] = {-1, -1, -1};
-	int video_rtcp_fd = -1;
-	if(dovideo) {
-		video_fd[0] = janus_streaming_create_fd(vport, vmcast ? inet_addr(vmcast) : INADDR_ANY, viface,
-			"Video", "video", name ? name : tempname, vport == 0);
-		if(video_fd[0] < 0) {
-			JANUS_LOG(LOG_ERR, "Can't bind to port %d for video...\n", vport);
-			if(audio_fd > -1)
-				close(audio_fd);
-			if(audio_rtcp_fd > -1)
-				close(audio_rtcp_fd);
-			janus_mutex_lock(&mountpoints_mutex);
-			g_hash_table_remove(mountpoints_temp, &id);
-			janus_mutex_unlock(&mountpoints_mutex);
-			return NULL;
-		}
-		vport = janus_streaming_get_fd_port(video_fd[0]);
-		if(vrtcpport > 0) {
-			video_rtcp_fd = janus_streaming_create_fd(vrtcpport, vmcast ? inet_addr(vmcast) : INADDR_ANY, viface,
-				"Video", "video", name ? name : tempname, FALSE);
-			if(video_rtcp_fd < 0) {
-				JANUS_LOG(LOG_ERR, "Can't bind to port %d for video RTCP...\n", vrtcpport);
-				if(audio_fd > -1)
-					close(audio_fd);
-				if(audio_rtcp_fd > -1)
-					close(audio_rtcp_fd);
-				if(video_fd[0] > -1)
-					close(video_fd[0]);
-				janus_mutex_lock(&mountpoints_mutex);
-				g_hash_table_remove(mountpoints_temp, &id);
-				janus_mutex_unlock(&mountpoints_mutex);
+		if(port3 > 0) {
+			fd[2] = janus_streaming_create_fd(port3, mcast ? inet_addr(mcast) : INADDR_ANY, iface, type, name, FALSE);
+			if(fd[2] < 0) {
+				JANUS_LOG(LOG_ERR, "[%s] Can't bind to port %d for video (3rd port)...\n", name, port3);
+				if(fd[0] > -1)
+					close(fd[0]);
+				if(fd[1] > -1)
+					close(fd[1]);
+				if(rtcp_fd > -1)
+					close(rtcp_fd);
 				return NULL;
 			}
+			port3 = janus_streaming_get_fd_port(fd[2]);
 		}
-		if(simulcast) {
-			if(vport2 > 0) {
-				video_fd[1] = janus_streaming_create_fd(vport2, vmcast ? inet_addr(vmcast) : INADDR_ANY, viface,
-					"Video", "video", name ? name : tempname, FALSE);
-				if(video_fd[1] < 0) {
-					JANUS_LOG(LOG_ERR, "Can't bind to port %d for video (2nd port)...\n", vport2);
-					if(audio_fd > -1)
-						close(audio_fd);
-					if(audio_rtcp_fd > -1)
-						close(audio_rtcp_fd);
-					if(video_fd[0] > -1)
-						close(video_fd[0]);
-					if(video_rtcp_fd > -1)
-						close(video_rtcp_fd);
-					janus_mutex_lock(&mountpoints_mutex);
-					g_hash_table_remove(mountpoints_temp, &id);
-					janus_mutex_unlock(&mountpoints_mutex);
-					return NULL;
-				}
-				vport2 = janus_streaming_get_fd_port(video_fd[1]);
-			}
-			if(vport3 > 0) {
-				video_fd[2] = janus_streaming_create_fd(vport3, vmcast ? inet_addr(vmcast) : INADDR_ANY, viface,
-					"Video", "video", name ? name : tempname, FALSE);
-				if(video_fd[2] < 0) {
-					JANUS_LOG(LOG_ERR, "Can't bind to port %d for video (3rd port)...\n", vport3);
-					if(audio_fd > -1)
-						close(audio_fd);
-					if(audio_rtcp_fd > -1)
-						close(audio_rtcp_fd);
-					if(video_rtcp_fd > -1)
-						close(video_rtcp_fd);
-					if(video_fd[0] > -1)
-						close(video_fd[0]);
-					if(video_fd[1] > -1)
-						close(video_fd[1]);
-					janus_mutex_lock(&mountpoints_mutex);
-					g_hash_table_remove(mountpoints_temp, &id);
-					janus_mutex_unlock(&mountpoints_mutex);
-					return NULL;
-				}
-				vport3 = janus_streaming_get_fd_port(video_fd[2]);
+	}
+	/* Create the stream */
+	janus_network_address nil;
+	janus_network_address_nullify(&nil);
+
+	janus_streaming_rtp_source_stream *stream = g_malloc0(sizeof(janus_streaming_rtp_source_stream));
+	stream->mindex = mindex;
+	stream->type = mtype;
+	stream->mid = g_strdup(mid);
+	stream->label = g_strdup(label);
+	stream->codecs.pt = (mtype != JANUS_STREAMING_MEDIA_DATA ? codec : -1);
+	stream->codecs.rtpmap = rtpmap ? g_strdup(rtpmap) : NULL;
+	stream->codecs.fmtp = fmtp ? g_strdup(fmtp) : NULL;
+	stream->codecs.video_codec = JANUS_VIDEOCODEC_NONE;
+	if(mtype == JANUS_STREAMING_MEDIA_VIDEO) {
+		if(strstr(rtpmap, "vp8") || strstr(rtpmap, "VP8"))
+			stream->codecs.video_codec = JANUS_VIDEOCODEC_VP8;
+		else if(strstr(rtpmap, "vp9") || strstr(rtpmap, "VP9"))
+			stream->codecs.video_codec = JANUS_VIDEOCODEC_VP9;
+		else if(strstr(rtpmap, "h264") || strstr(rtpmap, "H264"))
+			stream->codecs.video_codec = JANUS_VIDEOCODEC_H264;
+		if(svc) {
+			if(stream->codecs.video_codec == JANUS_VIDEOCODEC_VP9) {
+				stream->svc = TRUE;
+			} else {
+				JANUS_LOG(LOG_WARN, "[%s] SVC is only supported, in an experimental way, for VP9-SVC mountpoints: disabling it...\n", name);
 			}
 		}
 	}
-	int data_fd = -1;
-	if(dodata) {
-#ifdef HAVE_SCTP
-		data_fd = janus_streaming_create_fd(dport, INADDR_ANY, diface,
-			"Data", "data", name ? name : tempname, FALSE);
-		if(data_fd < 0) {
-			JANUS_LOG(LOG_ERR, "Can't bind to port %d for data...\n", dport);
-			if(audio_fd > -1)
-				close(audio_fd);
-			if(audio_rtcp_fd > -1)
-				close(audio_rtcp_fd);
-			if(video_rtcp_fd > -1)
-				close(video_rtcp_fd);
-			if(video_fd[0] > -1)
-				close(video_fd[0]);
-			if(video_fd[1] > -1)
-				close(video_fd[1]);
-			if(video_fd[2] > -1)
-				close(video_fd[2]);
-			janus_mutex_lock(&mountpoints_mutex);
-			g_hash_table_remove(mountpoints_temp, &id);
-			janus_mutex_unlock(&mountpoints_mutex);
-			return NULL;
-		}
-		dport = janus_streaming_get_fd_port(data_fd);
-#else
-		JANUS_LOG(LOG_WARN, "Mountpoint wants to do datachannel relaying, but datachannels support was not compiled...\n");
-		dodata = FALSE;
-#endif
+	stream->simulcast = (mtype == JANUS_STREAMING_MEDIA_VIDEO && simulcast);
+	stream->mcast = mcast ? inet_addr(mcast) : INADDR_ANY;
+	stream->mcast_str = mcast ? g_strdup(mcast) : NULL;
+	stream->port[0] = port;
+	stream->rtcp_port = rtcpport ? rtcpport : -1;
+	stream->port[1] = stream->simulcast ? port2 : -1;
+	stream->port[2] = stream->simulcast ? port3 : -1;
+	stream->iface = !janus_network_address_is_null(iface) ? *iface : nil;
+	stream->iface_str = miface ? g_strdup(miface) : NULL;
+	stream->skew = doskew;
+	stream->rc = NULL;
+	janus_rtp_switching_context_reset(&stream->context[0]);
+	janus_rtp_switching_context_reset(&stream->context[1]);
+	janus_rtp_switching_context_reset(&stream->context[2]);
+	stream->fd[0] = fd[0];
+	stream->fd[1] = fd[1];
+	stream->fd[2] = fd[2];
+	stream->rtcp_fd = rtcp_fd;
+	stream->last_received = janus_get_monotonic_time();
+	if(mtype == JANUS_STREAMING_MEDIA_VIDEO) {
+		stream->keyframe.enabled = bufferkf;
+		stream->keyframe.latest_keyframe = NULL;
+		stream->keyframe.temp_keyframe = NULL;
+		stream->keyframe.temp_ts = 0;
+		janus_mutex_init(&stream->keyframe.mutex);
+	} else if(mtype == JANUS_STREAMING_MEDIA_DATA) {
+		stream->buffermsg = buffermsg;
+		stream->last_msg = NULL;
+		janus_mutex_init(&stream->buffermsg_mutex);
 	}
-	/* Create the mountpoint */
-	janus_network_address nil;
-	janus_network_address_nullify(&nil);
+	janus_refcount_init(&stream->ref, janus_streaming_rtp_source_stream_free);
+	return stream;
+}
 
+janus_streaming_mountpoint *janus_streaming_create_rtp_source(
+		uint64_t id, char *name, char *desc, GList *media,
+		int srtpsuite, char *srtpcrypto, int threads, int rtp_collision) {
+	char tempname[255];
+	if(name == NULL) {
+		JANUS_LOG(LOG_VERB, "Missing name, will generate a random one...\n");
+		memset(tempname, 0, 255);
+		g_snprintf(tempname, 255, "mp-%"SCNu64, id);
+	} else if(atoi(name) != 0) {
+		JANUS_LOG(LOG_VERB, "Names can't start with a number, prefixing it...\n");
+		memset(tempname, 0, 255);
+		g_snprintf(tempname, 255, "mp-%s", name);
+		name = NULL;
+	}
+	if(!media || g_list_length(media) == 0) {
+		JANUS_LOG(LOG_ERR, "Can't add 'rtp' mountpoint, no audio, video or data have to be streamed...\n");
+		janus_mutex_unlock(&mountpoints_mutex);
+		return NULL;
+	}
+	/* Create the mountpoint */
 	janus_streaming_mountpoint *live_rtp = g_malloc0(sizeof(janus_streaming_mountpoint));
 	live_rtp->id = id;
 	live_rtp->name = g_strdup(name ? name : tempname);
@@ -5068,9 +5843,6 @@ janus_streaming_mountpoint *janus_streaming_create_rtp_source(
 	live_rtp->description = description;
 	live_rtp->enabled = TRUE;
 	live_rtp->active = FALSE;
-	live_rtp->audio = doaudio;
-	live_rtp->video = dovideo;
-	live_rtp->data = dodata;
 	live_rtp->streaming_type = janus_streaming_type_live;
 	live_rtp->streaming_source = janus_streaming_source_rtp;
 	janus_streaming_rtp_source *live_rtp_source = g_malloc0(sizeof(janus_streaming_rtp_source));
@@ -5082,20 +5854,6 @@ janus_streaming_mountpoint *janus_streaming_create_rtp_source(
 		if(len < SRTP_MASTER_LENGTH) {
 			JANUS_LOG(LOG_ERR, "Invalid SRTP crypto (%s)\n", srtpcrypto);
 			g_free(decoded);
-			if(audio_fd > -1)
-				close(audio_fd);
-			if(audio_rtcp_fd > -1)
-				close(audio_rtcp_fd);
-			if(video_rtcp_fd > -1)
-				close(video_rtcp_fd);
-			if(video_fd[0] > -1)
-				close(video_fd[0]);
-			if(video_fd[1] > -1)
-				close(video_fd[1]);
-			if(video_fd[2] > -1)
-				close(video_fd[2]);
-			if(data_fd > -1)
-				close(data_fd);
 			janus_mutex_lock(&mountpoints_mutex);
 			g_hash_table_remove(mountpoints_temp, &id);
 			janus_mutex_unlock(&mountpoints_mutex);
@@ -5122,20 +5880,6 @@ janus_streaming_mountpoint *janus_streaming_create_rtp_source(
 			/* Something went wrong... */
 			JANUS_LOG(LOG_ERR, "Error creating forwarder SRTP session: %d (%s)\n", res, janus_srtp_error_str(res));
 			g_free(decoded);
-			if(audio_fd > -1)
-				close(audio_fd);
-			if(audio_rtcp_fd > -1)
-				close(audio_rtcp_fd);
-			if(video_rtcp_fd > -1)
-				close(video_rtcp_fd);
-			if(video_fd[0] > -1)
-				close(video_fd[0]);
-			if(video_fd[1] > -1)
-				close(video_fd[1]);
-			if(video_fd[2] > -1)
-				close(video_fd[2]);
-			if(data_fd > -1)
-				close(data_fd);
 			janus_mutex_lock(&mountpoints_mutex);
 			g_hash_table_remove(mountpoints_temp, &id);
 			janus_mutex_unlock(&mountpoints_mutex);
@@ -5149,74 +5893,41 @@ janus_streaming_mountpoint *janus_streaming_create_rtp_source(
 		live_rtp_source->srtpsuite = srtpsuite;
 		live_rtp_source->srtpcrypto = g_strdup(srtpcrypto);
 	}
-	live_rtp_source->audio_mcast = doaudio ? (amcast ? inet_addr(amcast) : INADDR_ANY) : INADDR_ANY;
-	live_rtp_source->audio_iface = doaudio && !janus_network_address_is_null(aiface) ? *aiface : nil;
-	live_rtp_source->audio_port = doaudio ? aport : -1;
-	live_rtp_source->audio_rtcp_port = artcpport;
-	live_rtp_source->askew = doaskew;
-	live_rtp_source->video_mcast = dovideo ? (vmcast ? inet_addr(vmcast) : INADDR_ANY) : INADDR_ANY;
-	live_rtp_source->video_port[0] = dovideo ? vport : -1;
-	live_rtp_source->video_rtcp_port = vrtcpport;
-	live_rtp_source->simulcast = dovideo && simulcast;
-	live_rtp_source->video_port[1] = live_rtp_source->simulcast ? vport2 : -1;
-	live_rtp_source->video_port[2] = live_rtp_source->simulcast ? vport3 : -1;
-	live_rtp_source->video_iface = dovideo && !janus_network_address_is_null(viface) ? *viface : nil;
-	live_rtp_source->vskew = dovskew;
-	live_rtp_source->data_port = dodata ? dport : -1;
-	live_rtp_source->data_iface = dodata && !janus_network_address_is_null(diface) ? *diface : nil;
-	live_rtp_source->arc = NULL;
-	live_rtp_source->vrc = NULL;
-	live_rtp_source->drc = NULL;
-	janus_rtp_switching_context_reset(&live_rtp_source->context[0]);
-	janus_rtp_switching_context_reset(&live_rtp_source->context[1]);
-	janus_rtp_switching_context_reset(&live_rtp_source->context[2]);
-	janus_mutex_init(&live_rtp_source->rec_mutex);
-	live_rtp_source->audio_fd = audio_fd;
-	live_rtp_source->audio_rtcp_fd = audio_rtcp_fd;
-	live_rtp_source->video_rtcp_fd = video_rtcp_fd;
-	live_rtp_source->video_fd[0] = video_fd[0];
-	live_rtp_source->video_fd[1] = video_fd[1];
-	live_rtp_source->video_fd[2] = video_fd[2];
-	live_rtp_source->data_fd = data_fd;
+	live_rtp->audio = FALSE;
+	live_rtp->video = FALSE;
+	live_rtp->data = FALSE;
+	live_rtp_source->media = media;
+	live_rtp_source->media_byid = g_hash_table_new(NULL, NULL);
+	live_rtp_source->media_byfd = g_hash_table_new(NULL, NULL);
+	GList *temp = media;
+	while(temp) {
+		janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+		if(stream->type == JANUS_STREAMING_MEDIA_AUDIO)
+			live_rtp->audio = TRUE;
+		else if(stream->type == JANUS_STREAMING_MEDIA_VIDEO)
+			live_rtp->video = TRUE;
+		else if(stream->type == JANUS_STREAMING_MEDIA_DATA)
+			live_rtp->data = TRUE;
+		/* Map this stream by mindex */
+		g_hash_table_insert(live_rtp_source->media_byid, GINT_TO_POINTER(stream->mindex), stream);
+		/* Map this stream by all its file descriptors */
+		if(stream->fd[0] != -1)
+			g_hash_table_insert(live_rtp_source->media_byfd, GINT_TO_POINTER(stream->fd[0]), stream);
+		if(stream->fd[1] != -1)
+			g_hash_table_insert(live_rtp_source->media_byfd, GINT_TO_POINTER(stream->fd[1]), stream);
+		if(stream->fd[2] != -1)
+			g_hash_table_insert(live_rtp_source->media_byfd, GINT_TO_POINTER(stream->fd[2]), stream);
+		if(stream->rtcp_fd != -1)
+			g_hash_table_insert(live_rtp_source->media_byfd, GINT_TO_POINTER(stream->rtcp_fd), stream);
+		temp = temp->next;
+	}
 	live_rtp_source->pipefd[0] = -1;
 	live_rtp_source->pipefd[1] = -1;
 	pipe(live_rtp_source->pipefd);
-	live_rtp_source->last_received_audio = janus_get_monotonic_time();
-	live_rtp_source->last_received_video = janus_get_monotonic_time();
-	live_rtp_source->last_received_data = janus_get_monotonic_time();
-	live_rtp_source->keyframe.enabled = bufferkf;
-	live_rtp_source->keyframe.latest_keyframe = NULL;
-	live_rtp_source->keyframe.temp_keyframe = NULL;
-	live_rtp_source->keyframe.temp_ts = 0;
-	janus_mutex_init(&live_rtp_source->keyframe.mutex);
+	janus_mutex_init(&live_rtp_source->rec_mutex);
 	live_rtp_source->rtp_collision = rtp_collision;
-	live_rtp_source->buffermsg = buffermsg;
-	live_rtp_source->last_msg = NULL;
-	janus_mutex_init(&live_rtp_source->buffermsg_mutex);
 	live_rtp->source = live_rtp_source;
-	live_rtp->source_destroy = (GDestroyNotify) janus_streaming_rtp_source_free;
-	live_rtp->codecs.audio_pt = doaudio ? acodec : -1;
-	live_rtp->codecs.audio_rtpmap = doaudio ? g_strdup(artpmap) : NULL;
-	live_rtp->codecs.audio_fmtp = doaudio ? (afmtp ? g_strdup(afmtp) : NULL) : NULL;
-	live_rtp->codecs.video_codec = JANUS_VIDEOCODEC_NONE;
-	if(dovideo) {
-		if(strstr(vrtpmap, "vp8") || strstr(vrtpmap, "VP8"))
-			live_rtp->codecs.video_codec = JANUS_VIDEOCODEC_VP8;
-		else if(strstr(vrtpmap, "vp9") || strstr(vrtpmap, "VP9"))
-			live_rtp->codecs.video_codec = JANUS_VIDEOCODEC_VP9;
-		else if(strstr(vrtpmap, "h264") || strstr(vrtpmap, "H264"))
-			live_rtp->codecs.video_codec = JANUS_VIDEOCODEC_H264;
-	}
-	if(svc) {
-		if(live_rtp->codecs.video_codec == JANUS_VIDEOCODEC_VP9) {
-			live_rtp_source->svc = TRUE;
-		} else {
-			JANUS_LOG(LOG_WARN, "SVC is only supported, in an experimental way, for VP9-SVC mountpoints: disabling it...\n");
-		}
-	}
-	live_rtp->codecs.video_pt = dovideo ? vcodec : -1;
-	live_rtp->codecs.video_rtpmap = dovideo ? g_strdup(vrtpmap) : NULL;
-	live_rtp->codecs.video_fmtp = dovideo ? (vfmtp ? g_strdup(vfmtp) : NULL) : NULL;
+	live_rtp->source_destroy = (GDestroyNotify)janus_streaming_rtp_source_free;
 	live_rtp->viewers = NULL;
 	g_atomic_int_set(&live_rtp->destroyed, 0);
 	janus_refcount_init(&live_rtp->ref, janus_streaming_mountpoint_free);
@@ -5245,6 +5956,7 @@ janus_streaming_mountpoint *janus_streaming_create_rtp_source(
 			if(error != NULL) {
 				JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the helper thread...\n",
 					error->code, error->message ? error->message : "??");
+				g_error_free(error);
 				janus_refcount_decrease(&live_rtp->ref);	/* This is for the helper thread */
 				janus_streaming_mountpoint_destroy(live_rtp);
 				return NULL;
@@ -5326,12 +6038,10 @@ janus_streaming_mountpoint *janus_streaming_create_file_source(
 	file_source->streaming_source = janus_streaming_source_file;
 	janus_streaming_file_source *file_source_source = g_malloc0(sizeof(janus_streaming_file_source));
 	file_source_source->filename = g_strdup(filename);
+	file_source_source->codecs.pt = strstr(filename, ".alaw") ? 8 : 0;
+	file_source_source->codecs.rtpmap = g_strdup(strstr(filename, ".alaw") ? "PCMA/8000" : "PCMU/8000");
 	file_source->source = file_source_source;
-	file_source->source_destroy = (GDestroyNotify) janus_streaming_file_source_free;
-	file_source->codecs.audio_pt = strstr(filename, ".alaw") ? 8 : 0;
-	file_source->codecs.audio_rtpmap = g_strdup(strstr(filename, ".alaw") ? "PCMA/8000" : "PCMU/8000");
-	file_source->codecs.video_pt = -1;	/* FIXME We don't support video for this type yet */
-	file_source->codecs.video_rtpmap = NULL;
+	file_source->source_destroy = (GDestroyNotify)janus_streaming_file_source_free;
 	file_source->viewers = NULL;
 	g_atomic_int_set(&file_source->destroyed, 0);
 	janus_refcount_init(&file_source->ref, janus_streaming_mountpoint_free);
@@ -5348,6 +6058,7 @@ janus_streaming_mountpoint *janus_streaming_create_file_source(
 		file_source->thread = g_thread_try_new(tname, &janus_streaming_filesource_thread, file_source, &error);
 		if(error != NULL) {
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the live filesource thread...\n", error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			janus_refcount_decrease(&file_source->ref);		/* This is for the failed thread */
 			janus_refcount_decrease(&file_source->ref);
 			return NULL;
@@ -5518,13 +6229,14 @@ static int janus_streaming_rtsp_connect_to_server(janus_streaming_mountpoint *mp
 
 	janus_mutex_lock(&mountpoints_mutex);
 	/* Parse both video and audio first before proceed to setup as curldata will be reused */
+	janus_network_address audio_iface, video_iface;
+	uint32_t audio_ssrc = 0, video_ssrc = 0;
 	int vresult;
 	vresult = janus_streaming_rtsp_parse_sdp(curldata->buffer, name, "video", &vpt,
-		vtransport, vhost, vrtpmap, vfmtp, vcontrol, &source->video_iface, &video_fds);
-
+		vtransport, vhost, vrtpmap, vfmtp, vcontrol, &video_iface, &video_fds);
 	int aresult;
 	aresult = janus_streaming_rtsp_parse_sdp(curldata->buffer, name, "audio", &apt,
-		atransport, ahost, artpmap, afmtp, acontrol, &source->audio_iface, &audio_fds);
+		atransport, ahost, artpmap, afmtp, acontrol, &audio_iface, &audio_fds);
 	janus_mutex_unlock(&mountpoints_mutex);
 
 	if(vresult == -1 && aresult == -1) {
@@ -5586,9 +6298,8 @@ static int janus_streaming_rtsp_connect_to_server(janus_streaming_mountpoint *mp
 		/* Get the RTP server port, which we'll need for the latching packet */
 		const char *vssrc = strstr(curldata->buffer, ";ssrc=");
 		if(vssrc != NULL) {
-			uint32_t ssrc = strtol(vssrc+strlen(";ssrc="), NULL, 16);
-			JANUS_LOG(LOG_VERB, "  -- SSRC (video): %"SCNu32"\n", ssrc);
-			source->video_ssrc = ssrc;
+			video_ssrc = strtol(vssrc+strlen(";ssrc="), NULL, 16);
+			JANUS_LOG(LOG_VERB, "  -- SSRC (video): %"SCNu32"\n", video_ssrc);
 		}
 		const char *server_ports = strstr(curldata->buffer, ";server_port=");
 		if(server_ports != NULL) {
@@ -5656,9 +6367,8 @@ static int janus_streaming_rtsp_connect_to_server(janus_streaming_mountpoint *mp
 		/* Get the RTP server port, which we'll need for the latching packet */
 		const char *assrc = strstr(curldata->buffer, ";ssrc=");
 		if(assrc != NULL) {
-			uint32_t ssrc = strtol(assrc+strlen(";ssrc="), NULL, 16);
-			JANUS_LOG(LOG_VERB, "  -- SSRC (audio): %"SCNu32"\n", ssrc);
-			source->audio_ssrc = ssrc;
+			audio_ssrc = strtol(assrc+strlen(";ssrc="), NULL, 16);
+			JANUS_LOG(LOG_VERB, "  -- SSRC (audio): %"SCNu32"\n", audio_ssrc);
 		}
 		const char *server_ports = strstr(curldata->buffer, ";server_port=");
 		if(server_ports != NULL) {
@@ -5670,33 +6380,74 @@ static int janus_streaming_rtsp_connect_to_server(janus_streaming_mountpoint *mp
 		}
 	}
 
-	/* Update the source (but check if ptype/rtpmap/fmtp need to be overridden) */
-	if(mp->codecs.audio_pt == -1)
-		mp->codecs.audio_pt = doaudio ? apt : -1;
-	if(mp->codecs.audio_rtpmap == NULL)
-		mp->codecs.audio_rtpmap = doaudio ? g_strdup(artpmap) : NULL;
-	if(mp->codecs.audio_fmtp == NULL)
-		mp->codecs.audio_fmtp = doaudio ? g_strdup(afmtp) : NULL;
-	if(mp->codecs.video_pt == -1)
-		mp->codecs.video_pt = dovideo ? vpt : -1;
-	if(mp->codecs.video_rtpmap == NULL)
-		mp->codecs.video_rtpmap = dovideo ? g_strdup(vrtpmap) : NULL;
-	if(mp->codecs.video_fmtp == NULL)
-		mp->codecs.video_fmtp = dovideo ? g_strdup(vfmtp) : NULL;
-	source->audio_fd = audio_fds.fd;
-	source->audio_rtcp_fd = audio_fds.rtcp_fd;
-	source->remote_audio_port = asport;
-	source->remote_audio_rtcp_port = asport_rtcp;
-	g_free(source->rtsp_ahost);
-	if(asport > 0)
-		source->rtsp_ahost = g_strdup(ahost);
-	source->video_fd[0] = video_fds.fd;
-	source->video_rtcp_fd = video_fds.rtcp_fd;
-	source->remote_video_port = vsport;
-	source->remote_video_rtcp_port = vsport_rtcp;
-	g_free(source->rtsp_vhost);
-	if(vsport > 0)
-		source->rtsp_vhost = g_strdup(vhost);
+	/* Do we have media streams already? */
+	if(source->media == NULL) {
+		if(doaudio) {
+			janus_streaming_rtp_source_stream *stream = g_malloc0(sizeof(janus_streaming_rtp_source_stream));
+			stream->mindex = g_list_length(source->media);
+			stream->type = JANUS_STREAMING_MEDIA_AUDIO;
+			stream->mid = g_strdup("a");
+			stream->label = g_strdup("RTSP audio");
+			source->media = g_list_append(source->media, stream);
+			janus_refcount_init(&stream->ref, janus_streaming_rtp_source_stream_free);
+		}
+		if(dovideo) {
+			janus_streaming_rtp_source_stream *stream = g_malloc0(sizeof(janus_streaming_rtp_source_stream));
+			stream->mindex = g_list_length(source->media);
+			stream->type = JANUS_STREAMING_MEDIA_VIDEO;
+			stream->mid = g_strdup("v");
+			stream->label = g_strdup("RTSP video");
+			source->media = g_list_append(source->media, stream);
+			janus_refcount_init(&stream->ref, janus_streaming_rtp_source_stream_free);
+		}
+	}
+
+	/* Update the source (but check if rtpmap/fmtp need to be overridden) */
+	GList *temp = source->media;
+	while(temp) {
+		janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+		if(stream->type == JANUS_STREAMING_MEDIA_AUDIO) {
+			stream->codecs.pt = source->rtsp_acodecs.pt != -1 ? source->rtsp_acodecs.pt : apt;
+			g_free(stream->codecs.rtpmap);
+			stream->codecs.rtpmap = source->rtsp_acodecs.rtpmap ? g_strdup(source->rtsp_acodecs.rtpmap) : g_strdup(artpmap);
+			g_free(stream->codecs.fmtp);
+			stream->codecs.fmtp = source->rtsp_acodecs.fmtp ? g_strdup(source->rtsp_acodecs.fmtp) : g_strdup(afmtp);
+			g_free(source->rtsp_ahost);
+			source->rtsp_ahost = asport > 0 ? g_strdup(ahost) : NULL;
+			stream->remote_port = asport;
+			stream->remote_rtcp_port = asport_rtcp;
+			/* Map this stream by all its file descriptors */
+			stream->fd[0] = audio_fds.fd;
+			stream->fd[1] = -1;
+			stream->fd[2] = -1;
+			stream->rtcp_fd = audio_fds.rtcp_fd;
+			if(stream->fd[0] != -1)
+				g_hash_table_insert(source->media_byfd, GINT_TO_POINTER(stream->fd[0]), stream);
+			if(stream->rtcp_fd != -1)
+				g_hash_table_insert(source->media_byfd, GINT_TO_POINTER(stream->rtcp_fd), stream);
+		}
+		if(stream->type == JANUS_STREAMING_MEDIA_VIDEO) {
+			stream->codecs.pt = source->rtsp_vcodecs.pt != -1 ? source->rtsp_vcodecs.pt : vpt;
+			g_free(stream->codecs.rtpmap);
+			stream->codecs.rtpmap = source->rtsp_vcodecs.rtpmap ? g_strdup(source->rtsp_vcodecs.rtpmap) : g_strdup(vrtpmap);
+			g_free(stream->codecs.fmtp);
+			stream->codecs.fmtp = source->rtsp_vcodecs.fmtp ? g_strdup(source->rtsp_vcodecs.fmtp) : g_strdup(vfmtp);
+			g_free(source->rtsp_vhost);
+			source->rtsp_vhost = vsport > 0 ? g_strdup(vhost) : NULL;
+			stream->remote_port = vsport;
+			stream->remote_rtcp_port = vsport_rtcp;
+			/* Map this stream by all its file descriptors */
+			stream->fd[0] = video_fds.fd;
+			stream->fd[1] = -1;
+			stream->fd[2] = -1;
+			stream->rtcp_fd = video_fds.rtcp_fd;
+			if(stream->fd[0] != -1)
+				g_hash_table_insert(source->media_byfd, GINT_TO_POINTER(stream->fd[0]), stream);
+			if(stream->rtcp_fd != -1)
+				g_hash_table_insert(source->media_byfd, GINT_TO_POINTER(stream->rtcp_fd), stream);
+		}
+		temp = temp->next;
+	}
 	source->curl = curl;
 	source->curldata = curldata;
 	source->ka_timeout = ka_timeout;
@@ -5753,25 +6504,21 @@ static int janus_streaming_rtsp_play(janus_streaming_rtp_source *source) {
 		return -1;
 	/* First of all, send a latching packet to the RTSP server port(s) */
 	struct sockaddr_in6 remote;
-	if(source->remote_audio_port > 0 && source->audio_fd >= 0) {
-		JANUS_LOG(LOG_VERB, "RTSP audio latching: %s:%d\n", source->rtsp_ahost, source->remote_audio_port);
-		janus_streaming_rtsp_latch(source->audio_fd, source->rtsp_ahost,
-			source->remote_audio_port, (struct sockaddr *)&remote);
-		if(source->remote_audio_rtcp_port > 0 && source->audio_rtcp_fd >= 0) {
-			JANUS_LOG(LOG_VERB, "  -- RTCP: %s:%d\n", source->rtsp_ahost, source->remote_audio_rtcp_port);
-			janus_streaming_rtsp_latch(source->audio_rtcp_fd, source->rtsp_ahost,
-				source->remote_audio_rtcp_port, (struct sockaddr *)&source->audio_rtcp_addr);
-		}
-	}
-	if(source->remote_video_port > 0 && source->video_fd[0] >= 0) {
-		JANUS_LOG(LOG_VERB, "RTSP video latching: %s:%d\n", source->rtsp_vhost, source->remote_video_port);
-		janus_streaming_rtsp_latch(source->video_fd[0], source->rtsp_vhost,
-			source->remote_video_port, (struct sockaddr *)&remote);
-		if(source->remote_video_rtcp_port > 0 && source->video_rtcp_fd >= 0) {
-			JANUS_LOG(LOG_VERB, "  -- RTCP: %s:%d\n", source->rtsp_vhost, source->remote_video_rtcp_port);
-			janus_streaming_rtsp_latch(source->video_rtcp_fd, source->rtsp_vhost,
-				source->remote_video_rtcp_port, (struct sockaddr *)&source->video_rtcp_addr);
+	GList *temp = source->media;
+	while(temp) {
+		janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+		if(stream->remote_port > 0 && stream->fd[0] >= 0) {
+			char *host = (stream->type == JANUS_STREAMING_MEDIA_AUDIO) ? source->rtsp_ahost : source->rtsp_vhost;
+			JANUS_LOG(LOG_VERB, "RTSP %s latching: %s:%d\n",
+				janus_streaming_media_str(stream->type), host, stream->remote_port);
+			janus_streaming_rtsp_latch(stream->fd[0], host, stream->remote_port, (struct sockaddr *)&remote);
+			if(stream->remote_rtcp_port > 0 && stream->rtcp_fd >= 0) {
+				JANUS_LOG(LOG_VERB, "  -- RTCP: %s:%d\n", host, stream->remote_rtcp_port);
+				janus_streaming_rtsp_latch(stream->rtcp_fd, host,
+					stream->remote_rtcp_port, (struct sockaddr *)&stream->rtcp_addr);
+			}
 		}
+		temp = temp->next;
 	}
 	/* Send an RTSP PLAY */
 	janus_mutex_lock(&source->rtsp_mutex);
@@ -5847,37 +6594,27 @@ janus_streaming_mountpoint *janus_streaming_create_rtsp_source(
 	live_rtsp_source->rtsp_url = g_strdup(url);
 	live_rtsp_source->rtsp_username = username ? g_strdup(username) : NULL;
 	live_rtsp_source->rtsp_password = password ? g_strdup(password) : NULL;
-	live_rtsp_source->arc = NULL;
-	live_rtsp_source->vrc = NULL;
-	live_rtsp_source->drc = NULL;
-	live_rtsp_source->audio_fd = -1;
-	live_rtsp_source->audio_rtcp_fd = -1;
-	live_rtsp_source->audio_iface = iface ? *iface : nil;
-	live_rtsp_source->video_fd[0] = -1;
-	live_rtsp_source->video_fd[1] = -1;
-	live_rtsp_source->video_fd[2] = -1;
-	live_rtsp_source->video_rtcp_fd = -1;
-	live_rtsp_source->video_iface = iface ? *iface : nil;
-	live_rtsp_source->data_fd = -1;
+	live_rtsp_source->media = NULL;		/* We'll prepare this later */
+	live_rtsp_source->media_byid = g_hash_table_new(NULL, NULL);
+	live_rtsp_source->media_byfd = g_hash_table_new(NULL, NULL);
 	live_rtsp_source->pipefd[0] = -1;
 	live_rtsp_source->pipefd[1] = -1;
 	pipe(live_rtsp_source->pipefd);
-	live_rtsp_source->data_iface = nil;
 	live_rtsp_source->reconnect_timer = 0;
 	janus_mutex_init(&live_rtsp_source->rtsp_mutex);
 	live_rtsp->source = live_rtsp_source;
-	live_rtsp->source_destroy = (GDestroyNotify) janus_streaming_rtp_source_free;
+	live_rtsp->source_destroy = (GDestroyNotify)janus_streaming_rtp_source_free;
 	live_rtsp->viewers = NULL;
 	g_atomic_int_set(&live_rtsp->destroyed, 0);
 	janus_refcount_init(&live_rtsp->ref, janus_streaming_mountpoint_free);
 	janus_mutex_init(&live_rtsp->mutex);
 	/* We may have to override the payload type and/or rtpmap and/or fmtp for audio and/or video */
-	live_rtsp->codecs.audio_pt = doaudio ? acodec : -1;
-	live_rtsp->codecs.audio_rtpmap = doaudio ? (artpmap ? g_strdup(artpmap) : NULL) : NULL;
-	live_rtsp->codecs.audio_fmtp = doaudio ? (afmtp ? g_strdup(afmtp) : NULL) : NULL;
-	live_rtsp->codecs.video_pt = dovideo ? vcodec : -1;
-	live_rtsp->codecs.video_rtpmap = dovideo ? (vrtpmap ? g_strdup(vrtpmap) : NULL) : NULL;
-	live_rtsp->codecs.video_fmtp = dovideo ? (vfmtp ? g_strdup(vfmtp) : NULL) : NULL;
+	live_rtsp_source->rtsp_acodecs.pt = doaudio ? acodec : -1;
+	live_rtsp_source->rtsp_acodecs.rtpmap = doaudio ? (artpmap ? g_strdup(artpmap) : NULL) : NULL;
+	live_rtsp_source->rtsp_acodecs.fmtp = doaudio ? (afmtp ? g_strdup(afmtp) : NULL) : NULL;
+	live_rtsp_source->rtsp_vcodecs.pt = dovideo ? vcodec : -1;
+	live_rtsp_source->rtsp_vcodecs.rtpmap = dovideo ? (vrtpmap ? g_strdup(vrtpmap) : NULL) : NULL;
+	live_rtsp_source->rtsp_vcodecs.fmtp = dovideo ? (vfmtp ? g_strdup(vfmtp) : NULL) : NULL;
 	/* If we need to return an error on failure, try connecting right now */
 	if(error_on_failure) {
 		/* Now connect to the RTSP server */
@@ -5907,6 +6644,7 @@ janus_streaming_mountpoint *janus_streaming_create_rtsp_source(
 	live_rtsp->thread = g_thread_try_new(tname, &janus_streaming_relay_thread, live_rtsp, &error);
 	if(error != NULL) {
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the RTSP thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		janus_mutex_lock(&mountpoints_mutex);
 		g_hash_table_remove(mountpoints_temp, &id);
 		janus_mutex_unlock(&mountpoints_mutex);
@@ -5991,7 +6729,7 @@ static void *janus_streaming_ondemand_thread(void *data) {
 	janus_rtp_header *header = (janus_rtp_header *)buf;
 	header->version = 2;
 	header->markerbit = 1;
-	header->type = mountpoint->codecs.audio_pt;
+	header->type = source->codecs.pt;
 	header->seq_number = htons(seq);
 	header->timestamp = htonl(ts);
 	header->ssrc = htonl(1);	/* The gateway will fix this anyway */
@@ -6043,6 +6781,7 @@ static void *janus_streaming_ondemand_thread(void *data) {
 			//~ header->type, ntohs(header->seq_number), ntohl(header->timestamp));
 		//~ JANUS_LOG(LOG_VERB, " ... Read %d bytes from the audio file...\n", read);
 		/* Relay on all sessions */
+		packet.mindex = -1;
 		packet.data = header;
 		packet.length = RTP_HEADER_SIZE + read;
 		packet.is_rtp = TRUE;
@@ -6111,7 +6850,7 @@ static void *janus_streaming_filesource_thread(void *data) {
 	janus_rtp_header *header = (janus_rtp_header *)buf;
 	header->version = 2;
 	header->markerbit = 1;
-	header->type = mountpoint->codecs.audio_pt;
+	header->type = source->codecs.pt;
 	header->seq_number = htons(seq);
 	header->timestamp = htonl(ts);
 	header->ssrc = htonl(1);	/* The Janus core will fix this anyway */
@@ -6163,6 +6902,7 @@ static void *janus_streaming_filesource_thread(void *data) {
 			// header->type, ntohs(header->seq_number), ntohl(header->timestamp));
 		// JANUS_LOG(LOG_VERB, " ... Read %d bytes from the audio file...\n", read);
 		/* Relay on all sessions */
+		packet.mindex = -1;
 		packet.data = header;
 		packet.length = RTP_HEADER_SIZE + read;
 		packet.is_rtp = TRUE;
@@ -6209,22 +6949,19 @@ static void *janus_streaming_relay_thread(void *data) {
 		janus_refcount_decrease(&mountpoint->ref);
 		return NULL;
 	}
-	int audio_fd = source->audio_fd;
-	int video_fd[3] = {source->video_fd[0], source->video_fd[1], source->video_fd[2]};
-	int data_fd = source->data_fd;
-	int pipe_fd = source->pipefd[0];
-	int audio_rtcp_fd = source->audio_rtcp_fd;
-	int video_rtcp_fd = source->video_rtcp_fd;
 	char *name = g_strdup(mountpoint->name ? mountpoint->name : "??");
 	/* Needed to fix seq and ts */
-	uint32_t ssrc = 0, a_last_ssrc = 0, v_last_ssrc[3] = {0, 0, 0};
+	uint32_t ssrc = 0;
 	/* File descriptors */
 	socklen_t addrlen;
 	struct sockaddr_storage remote;
 	int resfd = 0, bytes = 0;
-	struct pollfd fds[8];
+	struct pollfd fds[50];		/* FIXME We should make this dynamic */
 	char buffer[1500];
 	memset(buffer, 0, 1500);
+	/* We'll have a dynamic number of streams */
+	janus_streaming_rtp_source_stream *stream = NULL;
+	GList *temp = NULL;
 #ifdef HAVE_LIBCURL
 	/* In case this is an RTSP restreamer, we may have to send keep-alives from time to time */
 	gint64 now = janus_get_monotonic_time(), before = now, ka_timeout = 0;
@@ -6232,6 +6969,7 @@ static void *janus_streaming_relay_thread(void *data) {
 		source->reconnect_timer = now;
 		ka_timeout = ((gint64)source->ka_timeout*G_USEC_PER_SEC)/2;
 	}
+	gboolean connected = TRUE;
 #endif
 	/* Loop */
 	int num = 0;
@@ -6250,12 +6988,33 @@ static void *janus_streaming_relay_thread(void *data) {
 				/* 5 seconds passed and no media? Assume the RTSP server has gone and schedule a reconnect */
 				JANUS_LOG(LOG_WARN, "[%s] %"SCNi64"s passed with no media, trying to reconnect the RTSP stream\n",
 					name, (now - source->reconnect_timer)/G_USEC_PER_SEC);
-				audio_fd = -1;
-				video_fd[0] = -1;
-				video_fd[1] = -1;
-				video_fd[2] = -1;
-				data_fd = -1;
+				temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					if(stream->fd[0] > -1) {
+						g_hash_table_remove(source->media_byfd, GINT_TO_POINTER(stream->fd[0]));
+						close(stream->fd[0]);
+					}
+					stream->fd[0] = -1;
+					if(stream->fd[1] > -1) {
+						g_hash_table_remove(source->media_byfd, GINT_TO_POINTER(stream->fd[1]));
+						close(stream->fd[1]);
+					}
+					stream->fd[1] = -1;
+					if(stream->fd[2] > -1) {
+						g_hash_table_remove(source->media_byfd, GINT_TO_POINTER(stream->fd[2]));
+						close(stream->fd[2]);
+					}
+					stream->fd[2] = -1;
+					if(stream->rtcp_fd > -1) {
+						g_hash_table_remove(source->media_byfd, GINT_TO_POINTER(stream->rtcp_fd));
+						close(stream->rtcp_fd);
+					}
+					stream->rtcp_fd = -1;
+					temp = temp->next;
+				}
 				source->reconnect_timer = now;
+				connected = FALSE;
 				source->reconnecting = TRUE;
 				/* Let's clean up the source first */
 				curl_easy_cleanup(source->curl);
@@ -6264,34 +7023,6 @@ static void *janus_streaming_relay_thread(void *data) {
 					g_free(source->curldata->buffer);
 				g_free(source->curldata);
 				source->curldata = NULL;
-				if(source->audio_fd > -1) {
-					close(source->audio_fd);
-				}
-				source->audio_fd = -1;
-				if(source->video_fd[0] > -1) {
-					close(source->video_fd[0]);
-				}
-				source->video_fd[0] = -1;
-				if(source->video_fd[1] > -1) {
-					close(source->video_fd[1]);
-				}
-				source->video_fd[1] = -1;
-				if(source->video_fd[2] > -1) {
-					close(source->video_fd[2]);
-				}
-				source->video_fd[2] = -1;
-				if(source->data_fd > -1) {
-					close(source->data_fd);
-				}
-				source->data_fd = -1;
-				if(source->audio_rtcp_fd > -1) {
-					close(source->audio_rtcp_fd);
-				}
-				source->audio_rtcp_fd = -1;
-				if(source->video_rtcp_fd > -1) {
-					close(source->video_rtcp_fd);
-				}
-				source->video_rtcp_fd = -1;
 				/* Now let's try to reconnect */
 				if(janus_streaming_rtsp_connect_to_server(mountpoint) < 0) {
 					/* Reconnection failed? Let's try again later */
@@ -6302,14 +7033,10 @@ static void *janus_streaming_relay_thread(void *data) {
 						/* Error trying to play? Let's try again later */
 						JANUS_LOG(LOG_WARN, "[%s] RTSP PLAY failed, trying again in a few seconds...\n", name);
 					} else {
-						/* Everything should be back to normal, let's update the file descriptors */
+						/* Everything should be back to normal */
 						JANUS_LOG(LOG_INFO, "[%s] Reconnected to the RTSP server, streaming again\n", name);
-						audio_fd = source->audio_fd;
-						video_fd[0] = source->video_fd[0];
-						data_fd = source->data_fd;
-						audio_rtcp_fd = source->audio_rtcp_fd;
-						video_rtcp_fd = source->video_rtcp_fd;
 						ka_timeout = ((gint64)source->ka_timeout*G_USEC_PER_SEC)/2;
+						connected = TRUE;
 					}
 				}
 				source->reconnect_timer = janus_get_monotonic_time();
@@ -6317,9 +7044,9 @@ static void *janus_streaming_relay_thread(void *data) {
 				continue;
 			}
 		}
-		if(audio_fd < 0 && video_fd[0] < 0 && video_fd[1] < 0 && video_fd[2] < 0 && data_fd < 0) {
+		if(source->reconnecting || !connected) {
 			/* No socket, we may be in the process of reconnecting, or waiting to reconnect */
-			g_usleep(5000000);
+			g_usleep(1000000);
 			continue;
 		}
 		/* We may also need to occasionally send a OPTIONS request as a keep-alive */
@@ -6344,62 +7071,51 @@ static void *janus_streaming_relay_thread(void *data) {
 			}
 		}
 #endif
-		/* Any PLI and/or REMB we should send back to the source? */
-		if(g_atomic_int_get(&source->need_pli))
-			janus_streaming_rtcp_pli_send(source);
-		if(source->video_rtcp_fd > -1 && source->lowest_bitrate > 0) {
-			gint64 now = janus_get_monotonic_time();
-			if(source->remb_latest == 0)
-				source->remb_latest = now;
-			else if(now - source->remb_latest >= G_USEC_PER_SEC)
-				janus_streaming_rtcp_remb_send(source);
-		}
 		/* Prepare poll */
 		num = 0;
-		if(audio_fd != -1) {
-			fds[num].fd = audio_fd;
-			fds[num].events = POLLIN;
-			fds[num].revents = 0;
-			num++;
-		}
-		if(video_fd[0] != -1) {
-			fds[num].fd = video_fd[0];
-			fds[num].events = POLLIN;
-			fds[num].revents = 0;
-			num++;
-		}
-		if(video_fd[1] != -1) {
-			fds[num].fd = video_fd[1];
-			fds[num].events = POLLIN;
-			fds[num].revents = 0;
-			num++;
-		}
-		if(video_fd[2] != -1) {
-			fds[num].fd = video_fd[2];
-			fds[num].events = POLLIN;
-			fds[num].revents = 0;
-			num++;
-		}
-		if(data_fd != -1) {
-			fds[num].fd = data_fd;
-			fds[num].events = POLLIN;
-			fds[num].revents = 0;
-			num++;
-		}
-		if(pipe_fd != -1) {
-			fds[num].fd = pipe_fd;
-			fds[num].events = POLLIN;
-			fds[num].revents = 0;
-			num++;
-		}
-		if(audio_rtcp_fd != -1) {
-			fds[num].fd = audio_rtcp_fd;
-			fds[num].events = POLLIN;
-			fds[num].revents = 0;
-			num++;
+		temp = source->media;
+		while(temp) {
+			janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+			if(stream->fd[0] != -1) {
+				fds[num].fd = stream->fd[0];
+				fds[num].events = POLLIN;
+				fds[num].revents = 0;
+				num++;
+			}
+			if(stream->fd[1] != -1) {
+				fds[num].fd = stream->fd[1];
+				fds[num].events = POLLIN;
+				fds[num].revents = 0;
+				num++;
+			}
+			if(stream->fd[2] != -1) {
+				fds[num].fd = stream->fd[2];
+				fds[num].events = POLLIN;
+				fds[num].revents = 0;
+				num++;
+			}
+			if(stream->rtcp_fd != -1) {
+				fds[num].fd = stream->rtcp_fd;
+				fds[num].events = POLLIN;
+				fds[num].revents = 0;
+				num++;
+			}
+			/* Any PLI and/or REMB we should send back to the source? */
+			if(stream->type == JANUS_STREAMING_MEDIA_VIDEO) {
+				if(g_atomic_int_get(&stream->need_pli))
+					janus_streaming_rtcp_pli_send(stream);
+				if(stream->rtcp_fd > -1 && source->lowest_bitrate > 0) {
+					gint64 now = janus_get_monotonic_time();
+					if(source->remb_latest == 0)
+						source->remb_latest = now;
+					else if(now - source->remb_latest >= G_USEC_PER_SEC)
+						janus_streaming_rtcp_remb_send(source, stream);
+				}
+			}
+			temp = temp->next;
 		}
-		if(video_rtcp_fd != -1) {
-			fds[num].fd = video_rtcp_fd;
+		if(source->pipefd[0] != -1) {
+			fds[num].fd = source->pipefd[0];
 			fds[num].events = POLLIN;
 			fds[num].revents = 0;
 			num++;
@@ -6414,26 +7130,17 @@ static void *janus_streaming_relay_thread(void *data) {
 			JANUS_LOG(LOG_ERR, "[%s] Error polling... %d (%s)\n", name, errno, strerror(errno));
 			mountpoint->enabled = FALSE;
 			janus_mutex_lock(&source->rec_mutex);
-			if(source->arc) {
-				janus_recorder_close(source->arc);
-				JANUS_LOG(LOG_INFO, "[%s] Closed audio recording %s\n", mountpoint->name, source->arc->filename ? source->arc->filename : "??");
-				janus_recorder *tmp = source->arc;
-				source->arc = NULL;
-				janus_recorder_destroy(tmp);
-			}
-			if(source->vrc) {
-				janus_recorder_close(source->vrc);
-				JANUS_LOG(LOG_INFO, "[%s] Closed video recording %s\n", mountpoint->name, source->vrc->filename ? source->vrc->filename : "??");
-				janus_recorder *tmp = source->vrc;
-				source->vrc = NULL;
-				janus_recorder_destroy(tmp);
-			}
-			if(source->drc) {
-				janus_recorder_close(source->drc);
-				JANUS_LOG(LOG_INFO, "[%s] Closed data recording %s\n", mountpoint->name, source->drc->filename ? source->drc->filename : "??");
-				janus_recorder *tmp = source->drc;
-				source->drc = NULL;
+			GList *temp = source->media;
+			while(temp) {
+				janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+				janus_recorder_close(stream->rc);
+				JANUS_LOG(LOG_INFO, "[%s] Closed %s recording %s (%s)\n",
+					name, janus_streaming_media_str(stream->type),
+					stream->rc->filename ? stream->rc->filename : "??", stream->mid);
+				janus_recorder *tmp = stream->rc;
+				stream->rc = NULL;
 				janus_recorder_destroy(tmp);
+				break;
 			}
 			janus_mutex_unlock(&source->rec_mutex);
 			break;
@@ -6449,38 +7156,39 @@ static void *janus_streaming_relay_thread(void *data) {
 					fds[i].revents & POLLERR ? "POLLERR" : "POLLHUP", errno, strerror(errno));
 				mountpoint->enabled = FALSE;
 				janus_mutex_lock(&source->rec_mutex);
-				if(source->arc) {
-					janus_recorder_close(source->arc);
-					JANUS_LOG(LOG_INFO, "[%s] Closed audio recording %s\n", mountpoint->name, source->arc->filename ? source->arc->filename : "??");
-					janus_recorder *tmp = source->arc;
-					source->arc = NULL;
-					janus_recorder_destroy(tmp);
-				}
-				if(source->vrc) {
-					janus_recorder_close(source->vrc);
-					JANUS_LOG(LOG_INFO, "[%s] Closed video recording %s\n", mountpoint->name, source->vrc->filename ? source->vrc->filename : "??");
-					janus_recorder *tmp = source->vrc;
-					source->vrc = NULL;
-					janus_recorder_destroy(tmp);
-				}
-				if(source->drc) {
-					janus_recorder_close(source->drc);
-					JANUS_LOG(LOG_INFO, "[%s] Closed data recording %s\n", mountpoint->name, source->drc->filename ? source->drc->filename : "??");
-					janus_recorder *tmp = source->drc;
-					source->drc = NULL;
+				GList *temp = source->media;
+				while(temp) {
+					janus_streaming_rtp_source_stream *stream = (janus_streaming_rtp_source_stream *)temp->data;
+					janus_recorder_close(stream->rc);
+					JANUS_LOG(LOG_INFO, "[%s] Closed %s recording %s (%s)\n",
+						name, janus_streaming_media_str(stream->type),
+						stream->rc->filename ? stream->rc->filename : "??", stream->mid);
+					janus_recorder *tmp = stream->rc;
+					stream->rc = NULL;
 					janus_recorder_destroy(tmp);
+					break;
 				}
 				janus_mutex_unlock(&source->rec_mutex);
 				break;
 			} else if(fds[i].revents & POLLIN) {
 				/* Got an RTP or data packet */
-				if(pipe_fd != -1 && fds[i].fd == pipe_fd) {
+				stream = NULL;
+				if(fds[i].fd == source->pipefd[0]) {
 					/* We're done here */
 					int code = 0;
-					bytes = read(pipe_fd, &code, sizeof(int));
+					bytes = read(fds[i].fd, &code, sizeof(int));
 					JANUS_LOG(LOG_VERB, "[%s] Interrupting mountpoint\n", mountpoint->name);
 					break;
-				} else if(audio_fd != -1 && fds[i].fd == audio_fd) {
+				} else {
+					/* Check which stream this file descriptor belongs to */
+					stream = g_hash_table_lookup(source->media_byfd, GINT_TO_POINTER(fds[i].fd));
+				}
+				if(stream == NULL) {
+					/* No stream..? Shouldn't happen, read the bytes and dump them */
+					(void)recvfrom(fds[i].fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
+					continue;
+				}
+				if(stream->type == JANUS_STREAMING_MEDIA_AUDIO && fds[i].fd == stream->fd[0]) {
 					/* Got something audio (RTP) */
 					if(mountpoint->active == FALSE)
 						mountpoint->active = TRUE;
@@ -6489,27 +7197,27 @@ static void *janus_streaming_relay_thread(void *data) {
 					source->reconnect_timer = now;
 #endif
 					addrlen = sizeof(remote);
-					bytes = recvfrom(audio_fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
+					bytes = recvfrom(fds[i].fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
 					if(bytes < 0 || !janus_is_rtp(buffer, bytes)) {
 						/* Failed to read or not an RTP packet? */
 						continue;
 					}
 					janus_rtp_header *rtp = (janus_rtp_header *)buffer;
 					ssrc = ntohl(rtp->ssrc);
-					if(source->rtp_collision > 0 && a_last_ssrc && ssrc != a_last_ssrc &&
-							(now-source->last_received_audio) < (gint64)1000*source->rtp_collision) {
+					if(source->rtp_collision > 0 && stream->last_ssrc[0] && ssrc != stream->last_ssrc[0] &&
+							(now-stream->last_received) < (gint64)1000*source->rtp_collision) {
 						JANUS_LOG(LOG_WARN, "[%s] RTP collision on audio mountpoint, dropping packet (ssrc=%"SCNu32")\n", name, ssrc);
 						continue;
 					}
-					source->last_received_audio = now;
+					stream->last_received = now;
 					//~ JANUS_LOG(LOG_VERB, "************************\nGot %d bytes on the audio channel...\n", bytes);
 					/* Do we have a new stream? */
-					if(ssrc != a_last_ssrc) {
-						source->audio_ssrc = a_last_ssrc = ssrc;
-						JANUS_LOG(LOG_INFO, "[%s] New audio stream! (ssrc=%"SCNu32")\n", name, a_last_ssrc);
+					if(ssrc != stream->last_ssrc[0]) {
+						stream->ssrc = stream->last_ssrc[0] = ssrc;
+						JANUS_LOG(LOG_INFO, "[%s] New audio stream! (#%d, ssrc=%"SCNu32")\n", name, stream->mindex, stream->last_ssrc[0]);
 					}
 					/* If paused, ignore this packet */
-					if(!mountpoint->enabled && !source->arc)
+					if(!mountpoint->enabled && !stream->rc)
 						continue;
 					/* Is this SRTP? */
 					if(source->is_srtp) {
@@ -6528,53 +7236,50 @@ static void *janus_streaming_relay_thread(void *data) {
 					//~ JANUS_LOG(LOG_VERB, " ... parsed RTP packet (ssrc=%u, pt=%u, seq=%u, ts=%u)...\n",
 						//~ ntohl(rtp->ssrc), rtp->type, ntohs(rtp->seq_number), ntohl(rtp->timestamp));
 					/* Relay on all sessions */
+					packet.mindex = stream->mindex;
 					packet.data = rtp;
 					packet.length = bytes;
 					packet.is_rtp = TRUE;
 					packet.is_video = FALSE;
 					packet.is_keyframe = FALSE;
-					packet.data->type = mountpoint->codecs.audio_pt;
+					packet.data->type = stream->codecs.pt;
 					/* Is there a recorder? */
-					janus_rtp_header_update(packet.data, &source->context[0], FALSE, 0);
-					if(source->askew) {
-						int ret = janus_rtp_skew_compensate_audio(packet.data, &source->context[0], now);
+					janus_rtp_header_update(packet.data, &stream->context[0], FALSE);
+					if(stream->skew) {
+						int ret = janus_rtp_skew_compensate_audio(packet.data, &stream->context[0], now);
 						if(ret < 0) {
 							JANUS_LOG(LOG_WARN, "[%s] Dropping %d packets, audio source clock is too fast (ssrc=%"SCNu32")\n",
-								name, -ret, a_last_ssrc);
+								name, -ret, stream->last_ssrc[0]);
 							continue;
 						} else if(ret > 0) {
 							JANUS_LOG(LOG_WARN, "[%s] Jumping %d RTP sequence numbers, audio source clock is too slow (ssrc=%"SCNu32")\n",
-								name, ret, a_last_ssrc);
+								name, ret, stream->last_ssrc[0]);
 						}
 					}
-					if(source->arc) {
-						packet.data->ssrc = htonl((uint32_t)mountpoint->id);
-						janus_recorder_save_frame(source->arc, buffer, bytes);
-					}
-					if(mountpoint->enabled) {
-						packet.data->ssrc = htonl(ssrc);
-						/* Backup the actual timestamp and sequence number set by the restreamer, in case switching is involved */
-						packet.timestamp = ntohl(packet.data->timestamp);
-						packet.seq_number = ntohs(packet.data->seq_number);
-						/* Go! */
-
-						janus_mutex_lock(&mountpoint->mutex);
-						g_list_foreach(mountpoint->helper_threads == 0 ? mountpoint->viewers : mountpoint->threads,
-							mountpoint->helper_threads == 0 ? janus_streaming_relay_rtp_packet : janus_streaming_helper_rtprtcp_packet,
-							&packet);
-						janus_mutex_unlock(&mountpoint->mutex);
-					}
+					packet.data->ssrc = htonl((uint32_t)mountpoint->id);
+					janus_recorder_save_frame(stream->rc, buffer, bytes);
+					if(!mountpoint->enabled)
+						continue;
+					packet.data->ssrc = htonl(ssrc);
+					/* Backup the actual timestamp and sequence number set by the restreamer, in case switching is involved */
+					packet.timestamp = ntohl(packet.data->timestamp);
+					packet.seq_number = ntohs(packet.data->seq_number);
+					/* Go! */
+					janus_mutex_lock(&mountpoint->mutex);
+					g_list_foreach(mountpoint->helper_threads == 0 ? mountpoint->viewers : mountpoint->threads,
+						mountpoint->helper_threads == 0 ? janus_streaming_relay_rtp_packet : janus_streaming_helper_rtprtcp_packet,
+						&packet);
+					janus_mutex_unlock(&mountpoint->mutex);
 					continue;
-				} else if((video_fd[0] != -1 && fds[i].fd == video_fd[0]) ||
-						(video_fd[1] != -1 && fds[i].fd == video_fd[1]) ||
-						(video_fd[2] != -1 && fds[i].fd == video_fd[2])) {
+				} else if(stream->type == JANUS_STREAMING_MEDIA_VIDEO && ((fds[i].fd == stream->fd[0]) ||
+						(fds[i].fd == stream->fd[1]) || (fds[i].fd == stream->fd[2]))) {
 					/* Got something video (RTP) */
 					int index = -1;
-					if(fds[i].fd == video_fd[0])
+					if(fds[i].fd == stream->fd[0])
 						index = 0;
-					else if(fds[i].fd == video_fd[1])
+					else if(fds[i].fd == stream->fd[1])
 						index = 1;
-					else if(fds[i].fd == video_fd[2])
+					else if(fds[i].fd == stream->fd[2])
 						index = 2;
 					if(mountpoint->active == FALSE)
 						mountpoint->active = TRUE;
@@ -6590,21 +7295,21 @@ static void *janus_streaming_relay_thread(void *data) {
 					}
 					janus_rtp_header *rtp = (janus_rtp_header *)buffer;
 					ssrc = ntohl(rtp->ssrc);
-					if(source->rtp_collision > 0 && v_last_ssrc[index] && ssrc != v_last_ssrc[index] &&
-							(now-source->last_received_video) < (gint64)1000*source->rtp_collision) {
+					if(source->rtp_collision > 0 && stream->last_ssrc[index] && ssrc != stream->last_ssrc[index] &&
+							(now-stream->last_received) < (gint64)1000*source->rtp_collision) {
 						JANUS_LOG(LOG_WARN, "[%s] RTP collision on video mountpoint, dropping packet (ssrc=%"SCNu32")\n",
 							name, ssrc);
 						continue;
 					}
-					source->last_received_video = now;
+					stream->last_received = now;
 					//~ JANUS_LOG(LOG_VERB, "************************\nGot %d bytes on the video channel...\n", bytes);
 					/* Do we have a new stream? */
-					if(ssrc != v_last_ssrc[index]) {
-						v_last_ssrc[index] = ssrc;
+					if(ssrc != stream->last_ssrc[index]) {
+						stream->last_ssrc[index] = ssrc;
 						if(index == 0)
-							source->video_ssrc = ssrc;
-						JANUS_LOG(LOG_INFO, "[%s] New video stream! (ssrc=%"SCNu32", index %d)\n",
-							name, v_last_ssrc[index], index);
+							stream->ssrc = ssrc;
+						JANUS_LOG(LOG_INFO, "[%s] New video stream! (#%d, ssrc=%"SCNu32", index %d)\n",
+							name, stream->mindex, stream->last_ssrc[index], index);
 					}
 					/* Is this SRTP? */
 					if(source->is_srtp) {
@@ -6621,35 +7326,37 @@ static void *janus_streaming_relay_thread(void *data) {
 						bytes = buflen;
 					}
 					/* First of all, let's check if this is (part of) a keyframe that we may need to save it for future reference */
-					if(source->keyframe.enabled) {
-						if(source->keyframe.temp_ts > 0 && ntohl(rtp->timestamp) != source->keyframe.temp_ts) {
+					if(stream->keyframe.enabled) {
+						if(stream->keyframe.temp_ts > 0 && ntohl(rtp->timestamp) != stream->keyframe.temp_ts) {
 							/* We received the last part of the keyframe, get rid of the old one and use this from now on */
 							JANUS_LOG(LOG_HUGE, "[%s] ... ... last part of keyframe received! ts=%"SCNu32", %d packets\n",
-								name, source->keyframe.temp_ts, g_list_length(source->keyframe.temp_keyframe));
-							source->keyframe.temp_ts = 0;
-							janus_mutex_lock(&source->keyframe.mutex);
-							if(source->keyframe.latest_keyframe != NULL)
-								g_list_free_full(source->keyframe.latest_keyframe, (GDestroyNotify)janus_streaming_rtp_relay_packet_free);
-							source->keyframe.latest_keyframe = source->keyframe.temp_keyframe;
-							source->keyframe.temp_keyframe = NULL;
-							janus_mutex_unlock(&source->keyframe.mutex);
-						} else if(ntohl(rtp->timestamp) == source->keyframe.temp_ts) {
+								name, stream->keyframe.temp_ts, g_list_length(stream->keyframe.temp_keyframe));
+							stream->keyframe.temp_ts = 0;
+							janus_mutex_lock(&stream->keyframe.mutex);
+							if(stream->keyframe.latest_keyframe != NULL)
+								g_list_free_full(stream->keyframe.latest_keyframe, (GDestroyNotify)janus_streaming_rtp_relay_packet_free);
+							stream->keyframe.latest_keyframe = stream->keyframe.temp_keyframe;
+							stream->keyframe.temp_keyframe = NULL;
+							janus_mutex_unlock(&stream->keyframe.mutex);
+						} else if(ntohl(rtp->timestamp) == stream->keyframe.temp_ts) {
 							/* Part of the keyframe we're currently saving, store */
-							janus_mutex_lock(&source->keyframe.mutex);
-							JANUS_LOG(LOG_HUGE, "[%s] ... other part of keyframe received! ts=%"SCNu32"\n", name, source->keyframe.temp_ts);
+							janus_mutex_lock(&stream->keyframe.mutex);
+							JANUS_LOG(LOG_HUGE, "[%s] ... other part of keyframe received! ts=%"SCNu32"\n", name, stream->keyframe.temp_ts);
 							janus_streaming_rtp_relay_packet *pkt = g_malloc0(sizeof(janus_streaming_rtp_relay_packet));
+							pkt->mindex = stream->mindex;
 							pkt->data = g_malloc(bytes);
 							memcpy(pkt->data, buffer, bytes);
 							pkt->data->ssrc = htons(1);
-							pkt->data->type = mountpoint->codecs.video_pt;
-							packet.is_rtp = TRUE;
-							packet.is_video = TRUE;
-							packet.is_keyframe = TRUE;
+							pkt->data->type = stream->codecs.pt;
+							pkt->is_rtp = TRUE;
+							pkt->is_video = TRUE;
+							pkt->is_keyframe = TRUE;
+							pkt->codec = stream->codecs.video_codec;
 							pkt->length = bytes;
-							pkt->timestamp = source->keyframe.temp_ts;
+							pkt->timestamp = stream->keyframe.temp_ts;
 							pkt->seq_number = ntohs(rtp->seq_number);
-							source->keyframe.temp_keyframe = g_list_append(source->keyframe.temp_keyframe, pkt);
-							janus_mutex_unlock(&source->keyframe.mutex);
+							stream->keyframe.temp_keyframe = g_list_append(stream->keyframe.temp_keyframe, pkt);
+							janus_mutex_unlock(&stream->keyframe.mutex);
 						} else {
 							gboolean kf = FALSE;
 							/* Parse RTP header first */
@@ -6661,7 +7368,7 @@ static void *janus_streaming_relay_thread(void *data) {
 							int plen = 0;
 							char *payload = janus_rtp_payload(buffer, bytes, &plen);
 							if(payload) {
-								switch(mountpoint->codecs.video_codec) {
+								switch(stream->codecs.video_codec) {
 									case JANUS_VIDEOCODEC_VP8:
 										kf = janus_vp8_is_keyframe(payload, plen);
 										break;
@@ -6676,42 +7383,45 @@ static void *janus_streaming_relay_thread(void *data) {
 								}
 								if(kf) {
 									/* New keyframe, start saving it */
-									source->keyframe.temp_ts = ntohl(rtp->timestamp);
-									JANUS_LOG(LOG_HUGE, "[%s] New keyframe received! ts=%"SCNu32"\n", name, source->keyframe.temp_ts);
-									janus_mutex_lock(&source->keyframe.mutex);
+									stream->keyframe.temp_ts = ntohl(rtp->timestamp);
+									JANUS_LOG(LOG_HUGE, "[%s] New keyframe received! ts=%"SCNu32"\n", name, stream->keyframe.temp_ts);
+									janus_mutex_lock(&stream->keyframe.mutex);
 									janus_streaming_rtp_relay_packet *pkt = g_malloc0(sizeof(janus_streaming_rtp_relay_packet));
+									pkt->mindex = stream->mindex;
 									pkt->data = g_malloc(bytes);
 									memcpy(pkt->data, buffer, bytes);
 									pkt->data->ssrc = htons(1);
-									pkt->data->type = mountpoint->codecs.video_pt;
-									packet.is_rtp = TRUE;
-									packet.is_video = TRUE;
-									packet.is_keyframe = TRUE;
+									pkt->data->type = stream->codecs.pt;
+									pkt->is_rtp = TRUE;
+									pkt->is_video = TRUE;
+									pkt->is_keyframe = TRUE;
+									pkt->codec = stream->codecs.video_codec;
 									pkt->length = bytes;
-									pkt->timestamp = source->keyframe.temp_ts;
+									pkt->timestamp = stream->keyframe.temp_ts;
 									pkt->seq_number = ntohs(rtp->seq_number);
-									source->keyframe.temp_keyframe = g_list_append(source->keyframe.temp_keyframe, pkt);
-									janus_mutex_unlock(&source->keyframe.mutex);
+									stream->keyframe.temp_keyframe = g_list_append(stream->keyframe.temp_keyframe, pkt);
+									janus_mutex_unlock(&stream->keyframe.mutex);
 								}
 							}
 						}
 					}
 					/* If paused, ignore this packet */
-					if(!mountpoint->enabled && !source->vrc)
+					if(!mountpoint->enabled && !stream->rc)
 						continue;
 					//~ JANUS_LOG(LOG_VERB, " ... parsed RTP packet (ssrc=%u, pt=%u, seq=%u, ts=%u)...\n",
 						//~ ntohl(rtp->ssrc), rtp->type, ntohs(rtp->seq_number), ntohl(rtp->timestamp));
 					/* Relay on all sessions */
+					packet.mindex = stream->mindex;
 					packet.data = rtp;
 					packet.length = bytes;
 					packet.is_rtp = TRUE;
 					packet.is_video = TRUE;
 					packet.is_keyframe = FALSE;
-					packet.simulcast = source->simulcast;
+					packet.simulcast = stream->simulcast;
 					packet.substream = index;
-					packet.codec = mountpoint->codecs.video_codec;
+					packet.codec = stream->codecs.video_codec;
 					packet.svc = FALSE;
-					if(source->svc) {
+					if(stream->svc) {
 						/* We're doing SVC: let's parse this packet to see which layers are there */
 						int plen = 0;
 						char *payload = janus_rtp_payload(buffer, bytes, &plen);
@@ -6732,108 +7442,110 @@ static void *janus_streaming_relay_thread(void *data) {
 							}
 						}
 					}
-					packet.data->type = mountpoint->codecs.video_pt;
+					packet.data->type = stream->codecs.pt;
 					/* Is there a recorder? (FIXME notice we only record the first substream, if simulcasting) */
-					janus_rtp_header_update(packet.data, &source->context[index], TRUE, 0);
-					if(source->vskew) {
-						int ret = janus_rtp_skew_compensate_video(packet.data, &source->context[index], now);
+					janus_rtp_header_update(packet.data, &stream->context[index], TRUE);
+					if(stream->skew) {
+						int ret = janus_rtp_skew_compensate_video(packet.data, &stream->context[index], now);
 						if(ret < 0) {
 							JANUS_LOG(LOG_WARN, "[%s] Dropping %d packets, video source clock is too fast (ssrc=%"SCNu32", index %d)\n",
-								name, -ret, v_last_ssrc[index], index);
+								name, -ret, stream->last_ssrc[index], index);
 							continue;
 						} else if(ret > 0) {
 							JANUS_LOG(LOG_WARN, "[%s] Jumping %d RTP sequence numbers, video source clock is too slow (ssrc=%"SCNu32", index %d)\n",
-								name, ret, v_last_ssrc[index], index);
+								name, ret, stream->last_ssrc[index], index);
 						}
 					}
-					if(index == 0 && source->vrc) {
+					if(index == 0) {
 						packet.data->ssrc = htonl((uint32_t)mountpoint->id);
-						janus_recorder_save_frame(source->vrc, buffer, bytes);
-					}
-					if (mountpoint->enabled) {
+						janus_recorder_save_frame(stream->rc, buffer, bytes);
 						packet.data->ssrc = htonl(ssrc);
-						/* Backup the actual timestamp and sequence number set by the restreamer, in case switching is involved */
-						packet.timestamp = ntohl(packet.data->timestamp);
-						packet.seq_number = ntohs(packet.data->seq_number);
-						/* Take note of the simulcast SSRCs */
-						if(source->simulcast) {
-							packet.ssrc[0] = v_last_ssrc[0];
-							packet.ssrc[1] = v_last_ssrc[1];
-							packet.ssrc[2] = v_last_ssrc[2];
-						}
-						/* Go! */
-						janus_mutex_lock(&mountpoint->mutex);
-						g_list_foreach(mountpoint->helper_threads == 0 ? mountpoint->viewers : mountpoint->threads,
-							mountpoint->helper_threads == 0 ? janus_streaming_relay_rtp_packet : janus_streaming_helper_rtprtcp_packet,
-							&packet);
-						janus_mutex_unlock(&mountpoint->mutex);
 					}
+					if(!mountpoint->enabled)
+						continue;
+					/* Backup the actual timestamp and sequence number set by the restreamer, in case switching is involved */
+					packet.timestamp = ntohl(packet.data->timestamp);
+					packet.seq_number = ntohs(packet.data->seq_number);
+					/* Take note of the simulcast SSRCs */
+					if(stream->simulcast) {
+						packet.ssrc[0] = stream->last_ssrc[0];
+						packet.ssrc[1] = stream->last_ssrc[1];
+						packet.ssrc[2] = stream->last_ssrc[2];
+					}
+					/* Go! */
+					janus_mutex_lock(&mountpoint->mutex);
+					g_list_foreach(mountpoint->helper_threads == 0 ? mountpoint->viewers : mountpoint->threads,
+						mountpoint->helper_threads == 0 ? janus_streaming_relay_rtp_packet : janus_streaming_helper_rtprtcp_packet,
+						&packet);
+					janus_mutex_unlock(&mountpoint->mutex);
 					continue;
-				} else if(data_fd != -1 && fds[i].fd == data_fd) {
+				} else if(stream->type == JANUS_STREAMING_MEDIA_DATA && fds[i].fd == stream->fd[0]) {
 					/* Got something data (text) */
 					if(mountpoint->active == FALSE)
 						mountpoint->active = TRUE;
-					source->last_received_data = janus_get_monotonic_time();
+					stream->last_received = janus_get_monotonic_time();
 #ifdef HAVE_LIBCURL
 					source->reconnect_timer = janus_get_monotonic_time();
 #endif
 					addrlen = sizeof(remote);
-					bytes = recvfrom(data_fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
+					bytes = recvfrom(fds[i].fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
 					if(bytes < 1) {
 						/* Failed to read? */
 						continue;
 					}
-					if(!mountpoint->enabled && !source->drc)
+					if(!mountpoint->enabled && !stream->rc)
 						continue;
 					/* Get a string out of the data */
 					char *text = g_malloc(bytes+1);
 					memcpy(text, buffer, bytes);
 					*(text+bytes) = '\0';
 					/* Relay on all sessions */
+					packet.mindex = stream->mindex;
 					packet.data = (janus_rtp_header *)text;
 					packet.length = bytes+1;
 					packet.is_rtp = FALSE;
 					/* Is there a recorder? */
-					janus_recorder_save_frame(source->drc, text, strlen(text));
-					if(mountpoint->enabled) {
-						/* Are we keeping track of the last message being relayed? */
-						if(source->buffermsg) {
-							janus_mutex_lock(&source->buffermsg_mutex);
-							janus_streaming_rtp_relay_packet *pkt = g_malloc0(sizeof(janus_streaming_rtp_relay_packet));
-							pkt->data = g_malloc(bytes+1);
-							memcpy(pkt->data, text, bytes+1);
-							packet.is_rtp = FALSE;
-							pkt->length = bytes+1;
-							janus_mutex_unlock(&source->buffermsg_mutex);
-						}
-						/* Go! */
-						janus_mutex_lock(&mountpoint->mutex);
-						g_list_foreach(mountpoint->helper_threads == 0 ? mountpoint->viewers : mountpoint->threads,
-							mountpoint->helper_threads == 0 ? janus_streaming_relay_rtp_packet : janus_streaming_helper_rtprtcp_packet,
-							&packet);
-						janus_mutex_unlock(&mountpoint->mutex);
+					janus_recorder_save_frame(stream->rc, text, strlen(text));
+					if(!mountpoint->enabled)
+						continue;
+					/* Are we keeping track of the last message being relayed? */
+					if(stream->buffermsg) {
+						janus_mutex_lock(&stream->buffermsg_mutex);
+						janus_streaming_rtp_relay_packet *pkt = g_malloc0(sizeof(janus_streaming_rtp_relay_packet));
+						pkt->data = g_malloc(bytes+1);
+						memcpy(pkt->data, text, bytes+1);
+						packet.is_rtp = FALSE;
+						pkt->length = bytes+1;
+						janus_mutex_unlock(&stream->buffermsg_mutex);
 					}
+					/* Go! */
+					janus_mutex_lock(&mountpoint->mutex);
+					g_list_foreach(mountpoint->helper_threads == 0 ? mountpoint->viewers : mountpoint->threads,
+						mountpoint->helper_threads == 0 ? janus_streaming_relay_rtp_packet : janus_streaming_helper_rtprtcp_packet,
+						&packet);
+					janus_mutex_unlock(&mountpoint->mutex);
 					packet.data = NULL;
 					g_free(text);
 					continue;
-				} else if(audio_rtcp_fd != -1 && fds[i].fd == audio_rtcp_fd) {
+				} else if(stream->type == JANUS_STREAMING_MEDIA_AUDIO && fds[i].fd == stream->rtcp_fd) {
 					addrlen = sizeof(remote);
-					bytes = recvfrom(audio_rtcp_fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
+					bytes = recvfrom(fds[i].fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
 					if(bytes < 0 || (!janus_is_rtp(buffer, bytes) && !janus_is_rtcp(buffer, bytes))) {
 						/* For latching we need an RTP or RTCP packet */
 						continue;
 					}
 					if(!mountpoint->enabled)
 						continue;
-					memcpy(&source->audio_rtcp_addr, &remote, addrlen);
+					memcpy(&stream->rtcp_addr, &remote, addrlen);
 					if(!janus_is_rtcp(buffer, bytes)) {
 						/* Failed to read or not an RTCP packet? */
 						continue;
 					}
-					JANUS_LOG(LOG_HUGE, "[%s] Got audio RTCP feedback: SSRC %"SCNu32"\n",
-						name, janus_rtcp_get_sender_ssrc(buffer, bytes));
+					JANUS_LOG(LOG_HUGE, "[%s] Got audio RTCP feedback: #%d, SSRC %"SCNu32"\n",
+						name, stream->mindex, janus_rtcp_get_sender_ssrc(buffer, bytes));
 					/* Relay on all sessions */
 					packet.is_video = FALSE;
+					packet.mindex = stream->mindex;
 					packet.data = (janus_rtp_header *)buffer;
 					packet.length = bytes;
 					/* Go! */
@@ -6842,24 +7554,25 @@ static void *janus_streaming_relay_thread(void *data) {
 						mountpoint->helper_threads == 0 ? janus_streaming_relay_rtcp_packet : janus_streaming_helper_rtprtcp_packet,
 						&packet);
 					janus_mutex_unlock(&mountpoint->mutex);
-				} else if(video_rtcp_fd != -1 && fds[i].fd == video_rtcp_fd) {
+				} else if(stream->type == JANUS_STREAMING_MEDIA_VIDEO && fds[i].fd == stream->rtcp_fd) {
 					addrlen = sizeof(remote);
-					bytes = recvfrom(video_rtcp_fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
+					bytes = recvfrom(fds[i].fd, buffer, 1500, 0, (struct sockaddr *)&remote, &addrlen);
 					if(bytes < 0 || (!janus_is_rtp(buffer, bytes) && !janus_is_rtcp(buffer, bytes))) {
 						/* For latching we need an RTP or RTCP packet */
 						continue;
 					}
 					if(!mountpoint->enabled)
 						continue;
-					memcpy(&source->video_rtcp_addr, &remote, addrlen);
+					memcpy(&stream->rtcp_addr, &remote, addrlen);
 					if(!janus_is_rtcp(buffer, bytes)) {
 						/* Failed to read or not an RTCP packet? */
 						continue;
 					}
-					JANUS_LOG(LOG_HUGE, "[%s] Got video RTCP feedback: SSRC %"SCNu32"\n",
-						name, janus_rtcp_get_sender_ssrc(buffer, bytes));
+					JANUS_LOG(LOG_HUGE, "[%s] Got video RTCP feedback: #%d, SSRC %"SCNu32"\n",
+						name, stream->mindex, janus_rtcp_get_sender_ssrc(buffer, bytes));
 					/* Relay on all sessions */
 					packet.is_video = TRUE;
+					packet.mindex = stream->mindex;
 					packet.data = (janus_rtp_header *)buffer;
 					packet.length = bytes;
 					/* Go! */
@@ -6932,48 +7645,57 @@ static void janus_streaming_relay_rtp_packet(gpointer data, gpointer user_data)
 		//~ JANUS_LOG(LOG_ERR, "Streaming not started yet for this session...\n");
 		return;
 	}
+	janus_streaming_session_stream *s = g_hash_table_lookup(session->streams_byid, GINT_TO_POINTER(packet->mindex));
+	if(s == NULL) {
+		/* No session stream for this mindex: maybe the viewer did not subscribe to it */
+		return;
+	}
+	/* Make sure we're allowed to send packets from this stream */
+	if(!s->send)
+		return;
+	janus_streaming_rtp_source_stream *stream = s->stream;
 
 	if(packet->is_rtp) {
 		/* Make sure there hasn't been a video source switch by checking the SSRC */
 		if(packet->is_video) {
-			if(!session->video)
-				return;
 			/* Check if there's any SVC info to take into account */
 			if(packet->svc) {
 				/* There is: check if this is a layer that can be dropped for this viewer
 				 * Note: Following core inspired by the excellent job done by Sergio Garcia Murillo here:
 				 * https://github.com/medooze/media-server/blob/master/src/vp9/VP9LayerSelector.cpp */
 				gboolean override_mark_bit = FALSE, has_marker_bit = packet->data->markerbit;
-				int temporal_layer = session->temporal_layer;
-				if(session->target_temporal_layer > session->temporal_layer) {
+				int temporal_layer = s->temporal_layer;
+				if(s->target_temporal_layer > s->temporal_layer) {
 					/* We need to upscale */
 					JANUS_LOG(LOG_HUGE, "We need to upscale temporally:\n");
-					if(packet->ubit && packet->bbit && packet->temporal_layer <= session->target_temporal_layer) {
+					if(packet->ubit && packet->bbit && packet->temporal_layer <= s->target_temporal_layer) {
 						JANUS_LOG(LOG_HUGE, "  -- Upscaling temporal layer: %u --> %u\n",
-							packet->temporal_layer, session->target_temporal_layer);
-						session->temporal_layer = packet->temporal_layer;
-						temporal_layer = session->temporal_layer;
+							packet->temporal_layer, s->target_temporal_layer);
+						s->temporal_layer = packet->temporal_layer;
+						temporal_layer = s->temporal_layer;
 						/* Notify the viewer */
 						json_t *event = json_object();
 						json_object_set_new(event, "streaming", json_string("event"));
 						json_t *result = json_object();
-						json_object_set_new(result, "temporal_layer", json_integer(session->temporal_layer));
+						json_object_set_new(result, "mid", json_string(stream->mid));
+						json_object_set_new(result, "temporal_layer", json_integer(s->temporal_layer));
 						json_object_set_new(event, "result", result);
 						gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
 						json_decref(event);
 					}
-				} else if(session->target_temporal_layer < session->temporal_layer) {
+				} else if(s->target_temporal_layer < s->temporal_layer) {
 					/* We need to downscale */
 					JANUS_LOG(LOG_HUGE, "We need to downscale temporally:\n");
 					if(packet->ebit) {
 						JANUS_LOG(LOG_HUGE, "  -- Downscaling temporal layer: %u --> %u\n",
-							session->temporal_layer, session->target_temporal_layer);
-						session->temporal_layer = session->target_temporal_layer;
+							s->temporal_layer, s->target_temporal_layer);
+						s->temporal_layer = s->target_temporal_layer;
 						/* Notify the viewer */
 						json_t *event = json_object();
 						json_object_set_new(event, "streaming", json_string("event"));
 						json_t *result = json_object();
-						json_object_set_new(result, "temporal_layer", json_integer(session->temporal_layer));
+						json_object_set_new(result, "mid", json_string(stream->mid));
+						json_object_set_new(result, "temporal_layer", json_integer(s->temporal_layer));
 						json_object_set_new(event, "result", result);
 						gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
 						json_decref(event);
@@ -6982,39 +7704,41 @@ static void janus_streaming_relay_rtp_packet(gpointer data, gpointer user_data)
 				if(temporal_layer < packet->temporal_layer) {
 					/* Drop the packet: update the context to make sure sequence number is increased normally later */
 					JANUS_LOG(LOG_HUGE, "Dropping packet (temporal layer %d < %d)\n", temporal_layer, packet->temporal_layer);
-					session->context.v_base_seq++;
+					s->context.base_seq++;
 					return;
 				}
-				int spatial_layer = session->spatial_layer;
-				if(session->target_spatial_layer > session->spatial_layer) {
+				int spatial_layer = s->spatial_layer;
+				if(s->target_spatial_layer > s->spatial_layer) {
 					JANUS_LOG(LOG_HUGE, "We need to upscale spatially:\n");
 					/* We need to upscale */
-					if(packet->pbit == 0 && packet->bbit && packet->spatial_layer == session->spatial_layer+1) {
+					if(packet->pbit == 0 && packet->bbit && packet->spatial_layer == s->spatial_layer+1) {
 						JANUS_LOG(LOG_HUGE, "  -- Upscaling spatial layer: %u --> %u\n",
-							packet->spatial_layer, session->target_spatial_layer);
-						session->spatial_layer = packet->spatial_layer;
-						spatial_layer = session->spatial_layer;
+							packet->spatial_layer, s->target_spatial_layer);
+						s->spatial_layer = packet->spatial_layer;
+						spatial_layer = s->spatial_layer;
 						/* Notify the viewer */
 						json_t *event = json_object();
 						json_object_set_new(event, "streaming", json_string("event"));
 						json_t *result = json_object();
-						json_object_set_new(result, "spatial_layer", json_integer(session->spatial_layer));
+						json_object_set_new(result, "mid", json_string(stream->mid));
+						json_object_set_new(result, "spatial_layer", json_integer(s->spatial_layer));
 						json_object_set_new(event, "result", result);
 						gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
 						json_decref(event);
 					}
-				} else if(session->target_spatial_layer < session->spatial_layer) {
+				} else if(s->target_spatial_layer < s->spatial_layer) {
 					/* We need to downscale */
 					JANUS_LOG(LOG_HUGE, "We need to downscale spatially:\n");
 					if(packet->ebit) {
 						JANUS_LOG(LOG_HUGE, "  -- Downscaling spatial layer: %u --> %u\n",
-							session->spatial_layer, session->target_spatial_layer);
-						session->spatial_layer = session->target_spatial_layer;
+							s->spatial_layer, s->target_spatial_layer);
+						s->spatial_layer = s->target_spatial_layer;
 						/* Notify the viewer */
 						json_t *event = json_object();
 						json_object_set_new(event, "streaming", json_string("event"));
 						json_t *result = json_object();
-						json_object_set_new(result, "spatial_layer", json_integer(session->spatial_layer));
+						json_object_set_new(result, "mid", json_string(stream->mid));
+						json_object_set_new(result, "spatial_layer", json_integer(s->spatial_layer));
 						json_object_set_new(event, "result", result);
 						gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
 						json_decref(event);
@@ -7023,7 +7747,7 @@ static void janus_streaming_relay_rtp_packet(gpointer data, gpointer user_data)
 				if(spatial_layer < packet->spatial_layer) {
 					/* Drop the packet: update the context to make sure sequence number is increased normally later */
 					JANUS_LOG(LOG_HUGE, "Dropping packet (spatial layer %d < %d)\n", spatial_layer, packet->spatial_layer);
-					session->context.v_base_seq++;
+					s->context.base_seq++;
 					return;
 				} else if(packet->ebit && spatial_layer == packet->spatial_layer) {
 					/* If we stop at layer 0, we need a marker bit now, as the one from layer 1 will not be received */
@@ -7034,12 +7758,14 @@ static void janus_streaming_relay_rtp_packet(gpointer data, gpointer user_data)
 				JANUS_LOG(LOG_HUGE, "Sending packet (spatial=%d, temporal=%d)\n",
 					packet->spatial_layer, packet->temporal_layer);
 				/* Fix sequence number and timestamp (video source switching may be involved) */
-				janus_rtp_header_update(packet->data, &session->context, TRUE, 4500);
+				janus_rtp_header_update(packet->data, &s->context, TRUE);
 				if(override_mark_bit && !has_marker_bit) {
 					packet->data->markerbit = 1;
 				}
-				if(gateway != NULL)
-					gateway->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+				if(gateway != NULL) {
+					gateway->relay_rtp(session->handle, s->mindex,
+						packet->is_video, (char *)packet->data, packet->length);
+				}
 				if(override_mark_bit && !has_marker_bit) {
 					packet->data->markerbit = 0;
 				}
@@ -7053,54 +7779,52 @@ static void janus_streaming_relay_rtp_packet(gpointer data, gpointer user_data)
 				if(payload == NULL)
 					return;
 				/* Process this packet: don't relay if it's not the SSRC/layer we wanted to handle */
-				gboolean relay = janus_rtp_simulcasting_context_process_rtp(&session->sim_context,
-					(char *)packet->data, packet->length, packet->ssrc, NULL, packet->codec, &session->context);
+				gboolean relay = janus_rtp_simulcasting_context_process_rtp(&s->sim_context,
+					(char *)packet->data, packet->length, packet->ssrc, NULL, packet->codec, &s->context);
 				/* Do we need to drop this? */
 				if(!relay)
 					return;
 				/* Any event we should notify? */
-				if(session->sim_context.changed_substream) {
+				if(s->sim_context.changed_substream) {
 					/* Notify the user about the substream change */
 					json_t *event = json_object();
 					json_object_set_new(event, "streaming", json_string("event"));
 					json_t *result = json_object();
-					json_object_set_new(result, "substream", json_integer(session->sim_context.substream));
+					json_object_set_new(result, "substream", json_integer(s->sim_context.substream));
 					json_object_set_new(event, "result", result);
 					gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
 					json_decref(event);
 				}
-				if(session->sim_context.need_pli) {
+				if(s->sim_context.need_pli) {
 					/* Schedule a PLI */
 					JANUS_LOG(LOG_VERB, "We need a PLI for the simulcast context\n");
-					if(session->mountpoint != NULL) {
-						janus_streaming_rtp_source *source = session->mountpoint->source;
-						if(source != NULL)
-							g_atomic_int_set(&source->need_pli, 1);
-					}
+					g_atomic_int_set(&stream->need_pli, 1);
 				}
-				if(session->sim_context.changed_temporal) {
+				if(s->sim_context.changed_temporal) {
 					/* Notify the user about the temporal layer change */
 					json_t *event = json_object();
 					json_object_set_new(event, "streaming", json_string("event"));
 					json_t *result = json_object();
-					json_object_set_new(result, "temporal", json_integer(session->sim_context.templayer));
+					json_object_set_new(result, "temporal", json_integer(s->sim_context.templayer));
 					json_object_set_new(event, "result", result);
 					gateway->push_event(session->handle, &janus_streaming_plugin, NULL, event, NULL);
 					json_decref(event);
 				}
 				/* If we got here, update the RTP header and send the packet */
-				janus_rtp_header_update(packet->data, &session->context, TRUE, 4500);
+				janus_rtp_header_update(packet->data, &s->context, TRUE);
 				char vp8pd[6];
 				if(packet->codec == JANUS_VIDEOCODEC_VP8) {
 					/* For VP8, we save the original payload descriptor, to restore it after */
 					memcpy(vp8pd, payload, sizeof(vp8pd));
-					janus_vp8_simulcast_descriptor_update(payload, plen, &session->vp8_context,
-						session->sim_context.changed_substream);
+					janus_vp8_simulcast_descriptor_update(payload, plen, &s->vp8_context,
+						s->sim_context.changed_substream);
 				}
 				/* Send the packet */
-				if(gateway != NULL)
-					gateway->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
-				/* Restore the timestamp and sequence number to what the publisher set them to */
+				if(gateway != NULL) {
+					gateway->relay_rtp(session->handle, s->mindex,
+						packet->is_video, (char *)packet->data, packet->length);
+				}
+				/* Restore the timestamp and sequence number to what the video source set them to */
 				packet->data->timestamp = htonl(packet->timestamp);
 				packet->data->seq_number = htons(packet->seq_number);
 				if(packet->codec == JANUS_VIDEOCODEC_VP8) {
@@ -7109,28 +7833,28 @@ static void janus_streaming_relay_rtp_packet(gpointer data, gpointer user_data)
 				}
 			} else {
 				/* Fix sequence number and timestamp (switching may be involved) */
-				janus_rtp_header_update(packet->data, &session->context, TRUE, 0);
-				if(gateway != NULL)
-					gateway->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+				janus_rtp_header_update(packet->data, &s->context, TRUE);
+				if(gateway != NULL) {
+					gateway->relay_rtp(session->handle, s->mindex,
+						packet->is_video, (char *)packet->data, packet->length);
+				}
 				/* Restore the timestamp and sequence number to what the video source set them to */
 				packet->data->timestamp = htonl(packet->timestamp);
 				packet->data->seq_number = htons(packet->seq_number);
 			}
 		} else {
-			if(!session->audio)
-				return;
 			/* Fix sequence number and timestamp (switching may be involved) */
-			janus_rtp_header_update(packet->data, &session->context, FALSE, 0);
-			if(gateway != NULL)
-				gateway->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+			janus_rtp_header_update(packet->data, &s->context, FALSE);
+			if(gateway != NULL) {
+				gateway->relay_rtp(session->handle, s->mindex,
+					packet->is_video, (char *)packet->data, packet->length);
+			}
 			/* Restore the timestamp and sequence number to what the video source set them to */
 			packet->data->timestamp = htonl(packet->timestamp);
 			packet->data->seq_number = htons(packet->seq_number);
 		}
 	} else {
 		/* We're broadcasting a data channel message */
-		if(!session->data)
-			return;
 		char *text = (char *)packet->data;
 		if(gateway != NULL && text != NULL)
 			gateway->relay_data(session->handle, NULL, text, strlen(text));
@@ -7155,9 +7879,16 @@ static void janus_streaming_relay_rtcp_packet(gpointer data, gpointer user_data)
 		//~ JANUS_LOG(LOG_ERR, "Streaming not started yet for this session...\n");
 		return;
 	}
+	janus_streaming_session_stream *s = g_hash_table_lookup(session->streams_byid, GINT_TO_POINTER(packet->mindex));
+	if(packet->mindex != -1 && s == NULL) {
+		JANUS_LOG(LOG_WARN, "No session stream for mindex %d...\n", packet->mindex);
+		return;
+	}
 
-	if(gateway != NULL)
-		gateway->relay_rtcp(session->handle, (packet->is_video ? 1 : 0), (char*)packet->data, packet->length);
+	if(gateway != NULL) {
+		gateway->relay_rtcp(session->handle, s->mindex,
+			packet->is_video, (char*)packet->data, packet->length);
+	}
 
 	return;
 }
@@ -7175,6 +7906,7 @@ static void janus_streaming_helper_rtprtcp_packet(gpointer data, gpointer user_d
 	}
 	/* Clone the packet and queue it for delivery on the helper thread */
 	janus_streaming_rtp_relay_packet *copy = g_malloc0(sizeof(janus_streaming_rtp_relay_packet));
+	copy->mindex = packet->mindex;
 	copy->data = g_malloc(packet->length);
 	memcpy(copy->data, packet->data, packet->length);
 	copy->length = packet->length;
diff --git a/plugins/janus_textroom.c b/plugins/janus_textroom.c
index cf22676c63..45447fdce8 100644
--- a/plugins/janus_textroom.c
+++ b/plugins/janus_textroom.c
@@ -510,8 +510,8 @@ post = <optional backend to contact via HTTP post for all incoming messages>
 
 
 /* Plugin information */
-#define JANUS_TEXTROOM_VERSION			2
-#define JANUS_TEXTROOM_VERSION_STRING	"0.0.2"
+#define JANUS_TEXTROOM_VERSION			3
+#define JANUS_TEXTROOM_VERSION_STRING	"0.0.3"
 #define JANUS_TEXTROOM_DESCRIPTION		"This is a plugin implementing a text-only room for Janus, using DataChannels."
 #define JANUS_TEXTROOM_NAME				"JANUS TextRoom plugin"
 #define JANUS_TEXTROOM_AUTHOR			"Meetecho s.r.l."
@@ -532,10 +532,10 @@ void janus_textroom_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_textroom_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 json_t *janus_textroom_handle_admin_message(json_t *message);
 void janus_textroom_setup_media(janus_plugin_session *handle);
-void janus_textroom_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_textroom_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_textroom_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_textroom_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_textroom_incoming_data(janus_plugin_session *handle, char *label, char *buf, int len);
-void janus_textroom_slow_link(janus_plugin_session *handle, int uplink, int video);
+void janus_textroom_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink);
 void janus_textroom_hangup_media(janus_plugin_session *handle);
 void janus_textroom_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_textroom_query_session(janus_plugin_session *handle);
@@ -576,64 +576,64 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter transaction_parameters[] = {
-	{"textroom", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"transaction", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"textroom", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"transaction", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter room_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter adminkey_parameters[] = {
-	{"admin_key", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"admin_key", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter create_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"description", JSON_STRING, 0},
-	{"secret", JSON_STRING, 0},
-	{"pin", JSON_STRING, 0},
-	{"post", JSON_STRING, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"description", JANUS_JSON_STRING, 0},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"pin", JANUS_JSON_STRING, 0},
+	{"post", JANUS_JSON_STRING, 0},
 	{"is_private", JANUS_JSON_BOOL, 0},
-	{"allowed", JSON_ARRAY, 0},
+	{"allowed", JANUS_JSON_ARRAY, 0},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter destroy_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter edit_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"new_description", JSON_STRING, 0},
-	{"new_secret", JSON_STRING, 0},
-	{"new_pin", JSON_STRING, 0},
-	{"new_post", JSON_STRING, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"new_description", JANUS_JSON_STRING, 0},
+	{"new_secret", JANUS_JSON_STRING, 0},
+	{"new_pin", JANUS_JSON_STRING, 0},
+	{"new_post", JANUS_JSON_STRING, 0},
 	{"new_is_private", JANUS_JSON_BOOL, 0},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter allowed_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"action", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"allowed", JSON_ARRAY, 0}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"action", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"allowed", JANUS_JSON_ARRAY, 0}
 };
 static struct janus_json_parameter kick_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"username", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"username", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter join_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"username", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"pin", JSON_STRING, 0},
-	{"display", JSON_STRING, 0}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"username", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"pin", JANUS_JSON_STRING, 0},
+	{"display", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter message_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"text", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"to", JSON_STRING, 0},
-	{"tos", JSON_ARRAY, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"text", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"to", JANUS_JSON_STRING, 0},
+	{"tos", JANUS_JSON_ARRAY, 0},
 	{"ack", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter announcement_parameters[] = {
@@ -968,6 +968,7 @@ int janus_textroom_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the TextRoom handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_TEXTROOM_NAME);
@@ -1297,11 +1298,11 @@ void janus_textroom_setup_media(janus_plugin_session *handle) {
 	janus_mutex_unlock(&sessions_mutex);
 }
 
-void janus_textroom_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_textroom_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	/* We don't do audio/video */
 }
 
-void janus_textroom_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_textroom_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	/* We don't do audio/video */
 }
 
@@ -2510,7 +2511,7 @@ janus_plugin_result *janus_textroom_handle_incoming_request(janus_plugin_session
 	return NULL;
 }
 
-void janus_textroom_slow_link(janus_plugin_session *handle, int uplink, int video) {
+void janus_textroom_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink) {
 	/* We don't do audio/video */
 }
 
diff --git a/plugins/janus_videocall.c b/plugins/janus_videocall.c
index 9d0fb4cbf7..5d6c1ef653 100644
--- a/plugins/janus_videocall.c
+++ b/plugins/janus_videocall.c
@@ -267,8 +267,8 @@
 
 
 /* Plugin information */
-#define JANUS_VIDEOCALL_VERSION			6
-#define JANUS_VIDEOCALL_VERSION_STRING	"0.0.6"
+#define JANUS_VIDEOCALL_VERSION			7
+#define JANUS_VIDEOCALL_VERSION_STRING	"0.0.7"
 #define JANUS_VIDEOCALL_DESCRIPTION		"This is a simple video call plugin for Janus, allowing two WebRTC peers to call each other through a server."
 #define JANUS_VIDEOCALL_NAME			"JANUS VideoCall plugin"
 #define JANUS_VIDEOCALL_AUTHOR			"Meetecho s.r.l."
@@ -288,10 +288,10 @@ const char *janus_videocall_get_package(void);
 void janus_videocall_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_videocall_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 void janus_videocall_setup_media(janus_plugin_session *handle);
-void janus_videocall_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_videocall_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_videocall_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_videocall_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_videocall_incoming_data(janus_plugin_session *handle, char *label, char *buf, int len);
-void janus_videocall_slow_link(janus_plugin_session *handle, int uplink, int video);
+void janus_videocall_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink);
 void janus_videocall_hangup_media(janus_plugin_session *handle);
 void janus_videocall_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_videocall_query_session(janus_plugin_session *handle);
@@ -330,17 +330,17 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter username_parameters[] = {
-	{"username", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"username", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter set_parameters[] = {
 	{"audio", JANUS_JSON_BOOL, 0},
 	{"video", JANUS_JSON_BOOL, 0},
-	{"bitrate", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"bitrate", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"record", JANUS_JSON_BOOL, 0},
-	{"filename", JSON_STRING, 0},
+	{"filename", JANUS_JSON_STRING, 0},
 	{"restart", JANUS_JSON_BOOL, 0}
 };
 
@@ -493,6 +493,7 @@ int janus_videocall_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the VideoCall handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_VIDEOCALL_NAME);
@@ -649,9 +650,9 @@ json_t *janus_videocall_query_session(janus_plugin_session *handle) {
 			json_object_set_new(recording, "data", json_string(session->drc->filename));
 		json_object_set_new(info, "recording", recording);
 	}
-	json_object_set_new(info, "incall", json_integer(g_atomic_int_get(&session->incall)));
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "incall", g_atomic_int_get(&session->incall) ? json_true() : json_false());
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -692,7 +693,7 @@ void janus_videocall_setup_media(janus_plugin_session *handle) {
 	/* We really don't care, as we only relay RTP/RTCP we get in the first place anyway */
 }
 
-void janus_videocall_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_videocall_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	if(gateway) {
@@ -728,7 +729,7 @@ void janus_videocall_incoming_rtp(janus_plugin_session *handle, int video, char
 				char rtcpbuf[12];
 				memset(rtcpbuf, 0, 12);
 				janus_rtcp_pli((char *)&rtcpbuf, 12);
-				gateway->relay_rtcp(session->handle, 1, rtcpbuf, 12);
+				gateway->relay_rtcp(session->handle, -1, TRUE, rtcpbuf, 12);
 			}
 			/* Any event we should notify? */
 			if(peer->sim_context.changed_substream) {
@@ -756,7 +757,7 @@ void janus_videocall_incoming_rtp(janus_plugin_session *handle, int video, char
 				json_decref(event);
 			}
 			/* If we got here, update the RTP header and send the packet */
-			janus_rtp_header_update(header, &peer->context, TRUE, 4500);
+			janus_rtp_header_update(header, &peer->context, TRUE);
 			if(session->vcodec == JANUS_VIDEOCODEC_VP8) {
 				int plen = 0;
 				char *payload = janus_rtp_payload(buf, len, &plen);
@@ -766,7 +767,7 @@ void janus_videocall_incoming_rtp(janus_plugin_session *handle, int video, char
 			header->ssrc = htonl(1);
 			janus_recorder_save_frame(session->vrc, buf, len);
 			/* Send the frame back */
-			gateway->relay_rtp(peer->handle, video, buf, len);
+			gateway->relay_rtp(peer->handle, mindex, video, buf, len);
 			/* Restore header or core statistics will be messed up */
 			header->ssrc = htonl(ssrc);
 			header->timestamp = htonl(timestamp);
@@ -776,13 +777,13 @@ void janus_videocall_incoming_rtp(janus_plugin_session *handle, int video, char
 				/* Save the frame if we're recording */
 				janus_recorder_save_frame(video ? session->vrc : session->arc, buf, len);
 				/* Forward the packet to the peer */
-				gateway->relay_rtp(peer->handle, video, buf, len);
+				gateway->relay_rtp(peer->handle, mindex, video, buf, len);
 			}
 		}
 	}
 }
 
-void janus_videocall_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_videocall_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	if(gateway) {
@@ -805,10 +806,10 @@ void janus_videocall_incoming_rtcp(janus_plugin_session *handle, int video, char
 				janus_rtcp_cap_remb(buf, len, 10000000);
 			else
 				janus_rtcp_cap_remb(buf, len, session->bitrate);
-			gateway->relay_rtcp(peer->handle, 1, buf, len);
+			gateway->relay_rtcp(peer->handle, -1, TRUE, buf, len);
 			return;
 		}
-		gateway->relay_rtcp(peer->handle, video, buf, len);
+		gateway->relay_rtcp(peer->handle, mindex, video, buf, len);
 	}
 }
 
@@ -842,7 +843,7 @@ void janus_videocall_incoming_data(janus_plugin_session *handle, char *label, ch
 	}
 }
 
-void janus_videocall_slow_link(janus_plugin_session *handle, int uplink, int video) {
+void janus_videocall_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink) {
 	/* The core is informing us that our peer got or sent too many NACKs, are we pushing media too hard? */
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
@@ -1202,12 +1203,19 @@ static void *janus_videocall_handler(void *data) {
 				JANUS_LOG(LOG_VERB, "This is involving a negotiation (%s) as well:\n%s\n", msg_sdp_type, msg_sdp);
 				/* Check if this user will simulcast */
 				json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
-				if(msg_simulcast) {
-					JANUS_LOG(LOG_VERB, "VideoCall caller (%s) is going to do simulcasting\n", session->username);
-					int rid_ext_id = -1, framemarking_ext_id = -1;
-					janus_rtp_simulcasting_prepare(msg_simulcast, &rid_ext_id, &framemarking_ext_id, session->ssrc, session->rid);
-					session->sim_context.rid_ext_id = rid_ext_id;
-					session->sim_context.framemarking_ext_id = framemarking_ext_id;
+				if(msg_simulcast && json_array_size(msg_simulcast) > 0 && janus_get_codec_pt(msg_sdp, "vp8") > 0) {
+					size_t i = 0;
+					for(i=0; i<json_array_size(msg_simulcast); i++) {
+						json_t *s = json_array_get(msg_simulcast, i);
+						int mindex = json_integer_value(json_object_get(s, "mindex"));
+						JANUS_LOG(LOG_VERB, "VideoCall caller (%s) is going to do simulcasting (#%d)\n", session->username, mindex);
+						int rid_ext_id = -1, framemarking_ext_id = -1;
+						janus_rtp_simulcasting_prepare(s, &rid_ext_id, &framemarking_ext_id, session->ssrc, session->rid);
+						session->sim_context.rid_ext_id = rid_ext_id;
+						session->sim_context.framemarking_ext_id = framemarking_ext_id;
+						/* FIXME We're stopping at the first item, there may be more */
+						break;
+					}
 				}
 				/* Send SDP to our peer */
 				json_t *call = json_object();
@@ -1286,8 +1294,9 @@ static void *janus_videocall_handler(void *data) {
 			}
 			/* Check which codecs we ended up using */
 			const char *acodec = NULL, *vcodec = NULL;
-			janus_sdp_find_first_codecs(answer, &acodec, &vcodec);
+			janus_sdp_find_first_codecs(answer, JANUS_SDP_AUDIO, -1, &acodec);
 			session->acodec = janus_audiocodec_from_name(acodec);
+			janus_sdp_find_first_codecs(answer, JANUS_SDP_VIDEO, -1, &vcodec);
 			session->vcodec = janus_videocodec_from_name(vcodec);
 			if(session->acodec == JANUS_AUDIOCODEC_NONE) {
 				session->has_audio = FALSE;
@@ -1374,7 +1383,7 @@ static void *janus_videocall_handler(void *data) {
 					JANUS_LOG(LOG_VERB, "Just (re-)enabled video, sending a PLI to recover it\n");
 					char buf[12];
 					janus_rtcp_pli((char *)&buf, 12);
-					gateway->relay_rtcp(session->handle, 1, buf, 12);
+					gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 				}
 				session->video_active = json_is_true(video);
 				JANUS_LOG(LOG_VERB, "Setting video property: %s\n", session->video_active ? "true" : "false");
@@ -1387,7 +1396,7 @@ static void *janus_videocall_handler(void *data) {
 					char buf[24];
 					janus_rtcp_remb((char *)&buf, 24, session->bitrate);
 					JANUS_LOG(LOG_VERB, "Sending REMB\n");
-					gateway->relay_rtcp(session->handle, 1, buf, 24);
+					gateway->relay_rtcp(session->handle, -1, TRUE, buf, 24);
 					/* FIXME How should we handle a subsequent "no limit" bitrate? */
 				}
 			}
@@ -1414,7 +1423,7 @@ static void *janus_videocall_handler(void *data) {
 					memset(buf, 0, 12);
 					janus_rtcp_pli((char *)&buf, 12);
 					if(peer && peer->handle)
-						gateway->relay_rtcp(session->handle, 1, buf, 12);
+						gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 				}
 			}
 			if(temporal) {
@@ -1439,7 +1448,7 @@ static void *janus_videocall_handler(void *data) {
 					memset(buf, 0, 12);
 					janus_rtcp_pli((char *)&buf, 12);
 					if(peer && peer->handle)
-						gateway->relay_rtcp(session->handle, 1, buf, 12);
+						gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 				}
 			}
 			if(record) {
@@ -1510,7 +1519,7 @@ static void *janus_videocall_handler(void *data) {
 						JANUS_LOG(LOG_VERB, "Recording video, sending a PLI to kickstart it\n");
 						char buf[12];
 						janus_rtcp_pli((char *)&buf, 12);
-						gateway->relay_rtcp(session->handle, 1, buf, 12);
+						gateway->relay_rtcp(session->handle, -1, TRUE, buf, 12);
 					}
 					if(session->has_data) {
 						memset(filename, 0, 255);
diff --git a/plugins/janus_videoroom.c b/plugins/janus_videoroom.c
index cf9ca833b3..0f6c02aa06 100644
--- a/plugins/janus_videoroom.c
+++ b/plugins/janus_videoroom.c
@@ -14,30 +14,48 @@
  * can join and leave at any time. This room is based on a Publish/Subscribe
  * pattern. Each peer can publish his/her own live audio/video feeds: this
  * feed becomes an available stream in the room the other participants can
- * attach to. This means that this plugin allows the realization of several
+ * subscribe to. This means that this plugin allows the realization of several
  * different scenarios, ranging from a simple webinar (one speaker, several
  * watchers) to a fully meshed video conference (each peer sending and
  * receiving to and from all the others).
  *
- * Considering that this plugin allows for several different WebRTC PeerConnections
- * to be on at the same time for the same peer (specifically, each peer
- * potentially has 1 PeerConnection on for publishing and N on for subscriptions
- * from other peers), each peer may need to attach several times to the same
- * plugin for every stream: this means that each peer needs to have at least one
- * handle active for managing its relation with the plugin (joining a room,
+ * Notice that, since Janus now supports multistream via Unified Plan,
+ * subscriptions can be done either in "bulks" (you use a single PeerConnection
+ * to subscribe to multiple streams from one or more publishers) or
+ * separately (each PeerConnections represents a subscription to a single
+ * publisher). Same thing for publishers: you may choose to publish, e.g.,
+ * audio and video on one PeerConnection, and share your screen on another,
+ * or publish everything on the same PeerConnection instead. While
+ * functionally both approaches (multistream vs. legacy mode) are the same
+ * (the same media flows in both cases), the differences are in how
+ * resources are used, and in how the client has to handle incoming and
+ * outgoing connections. Besides, one approach might make more sense in
+ * some scenarios, and the other make more sense in different use cases.
+ * As such, the approach to follow is left to the developer and the application.
+ *
+ * What is important to point out, though, is that publishers and subscribers
+ * will in all cases require different PeerConnections. This means that,
+ * even with multistream, you won't be able to use a single PeerConnection
+ * to send your contributions and receive those from everyone else. This
+ * is a choice done by design, to avoid the issues that would inevitably
+ * arise when doing, for instance, renegotiations to update the streams.
+ *
+ * On a more general note and to give some more context with respect to the
+ * core functionality in Janus, notice that, considering this plugin allows
+ * for several different WebRTC PeerConnections to be on at the same time
+ * for the same peer (different publishers and subscribers for sure, and
+ * potentially more than one of each if multistream is not in use), each
+ * peer will often need to attach several times to the same plugin for each
+ * stream: this means that each peer needs to have at least one handle active
+ * for managing its relation with the plugin (joining a room,
  * leaving a room, muting/unmuting, publishing, receiving events), and needs
- * to open a new one each time he/she wants to subscribe to a feed from
- * another publisher participant. The handle used for a subscription,
- * however, would be logically a "slave" to the master one used for
- * managing the room: this means that it cannot be used, for instance,
- * to unmute in the room, as its only purpose would be to provide a
- * context in which creating the recvonly PeerConnection for the
- * subscription to an active publisher participant.
- *
- * \note Work is going on to implement SSRC multiplexing (Unified Plan),
- * meaning that in the future you'll be able to use the same
- * Janus handle/VideoRoom subscriber/PeerConnection to receive multiple
- * publishers at the same time.
+ * to open others when they want to subscribe to a feed from other participants
+ * (the number depends on the subscription approach of choice). Handles
+ * used for subscriptions, though, would be logically "subjects" to the master one used for
+ * managing the room: this means that they cannot be used, for instance,
+ * to unmute in the room, as their only purpose would be to provide a
+ * context in which creating the recvonly PeerConnections for the
+ * subscription(s).
  *
  * Rooms to make available are listed in the plugin configuration file.
  * A pre-filled configuration file is provided in \c conf/janus.plugin.videoroom.jcfg
@@ -59,7 +77,7 @@ room-<unique room ID>: {
 	publishers = <max number of concurrent senders> (e.g., 6 for a video
 				 conference or 1 for a webinar, default=3)
 	bitrate = <max video bitrate for senders> (e.g., 128000)
-	fir_freq = <send a FIR to publishers every fir_freq seconds> (0=disable)
+	pli_freq = <send a PLI to publishers every pli_freq seconds> (0=disable)
 	audiocodec = opus|g722|pcmu|pcma|isac32|isac16 (audio codec to force on publishers, default=opus
 				can be a comma separated list in order of preference, e.g., opus,pcmu)
 	videocodec = vp8|vp9|h264 (video codec to force on publishers, default=vp8
@@ -201,7 +219,7 @@ room-<unique room ID>: {
 	"new_is_private" : <true|false, whether the room should appear in a list request>,
 	"new_require_pvtid" : <true|false, whether the room should require private_id from subscribers>,
 	"new_bitrate" : <new bitrate cap to force on all publishers (except those with custom overrides)>,
-	"new_fir_freq" : <new period for regular PLI keyframe requests to publishers>,
+	"new_pli_freq" : <new period for regular PLI keyframe requests to publishers>,
 	"new_publishers" : <new cap on the number of concurrent active WebRTC publishers>,
 	"permanent" : <true|false, whether the room should be also removed from the config file, default=false>
 }
@@ -342,12 +360,12 @@ room-<unique room ID>: {
 			"max_publishers" : <how many publishers can actually publish via WebRTC at the same time>,
 			"bitrate" : <bitrate cap that should be forced (via REMB) on all publishers by default>,
 			"bitrate_cap" : <true|false, whether the above cap should act as a limit to dynamic bitrate changes by publishers>,
-			"fir_freq" : <how often a keyframe request is sent via PLI/FIR to active publishers>,
+			"pli_freq" : <how often a keyframe request is sent via PLI to active publishers>,
 			"audiocodec" : "<comma separated list of allowed audio codecs>",
 			"videocodec" : "<comma separated list of allowed video codecs>",
 			"record" : <true|false, whether the room is being recorded>,
 			"record_dir" : "<if recording, the path where the .mjr files are being saved>",
-			"num_participants" : <count of the participants (publishers, active or not; not subscribers)>
+			"num_participants" : <count of the participants (publisher role instances, active or not; not subscribers)>
 		},
 		// Other rooms
 	]
@@ -376,9 +394,8 @@ room-<unique room ID>: {
 		{	// Participant #1
 			"id" : <unique numeric ID of the participant>,
 			"display" : "<display name of the participant, if any; optional>",
-			"talking" : <true|false, whether user is talking or not (only if audio levels are used)>,
-			"internal_audio_ssrc" : <audio SSRC used internally for this active publisher>,
-			"internal_video_ssrc" : <video SSRC used internally for this active publisher>
+			"publisher" : <true|false, whether user is an active publisher or not>,
+			"talking" : <true|false, whether user is talking or not (only if audio levels are used)>
 		},
 		// Other participants
 	]
@@ -441,10 +458,20 @@ room-<unique room ID>: {
 		{
 			"id" : <unique ID of active publisher #1>,
 			"display" : "<display name of active publisher #1, if any>",
-			"audio_codec" : "<audio codec used by active publisher #1, if any>",
-			"video_codec" : "<video codec used by active publisher #1, if any>",
-			"simulcast" : "<true if the publisher uses simulcast (VP8 and H.264 only)>",
-			"talking" : <true|false, whether the publisher is talking or not (only if audio levels are used)>,
+			"streams" : [
+				{
+					"type" : "<type of published stream #1 (audio|video|data)">,
+					"mindex" : "<unique mindex of published stream #1>",
+					"mid" : "<unique mid of of published stream #1>",
+					"codec" : "<codec used for published stream #1>",
+					"description" : "<text description of published stream #1, if any>",
+					"simulcast" : "<true if published stream #1 uses simulcast (VP8 and H.264 only)>",
+					"svc" : "<true if published stream #1 uses SVC (VP9 only)>",
+					"talking" : <true|false, whether the publisher stream has audio activity or not (only if audio levels are used)>,
+				},
+				// Other streams, if any
+			],
+			"talking" : <true|false, whether the publisher is talking or not (only if audio levels are used); deprecated, use the stream specific ones>,
 		},
 		// Other active publishers
 	],
@@ -492,7 +519,7 @@ room-<unique room ID>: {
  * to the room configuration (e.g., to make sure the codecs you negotiated
  * are allowed in the room), and will reply with a JSEP SDP answer to
  * close the circle and complete the setup of the PeerConnection. As soon
- * as the PeerConnection has been establisher, the publisher will become
+ * as the PeerConnection has been established, the publisher will become
  * active, and a new active feed other participants can subscribe to.
  *
  * The syntax of a \c publish request is the following:
@@ -500,22 +527,37 @@ room-<unique room ID>: {
 \verbatim
 {
 	"request" : "publish",
-	"audio" : <true|false, depending on whether or not audio should be relayed; true by default>,
-	"video" : <true|false, depending on whether or not video should be relayed; true by default>,
-	"data" : <true|false, depending on whether or not data should be relayed; true by default>,
 	"audiocodec" : "<audio codec to prefer among the negotiated ones; optional>",
 	"videocodec" : "<video codec to prefer among the negotiated ones; optional>",
 	"bitrate" : <bitrate cap to return via REMB; optional, overrides the global room value if present>,
 	"record" : <true|false, whether this publisher should be recorded or not; optional>,
 	"filename" : "<if recording, the base path/file to use for the recording files; optional>",
-	"display" : "<new display name to use in the room; optional>"
+	"display" : "<new display name to use in the room; optional>",
+	"descriptions" : [	// Optional
+		{
+			"mid" : "<unique mid of a stream being published>",
+			"description" : "<text description of the stream (e.g., My front webcam)>"
+		},
+		// Other descriptions, if any
+	]
 }
 \endverbatim
  *
  * As anticipated, since this is supposed to be accompanied by a JSEP SDP
  * offer describing the publisher's media streams, the plugin will negotiate
- * and prepare a matching JSEP SDP answer. If successful, a \c configured
- * event will be sent back, formatted like this:
+ * and prepare a matching JSEP SDP answer. Notice that, in principle, all
+ * published streams will be only identifier by their unique \c mid and
+ * by their type (e.g., audio or video). In case you want to provide more
+ * information about the streams being published (e.g., to let other
+ * participants know that the first video is a camera, while the second
+ * video is a screen share), you can use the \c descriptions array for
+ * the purpose: each object in the array can be used to add a text description
+ * to associate to a specific mid, in order to help with the UI rendering.
+ * The \c descriptions property is optional, so no text will be provided
+ * by default: notice these descriptions can be updated dynamically via
+ * \c configure requests.
+ *
+ * If successful, a \c configured event will be sent back, formatted like this:
  *
 \verbatim
 {
@@ -557,10 +599,20 @@ room-<unique room ID>: {
 		{
 			"id" : <unique ID of the new publisher>,
 			"display" : "<display name of the new publisher, if any>",
-			"audio_codec" : "<audio codec used the new publisher, if any>",
-			"video_codec" : "<video codec used by the new publisher, if any>",
-			"simulcast" : "<true if the publisher uses simulcast (VP8 and H.264 only)>",
-			"talking" : <true|false, whether the publisher is talking or not (only if audio levels are used)>,
+			"streams" : [
+				{
+					"type" : "<type of published stream #1 (audio|video|data)">,
+					"mindex" : "<unique mindex of published stream #1>",
+					"mid" : "<unique mid of of published stream #1>",
+					"codec" : "<codec used for published stream #1>",
+					"description" : "<text description of published stream #1, if any>",
+					"simulcast" : "<true if published stream #1 uses simulcast (VP8 and H.264 only)>",
+					"svc" : "<true if published stream #1 uses SVC (VP9 only)>",
+					"talking" : <true|false, whether the publisher stream has audio activity or not (only if audio levels are used)>,
+				},
+				// Other streams, if any
+			],
+			"talking" : <true|false, whether the publisher is talking or not (only if audio levels are used); deprecated, use the stream specific ones>,
 		}
 	]
 }
@@ -611,14 +663,16 @@ room-<unique room ID>: {
 \verbatim
 {
 	"request" : "configure",
-	"audio" : <true|false, depending on whether or not audio should be relayed; true by default>,
-	"video" : <true|false, depending on whether or not video should be relayed; true by default>,
-	"data" : <true|false, depending on whether or not data should be relayed; true by default>,
 	"bitrate" : <bitrate cap to return via REMB; optional, overrides the global room value if present (unless bitrate_cap is set)>,
 	"keyframe" : <true|false, whether we should send this publisher a keyframe request>,
 	"record" : <true|false, whether this publisher should be recorded or not; optional>,
 	"filename" : "<if recording, the base path/file to use for the recording files; optional>",
-	"display" : "<new display name to use in the room; optional>"
+	"display" : "<new display name to use in the room; optional>",
+	"mid" : <mid of the m-line to refer to for this configure request; optional>,
+	"send" : <true|false, depending on whether the media addressed by the above mid should be relayed or not; optional>,
+	"descriptions" : [
+		// Updated descriptions for the published streams; see "publish" for syntax; optional
+	]
 }
 \endverbatim
  *
@@ -659,7 +713,9 @@ room-<unique room ID>: {
  * whatever else makes sense in this context. This is made possible by
  * a request called \c rtp_forward which, as the name suggests, simply
  * forwards in real-time the media sent by a publisher via RTP (plain
- * or encrypted) to a remote backend.
+ * or encrypted) to a remote backend. Notice that, although we're using
+ * the term "RTP forwarder", this feature can be used to forward data
+ * channel messages as well.
  *
  * You can add a new RTP forwarder for an existing publisher using the
  * \c rtp_forward request, which has to be formatted as follows:
@@ -671,25 +727,40 @@ room-<unique room ID>: {
 	"publisher_id" : <unique numeric ID of the publisher to relay externally>,
 	"host" : "<host address to forward the RTP and data packets to>",
 	"host_family" : "<ipv4|ipv6, if we need to resolve the host address to an IP; by default, whatever we get>",
-	"audio_port" : <port to forward the audio RTP packets to>,
-	"audio_ssrc" : <audio SSRC to use to use when streaming; optional>,
-	"audio_pt" : <audio payload type to use when streaming; optional>,
-	"audio_rtcp_port" : <port to contact to receive audio RTCP feedback from the recipient; optional, and currently unused for audio>,
-	"video_port" : <port to forward the video RTP packets to>,
-	"video_ssrc" : <video SSRC to use to use when streaming; optional>,
-	"video_pt" : <video payload type to use when streaming; optional>,
-	"video_rtcp_port" : <port to contact to receive video RTCP feedback from the recipient; optional>,
-	"video_port_2" : <if simulcasting, port to forward the video RTP packets from the second substream/layer to>,
-	"video_ssrc_2" : <if simulcasting, video SSRC to use to use the second substream/layer; optional>,
-	"video_pt_2" : <if simulcasting, video payload type to use the second substream/layer; optional>,
-	"video_port_3" : <if simulcasting, port to forward the video RTP packets from the third substream/layer to>,
-	"video_ssrc_3" : <if simulcasting, video SSRC to use to use the third substream/layer; optional>,
-	"video_pt_3" : <if simulcasting, video payload type to use the third substream/layer; optional>,
-	"data_port" : <port to forward the datachannel messages to>,
+	"streams" : [
+		{
+			"mid" : "<mid of publisher stream to forward>",
+			"host" : "<host address to forward the packets to; optional, will use global one if missing>",
+			"host_family" : "<optional, will use global one if missing>",
+			"port" : <port to forward the packets to>,
+			"ssrc" : <SSRC to use to use when forwarding; optional, and only for RTP streams, not data>,
+			"pt" : <payload type to use when forwarding; optional, and only for RTP streams, not data>,
+			"rtcp_port" : <port to contact to receive RTCP feedback from the recipient; optional, and only for RTP streams, not data>,
+			"simulcast" : <whether we need to take simulcasting into account, for this video forwarder>,
+			"port_2" : <if video and simulcasting, port to forward the packets from the second substream/layer to>,
+			"ssrc_2" : <if video and simulcasting, SSRC to use to use the second substream/layer; optional>,
+			"pt_2" : <if video and simulcasting, payload type to use the second substream/layer; optional>,
+			"port_3" : <if video and simulcasting, port to forward the packets from the third substream/layer to>,
+			"ssrc_3" : <if video and simulcasting, SSRC to use to use the third substream/layer; optional>,
+			"pt_3" : <if video and simulcasting, payload type to use the third substream/layer; optional>,
+		},
+		{
+			.. other streams, if needed..
+		}
+	],
 	"srtp_suite" : <length of authentication tag (32 or 80); optional>,
 	"srtp_crypto" : "<key to use as crypto (base64 encoded key as in SDES); optional>"
 }
 \endverbatim
+ *
+ * As you can see, you basically configure each stream to forward in a
+ * dedicated object of the \c streams array: for RTP streams (audio, video)
+ * this includes optionally overriding payload type or SSRC; simulcast
+ * streams can be forwarded separately for each layer. The only parameters
+ * you MUST specify are the host and port to send the packets to: the host
+ * part can be put in the global part of the request, if all streams will
+ * be sent to the same IP address, while the port must be specific to the
+ * stream itself.
  *
  * Notice that, as explained above, in case you configured an \c admin_key
  * property and extended it to RTP forwarding as well, you'll need to provide
@@ -704,21 +775,21 @@ room-<unique room ID>: {
 	"videoroom" : "rtp_forward",
 	"room" : <unique numeric ID, same as request>,
 	"publisher_id" : <unique numeric ID, same as request>,
-	"rtp_stream" : {
-		"host" : "<host this forwarder is streaming to, same as request if not resolved>",
-		"audio" : <audio RTP port, same as request if configured>,
-		"audio_rtcp" : <audio RTCP port, same as request if configured>,
-		"audio_stream_id" : <unique numeric ID assigned to the audio RTP forwarder, if any>,
-		"video" : <video RTP port, same as request if configured>,
-		"video_rtcp" : <video RTCP port, same as request if configured>,
-		"video_stream_id" : <unique numeric ID assigned to the main video RTP forwarder, if any>,
-		"video_2" : <second video port, same as request if configured>,
-		"video_stream_id_2" : <unique numeric ID assigned to the second video RTP forwarder, if any>,
-		"video_3" : <third video port, same as request if configured>,
-		"video_stream_id_3" : <unique numeric ID assigned to the third video RTP forwarder, if any>,
-		"data" : <data port, same as request if configured>,
-		"data_stream_id" : <unique numeric ID assigned to datachannel messages forwarder, if any>
-	}
+	"forwarders" : [
+		{
+			"stream_id" : <unique numeric ID assigned to this forwarder, if any>,
+			"type" : "<audio|video|data>",
+			"ip" : "<IP this forwarder is streaming to, same as request if not resolved>",
+			"port" : <port this forwarder is streaming to, same as request if configured>,
+			"local_rtcp_port" : <local port this forwarder is using to get RTCP feedback, if any>,
+			"remote_rtcp_port" : <remote port this forwarder getting RTCP feedback from, if any>,
+			"ssrc" : <SSRC this forwarder is using, same as request if configured>,
+			"pt" : <payload type this forwarder is using, same as request if configured>,
+			"substream" : <video substream this video forwarder is relaying, if any>,
+			"srtp" : <true|false, whether the RTP stream is encrypted>
+		},
+		// Other forwarders, if configured
+	]
 }
 \endverbatim
  *
@@ -764,17 +835,17 @@ room-<unique room ID>: {
 {
 	"videoroom" : "forwarders",
 	"room" : <unique numeric ID of the room>,
-	"rtp_forwarders" : [		// Array of publishers with RTP forwarders
+	"publishers" : [		// Array of publishers with RTP forwarders
 		{	// Publisher #1
 			"publisher_id" : <unique numeric ID of publisher #1>,
-			"rtp_forwarders" : [		// Array of RTP forwarders
+			"forwarders" : [		// Array of RTP forwarders
 				{	// RTP forwarder #1
-					"audio_stream_id" : <unique numeric ID assigned to this audio RTP forwarder, if any>,
-					"video_stream_id" : <unique numeric ID assigned to this video RTP forwarder, if any>,
-					"data_stream_id" : <unique numeric ID assigned to this datachannel messages forwarder, if any>
+					"stream_id" : <unique numeric ID assigned to this forwarder, if any>,
+					"type" : "<audio|video|data>",
 					"ip" : "<IP this forwarder is streaming to>",
 					"port" : <port this forwarder is streaming to>,
-					"rtcp_port" : <local port this forwarder is using to get RTCP feedback, if any>,
+					"local_rtcp_port" : <local port this forwarder is using to get RTCP feedback, if any>,
+					"remote_rtcp_port" : <remote port this forwarder getting RTCP feedback from, if any>,
 					"ssrc" : <SSRC this forwarder is using, if any>,
 					"pt" : <payload type this forwarder is using, if any>,
 					"substream" : <video substream this video forwarder is relaying, if any>,
@@ -823,14 +894,15 @@ room-<unique room ID>: {
  * \subsection vroomsub VideoRoom Subscribers
  *
  * In a VideoRoom, subscribers are NOT participants, but simply handles
- * that will be used exclusively to receive media from a specific publisher
+ * that will be used exclusively to receive media from one or more publishers
  * in the room. Since they're not participants per se, they're basically
  * streams that can be (and typically are) associated to publisher handles
  * as the ones we introduced in the previous section, whether active or not.
  * In fact, the typical use case is publishers being notified about new
  * participants becoming active in the room, and as a result new subscriber
  * sessions being created to receive their media streams; as soon as the
- * publisher goes away, the subscriber handle is removed as well. As such,
+ * publisher goes away, other participants are notified so that the related
+ * subscriber handles can be removed/updated accordingly as well. As such,
  * these subscriber sessions are dependent on feedback obtained by
  * publishers, and can't exist on their own, unless you feed them the
  * right info out of band.
@@ -846,37 +918,46 @@ room-<unique room ID>: {
 	"room" : <unique ID of the room to subscribe in>,
 	"feed" : <unique ID of the publisher to subscribe to; mandatory>,
 	"private_id" : <unique ID of the publisher that originated this request; optional, unless mandated by the room configuration>,
-	"close_pc" : <true|false, depending on whether or not the PeerConnection should be automatically closed when the publisher leaves; true by default>,
-	"audio" : <true|false, depending on whether or not audio should be relayed; true by default>,
-	"video" : <true|false, depending on whether or not video should be relayed; true by default>,
-	"data" : <true|false, depending on whether or not data should be relayed; true by default>,
-	"offer_audio" : <true|false; whether or not audio should be negotiated; true by default if the publisher has audio>,
-	"offer_video" : <true|false; whether or not video should be negotiated; true by default if the publisher has video>,
-	"offer_data" : <true|false; whether or not datachannels should be negotiated; true by default if the publisher has datachannels>,
-	"substream" : <substream to receive (0-2), in case simulcasting is enabled; optional>,
-	"temporal" : <temporal layers to receive (0-2), in case simulcasting is enabled; optional>,
-	"spatial_layer" : <spatial layer to receive (0-2), in case VP9-SVC is enabled; optional>,
-	"temporal_layer" : <temporal layers to receive (0-2), in case VP9-SVC is enabled; optional>
+	"streams" : [
+		{
+			"feed_id" : <unique ID of publisher owning the stream to subscribe to>,
+			"mid" : "<unique mid of the publisher stream to subscribe to; optional>"
+			// Optionally, simulcast or SVC targets (defaults if missing)
+		},
+		// Other streams to subscribe to
+	]
 }
 \endverbatim
  *
- * As you can see, it's just a matter of specifying the ID of the publisher to
- * subscribe to and, if needed, your own \c private_id (if mandated by the room).
- * The \c offer_audio , \c offer_video and \c offer_data are
- * also particularly interesting, though, as they allow you to only subscribe
- * to a subset of the mountpoint media. By default, in fact, this \c join
- * request will result in the plugin preparing a new SDP offer trying to
- * negotiate all the media streams made available by the publisher; in case
- * the subscriber knows they don't support one of the mountpoint codecs, though
- * (e.g., the video in the mountpoint is VP8, but they only support H.264),
- * or are not interested in getting all the media (e.g., they're ok with
- * just audio and not video, or don't have enough bandwidth for both),
- * they can use those properties to shape the SDP offer to their needs.
- * In case the publisher to subscribe to is simulcasting or doing VP9 SVC,
- * you can choose in advance which substream you're interested in, e.g.,
- * to only get the medium quality at best, instead of higher options if
- * available. As we'll see later, this can be changed dynamically at any
- * time using a subsequent \c configure request.
+ * As you can see, it's just a matter of specifying the list of streams to
+ * subscribe to: in particular, you have to provide an array of objects,
+ * where each objects represents a specific stream (or group of streams)
+ * you're interested in. For each object, the \c feed_id indicating the
+ * publisher owning the stream(s) is mandatory, while the related \c mid
+ * is optional: this gives you some flexibility when subscribing, as
+ * only providing a \c feed_id will indicate you're interested in ALL
+ * the stream from that publisher, while providing a \c mid as well will
+ * indicate you're interested in a stream in particular. Since you can
+ * provide an array of streams, just specifying the \c feed_id or explicitly
+ * listing all the \c feed_id + \c mid combinations is equivalent: of
+ * course, different objects in the array can indicate different publishers,
+ * allowing you to combine streams from different sources in the same subscription.
+ *
+ * Depending on whether the subscription will refer to a
+ * single publisher (legacy approach) or to streams coming from different
+ * publishers (multistream), the list of streams may differ. The ability
+ * to single out the streams to subscribe to is particularly useful in
+ * case you don't want to, or can't, subscribe to all available media:
+ * e.g., you know a publisher is sending both audio and video, but video
+ * is in a codec you don't support or you don't have bandwidth for both;
+ * or maybe there are 10 participants in the room, but you only want video
+ * from the 3 most active speakers; and so on. The content of the \c streams
+ * array will shape what the SDP offer the plugin will send will look like,
+ * so that eventually a subscription for the specified streams will take place.
+ * Notice that, while for backwards compatibility you can still use the
+ * old \c feed, \c audio, \c video, \c data, \c offer_audio, \c offer_video and
+ * \c offer_data named properties, they're now deprecated and so you're
+ * highly encouraged to use this new drill-down \c streams list instead.
  *
  * As anticipated, if successful this request will generate a new JSEP SDP
  * offer, which will accompany an \c attached event:
@@ -885,10 +966,24 @@ room-<unique room ID>: {
 {
 	"videoroom" : "attached",
 	"room" : <room ID>,
-	"feed" : <publisher ID>,
-	"display" : "<the display name of the publisher, if any>"
+	"streams" : [
+		{
+			"mindex" : <unique m-index of this stream>,
+			"mid" : "<unique mid of this stream>",
+			"type" : "<type of this stream's media (audio|video|data)>",
+			"feed_id" : <unique ID of the publisher originating this stream>,
+			"feed_mid" : "<unique mid of this publisher's stream>",
+			"feed_display" : "<display name of this publisher, if any>",
+			"send" : <true|false; whether we configured the stream to relay media>,
+			"ready" : <true|false; whether this stream is ready to start sending media (will be false at the beginning)>
+		},
+		// Other streams in the subscription, if any
+	]
 }
 \endverbatim
+ *
+ * As you can see, a summary of the streams we subscribed to will be sent back,
+ * which will be useful on the client side for both mapping and rendering purposes.
  *
  * At this stage, to complete the setup of the PeerConnection the subscriber is
  * supposed to send a JSEP SDP answer back to the plugin. This is done
@@ -911,14 +1006,116 @@ room-<unique room ID>: {
 \endverbatim
  *
  * Once this is done, all that's needed is waiting for the WebRTC PeerConnection
- * establishment to succeed. As soon as that happens, the Streaming plugin
- * can start relaying media from the mountpoint the viewer subscribed to
- * to the viewer themselves.
+ * establishment to succeed. As soon as that happens, the VideoRoom plugin
+ * can start relaying media the recipient subscribed to.
+ *
+ * Once a WebRTC PeerConnection has been established for a subscriber, in
+ * case you want to update a subscription you have to use the \c subscribe
+ * and \c unsubscribe methods: as the names of the requests suggest, the
+ * former allows you to add more streams to subscribe to, while the latter
+ * instructs the plugin to remove streams you're currently subscribe to.
+ * Both requests will trigger a renegotiation, if they were successful,
+ * meaning the plugin will send you a new JSEP offer you'll have to reply
+ * to with an answer: to send the answer, just use the same \c start request
+ * we already described above. Notice that renegotiations may not be
+ * triggered right away, e.g., whenever you're trying to update a session
+ * and the plugin is still in the process of renegoting a previous update
+ * for the same subscription: in that case, an update will be scheduled
+ * and a renegotiation will be triggered as soon as it's viable.
+ *
+ * The syntax of the \c subscribe mirrors the one for new subscriptions,
+ * meaning you use the same \c streams array to address the new streams
+ * you want to receive, and formatted the same way:
+ *
+\verbatim
+{
+	"request" : "subscribe",
+	"streams" : [
+		{
+			"feed_id" : <unique ID of publisher owning the new stream to subscribe to>,
+			"mid" : "<unique mid of the publisher stream to subscribe to; optional>"
+			// Optionally, simulcast or SVC targets (defaults if missing)
+		},
+		// Other new streams to subscribe to
+	]
+}
+\endverbatim
+ *
+ * This means the exact same considerations we made on \c streams before
+ * apply here as well: whatever they represent, will indicate the willingness
+ * to subscribe to the related stream. Notice that if you were already
+ * subscribed to one of the new streams indicated here, you'll subscribe
+ * to it again in a different m-line, so it's up to you to ensure you
+ * avoid duplicates (unless that's what you wanted, e.g., for testing
+ * purposes). In case the update was successful, you'll get an \c updated
+ * event, containing the updated layout of all subscriptions (pre-existing
+ * and new ones), and a new JSEP offer to renegotiate the session:
+ *
+\verbatim
+{
+	"videoroom" : "updated",
+	"room" : <room ID>,
+	"streams": [
+		{
+			"mindex" : <unique m-index of this stream>,
+			"mid" : "<unique mid of this stream>",
+			"type" : "<type of this stream's media (audio|video|data)>",
+			"feed_id" : <unique ID of the publisher originating this stream>,
+			"feed_mid" : "<unique mid of this publisher's stream>",
+			"feed_display" : "<display name of this publisher, if any>",
+			"send" : <true|false; whether we configured the stream to relay media>,
+			"ready" : <true|false; whether this stream is ready to start sending media (will be false at the beginning)>
+		},
+		// Other streams in the subscription, if any; old and new
+	]
+}
+\endverbatim
+ *
+ * As explained before, in case the message contains a JSEP offer (which may
+ * not be the case if no change occurred), then clients will need to send
+ * a new JSEP answer with a \c start request to close this renegotiation.
+ *
+ * The \c unsubscribe request works pretty much the same way, with the
+ * difference that the \c streams array you provide to specify what to
+ * unsubscribe from may look different. Specifically, the syntax looks
+ * like this:
  *
- * Notice that the same exact steps we just went through (\c watch request,
- * followed by JSEP offer by the plugin, followed by \c start request with
- * JSEP answer by the viewer) is what you also use when renegotiations are
- * needed, e.g., for the purpose of ICE restarts.
+\verbatim
+{
+	"request" : "unsubscribe",
+	"streams" : [
+		{
+			"feed_id" : <unique ID of publisher owning the new stream to unsubscribe from; optional>,
+			"mid" : "<unique mid of the publisher stream to unsubscribe from; optional>"
+			"sub_mid" : "<unique mid of the subscriber stream to unsubscribe; optional>"
+		},
+		// Other streams to unsubscribe from
+	]
+}
+\endverbatim
+ *
+ * This means that you have different ways to specify what to unsubscribe from:
+ * if an object only specifies \c feed_id, then all the subscription streams that
+ * were receiving media from that publisher will be removed; if an object
+ * specifies \c feed_id and \c mid, then all the subscription streams that
+ * were receiving media from the publisher stream with the related mid will be
+ * removed; finally, if an object only specifies \c sub_mid instead, then
+ * only the stream in the subscription that is addressed by the related mid
+ * (subscription mid, no relation to the publishers') will be removed. As
+ * such, you have a great deal of flexibility in how to unsubscribe from
+ * media. Notice that multiple streams may be removed in case you refer
+ * to the "source" ( \c feed_id ), rather than the "sink" ( \c sub_mid ),
+ * especially in case the subscription contained duplicates or multiple
+ * streams from the same publisher.
+ *
+ * A successful \c unsubscribe will result in exactly the same \c updated
+ * event \c subscribe triggers, so the same considerations apply with
+ * respect to the potential need of a renegotiation and how to complete
+ * it with a \c start along a JSEP answer.
+ *
+ * Notice that, in case you want to trigger an ICE restart rather than
+ * updating a subscription, you'll have to use a different request, named
+ * \c configure: this will be explained in a few paragraphs.
  *
  * As a subscriber, you can temporarily pause and resume the whole media delivery
  * with a \c pause and, again, \c start request (in this case without any JSEP
@@ -963,70 +1160,119 @@ room-<unique room ID>: {
 \verbatim
 {
 	"request" : "configure",
-	"audio" : <true|false, depending on whether audio should be relayed or not; optional>,
-	"video" : <true|false, depending on whether video should be relayed or not; optional>,
-	"data" : <true|false, depending on whether datachannel messages should be relayed or not; optional>,
+	"mid" : <mid of the m-line to refer to for this configure request; optional>,
+	"send" : <true|false, depending on whether the mindex media should be relayed or not; optional>,
 	"substream" : <substream to receive (0-2), in case simulcasting is enabled; optional>,
 	"temporal" : <temporal layers to receive (0-2), in case simulcasting is enabled; optional>,
 	"spatial_layer" : <spatial layer to receive (0-2), in case VP9-SVC is enabled; optional>,
-	"temporal_layer" : <temporal layers to receive (0-2), in case VP9-SVC is enabled; optional>
+	"temporal_layer" : <temporal layers to receive (0-2), in case VP9-SVC is enabled; optional>,
+	"restart" : <trigger an ICE restart; optional>
 }
 \endverbatim
  *
- * As you can see, the \c audio , \c video and \c data properties can be
- * used as a media-level pause/resume functionality, whereas \c pause
+ * As you can see, the \c mid and \c send properties can be used as a media-level
+ * pause/resume functionality ("only mute/unmute this mid"), whereas \c pause
  * and \c start simply pause and resume all streams at the same time.
  * The \c substream and \c temporal properties, instead, only make sense
  * when the mountpoint is configured with video simulcasting support, and
  * as such the viewer is interested in receiving a specific substream
- * or temporal layer, rather than any other of the available ones.
+ * or temporal layer, rather than any other of the available ones: notice
+ * that for them to work you'll have to specify the \c mid as well, as the same
+ * subscription may be receiving simulcast stream from multiple publishers.
  * The \c spatial_layer and \c temporal_layer have exactly the same meaning,
  * but within the context of VP9-SVC publishers, and will have no effect
  * on subscriptions associated to regular publishers.
  *
+ * As anticipated, \c configure is also the request you use when you want
+ * to trigger an ICE restart for a subscriber: in fact, while publishers
+ * can force a restart themselves by providing the right JSEP offer, subscribers
+ * always receive an offer from Janus instead, and as such have to
+ * explicitly ask for a dedicated offer when an ICE restart is needed;
+ * in that case, just set \c restart to \c true in a \c configure request,
+ * and a new JSEP offer with ICE restart information will be sent to the
+ * client, to which the client will have to reply, as usual, via \c start
+ * along a JSEP answer. This documentation doesn't explain when or why
+ * an ICE restart is needed or appropriate: please refer to the ICE RFC
+ * or other sources of information for that.
+ *
  * Another interesting feature that subscribers can take advantage of is the
- * so-called publisher "switching". Basically, when subscribed to a specific
- * publisher and receiving media from them, you can at any time "switch"
- * to a different publisher, and as such start receiving media from that
- * other mountpoint instead. Think of it as changing channel on a TV: you
- * keep on using the same PeerConnection, the plugin simply changes the
- * source of the media transparently. Of course, while powerful and effective
- * this request has some limitations. First of all, it switches both audio
- * and video, meaning you can't just switch video and keep the audio from
- * the previous publisher, for instance; besides, the two publishers
- * must have the same media configuration, that is, use the same codecs,
- * the same payload types, etc. In fact, since the same PeerConnection is
- * used for this feature, switching to a publisher with a different
- * configuration might result in media incompatible with the PeerConnection
- * setup being relayed to the subscriber, and as such in no audio/video being
- * played. That said, a \c switch request must be formatted like this:
+ * so-called publisher "switching". Basically, when subscribed to one or more
+ * publishers and receiving media from them, you can at any time "switch"
+ * any of the subscription streams to a different publisher, and as such
+ * start receiving media on the related m-line from that publisher instead,
+ * all without doing a new \c subscribe or \c unsubscribe, and so without
+ * the need of doing any renegotiation at all; just some logic changes.
+ * Think of it as changing channel on a TV: you keep on using the same
+ * PeerConnection, the plugin simply changes the source of the media
+ * transparently. Of course, while powerful and effective this request has
+ * some limitations: in fact, the source (audio or video) that you switch
+ * to must have the same media configuration (e.g., same codec) as the source
+ * you're replacing. In fact, since the same PeerConnection is used for this
+ * feature and no renegotiation is taking place, switching to a stream with
+ * a different configuration would result in media incompatible with the
+ * PeerConnection setup being relayed to the subscriber (e.g., negotiated
+ * VP9, but new source is H.264), and as such in no audio/video being played;
+ * in that case, you'll need a \c subscribe instead, and a new m-line.
+ *
+ * That said, a \c switch request must be formatted like this:
  *
 \verbatim
 {
 	"request" : "switch",
-	"feed" : <unique ID of the new publisher to switch to; mandatory>,
-	"audio" : <true|false, depending on whether audio should be relayed or not; optional>,
-	"video" : <true|false, depending on whether video should be relayed or not; optional>,
-	"data" : <true|false, depending on whether datachannel messages should be relayed or not; optional>
+	"streams" : [
+		{
+			"feed" : <unique ID of the publisher the new source is from>,
+			"mid" : "<unique mid of the source we want to switch to>",
+			"sub_mid" : "<unique mid of the stream we want to pipe the new source to>"
+		},
+		{
+			// Other updates, if any
+		}
+	]
 }
 \endverbatim
  *
- * If successful, you'll be unsubscribed from the previous publisher,
- * and subscribed to the new publisher instead. The event to confirm
- * the switch was successful will look like this:
+ * While apparently convoluted, this is actually a quite effective and powerful
+ * way of updating subscriptions without renegotiating. In fact, it allows for
+ * full or partial switches: for instance, sometimes you may want to replace all
+ * audio and video streams (e.g., switching from Bob to Alice in a "legacy"
+ * VideoRoom usage, where each PeerConnection subscription is a different
+ * publisher), or just replace a subset of them (e.g., you have a subscription
+ * with three video slots, and you change one of them depending on the loudest
+ * speaker). What to replace is dictated by the \c streams array, where each
+ * object in the array contains all the info needed for the switch to take
+ * place: in particular, you must specify which of your subscription m-lines
+ * you're going to update, via \c sub_mid , and which publisher stream should
+ * now start to feed it via \c feed and \c mid.
+ *
+ * If successful, the specified subscriptions will be updated, meaning they'll
+ * be unsubscribed from the previous publisher stream, and subscribed to the
+ * new publisher stream instead, all without a renegotiation (so no new SDP
+ * offer/answer exchange to take care of). The event to confirm the switch
+ * was successful will look like this:
  *
 \verbatim
 {
 	"videoroom" : "event",
 	"switched" : "ok",
 	"room" : <room ID>,
-	"id" : <unique ID of the new publisher>
+	"changes" : <number of successful changes (may be smaller than the size of the streams array provided in the request)>,
+	"streams" : [
+		// Current configuration of the subscription, same format as when subscribing
+		// Will contain info on all streams, not only those that have been updated
+	]
 }
 \endverbatim
  *
- * Finally, to stop the subscription to the mountpoint and tear down the
- * related PeerConnection, you can use the \c leave request. Since context
- * is implicit, no other argument is required:
+ * Notice that, while a \c switch request usually doesn't require a renegotiation,
+ * it \b MIGHT trigger one nevertheless: in fact, if a "switch" request assigns
+ * a new publisher stream to a previously inactive subscriber stream, then
+ * a renegotiation to re-activate that stream will be needed as well, as
+ * otherwise the packets from the new source will not be relayed.
+ *
+ * Finally, to close a subscription and tear down the related PeerConnection,
+ * you can use the \c leave request. Since context is implicit, no other
+ * argument is required:
  *
 \verbatim
 {
@@ -1066,8 +1312,8 @@ room-<unique room ID>: {
 
 
 /* Plugin information */
-#define JANUS_VIDEOROOM_VERSION			9
-#define JANUS_VIDEOROOM_VERSION_STRING	"0.0.9"
+#define JANUS_VIDEOROOM_VERSION			10
+#define JANUS_VIDEOROOM_VERSION_STRING	"0.0.10"
 #define JANUS_VIDEOROOM_DESCRIPTION		"This is a plugin implementing a videoconferencing SFU (Selective Forwarding Unit) for Janus, that is an audio/video router."
 #define JANUS_VIDEOROOM_NAME			"JANUS VideoRoom plugin"
 #define JANUS_VIDEOROOM_AUTHOR			"Meetecho s.r.l."
@@ -1088,10 +1334,10 @@ void janus_videoroom_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_videoroom_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 json_t *janus_videoroom_handle_admin_message(json_t *message);
 void janus_videoroom_setup_media(janus_plugin_session *handle);
-void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_videoroom_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_videoroom_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_videoroom_incoming_data(janus_plugin_session *handle, char *label, char *buf, int len);
-void janus_videoroom_slow_link(janus_plugin_session *handle, int uplink, int video);
+void janus_videoroom_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink);
 void janus_videoroom_hangup_media(janus_plugin_session *handle);
 void janus_videoroom_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_videoroom_query_session(janus_plugin_session *handle);
@@ -1131,158 +1377,212 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter adminkey_parameters[] = {
-	{"admin_key", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"admin_key", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 static struct janus_json_parameter create_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"description", JSON_STRING, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"description", JANUS_JSON_STRING, 0},
 	{"is_private", JANUS_JSON_BOOL, 0},
-	{"allowed", JSON_ARRAY, 0},
-	{"secret", JSON_STRING, 0},
-	{"pin", JSON_STRING, 0},
+	{"allowed", JANUS_JSON_ARRAY, 0},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"pin", JANUS_JSON_STRING, 0},
 	{"require_pvtid", JANUS_JSON_BOOL, 0},
-	{"bitrate", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"bitrate", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"bitrate_cap", JANUS_JSON_BOOL, 0},
-	{"fir_freq", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"publishers", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"audiocodec", JSON_STRING, 0},
-	{"videocodec", JSON_STRING, 0},
+	{"pli_freq", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"fir_freq", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},	/* Deprecated! */
+	{"publishers", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audiocodec", JANUS_JSON_STRING, 0},
+	{"videocodec", JANUS_JSON_STRING, 0},
 	{"opus_fec", JANUS_JSON_BOOL, 0},
 	{"video_svc", JANUS_JSON_BOOL, 0},
 	{"audiolevel_ext", JANUS_JSON_BOOL, 0},
 	{"audiolevel_event", JANUS_JSON_BOOL, 0},
-	{"audio_active_packets", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"audio_level_average", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audio_active_packets", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audio_level_average", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"videoorient_ext", JANUS_JSON_BOOL, 0},
 	{"playoutdelay_ext", JANUS_JSON_BOOL, 0},
 	{"transport_wide_cc_ext", JANUS_JSON_BOOL, 0},
 	{"record", JANUS_JSON_BOOL, 0},
-	{"rec_dir", JSON_STRING, 0},
+	{"rec_dir", JANUS_JSON_STRING, 0},
 	{"permanent", JANUS_JSON_BOOL, 0},
 	{"notify_joining", JANUS_JSON_BOOL, 0},
 };
 static struct janus_json_parameter edit_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"new_description", JSON_STRING, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"new_description", JANUS_JSON_STRING, 0},
 	{"new_is_private", JANUS_JSON_BOOL, 0},
-	{"new_secret", JSON_STRING, 0},
-	{"new_pin", JSON_STRING, 0},
+	{"new_secret", JANUS_JSON_STRING, 0},
+	{"new_pin", JANUS_JSON_STRING, 0},
 	{"new_require_pvtid", JANUS_JSON_BOOL, 0},
-	{"new_bitrate", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"new_fir_freq", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"new_publishers", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"new_bitrate", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"new_pli_freq", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"new_fir_freq", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},	/* Deprecated! */
+	{"new_publishers", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter room_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter destroy_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
 	{"permanent", JANUS_JSON_BOOL, 0}
 };
 static struct janus_json_parameter allowed_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"action", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"allowed", JSON_ARRAY, 0}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"action", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"allowed", JANUS_JSON_ARRAY, 0}
 };
 static struct janus_json_parameter kick_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"secret", JSON_STRING, 0},
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"secret", JANUS_JSON_STRING, 0},
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter join_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"ptype", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"audio", JANUS_JSON_BOOL, 0},
-	{"video", JANUS_JSON_BOOL, 0},
-	{"data", JANUS_JSON_BOOL, 0},
-	{"bitrate", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"ptype", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"descriptions", JANUS_JSON_ARRAY, 0},
+	{"audio", JANUS_JSON_BOOL, 0},	/* Deprecated! */
+	{"video", JANUS_JSON_BOOL, 0},	/* Deprecated! */
+	{"data", JANUS_JSON_BOOL, 0},	/* Deprecated! */
+	{"bitrate", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"record", JANUS_JSON_BOOL, 0},
-	{"filename", JSON_STRING, 0},
-	{"token", JSON_STRING, 0}
+	{"filename", JANUS_JSON_STRING, 0},
+	{"token", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter publish_parameters[] = {
-	{"audio", JANUS_JSON_BOOL, 0},
-	{"audiocodec", JSON_STRING, 0},
-	{"video", JANUS_JSON_BOOL, 0},
-	{"videocodec", JSON_STRING, 0},
-	{"data", JANUS_JSON_BOOL, 0},
-	{"bitrate", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"descriptions", JANUS_JSON_ARRAY, 0},
+	{"audiocodec", JANUS_JSON_STRING, 0},
+	{"videocodec", JANUS_JSON_STRING, 0},
+	{"bitrate", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"keyframe", JANUS_JSON_BOOL, 0},
 	{"record", JANUS_JSON_BOOL, 0},
-	{"filename", JSON_STRING, 0},
-	{"display", JSON_STRING, 0},
+	{"filename", JANUS_JSON_STRING, 0},
+	{"display", JANUS_JSON_STRING, 0},
+	/* Only needed when configuring, to make a stream active/inactive */
+	{"mid", JANUS_JSON_STRING, 0},
+	{"send", JANUS_JSON_BOOL, 0},
+	/* Deprecated, use mid+send instead */
+	{"audio", JANUS_JSON_BOOL, 0},	/* Deprecated! */
+	{"video", JANUS_JSON_BOOL, 0},	/* Deprecated! */
+	{"data", JANUS_JSON_BOOL, 0},	/* Deprecated! */
 	/* The following are just to force a renegotiation and/or an ICE restart */
 	{"update", JANUS_JSON_BOOL, 0},
 	{"restart", JANUS_JSON_BOOL, 0}
 };
+static struct janus_json_parameter publish_desc_parameters[] = {
+	{"mid", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"description", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+};
 static struct janus_json_parameter rtp_forward_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"publisher_id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"video_port", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_rtcp_port", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_ssrc", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_pt", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_port_2", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_ssrc_2", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_pt_2", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_port_3", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_ssrc_3", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"video_pt_3", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"audio_port", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"audio_rtcp_port", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"audio_ssrc", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"audio_pt", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"data_port", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"host", JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
-	{"host_family", JSON_STRING, 0},
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"publisher_id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"host", JANUS_JSON_STRING, 0},
+	{"host_family", JANUS_JSON_STRING, 0},
+	{"simulcast", JANUS_JSON_BOOL, 0},
+	{"srtp_suite", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"srtp_crypto", JANUS_JSON_STRING, 0},
+	{"streams", JANUS_JSON_ARRAY, 0},
+	/* Deprecated parameters, use the streams array instead */
+	{"video_port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_rtcp_port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_ssrc", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_pt", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_port_2", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_ssrc_2", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_pt_2", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_port_3", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_ssrc_3", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"video_pt_3", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audio_port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audio_rtcp_port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audio_ssrc", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"audio_pt", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"data_port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+};
+static struct janus_json_parameter rtp_forward_stream_parameters[] = {
+	{"host", JANUS_JSON_STRING, 0},
+	{"host_family", JANUS_JSON_STRING, 0},
+	{"port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"rtcp_port", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"ssrc", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"pt", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"simulcast", JANUS_JSON_BOOL, 0},
-	{"srtp_suite", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"srtp_crypto", JSON_STRING, 0}
+	{"port_2", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"ssrc_2", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"pt_2", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"port_3", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"ssrc_3", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"pt_3", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter stop_rtp_forward_parameters[] = {
-	{"room", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"publisher_id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"stream_id", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
+	{"room", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"publisher_id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"stream_id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE}
 };
 static struct janus_json_parameter publisher_parameters[] = {
-	{"id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"display", JSON_STRING, 0}
+	{"id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"display", JANUS_JSON_STRING, 0}
 };
 static struct janus_json_parameter configure_parameters[] = {
-	{"audio", JANUS_JSON_BOOL, 0},
-	{"video", JANUS_JSON_BOOL, 0},
-	{"data", JANUS_JSON_BOOL, 0},
+	{"mid", JANUS_JSON_STRING, 0},
+	{"send", JANUS_JSON_BOOL, 0},
 	/* For VP8 (or H.264) simulcast */
-	{"substream", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"temporal", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"substream", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"temporal", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	/* For VP9 SVC */
-	{"spatial_layer", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"temporal_layer", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"spatial_layer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"temporal_layer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	/* The following is to handle a renegotiation */
 	{"update", JANUS_JSON_BOOL, 0},
+	/* Deprecated properties, use mid+send instead */
+	{"audio", JANUS_JSON_BOOL, 0},	/* Deprecated */
+	{"video", JANUS_JSON_BOOL, 0},	/* Deprecated */
+	{"data", JANUS_JSON_BOOL, 0}	/* Deprecated */
 };
 static struct janus_json_parameter subscriber_parameters[] = {
-	{"feed", JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
-	{"private_id", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"streams", JANUS_JSON_ARRAY, 0},
+	{"feed", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},	/* Deprecated! Use feed in streams instead */
+	{"private_id", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	{"close_pc", JANUS_JSON_BOOL, 0},
+	/* All the following parameters are deprecated: use streams instead */
 	{"audio", JANUS_JSON_BOOL, 0},
 	{"video", JANUS_JSON_BOOL, 0},
 	{"data", JANUS_JSON_BOOL, 0},
 	{"offer_audio", JANUS_JSON_BOOL, 0},
 	{"offer_video", JANUS_JSON_BOOL, 0},
-	{"offer_data", JANUS_JSON_BOOL, 0},
+	{"offer_data", JANUS_JSON_BOOL, 0}
+};
+static struct janus_json_parameter subscriber_stream_parameters[] = {
+	{"feed", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"mid", JANUS_JSON_STRING, 0},
 	/* For VP8 (or H.264) simulcast */
-	{"substream", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"temporal", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"substream", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"temporal", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
 	/* For VP9 SVC */
-	{"spatial_layer", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
-	{"temporal_layer", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"spatial_layer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"temporal_layer", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE}
+};
+static struct janus_json_parameter subscriber_update_parameters[] = {
+	{"streams", JANUS_JSON_ARRAY, JANUS_JSON_PARAM_REQUIRED}
+};
+static struct janus_json_parameter subscriber_remove_parameters[] = {
+	{"feed", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
+	{"mid", JANUS_JSON_STRING, 0},
+	{"sub_mid", JANUS_JSON_STRING, 0}
+};
+static struct janus_json_parameter switch_parameters[] = {
+	{"streams", JANUS_JSON_ARRAY, 0}
+};
+static struct janus_json_parameter switch_update_parameters[] = {
+	{"feed", JANUS_JSON_INTEGER, JANUS_JSON_PARAM_REQUIRED | JANUS_JSON_PARAM_POSITIVE},
+	{"mid", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED},
+	{"sub_mid", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 
 /* Static configuration instance */
@@ -1306,6 +1606,35 @@ typedef enum janus_videoroom_p_type {
 	janus_videoroom_p_type_publisher,			/* Participant (for receiving events) and optionally publisher */
 } janus_videoroom_p_type;
 
+typedef enum janus_videoroom_media {
+	JANUS_VIDEOROOM_MEDIA_NONE = 0,
+	JANUS_VIDEOROOM_MEDIA_AUDIO,
+	JANUS_VIDEOROOM_MEDIA_VIDEO,
+	JANUS_VIDEOROOM_MEDIA_DATA
+} janus_videoroom_media;
+static const char *janus_videoroom_media_str(janus_videoroom_media type) {
+	switch(type) {
+		case JANUS_VIDEOROOM_MEDIA_AUDIO: return "audio";
+		case JANUS_VIDEOROOM_MEDIA_VIDEO: return "video";
+		case JANUS_VIDEOROOM_MEDIA_DATA: return "data";
+		case JANUS_VIDEOROOM_MEDIA_NONE:
+		default:
+			break;
+	}
+	return NULL;
+}
+static janus_sdp_mtype janus_videoroom_media_sdptype(janus_videoroom_media type) {
+	switch(type) {
+		case JANUS_VIDEOROOM_MEDIA_AUDIO: return JANUS_SDP_AUDIO;
+		case JANUS_VIDEOROOM_MEDIA_VIDEO: return JANUS_SDP_VIDEO;
+		case JANUS_VIDEOROOM_MEDIA_DATA: return JANUS_SDP_APPLICATION;
+		case JANUS_VIDEOROOM_MEDIA_NONE:
+		default:
+			break;
+	}
+	return JANUS_SDP_OTHER;
+}
+
 typedef struct janus_videoroom_message {
 	janus_plugin_session *handle;
 	char *transaction;
@@ -1326,7 +1655,7 @@ typedef struct janus_videoroom {
 	int max_publishers;			/* Maximum number of concurrent publishers */
 	uint32_t bitrate;			/* Global bitrate limit */
 	gboolean bitrate_cap;		/* Whether the above limit is insormountable */
-	uint16_t fir_freq;			/* Regular FIR frequency (0=disabled) */
+	uint16_t pli_freq;			/* Regular PLI frequency (0=disabled) */
 	janus_audiocodec acodec[3];	/* Audio codec(s) to force on publishers */
 	janus_videocodec vcodec[3];	/* Video codec(s) to force on publishers */
 	gboolean do_opusfec;		/* Whether inband FEC must be negotiated (note: only available for Opus) */
@@ -1374,6 +1703,7 @@ static janus_mutex sessions_mutex = JANUS_MUTEX_INITIALIZER;
 typedef struct janus_videoroom_srtp_context janus_videoroom_srtp_context;
 typedef struct janus_videoroom_rtp_forwarder {
 	void *source;
+	guint32 stream_id;
 	gboolean is_video;
 	gboolean is_data;
 	uint32_t ssrc;
@@ -1448,89 +1778,130 @@ static void *janus_videoroom_rtp_forwarder_rtcp_thread(void *data);
 
 typedef struct janus_videoroom_publisher {
 	janus_videoroom_session *session;
-	janus_videoroom *room;	/* Room */
-	guint64 room_id;	/* Unique room ID */
-	guint64 user_id;	/* Unique ID in the room */
-	guint32 pvt_id;		/* This is sent to the publisher for mapping purposes, but shouldn't be shared with others */
-	gchar *display;		/* Display name (just for fun) */
-	gchar *sdp;			/* The SDP this publisher negotiated, if any */
-	gboolean audio, video, data;		/* Whether audio, video and/or data is going to be sent by this publisher */
-	janus_audiocodec acodec;	/* Audio codec this publisher is using */
-	janus_videocodec vcodec;	/* Video codec this publisher is using */
-	guint32 audio_pt;		/* Audio payload type (Opus) */
-	guint32 video_pt;		/* Video payload type (depends on room configuration) */
-	guint32 audio_ssrc;		/* Audio SSRC of this publisher */
-	guint32 video_ssrc;		/* Video SSRC of this publisher */
-	gboolean do_opusfec;	/* Whether this publisher is sending inband Opus FEC */
-	uint32_t ssrc[3];		/* Only needed in case VP8 (or H.264) simulcasting is involved */
-	char *rid[3];			/* Only needed if simulcasting is rid-based */
-	int rid_extmap_id;		/* rid extmap ID */
-	int framemarking_ext_id;			/* Frame marking extmap ID */
-	guint8 audio_level_extmap_id;		/* Audio level extmap ID */
-	guint8 video_orient_extmap_id;		/* Video orientation extmap ID */
-	guint8 playout_delay_extmap_id;		/* Playout delay extmap ID */
-	gboolean audio_active;
-	gboolean video_active;
-	int audio_dBov_level;		/* Value in dBov of the audio level (last value from extension) */
-	int audio_active_packets;	/* Participant's number of audio packets to accumulate */
-	int audio_dBov_sum;			/* Participant's accumulated dBov value for audio level*/
-	gboolean talking;			/* Whether this participant is currently talking (uses audio levels extension) */
-	gboolean data_active;
-	gboolean firefox;	/* We send Firefox users a different kind of FIR */
+	janus_videoroom *room;		/* Room */
+	guint64 room_id;			/* Unique room ID */
+	guint64 user_id;			/* Unique ID in the room */
+	guint32 pvt_id;				/* This is sent to the publisher for mapping purposes, but shouldn't be shared with others */
+	gchar *display;				/* Display name (just for fun) */
+	gboolean firefox;			/* Firefox and Chrome use different b= attributes (TIAS vs AS) */
+	GList *streams;				/* List of media streams sent by this publisher (audio, video and/or data) */
+	GHashTable *streams_byid;	/* As above, indexed by mindex */
+	GHashTable *streams_bymid;	/* As above, indexed by mid */
+	int data_mindex;			/* We keep track of the mindex for data, as there can only be one */
+	janus_mutex streams_mutex;
 	uint32_t bitrate;
-	gint64 remb_startup;/* Incremental changes on REMB to reach the target at startup */
-	gint64 remb_latest;	/* Time of latest sent REMB (to avoid flooding) */
-	gint64 fir_latest;	/* Time of latest sent FIR (to avoid flooding) */
-	gint fir_seq;		/* FIR sequence number */
+	gint64 remb_startup;		/* Incremental changes on REMB to reach the target at startup */
+	gint64 remb_latest;			/* Time of latest sent REMB (to avoid flooding) */
 	gboolean recording_active;	/* Whether this publisher has to be recorded or not */
-	gchar *recording_base;	/* Base name for the recording (e.g., /path/to/filename, will generate /path/to/filename-audio.mjr and/or /path/to/filename-video.mjr */
-	janus_recorder *arc;	/* The Janus recorder instance for this publisher's audio, if enabled */
-	janus_recorder *vrc;	/* The Janus recorder instance for this user's video, if enabled */
-	janus_recorder *drc;	/* The Janus recorder instance for this publisher's data, if enabled */
-	janus_rtp_switching_context rec_ctx;
-	janus_rtp_simulcasting_context rec_simctx;
-	janus_mutex rec_mutex;	/* Mutex to protect the recorders from race conditions */
-	GSList *subscribers;	/* Subscriptions to this publisher (who's watching this publisher)  */
-	GSList *subscriptions;	/* Subscriptions this publisher has created (who this publisher is watching) */
+	gchar *recording_base;		/* Base name for the recording (e.g., /path/to/filename, will generate /path/to/filename-audio.mjr and/or /path/to/filename-video.mjr */
+	janus_mutex rec_mutex;		/* Mutex to protect the recorders from race conditions */
+	GSList *subscriptions;		/* Subscriptions this publisher has created (who this publisher is watching) */
 	janus_mutex subscribers_mutex;
+	GHashTable *srtp_contexts;	/* SRTP contexts that we can share among RTP forwarders */
+	/* Index of RTP (or data) forwarders for this stream, if any */
 	GHashTable *rtp_forwarders;
-	GHashTable *srtp_contexts;
 	janus_mutex rtp_forwarders_mutex;
-	int udp_sock; /* The udp socket on which to forward rtp packets */
-	gboolean kicked;	/* Whether this participant has been kicked */
+	int udp_sock; 				/* The udp socket on which to forward rtp packets */
+	gboolean kicked;			/* Whether this participant has been kicked */
 	volatile gint destroyed;
 	janus_refcount ref;
 } janus_videoroom_publisher;
-static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publisher *p,
+/* Each VideoRoom publisher can share multiple streams, so each stream is its own structure */
+typedef struct janus_videoroom_publisher_stream {
+	janus_videoroom_publisher *publisher;	/* Publisher instance this stream belongs to */
+	janus_videoroom_media type;				/* Type of this stream (audio, video or data) */
+	int mindex;								/* mindex of this stream */
+	char *mid;								/* mid of this stream */
+	char *description;						/* Description of this stream (user provided) */
+	gboolean active;						/* Whether this stream is active or not */
+	janus_audiocodec acodec;				/* Audio codec this publisher is using (if audio) */
+	janus_videocodec vcodec;				/* Video codec this publisher is using (if video) */
+	int pt;									/* Payload type of this stream (if audio or video) */
+	guint32 ssrc;							/* Internal SSRC of this stream */
+	gint64 pli_latest;						/* Time of latest sent PLI (to avoid flooding) */
+	gboolean opusfec;						/* Whether this stream is sending inband Opus FEC */
+	gboolean simulcast, svc;				/* Whether this stream uses simulcast or VP9 SVC */
+	uint32_t vssrc[3];						/* Only needed in case VP8 (or H.264) simulcasting is involved */
+	char *rid[3];							/* Only needed if simulcasting is rid-based */
+	int rid_extmap_id;						/* rid extmap ID */
+	int framemarking_ext_id;				/* Frame marking extmap ID */
+	/* RTP extensions, if negotiated */
+	guint8 audio_level_extmap_id;			/* Audio level extmap ID */
+	guint8 video_orient_extmap_id;			/* Video orientation extmap ID */
+	guint8 playout_delay_extmap_id;			/* Playout delay extmap ID */
+	janus_sdp_mdirection audio_level_mdir, video_orient_mdir, playout_delay_mdir;
+	/* Audio level processing, if enabled */
+	int audio_dBov_level;					/* Value in dBov of the audio level (last value from extension) */
+	int audio_active_packets;				/* Participant's number of audio packets to accumulate */
+	int audio_dBov_sum;						/* Participant's accumulated dBov value for audio level */
+	gboolean talking;						/* Whether this participant is currently talking (uses audio levels extension) */
+	/* Recording related stuff, if enabled */
+	janus_recorder *rc;
+	janus_rtp_switching_context rec_ctx;
+	janus_rtp_simulcasting_context rec_simctx;
+	/* RTP (or data) forwarders for this stream, if any */
+	GHashTable *rtp_forwarders;
+	janus_mutex rtp_forwarders_mutex;
+	/* Subscriptions to this publisher stream (who's receiving it)  */
+	GSList *subscribers;
+	janus_mutex subscribers_mutex;
+	volatile gint destroyed;
+	janus_refcount ref;
+} janus_videoroom_publisher_stream;
+static janus_videoroom_rtp_forwarder *janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publisher *p,
+	janus_videoroom_publisher_stream *stream,
 	const gchar *host, int port, int rtcp_port, int pt, uint32_t ssrc,
 	gboolean simulcast, int srtp_suite, const char *srtp_crypto,
 	int substream, gboolean is_video, gboolean is_data);
+static json_t *janus_videoroom_rtp_forwarder_summary(janus_videoroom_rtp_forwarder *f);
 
 typedef struct janus_videoroom_subscriber {
 	janus_videoroom_session *session;
-	janus_videoroom *room;	/* Room */
-	guint64 room_id;		/* Unique room ID */
-	janus_videoroom_publisher *feed;	/* Participant this subscriber is subscribed to */
-	gboolean close_pc;		/* Whether we should automatically close the PeerConnection when the publisher goes away */
-	guint32 pvt_id;			/* Private ID of the participant that is subscribing (if available/provided) */
-	janus_sdp *sdp;			/* Offer we sent this listener (may be updated within renegotiations) */
-	janus_rtp_switching_context context;	/* Needed in case there are publisher switches on this subscriber */
+	janus_videoroom *room;		/* Room */
+	guint64 room_id;			/* Unique room ID */
+	GList *streams;				/* List of media stream subscriptions originated by this subscriber (audio, video and/or data) */
+	GHashTable *streams_byid;	/* As above, indexed by mindex */
+	GHashTable *streams_bymid;	/* As above, indexed by mid */
+	janus_mutex streams_mutex;
+	gboolean close_pc;			/* Whether we should automatically close the PeerConnection when the last stream goes away */
+	guint32 pvt_id;				/* Private ID of the participant that is subscribing (if available/provided) */
+	gboolean paused;
+	gboolean kicked;			/* Whether this subscription belongs to a participant that has been kicked */
+	volatile gint answered, pending_offer, pending_restart;
+	volatile gint destroyed;
+	janus_refcount ref;
+} janus_videoroom_subscriber;
+/* Each VideoRoom subscriber can be subscribed to multiple streams, belonging to
+ * the same or different publishers: as such, each stream is its own structure */
+typedef struct janus_videoroom_subscriber_stream {
+	janus_videoroom_subscriber *subscriber;			/* Subscriber instance this stream belongs to */
+	GSList *publisher_streams;						/* Complete list of publisher streams (e.g., when this is data) */
+	int mindex;				/* The media index of this stream (may not be the same as the publisher stream) */
+	char *mid;				/* The mid of this stream (may not be the same as the publisher stream) */
+	gboolean send;			/* Whether this stream media must be sent to this subscriber */
+	/* The following properties are copied from the source, in case this stream becomes inactive */
+	janus_videoroom_media type;			/* Type of this stream (audio, video or data) */
+	janus_audiocodec acodec;			/* Audio codec this publisher is using (if audio) */
+	janus_videocodec vcodec;			/* Video codec this publisher is using (if video) */
+	int pt;								/* Payload type of this stream (if audio or video) */
+	gboolean opusfec;					/* Whether this stream is using inband Opus FEC */
+	guint8 audio_level_extmap_id;		/* Audio level extmap ID */
+	guint8 video_orient_extmap_id;		/* Video orientation extmap ID */
+	guint8 playout_delay_extmap_id;		/* Playout delay extmap ID */
+	/* RTP and simulcasting contexts */
+	janus_rtp_switching_context context;
 	janus_rtp_simulcasting_context sim_context;
 	janus_vp8_simulcast_context vp8_context;
-	gboolean audio, video, data;		/* Whether audio, video and/or data must be sent to this subscriber */
-	/* As above, but can't change dynamically (says whether something was negotiated at all in SDP) */
-	gboolean audio_offered, video_offered, data_offered;
-	gboolean paused;
-	gboolean kicked;	/* Whether this subscription belongs to a participant that has been kicked */
 	/* The following are only relevant if we're doing VP9 SVC, and are not to be confused with plain
 	 * simulcast, which has similar info (substream/templayer) but in a completely different context */
 	int spatial_layer, target_spatial_layer;
 	int temporal_layer, target_temporal_layer;
-	volatile gint destroyed;
+	volatile gint ready, destroyed;
 	janus_refcount ref;
-} janus_videoroom_subscriber;
+} janus_videoroom_subscriber_stream;
 
 typedef struct janus_videoroom_rtp_relay_packet {
+	janus_videoroom_publisher_stream *source;
 	janus_rtp_header *data;
 	gint length;
 	gboolean is_video;
@@ -1544,8 +1915,33 @@ typedef struct janus_videoroom_rtp_relay_packet {
 	uint8_t pbit, dbit, ubit, bbit, ebit;
 } janus_videoroom_rtp_relay_packet;
 
+typedef struct janus_videoroom_data_relay_packet {
+	janus_videoroom_publisher_stream *source;
+	char *text;
+} janus_videoroom_data_relay_packet;
+
 
 /* Freeing stuff */
+static void janus_videoroom_subscriber_stream_destroy(janus_videoroom_subscriber_stream *s) {
+	if(s && g_atomic_int_compare_and_exchange(&s->destroyed, 0, 1))
+		janus_refcount_decrease(&s->ref);
+	/* TODO Should unref the subscriber instance? */
+}
+
+static void janus_videoroom_subscriber_stream_unref(janus_videoroom_subscriber_stream *s) {
+	/* Decrease the counter */
+	if(s)
+		janus_refcount_decrease(&s->ref);
+}
+
+static void janus_videoroom_subscriber_stream_free(const janus_refcount *s_ref) {
+	janus_videoroom_subscriber_stream *s = janus_refcount_containerof(s_ref, janus_videoroom_subscriber_stream, ref);
+	/* This subscriber stream can be destroyed, free all the resources */
+		/* TODO Anything else we should free? */
+	g_free(s->mid);
+	g_free(s);
+}
+
 static void janus_videoroom_subscriber_destroy(janus_videoroom_subscriber *s) {
 	if(s && g_atomic_int_compare_and_exchange(&s->destroyed, 0, 1))
 		janus_refcount_decrease(&s->ref);
@@ -1554,18 +1950,53 @@ static void janus_videoroom_subscriber_destroy(janus_videoroom_subscriber *s) {
 static void janus_videoroom_subscriber_free(const janus_refcount *s_ref) {
 	janus_videoroom_subscriber *s = janus_refcount_containerof(s_ref, janus_videoroom_subscriber, ref);
 	/* This subscriber can be destroyed, free all the resources */
-	janus_sdp_destroy(s->sdp);
+	/* TODO Get rid of all the streams */
+	g_list_free_full(s->streams, (GDestroyNotify)(janus_videoroom_subscriber_stream_destroy));
+	g_hash_table_unref(s->streams_byid);
+	g_hash_table_unref(s->streams_bymid);
+		/* TODO Unref the publisher stream? */
+
 	g_free(s);
 }
 
-static void janus_videoroom_publisher_dereference(janus_videoroom_publisher *p) {
-	/* This is used by g_pointer_clear and g_hash_table_new_full so that NULL is only possible if that was inserted into the hash table. */
-	janus_refcount_decrease(&p->ref);
+static void janus_videoroom_publisher_stream_destroy(janus_videoroom_publisher_stream *ps) {
+	if(ps && g_atomic_int_compare_and_exchange(&ps->destroyed, 0, 1)) {
+		if(ps->publisher)
+			janus_refcount_decrease(&ps->publisher->ref);
+		ps->publisher = NULL;
+		janus_refcount_decrease(&ps->ref);
+	}
+	/* TODO Should unref the publisher instance? */
+}
+
+static void janus_videoroom_publisher_stream_unref(janus_videoroom_publisher_stream *ps) {
+	/* Decrease the counter */
+	if(ps)
+		janus_refcount_decrease(&ps->ref);
 }
 
-static void janus_videoroom_publisher_dereference_by_subscriber(janus_videoroom_publisher *p) {
+static void janus_videoroom_publisher_stream_free(const janus_refcount *ps_ref) {
+	janus_videoroom_publisher_stream *ps = janus_refcount_containerof(ps_ref, janus_videoroom_publisher_stream, ref);
+	/* This publisher stream can be destroyed, free all the resources */
+		/* TODO Anything else we should free? */
+	g_free(ps->mid);
+	g_free(ps->description);
+	janus_recorder_destroy(ps->rc);
+	g_hash_table_destroy(ps->rtp_forwarders);
+	ps->rtp_forwarders = NULL;
+	janus_mutex_destroy(&ps->rtp_forwarders_mutex);
+	g_slist_free(ps->subscribers);
+	janus_mutex_destroy(&ps->subscribers_mutex);
+	int i=0;
+	for(i=0; i<3; i++) {
+		g_free(ps->rid[i]);
+		ps->rid[i] = NULL;
+	}
+	g_free(ps);
+}
+
+static void janus_videoroom_publisher_dereference(janus_videoroom_publisher *p) {
 	/* This is used by g_pointer_clear and g_hash_table_new_full so that NULL is only possible if that was inserted into the hash table. */
-	janus_refcount_decrease(&p->session->ref);
 	janus_refcount_decrease(&p->ref);
 }
 
@@ -1582,24 +2013,22 @@ static void janus_videoroom_publisher_free(const janus_refcount *p_ref) {
 	janus_videoroom_publisher *p = janus_refcount_containerof(p_ref, janus_videoroom_publisher, ref);
 	g_free(p->display);
 	p->display = NULL;
-	g_free(p->sdp);
-	p->sdp = NULL;
 	g_free(p->recording_base);
 	p->recording_base = NULL;
-	janus_recorder_destroy(p->arc);
-	janus_recorder_destroy(p->vrc);
-	janus_recorder_destroy(p->drc);
+	/* Get rid of all the streams */
+	g_list_free_full(p->streams, (GDestroyNotify)(janus_videoroom_publisher_stream_destroy));
+	g_hash_table_unref(p->streams_byid);
+	g_hash_table_unref(p->streams_bymid);
 
 	if(p->udp_sock > 0)
 		close(p->udp_sock);
 	g_hash_table_destroy(p->rtp_forwarders);
-	p->rtp_forwarders = NULL;
+	janus_mutex_destroy(&p->rtp_forwarders_mutex);
 	g_hash_table_destroy(p->srtp_contexts);
 	p->srtp_contexts = NULL;
-	g_slist_free(p->subscribers);
 
+	g_slist_free(p->subscriptions);
 	janus_mutex_destroy(&p->subscribers_mutex);
-	janus_mutex_destroy(&p->rtp_forwarders_mutex);
 	g_free(p);
 }
 
@@ -1688,18 +2117,15 @@ static void janus_videoroom_codecstr(janus_videoroom *videoroom, char *audio_cod
 	}
 }
 
-static void janus_videoroom_reqfir(janus_videoroom_publisher *publisher, const char *reason) {
-	/* Send a FIR */
-	char buf[20];
-	janus_rtcp_fir((char *)&buf, 20, &publisher->fir_seq);
-	JANUS_LOG(LOG_VERB, "%s sending FIR to %"SCNu64" (%s)\n", reason, publisher->user_id, publisher->display ? publisher->display : "??");
-	gateway->relay_rtcp(publisher->session->handle, 1, buf, 20);
-	/* Send a PLI too, just in case... */
+static void janus_videoroom_reqpli(janus_videoroom_publisher_stream *ps, const char *reason) {
+	/* Send a PLI */
+	char buf[12];
 	janus_rtcp_pli((char *)&buf, 12);
-	JANUS_LOG(LOG_VERB, "%s sending PLI to %"SCNu64" (%s)\n", reason, publisher->user_id, publisher->display ? publisher->display : "??");
-	gateway->relay_rtcp(publisher->session->handle, 1, buf, 12);
+	JANUS_LOG(LOG_VERB, "%s sending PLI to %"SCNu64" (%s)\n", reason,
+		ps->publisher->user_id, ps->publisher->display ? ps->publisher->display : "??");
+	gateway->relay_rtcp(ps->publisher->session->handle, ps->mindex, TRUE, buf, 12);
 	/* Update the time of when we last sent a keyframe request */
-	publisher->fir_latest = janus_get_monotonic_time();
+	ps->pli_latest = janus_get_monotonic_time();
 }
 
 /* Error codes */
@@ -1723,14 +2149,16 @@ static void janus_videoroom_reqfir(janus_videoroom_publisher *publisher, const c
 #define JANUS_VIDEOROOM_ERROR_INVALID_SDP		437
 
 
-static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publisher *p,
+/* RTP forwarder helpers */
+static janus_videoroom_rtp_forwarder *janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publisher *p,
+		janus_videoroom_publisher_stream *stream,
 		const gchar *host, int port, int rtcp_port, int pt, uint32_t ssrc,
 		gboolean simulcast, int srtp_suite, const char *srtp_crypto,
 		int substream, gboolean is_video, gboolean is_data) {
-	if(!p || !host) {
-		return 0;
+	if(!p || !stream || !host) {
+		return NULL;
 	}
-	janus_mutex_lock(&p->rtp_forwarders_mutex);
+	janus_mutex_lock(&stream->rtp_forwarders_mutex);
 	/* Do we need to bind to a port for RTCP? */
 	int fd = -1;
 	uint16_t local_rtcp_port = 0;
@@ -1740,7 +2168,7 @@ static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publishe
 			janus_mutex_unlock(&p->rtp_forwarders_mutex);
 			JANUS_LOG(LOG_ERR, "Error creating RTCP socket for new RTP forwarder... %d (%s)\n",
 				errno, strerror(errno));
-			return 0;
+			return NULL;
 		}
 		int v6only = 0;
 		if(setsockopt(fd, IPPROTO_IPV6, IPV6_V6ONLY, &v6only, sizeof(v6only)) != 0) {
@@ -1761,14 +2189,14 @@ static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publishe
 			JANUS_LOG(LOG_ERR, "Error binding RTCP socket for new RTP forwarder... %d (%s)\n",
 				errno, strerror(errno));
 			close(fd);
-			return 0;
+			return NULL;
 		}
 		local_rtcp_port = ntohs(address.sin6_port);
 		JANUS_LOG(LOG_VERB, "Bound local %s RTCP port: %"SCNu16"\n",
 			is_video ? "video" : "audio", local_rtcp_port);
 	}
 	janus_videoroom_rtp_forwarder *forward = g_malloc0(sizeof(janus_videoroom_rtp_forwarder));
-	forward->source = p;
+	forward->source = stream;
 	forward->rtcp_fd = fd;
 	forward->local_rtcp_port = local_rtcp_port;
 	forward->remote_rtcp_port = rtcp_port;
@@ -1797,14 +2225,14 @@ static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publishe
 			gsize len = 0;
 			guchar *decoded = g_base64_decode(srtp_crypto, &len);
 			if(len < SRTP_MASTER_LENGTH) {
-				janus_mutex_unlock(&p->rtp_forwarders_mutex);
+				janus_mutex_unlock(&stream->rtp_forwarders_mutex);
 				JANUS_LOG(LOG_ERR, "Invalid SRTP crypto (%s)\n", srtp_crypto);
 				g_free(decoded);
 				g_free(srtp_ctx);
 				if(forward->rtcp_fd > -1)
 					close(forward->rtcp_fd);
 				g_free(forward);
-				return 0;
+				return NULL;
 			}
 			/* Set SRTP policy */
 			srtp_policy_t *policy = &srtp_ctx->policy;
@@ -1821,7 +2249,7 @@ static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publishe
 			srtp_err_status_t res = srtp_create(&srtp_ctx->ctx, policy);
 			if(res != srtp_err_status_ok) {
 				/* Something went wrong... */
-				janus_mutex_unlock(&p->rtp_forwarders_mutex);
+				janus_mutex_unlock(&stream->rtp_forwarders_mutex);
 				JANUS_LOG(LOG_ERR, "Error creating forwarder SRTP session: %d (%s)\n", res, janus_srtp_error_str(res));
 				g_free(decoded);
 				policy->key = NULL;
@@ -1829,7 +2257,7 @@ static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publishe
 				if(forward->rtcp_fd > -1)
 					close(forward->rtcp_fd);
 				g_free(forward);
-				return 0;
+				return NULL;
 			}
 			srtp_ctx->contexts = p->srtp_contexts;
 			srtp_ctx->id = g_strdup(srtp_id);
@@ -1858,16 +2286,19 @@ static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publishe
 		forward->simulcast = TRUE;
 		janus_rtp_switching_context_reset(&forward->context);
 		janus_rtp_simulcasting_context_reset(&forward->sim_context);
-		forward->sim_context.rid_ext_id = p->rid_extmap_id;
+		forward->sim_context.rid_ext_id = stream->rid_extmap_id;
 		forward->sim_context.substream_target = 2;
 		forward->sim_context.templayer_target = 2;
 	}
 	janus_refcount_init(&forward->ref, janus_videoroom_rtp_forwarder_free);
 	guint32 stream_id = janus_random_uint32();
-	while(g_hash_table_lookup(p->rtp_forwarders, GUINT_TO_POINTER(stream_id)) != NULL) {
+	while(g_hash_table_lookup(stream->publisher->rtp_forwarders, GUINT_TO_POINTER(stream_id)) != NULL &&
+			g_hash_table_lookup(stream->rtp_forwarders, GUINT_TO_POINTER(stream_id)) != NULL) {
 		stream_id = janus_random_uint32();
 	}
-	g_hash_table_insert(p->rtp_forwarders, GUINT_TO_POINTER(stream_id), forward);
+	forward->stream_id = stream_id;
+	g_hash_table_insert(stream->rtp_forwarders, GUINT_TO_POINTER(stream_id), forward);
+	g_hash_table_insert(stream->publisher->rtp_forwarders, GUINT_TO_POINTER(stream_id), GUINT_TO_POINTER(stream_id));
 	if(fd > -1) {
 		/* We need RTCP: track this file descriptor, and ref the forwarder */
 		janus_refcount_increase(&forward->ref);
@@ -1901,10 +2332,54 @@ static guint32 janus_videoroom_rtp_forwarder_add_helper(janus_videoroom_publishe
 		(void)sendto(fd, &rtp, 12, 0, address, addrlen);
 		(void)sendto(fd, &rtp, 12, 0, address, addrlen);
 	}
-	janus_mutex_unlock(&p->rtp_forwarders_mutex);
+	janus_mutex_unlock(&stream->rtp_forwarders_mutex);
 	JANUS_LOG(LOG_VERB, "Added %s/%d rtp_forward to participant %"SCNu64" host: %s:%d stream_id: %"SCNu32"\n",
 		is_data ? "data" : (is_video ? "video" : "audio"), substream, p->user_id, host, port, stream_id);
-	return stream_id;
+	return forward;
+}
+
+static json_t *janus_videoroom_rtp_forwarder_summary(janus_videoroom_rtp_forwarder *f) {
+	if(f == NULL)
+		return NULL;
+	json_t *json = json_object();
+	json_object_set_new(json, "stream_id", json_integer(f->stream_id));
+	char address[100];
+	if(f->serv_addr.sin_family == AF_INET) {
+		json_object_set_new(json, "ip", json_string(
+			inet_ntop(AF_INET, &f->serv_addr.sin_addr, address, sizeof(address))));
+	} else {
+		json_object_set_new(json, "ip", json_string(
+			inet_ntop(AF_INET6, &f->serv_addr6.sin6_addr, address, sizeof(address))));
+	}
+	json_object_set_new(json, "port", json_integer(ntohs(f->serv_addr.sin_port)));
+	if(f->is_data) {
+		json_object_set_new(json, "type", json_string("data"));
+	} else if(f->is_video) {
+		json_object_set_new(json, "type", json_string("video"));
+		if(f->local_rtcp_port > 0)
+			json_object_set_new(json, "local_rtcp_port", json_integer(f->local_rtcp_port));
+		if(f->remote_rtcp_port > 0)
+			json_object_set_new(json, "remote_rtcp_port", json_integer(f->remote_rtcp_port));
+		if(f->payload_type)
+			json_object_set_new(json, "pt", json_integer(f->payload_type));
+		if(f->ssrc)
+			json_object_set_new(json, "ssrc", json_integer(f->ssrc));
+		if(f->substream)
+			json_object_set_new(json, "substream", json_integer(f->substream));
+	} else {
+		json_object_set_new(json, "type", json_string("audio"));
+		if(f->local_rtcp_port > 0)
+			json_object_set_new(json, "local_rtcp_port", json_integer(f->local_rtcp_port));
+		if(f->remote_rtcp_port > 0)
+			json_object_set_new(json, "remote_rtcp_port", json_integer(f->remote_rtcp_port));
+		if(f->payload_type)
+			json_object_set_new(json, "pt", json_integer(f->payload_type));
+		if(f->ssrc)
+			json_object_set_new(json, "ssrc", json_integer(f->ssrc));
+	}
+	if(f->is_srtp)
+		json_object_set_new(json, "srtp", json_true());
+	return json;
 }
 
 static void janus_videoroom_rtp_forwarder_destroy(janus_videoroom_rtp_forwarder *forward) {
@@ -1943,6 +2418,314 @@ static void janus_videoroom_srtp_context_free(gpointer data) {
 }
 
 
+/* Helpers for subscription streams */
+static janus_videoroom_subscriber_stream *janus_videoroom_subscriber_stream_add(janus_videoroom_subscriber *subscriber,
+		janus_videoroom_publisher_stream *ps,
+		gboolean legacy, gboolean do_audio, gboolean do_video, gboolean do_data) {
+	/* If this is a legacy subscription ("feed"), use the deprecated properties */
+	if(legacy && ((ps->type == JANUS_VIDEOROOM_MEDIA_AUDIO && !do_audio) ||
+			(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO && !do_video) ||
+			(ps->type == JANUS_VIDEOROOM_MEDIA_DATA && !do_data))) {
+		/* Skip this */
+		JANUS_LOG(LOG_WARN, "Skipping %s stream (legacy subscription)\n", janus_videoroom_media_str(ps->type));
+		return NULL;
+	}
+	/* Allocate a new subscriber stream instance */
+	janus_videoroom_subscriber_stream *stream = g_malloc0(sizeof(janus_videoroom_subscriber_stream));
+	stream->mindex = g_list_length(subscriber->streams);
+	stream->subscriber = subscriber;
+	stream->publisher_streams = g_slist_append(stream->publisher_streams, ps);
+	/* Copy properties from the source */
+	stream->type = ps->type;
+	stream->acodec = ps->acodec;
+	stream->vcodec = ps->vcodec;
+	stream->pt = ps->pt;
+	stream->opusfec = ps->opusfec;
+	char mid[5];
+	g_snprintf(mid, sizeof(mid), "%d", stream->mindex);
+	stream->mid = g_strdup(mid);
+	subscriber->streams = g_list_append(subscriber->streams, stream);
+	g_hash_table_insert(subscriber->streams_byid, GINT_TO_POINTER(stream->mindex), stream);
+	g_hash_table_insert(subscriber->streams_bymid, g_strdup(stream->mid), stream);
+	/* Initialize the stream */
+	janus_rtp_switching_context_reset(&stream->context);
+	stream->send = TRUE;
+	g_atomic_int_set(&stream->destroyed, 0);
+	janus_refcount_init(&stream->ref, janus_videoroom_subscriber_stream_free);
+	janus_refcount_increase(&stream->ref);	/* This is for the mid-indexed hashtable */
+	janus_rtp_simulcasting_context_reset(&stream->sim_context);
+	stream->sim_context.rid_ext_id = ps->rid_extmap_id;
+	stream->sim_context.substream_target = 2;
+	stream->sim_context.templayer_target = 2;
+	janus_vp8_simulcast_context_reset(&stream->vp8_context);
+	/* This stream may belong to a room where VP9 SVC has been enabled,
+	 * let's assume we're interested in all layers for the time being */
+	stream->spatial_layer = -1;
+	stream->target_spatial_layer = 1;		/* FIXME Chrome sends 0 and 1 */
+	stream->temporal_layer = -1;
+	stream->target_temporal_layer = 2;	/* FIXME Chrome sends 0, 1 and 2 */
+	janus_mutex_lock(&ps->subscribers_mutex);
+	ps->subscribers = g_slist_append(ps->subscribers, stream);
+	/* The two streams reference each other */
+	janus_refcount_increase(&stream->ref);
+	janus_refcount_increase(&ps->ref);
+	janus_mutex_unlock(&ps->subscribers_mutex);
+	return stream;
+}
+
+static janus_videoroom_subscriber_stream *janus_videoroom_subscriber_stream_add_or_replace(janus_videoroom_subscriber *subscriber,
+		janus_videoroom_publisher_stream *ps) {
+	if(subscriber == NULL || ps == NULL)
+		return NULL;
+	/* First of all, let's check if there's an m-line we can reuse */
+	gboolean found = FALSE;
+	janus_videoroom_subscriber_stream *stream = NULL;
+	GList *temp = subscriber->streams;
+	while(temp) {
+		stream = (janus_videoroom_subscriber_stream *)temp->data;
+		janus_videoroom_publisher_stream *stream_ps = stream->publisher_streams ? stream->publisher_streams->data : NULL;
+		if(stream_ps != NULL && stream_ps->type == ps->type && stream->type == JANUS_VIDEOROOM_MEDIA_DATA) {
+			/* We already have a datachannel m-line, no need for others: just update the subscribers list */
+			janus_mutex_lock(&ps->subscribers_mutex);
+			if(g_slist_find(ps->subscribers, stream) == NULL && g_slist_find(stream->publisher_streams, ps) == NULL) {
+				ps->subscribers = g_slist_append(ps->subscribers, stream);
+				stream->publisher_streams = g_slist_append(stream->publisher_streams, ps);
+				/* The two streams reference each other */
+				janus_refcount_increase(&stream->ref);
+				janus_refcount_increase(&ps->ref);
+			}
+			janus_mutex_unlock(&ps->subscribers_mutex);
+			return NULL;
+		}
+		if(stream_ps == NULL && stream->type == ps->type) {
+			/* There's an empty m-line of the right type, check if codecs match */
+			if(stream->type == JANUS_VIDEOROOM_MEDIA_DATA ||
+					(stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO && stream->acodec == ps->acodec) ||
+					(stream->type == JANUS_VIDEOROOM_MEDIA_VIDEO && stream->vcodec == ps->vcodec)) {
+				found = TRUE;
+				JANUS_LOG(LOG_VERB, "Reusing m-line %d for this subscription\n", stream->mindex);
+				stream->opusfec = ps->opusfec;
+				janus_rtp_simulcasting_context_reset(&stream->sim_context);
+				if(ps->simulcast) {
+					stream->sim_context.rid_ext_id = ps->rid_extmap_id;
+					stream->sim_context.substream_target = 2;
+					stream->sim_context.templayer_target = 2;
+				}
+				janus_vp8_simulcast_context_reset(&stream->vp8_context);
+				if(ps->svc) {
+					/* This stream belongs to a room where VP9 SVC has been enabled,
+					 * let's assume we're interested in all layers for the time being */
+					stream->spatial_layer = -1;
+					stream->target_spatial_layer = 1;		/* FIXME Chrome sends 0 and 1 */
+					stream->temporal_layer = -1;
+					stream->target_temporal_layer = 2;	/* FIXME Chrome sends 0, 1 and 2 */
+				}
+				janus_mutex_lock(&ps->subscribers_mutex);
+				if(g_slist_find(ps->subscribers, stream) == NULL && g_slist_find(stream->publisher_streams, ps) == NULL) {
+					ps->subscribers = g_slist_append(ps->subscribers, stream);
+					stream->publisher_streams = g_slist_append(stream->publisher_streams, ps);
+					/* The two streams reference each other */
+					janus_refcount_increase(&stream->ref);
+					janus_refcount_increase(&ps->ref);
+				}
+				janus_mutex_unlock(&ps->subscribers_mutex);
+				break;
+			}
+		}
+		temp = temp->next;
+	}
+	if(found)
+		return stream;
+	/* We couldn't find any, add a new one */
+	return janus_videoroom_subscriber_stream_add(subscriber, ps, FALSE, FALSE, FALSE, FALSE);
+}
+
+static void janus_videoroom_subscriber_stream_remove(janus_videoroom_subscriber_stream *s,
+		janus_videoroom_publisher_stream *ps, gboolean lock_ps) {
+	janus_videoroom_subscriber *subscriber = s->subscriber;
+	if(subscriber && subscriber->pvt_id > 0 && subscriber->room != NULL) {
+		janus_mutex_lock(&subscriber->room->mutex);
+		janus_videoroom_publisher *owner = g_hash_table_lookup(subscriber->room->private_ids, GUINT_TO_POINTER(subscriber->pvt_id));
+		if(owner != NULL) {
+			janus_mutex_lock(&owner->subscribers_mutex);
+			/* Note: we should refcount these subscription-publisher mappings as well */
+			owner->subscriptions = g_slist_remove(owner->subscriptions, s);
+			janus_mutex_unlock(&owner->subscribers_mutex);
+		}
+		janus_mutex_unlock(&subscriber->room->mutex);
+		//~ if(subscriber->room)
+			//~ g_clear_pointer(&subscriber->room, janus_videoroom_room_dereference);
+		//~ /* If the subscriber itself has no more active active subscriptions, should we close it? */
+		//~ if(subscriber->streams == NULL && subscriber->session && subscriber->close_pc)
+			//~ gateway->close_pc(subscriber->session->handle);
+	}
+	g_atomic_int_set(&s->ready, 0);
+	if(ps != NULL) {
+		/* Unsubscribe from this stream in particular (datachannels can have multiple sources) */
+		if(g_slist_find(s->publisher_streams, ps) != NULL) {
+			/* Remove the subscription from the list of recipients */
+			s->publisher_streams = g_slist_remove(s->publisher_streams, ps);
+			if(s->publisher_streams == NULL)
+				g_atomic_int_set(&s->ready, 0);
+			s->opusfec = FALSE;
+			if(lock_ps)
+				janus_mutex_lock(&ps->subscribers_mutex);
+			ps->subscribers = g_slist_remove(ps->subscribers, s);
+			if(lock_ps)
+				janus_mutex_unlock(&ps->subscribers_mutex);
+			/* Unref the two streams, as they're not related anymore */
+			janus_refcount_decrease(&ps->ref);
+			janus_refcount_decrease(&s->ref);
+		}
+	} else {
+		/* Unsubscribe from all sources (which will be one for audio/video, potentially more for datachannels) */
+		while(s->publisher_streams) {
+			ps = s->publisher_streams->data;
+			s->publisher_streams = g_slist_remove(s->publisher_streams, ps);
+			if(ps) {
+				/* Remove the subscription from the list of recipients */
+				if(s->publisher_streams == NULL)
+					g_atomic_int_set(&s->ready, 0);
+				s->opusfec = FALSE;
+				if(lock_ps)
+					janus_mutex_lock(&ps->subscribers_mutex);
+				ps->subscribers = g_slist_remove(ps->subscribers, s);
+				if(lock_ps)
+					janus_mutex_unlock(&ps->subscribers_mutex);
+				/* Unref the two streams, as they're not related anymore */
+				janus_refcount_decrease(&ps->ref);
+				janus_refcount_decrease(&s->ref);
+			}
+		}
+	}
+}
+
+static json_t *janus_videoroom_subscriber_streams_summary(janus_videoroom_subscriber *subscriber, gboolean legacy, json_t *event) {
+	json_t *media = json_array();
+	GList *temp = subscriber->streams;
+	while(temp) {
+		janus_videoroom_subscriber_stream *stream = (janus_videoroom_subscriber_stream *)temp->data;
+		janus_refcount_increase(&stream->ref);
+		janus_videoroom_publisher_stream *ps = stream->publisher_streams ? stream->publisher_streams->data : NULL;
+		if(ps)
+			janus_refcount_increase(&ps->ref);
+		json_t *m = json_object();
+		json_object_set_new(m, "type", json_string(janus_videoroom_media_str(stream->type)));
+		json_object_set_new(m, "active", (ps || stream->type == JANUS_VIDEOROOM_MEDIA_DATA) ? json_true() : json_false());
+		json_object_set_new(m, "mindex", json_integer(stream->mindex));
+		json_object_set_new(m, "mid", json_string(stream->mid));
+		json_object_set_new(m, "ready", g_atomic_int_get(&stream->ready) ? json_true() : json_false());
+		json_object_set_new(m, "send", stream->send ? json_true() : json_false());
+		if(ps && stream->type == JANUS_VIDEOROOM_MEDIA_DATA) {
+			json_object_set_new(m, "sources", json_integer(g_slist_length(stream->publisher_streams)));
+		} else if(ps && stream->type != JANUS_VIDEOROOM_MEDIA_DATA) {
+			if(ps->publisher) {
+				json_object_set_new(m, "feed_id", json_integer(ps->publisher->user_id));
+				if(ps->publisher->display)
+					json_object_set_new(m, "feed_display", json_string(ps->publisher->display));
+				/* If this is a legacy subscription, put the info in the generic part too */
+				if(legacy && event) {
+					json_object_set_new(event, "id", json_integer(ps->publisher->user_id));
+					if(ps->publisher->display)
+						json_object_set_new(event, "display", json_string(ps->publisher->display));
+				}
+			}
+			if(ps->mid)
+				json_object_set_new(m, "feed_mid", json_string(ps->mid));
+			if(ps->description)
+				json_object_set_new(m, "feed_description", json_string(ps->description));
+			if(ps->simulcast) {
+				json_t *simulcast = json_object();
+				json_object_set_new(simulcast, "substream", json_integer(stream->sim_context.substream));
+				json_object_set_new(simulcast, "substream-target", json_integer(stream->sim_context.substream_target));
+				json_object_set_new(simulcast, "temporal-layer", json_integer(stream->sim_context.templayer));
+				json_object_set_new(simulcast, "temporal-layer-target", json_integer(stream->sim_context.templayer_target));
+				json_object_set_new(m, "simulcast", simulcast);
+			}
+			if(ps->svc) {
+				json_t *svc = json_object();
+				json_object_set_new(svc, "spatial-layer", json_integer(stream->spatial_layer));
+				json_object_set_new(svc, "target-spatial-layer", json_integer(stream->target_spatial_layer));
+				json_object_set_new(svc, "temporal-layer", json_integer(stream->temporal_layer));
+				json_object_set_new(svc, "target-temporal-layer", json_integer(stream->target_temporal_layer));
+				json_object_set_new(m, "svc", svc);
+			}
+		}
+		if(ps)
+			janus_refcount_decrease(&ps->ref);
+		janus_refcount_decrease(&stream->ref);
+		json_array_append_new(media, m);
+		temp = temp->next;
+	}
+	return media;
+}
+
+/* Helper to generate a new offer with the subscriber streams */
+static json_t *janus_videoroom_subscriber_offer(janus_videoroom_subscriber *subscriber) {
+	g_atomic_int_set(&subscriber->answered, 0);
+	char s_name[100];
+	g_snprintf(s_name, sizeof(s_name), "VideoRoom %"SCNu64, subscriber->room->room_id);
+	janus_sdp *offer = janus_sdp_generate_offer(s_name, "0.0.0.0",
+		JANUS_SDP_OA_DONE);
+	GList *temp = subscriber->streams;
+	while(temp) {
+		janus_videoroom_subscriber_stream *stream = (janus_videoroom_subscriber_stream *)temp->data;
+		janus_videoroom_publisher_stream *ps = stream->publisher_streams ? stream->publisher_streams->data : NULL;
+		int pt = -1;
+		const char *codec = NULL;
+		if(stream->type != JANUS_VIDEOROOM_MEDIA_DATA) {
+			pt = stream->pt;
+			codec = (stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO ?
+				janus_audiocodec_name(stream->acodec) : janus_videocodec_name(stream->vcodec));
+		}
+		janus_sdp_generate_offer_mline(offer,
+			JANUS_SDP_OA_MLINE, janus_videoroom_media_sdptype(stream->type),
+			JANUS_SDP_OA_MID, stream->mid,
+			JANUS_SDP_OA_PT, pt,
+			JANUS_SDP_OA_CODEC, codec,
+			JANUS_SDP_OA_FMTP, (stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO && stream->opusfec ? "useinbandfec=1" : NULL),
+			JANUS_SDP_OA_DIRECTION, (ps || stream->type == JANUS_VIDEOROOM_MEDIA_DATA) ? JANUS_SDP_SENDONLY : JANUS_SDP_INACTIVE,
+			/* TODO Add other properties from original SDP */
+			JANUS_SDP_OA_DONE);
+		/* Add the extmap attributes, if needed */
+		if(ps) {
+			janus_sdp_mline *m = janus_sdp_mline_find_by_index(offer, stream->mindex);
+			if(ps->type == JANUS_VIDEOROOM_MEDIA_AUDIO && ps->audio_level_extmap_id > 0) {
+				if(m != NULL) {
+					janus_sdp_attribute *a = janus_sdp_attribute_create("extmap",
+						"%d %s\r\n", ps->audio_level_extmap_id, JANUS_RTP_EXTMAP_AUDIO_LEVEL);
+					janus_sdp_attribute_add_to_mline(m, a);
+				}
+			}
+			if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO && ps->video_orient_extmap_id > 0) {
+				if(m != NULL) {
+					janus_sdp_attribute *a = janus_sdp_attribute_create("extmap",
+						"%d %s\r\n", ps->video_orient_extmap_id, JANUS_RTP_EXTMAP_VIDEO_ORIENTATION);
+					janus_sdp_attribute_add_to_mline(m, a);
+				}
+			}
+			if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO && ps->playout_delay_extmap_id > 0) {
+				if(m != NULL) {
+					janus_sdp_attribute *a = janus_sdp_attribute_create("extmap",
+						"%d %s\r\n", ps->playout_delay_extmap_id, JANUS_RTP_EXTMAP_PLAYOUT_DELAY);
+					janus_sdp_attribute_add_to_mline(m, a);
+				}
+			}
+		}
+		temp = temp->next;
+	}
+	/* Update (or set) the SDP version */
+	subscriber->session->sdp_version++;
+	offer->o_version = subscriber->session->sdp_version;
+	char *sdp = janus_sdp_write(offer);
+	janus_sdp_destroy(offer);
+	json_t *jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp);
+	g_free(sdp);
+	/* Done */
+	return jsep;
+}
+
+
 /* Plugin implementation */
 int janus_videoroom_init(janus_callbacks *callback, const char *config_path) {
 	if(g_atomic_int_get(&stopping)) {
@@ -2011,7 +2794,9 @@ int janus_videoroom_init(janus_callbacks *callback, const char *config_path) {
 			janus_config_item *bitrate = janus_config_get(config, cat, janus_config_type_item, "bitrate");
 			janus_config_item *bitrate_cap = janus_config_get(config, cat, janus_config_type_item, "bitrate_cap");
 			janus_config_item *maxp = janus_config_get(config, cat, janus_config_type_item, "publishers");
-			janus_config_item *firfreq = janus_config_get(config, cat, janus_config_type_item, "fir_freq");
+			janus_config_item *plifreq = janus_config_get(config, cat, janus_config_type_item, "pli_freq");
+			if(plifreq == NULL)	/* For backwards compatibility, we accept fir_freq as well */
+				plifreq = janus_config_get(config, cat, janus_config_type_item, "fir_freq");
 			janus_config_item *audiocodec = janus_config_get(config, cat, janus_config_type_item, "audiocodec");
 			janus_config_item *videocodec = janus_config_get(config, cat, janus_config_type_item, "videocodec");
 			janus_config_item *fec = janus_config_get(config, cat, janus_config_type_item, "opus_fec");
@@ -2057,9 +2842,9 @@ int janus_videoroom_init(janus_callbacks *callback, const char *config_path) {
 			if(videoroom->bitrate > 0 && videoroom->bitrate < 64000)
 				videoroom->bitrate = 64000;	/* Don't go below 64k */
 			videoroom->bitrate_cap = bitrate_cap && bitrate_cap->value && janus_is_true(bitrate_cap->value);
-			videoroom->fir_freq = 0;
-			if(firfreq != NULL && firfreq->value != NULL)
-				videoroom->fir_freq = atol(firfreq->value);
+			videoroom->pli_freq = 0;
+			if(plifreq != NULL && plifreq->value != NULL)
+				videoroom->pli_freq = atol(plifreq->value);
 			/* By default, we force Opus as the only audio codec */
 			videoroom->acodec[0] = JANUS_AUDIOCODEC_OPUS;
 			videoroom->acodec[1] = JANUS_AUDIOCODEC_NONE;
@@ -2206,8 +2991,8 @@ int janus_videoroom_init(janus_callbacks *callback, const char *config_path) {
 		/* Compute a list of the supported codecs for the summary */
 		char audio_codecs[100], video_codecs[100];
 		janus_videoroom_codecstr(vr, audio_codecs, video_codecs, sizeof(audio_codecs), "|");
-		JANUS_LOG(LOG_VERB, "  ::: [%"SCNu64"][%s] %"SCNu32", max %d publishers, FIR frequency of %d seconds, %s audio codec(s), %s video codec(s)\n",
-			vr->room_id, vr->room_name, vr->bitrate, vr->max_publishers, vr->fir_freq,
+		JANUS_LOG(LOG_VERB, "  ::: [%"SCNu64"][%s] %"SCNu32", max %d publishers, PLI frequency of %d seconds, %s audio codec(s), %s video codec(s)\n",
+			vr->room_id, vr->room_name, vr->bitrate, vr->max_publishers, vr->pli_freq,
 			audio_codecs, video_codecs);
 	}
 	janus_mutex_unlock(&rooms_mutex);
@@ -2221,6 +3006,7 @@ int janus_videoroom_init(janus_callbacks *callback, const char *config_path) {
 		/* We show the error but it's not fatal */
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the VideoRoom RTCP thread for RTP forwarders...\n",
 			error->code, error->message ? error->message : "??");
+		g_error_free(error);
 	}
 
 	g_atomic_int_set(&initialized, 1);
@@ -2478,10 +3264,8 @@ void janus_videoroom_destroy_session(janus_plugin_session *handle, int *error) {
 		} else if(session->participant_type == janus_videoroom_p_type_subscriber) {
 			janus_videoroom_subscriber *s = (janus_videoroom_subscriber *)session->participant;
 			session->participant = NULL;
-			if(s->room) {
+			if(s->room)
 				janus_refcount_decrease(&s->room->ref);
-				janus_refcount_decrease(&s->ref);
-			}
 			janus_videoroom_subscriber_destroy(s);
 		}
 		g_hash_table_remove(sessions, handle);
@@ -2518,78 +3302,62 @@ json_t *janus_videoroom_query_session(janus_plugin_session *handle) {
 				json_object_set_new(info, "private_id", json_integer(participant->pvt_id));
 				if(participant->display)
 					json_object_set_new(info, "display", json_string(participant->display));
-				if(participant->subscribers)
-					json_object_set_new(info, "viewers", json_integer(g_slist_length(participant->subscribers)));
-				json_t *media = json_object();
-				json_object_set_new(media, "audio", participant->audio ? json_true() : json_false());
-				if(participant->audio)
-					json_object_set_new(media, "audio_codec", json_string(janus_audiocodec_name(participant->acodec)));
-				json_object_set_new(media, "video", participant->video ? json_true() : json_false());
-				if(participant->video)
-					json_object_set_new(media, "video_codec", json_string(janus_videocodec_name(participant->vcodec)));
-				json_object_set_new(media, "data", participant->data ? json_true() : json_false());
-				json_object_set_new(info, "media", media);
+				/* TODO Fix the summary of viewers, since everything is stream based now */
+				//~ if(participant->subscribers)
+					//~ json_object_set_new(info, "viewers", json_integer(g_slist_length(participant->subscribers)));
 				json_object_set_new(info, "bitrate", json_integer(participant->bitrate));
-				if(participant->ssrc[0] != 0 || participant->rid[0] != NULL)
-					json_object_set_new(info, "simulcast", json_true());
-				if(participant->arc || participant->vrc || participant->drc) {
-					json_t *recording = json_object();
-					if(participant->arc && participant->arc->filename)
-						json_object_set_new(recording, "audio", json_string(participant->arc->filename));
-					if(participant->vrc && participant->vrc->filename)
-						json_object_set_new(recording, "video", json_string(participant->vrc->filename));
-					if(participant->drc && participant->drc->filename)
-						json_object_set_new(recording, "data", json_string(participant->drc->filename));
-					json_object_set_new(info, "recording", recording);
-				}
-				if(participant->audio_level_extmap_id > 0) {
-					json_object_set_new(info, "audio-level-dBov", json_integer(participant->audio_dBov_level));
-					json_object_set_new(info, "talking", participant->talking ? json_true() : json_false());
+				json_t *media = json_array();
+				GList *temp = participant->streams;
+				while(temp) {
+					janus_videoroom_publisher_stream *stream = (janus_videoroom_publisher_stream *)temp->data;
+					janus_refcount_increase(&stream->ref);
+					json_t *m = json_object();
+					json_object_set_new(m, "type", json_string(janus_videoroom_media_str(stream->type)));
+					json_object_set_new(m, "mindex", json_integer(stream->mindex));
+					json_object_set_new(m, "mid", json_string(stream->mid));
+					if(stream->description)
+						json_object_set_new(m, "description", json_string(stream->description));
+					if(stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO)
+						json_object_set_new(m, "codec", json_string(janus_audiocodec_name(stream->acodec)));
+					else if(stream->type == JANUS_VIDEOROOM_MEDIA_VIDEO)
+						json_object_set_new(m, "codec", json_string(janus_videocodec_name(stream->vcodec)));
+					if(stream->simulcast)
+						json_object_set_new(m, "simulcast", json_true());
+					if(stream->svc)
+						json_object_set_new(m, "vp9-svc", json_true());
+					if(stream->rc && stream->rc->filename)
+						json_object_set_new(m, "recording", json_string(stream->rc->filename));
+					if(stream->audio_level_extmap_id > 0) {
+						json_object_set_new(m, "audio-level-dBov", json_integer(stream->audio_dBov_level));
+						json_object_set_new(m, "talking", stream->talking ? json_true() : json_false());
+					}
+					janus_mutex_lock(&stream->subscribers_mutex);
+					json_object_set_new(m, "subscribers", json_integer(g_slist_length(stream->subscribers)));
+					janus_mutex_unlock(&stream->subscribers_mutex);
+					janus_refcount_decrease(&stream->ref);
+					json_array_append_new(media, m);
+					temp = temp->next;
 				}
+				json_object_set_new(info, "streams", media);
 				janus_refcount_decrease(&participant->ref);
 			}
 		} else if(session->participant_type == janus_videoroom_p_type_subscriber) {
 			json_object_set_new(info, "type", json_string("subscriber"));
 			janus_videoroom_subscriber *participant = (janus_videoroom_subscriber *)session->participant;
 			if(participant && participant->room) {
-				janus_videoroom_publisher *feed = (janus_videoroom_publisher *)participant->feed;
-				if(feed && feed->room) {
-					janus_videoroom *room = feed->room;
-					json_object_set_new(info, "room", room ? json_integer(room->room_id) : NULL);
-					json_object_set_new(info, "private_id", json_integer(participant->pvt_id));
-					json_object_set_new(info, "feed_id", json_integer(feed->user_id));
-					if(feed->display)
-						json_object_set_new(info, "feed_display", json_string(feed->display));
-				}
-				json_t *media = json_object();
-				json_object_set_new(media, "audio", participant->audio ? json_true() : json_false());
-				json_object_set_new(media, "audio-offered", participant->audio_offered ? json_true() : json_false());
-				json_object_set_new(media, "video", participant->video ? json_true() : json_false());
-				json_object_set_new(media, "video-offered", participant->video_offered ? json_true() : json_false());
-				json_object_set_new(media, "data", participant->data ? json_true() : json_false());
-				json_object_set_new(media, "data-offered", participant->data_offered ? json_true() : json_false());
-				json_object_set_new(info, "media", media);
-				if(feed && (feed->ssrc[0] != 0 || feed->rid[0] != NULL)) {
-					json_t *simulcast = json_object();
-					json_object_set_new(simulcast, "substream", json_integer(participant->sim_context.substream));
-					json_object_set_new(simulcast, "substream-target", json_integer(participant->sim_context.substream_target));
-					json_object_set_new(simulcast, "temporal-layer", json_integer(participant->sim_context.templayer));
-					json_object_set_new(simulcast, "temporal-layer-target", json_integer(participant->sim_context.templayer_target));
-					json_object_set_new(info, "simulcast", simulcast);
-				}
-				if(participant->room && participant->room->do_svc) {
-					json_t *svc = json_object();
-					json_object_set_new(svc, "spatial-layer", json_integer(participant->spatial_layer));
-					json_object_set_new(svc, "target-spatial-layer", json_integer(participant->target_spatial_layer));
-					json_object_set_new(svc, "temporal-layer", json_integer(participant->temporal_layer));
-					json_object_set_new(svc, "target-temporal-layer", json_integer(participant->target_temporal_layer));
-					json_object_set_new(info, "svc", svc);
-				}
+				janus_videoroom *room = participant->room;
+				json_object_set_new(info, "room", room ? json_integer(room->room_id) : NULL);
+				json_object_set_new(info, "answered", g_atomic_int_get(&participant->answered) ? json_true() : json_false());
+				json_object_set_new(info, "pending_offer", g_atomic_int_get(&participant->pending_offer) ? json_true() : json_false());
+				json_object_set_new(info, "pending_restart", g_atomic_int_get(&participant->pending_restart) ? json_true() : json_false());
+				json_object_set_new(info, "paused", participant->paused ? json_true() : json_false());
+				json_t *media = janus_videoroom_subscriber_streams_summary(participant, FALSE, NULL);
+				json_object_set_new(info, "streams", media);
 			}
 		}
 	}
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -2681,7 +3449,9 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		json_t *pin = json_object_get(root, "pin");
 		json_t *bitrate = json_object_get(root, "bitrate");
 		json_t *bitrate_cap = json_object_get(root, "bitrate_cap");
-		json_t *fir_freq = json_object_get(root, "fir_freq");
+		json_t *pli_freq = json_object_get(root, "pli_freq");
+		if(pli_freq == NULL)	/* For backwards compatibility, we accept fir_freq as well */
+			pli_freq = json_object_get(root, "fir_freq");
 		json_t *publishers = json_object_get(root, "publishers");
 		json_t *allowed = json_object_get(root, "allowed");
 		json_t *audiocodec = json_object_get(root, "audiocodec");
@@ -2829,9 +3599,9 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		if(videoroom->bitrate > 0 && videoroom->bitrate < 64000)
 			videoroom->bitrate = 64000;	/* Don't go below 64k */
 		videoroom->bitrate_cap = bitrate_cap ? json_is_true(bitrate_cap) : FALSE;
-		videoroom->fir_freq = 0;
-		if(fir_freq)
-			videoroom->fir_freq = json_integer_value(fir_freq);
+		videoroom->pli_freq = 0;
+		if(pli_freq)
+			videoroom->pli_freq = json_integer_value(pli_freq);
 		/* By default, we force Opus as the only audio codec */
 		videoroom->acodec[0] = JANUS_AUDIOCODEC_OPUS;
 		videoroom->acodec[1] = JANUS_AUDIOCODEC_NONE;
@@ -2978,9 +3748,9 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 				janus_config_add(config, c, janus_config_item_create("bitrate_cap", "yes"));
 			g_snprintf(value, BUFSIZ, "%d", videoroom->max_publishers);
 			janus_config_add(config, c, janus_config_item_create("publishers", value));
-			if(videoroom->fir_freq) {
-				g_snprintf(value, BUFSIZ, "%"SCNu16, videoroom->fir_freq);
-				janus_config_add(config, c, janus_config_item_create("fir_freq", value));
+			if(videoroom->pli_freq) {
+				g_snprintf(value, BUFSIZ, "%"SCNu16, videoroom->pli_freq);
+				janus_config_add(config, c, janus_config_item_create("pli_freq", value));
 			}
 			char video_codecs[100];
 			char audio_codecs[100];
@@ -3032,7 +3802,7 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		g_hash_table_iter_init(&iter, rooms);
 		while (g_hash_table_iter_next(&iter, NULL, &value)) {
 			janus_videoroom *vr = value;
-			JANUS_LOG(LOG_VERB, "  ::: [%"SCNu64"][%s] %"SCNu32", max %d publishers, FIR frequency of %d seconds\n", vr->room_id, vr->room_name, vr->bitrate, vr->max_publishers, vr->fir_freq);
+			JANUS_LOG(LOG_VERB, "  ::: [%"SCNu64"][%s] %"SCNu32", max %d publishers, PLI frequency of %d seconds\n", vr->room_id, vr->room_name, vr->bitrate, vr->max_publishers, vr->pli_freq);
 		}
 		janus_mutex_unlock(&rooms_mutex);
 		/* Send info back */
@@ -3063,7 +3833,9 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		json_t *secret = json_object_get(root, "new_secret");
 		json_t *pin = json_object_get(root, "new_pin");
 		json_t *bitrate = json_object_get(root, "new_bitrate");
-		json_t *fir_freq = json_object_get(root, "new_fir_freq");
+		json_t *pli_freq = json_object_get(root, "new_pli_freq");
+		if(pli_freq == NULL)	/* For backwards compatibility, we accept new_fir_freq as well */
+			pli_freq = json_object_get(root, "new_fir_freq");
 		json_t *publishers = json_object_get(root, "new_publishers");
 		json_t *permanent = json_object_get(root, "permanent");
 		gboolean save = permanent ? json_is_true(permanent) : FALSE;
@@ -3098,8 +3870,8 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 			if(videoroom->bitrate > 0 && videoroom->bitrate < 64000)
 				videoroom->bitrate = 64000;	/* Don't go below 64k */
 		}
-		if(fir_freq)
-			videoroom->fir_freq = json_integer_value(fir_freq);
+		if(pli_freq)
+			videoroom->pli_freq = json_integer_value(pli_freq);
 		if(secret && strlen(json_string_value(secret)) > 0) {
 			char *old_secret = videoroom->room_secret;
 			char *new_secret = g_strdup(json_string_value(secret));
@@ -3135,9 +3907,9 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 				janus_config_add(config, c, janus_config_item_create("bitrate_cap", "yes"));
 			g_snprintf(value, BUFSIZ, "%d", videoroom->max_publishers);
 			janus_config_add(config, c, janus_config_item_create("publishers", value));
-			if(videoroom->fir_freq) {
-				g_snprintf(value, BUFSIZ, "%"SCNu16, videoroom->fir_freq);
-				janus_config_add(config, c, janus_config_item_create("fir_freq", value));
+			if(videoroom->pli_freq) {
+				g_snprintf(value, BUFSIZ, "%"SCNu16, videoroom->pli_freq);
+				janus_config_add(config, c, janus_config_item_create("pli_freq", value));
 			}
 			char audio_codecs[100];
 			char video_codecs[100];
@@ -3301,7 +4073,7 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 				json_object_set_new(rl, "bitrate", json_integer(room->bitrate));
 				if(room->bitrate_cap)
 					json_object_set_new(rl, "bitrate_cap", json_true());
-				json_object_set_new(rl, "fir_freq", json_integer(room->fir_freq));
+				json_object_set_new(rl, "pli_freq", json_integer(room->pli_freq));
 				json_object_set_new(rl, "require_pvtid", room->require_pvtid ? json_true() : json_false());
 				json_object_set_new(rl, "notify_joining", room->notify_joining ? json_true() : json_false());
 				char audio_codecs[100];
@@ -3346,66 +4118,6 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		}
 		json_t *room = json_object_get(root, "room");
 		json_t *pub_id = json_object_get(root, "publisher_id");
-		int video_port[3] = {-1, -1, -1}, video_rtcp_port = -1, video_pt[3] = {0, 0, 0};
-		uint32_t video_ssrc[3] = {0, 0, 0};
-		int audio_port = -1, audio_rtcp_port = -1, audio_pt = 0;
-		uint32_t audio_ssrc = 0;
-		int data_port = -1;
-		int srtp_suite = 0;
-		const char *srtp_crypto = NULL;
-		/* There may be multiple target video ports (e.g., publisher simulcasting) */
-		json_t *vid_port = json_object_get(root, "video_port");
-		if(vid_port) {
-			video_port[0] = json_integer_value(vid_port);
-			json_t *pt = json_object_get(root, "video_pt");
-			if(pt)
-				video_pt[0] = json_integer_value(pt);
-			json_t *ssrc = json_object_get(root, "video_ssrc");
-			if(ssrc)
-				video_ssrc[0] = json_integer_value(ssrc);
-		}
-		vid_port = json_object_get(root, "video_port_2");
-		if(vid_port) {
-			video_port[1] = json_integer_value(vid_port);
-			json_t *pt = json_object_get(root, "video_pt_2");
-			if(pt)
-				video_pt[1] = json_integer_value(pt);
-			json_t *ssrc = json_object_get(root, "video_ssrc_2");
-			if(ssrc)
-				video_ssrc[1] = json_integer_value(ssrc);
-		}
-		vid_port = json_object_get(root, "video_port_3");
-		if(vid_port) {
-			video_port[2] = json_integer_value(vid_port);
-			json_t *pt = json_object_get(root, "video_pt_3");
-			if(pt)
-				video_pt[2] = json_integer_value(pt);
-			json_t *ssrc = json_object_get(root, "video_ssrc_3");
-			if(ssrc)
-				video_ssrc[2] = json_integer_value(ssrc);
-		}
-		json_t *vid_rtcp_port = json_object_get(root, "video_rtcp_port");
-		if(vid_rtcp_port)
-			video_rtcp_port = json_integer_value(vid_rtcp_port);
-		/* Audio target */
-		json_t *au_port = json_object_get(root, "audio_port");
-		if(au_port) {
-			audio_port = json_integer_value(au_port);
-			json_t *pt = json_object_get(root, "audio_pt");
-			if(pt)
-				audio_pt = json_integer_value(pt);
-			json_t *ssrc = json_object_get(root, "audio_ssrc");
-			if(ssrc)
-				audio_ssrc = json_integer_value(ssrc);
-		}
-		json_t *au_rtcp_port = json_object_get(root, "audio_rtcp_port");
-		if(au_rtcp_port)
-			audio_rtcp_port = json_integer_value(au_rtcp_port);
-		/* Data target */
-		json_t *d_port = json_object_get(root, "data_port");
-		if(d_port) {
-			data_port = json_integer_value(d_port);
-		}
 		json_t *json_host = json_object_get(root, "host");
 		json_t *json_host_family = json_object_get(root, "host_family");
 		const char *host_family = json_string_value(json_host_family);
@@ -3422,30 +4134,6 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 				goto prepare_response;
 			}
 		}
-		/* Do we need to forward multiple simulcast streams to a single endpoint? */
-		gboolean simulcast = FALSE;
-		if(json_object_get(root, "simulcast") != NULL)
-			simulcast = json_is_true(json_object_get(root, "simulcast"));
-		if(simulcast) {
-			/* We do, disable the other video ports if they were requested */
-			video_port[1] = -1;
-			video_port[2] = -1;
-		}
-		/* Besides, we may need to SRTP-encrypt this stream */
-		json_t *s_suite = json_object_get(root, "srtp_suite");
-		json_t *s_crypto = json_object_get(root, "srtp_crypto");
-		if(s_suite && s_crypto) {
-			srtp_suite = json_integer_value(s_suite);
-			if(srtp_suite != 32 && srtp_suite != 80) {
-				JANUS_LOG(LOG_ERR, "Invalid SRTP suite (%d)\n", srtp_suite);
-				error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
-				g_snprintf(error_cause, 512, "Invalid SRTP suite (%d)", srtp_suite);
-				goto prepare_response;
-			}
-			srtp_crypto = json_string_value(s_crypto);
-		}
-		guint64 room_id = json_integer_value(room);
-		guint64 publisher_id = json_integer_value(pub_id);
 		const char *host = json_string_value(json_host), *resolved_host = NULL;
 		/* Check if we need to resolve this host address */
 		struct addrinfo *res = NULL, *start = NULL;
@@ -3478,155 +4166,487 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 			goto prepare_response;
 		}
 		host = resolved_host;
-		janus_mutex_lock(&rooms_mutex);
-		janus_videoroom *videoroom = NULL;
-		error_code = janus_videoroom_access_room(root, TRUE, FALSE, &videoroom, error_cause, sizeof(error_cause));
-		janus_mutex_unlock(&rooms_mutex);
-		if(error_code != 0)
-			goto prepare_response;
-		janus_refcount_increase(&videoroom->ref);
-		janus_mutex_lock(&videoroom->mutex);
-		janus_videoroom_publisher *publisher = g_hash_table_lookup(videoroom->participants, &publisher_id);
-		if(publisher == NULL) {
-			janus_mutex_unlock(&videoroom->mutex);
-			janus_refcount_decrease(&videoroom->ref);
-			JANUS_LOG(LOG_ERR, "No such publisher (%"SCNu64")\n", publisher_id);
-			error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
-			g_snprintf(error_cause, 512, "No such feed (%"SCNu64")", publisher_id);
-			goto prepare_response;
-		}
-		janus_refcount_increase(&publisher->ref);	/* This is just to handle the request for now */
-		if(publisher->udp_sock <= 0) {
-			publisher->udp_sock = socket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP);
-			int v6only = 0;
-			if(publisher->udp_sock <= 0 ||
-					setsockopt(publisher->udp_sock, IPPROTO_IPV6, IPV6_V6ONLY, &v6only, sizeof(v6only)) != 0) {
-				janus_refcount_decrease(&publisher->ref);
-				janus_mutex_unlock(&videoroom->mutex);
-				janus_refcount_decrease(&videoroom->ref);
-				JANUS_LOG(LOG_ERR, "Could not open UDP socket for RTP stream for publisher (%"SCNu64")\n", publisher_id);
-				error_code = JANUS_VIDEOROOM_ERROR_UNKNOWN_ERROR;
-				g_snprintf(error_cause, 512, "Could not open UDP socket for RTP stream");
+
+		json_t *streams = json_object_get(root, "streams");
+		if(streams == NULL || json_array_size(streams) == 0) {
+			/* No streams array, we'll use the legacy approach: make sure the host attribute was set */
+			if(host == NULL) {
+				error_code = JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT;
+				g_snprintf(error_cause, sizeof(error_cause), "Missing mandatory element host (deprecated API)");
 				goto prepare_response;
 			}
+		} else {
+			/* Iterate on the streams objects and validate them all */
+			size_t i = 0;
+			for(i=0; i<json_array_size(streams); i++) {
+				json_t *s = json_array_get(streams, i);
+				JANUS_VALIDATE_JSON_OBJECT(s, rtp_forward_stream_parameters,
+					error_code, error_cause, TRUE,
+					JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+				if(error_code != 0)
+					goto prepare_response;
+				/* Make sure we have a host attribute, either global or stream-specific */
+				json_t *stream_host = json_object_get(s, "host");
+				const char *s_host = json_string_value(stream_host);
+				if(host == NULL && s_host == NULL) {
+					error_code = JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT;
+					g_snprintf(error_cause, sizeof(error_cause), "Missing mandatory element host (global or local)");
+					goto prepare_response;
+				}
+				if(s_host != NULL) {
+					json_t *stream_host_family = json_object_get(s, "host_family");
+					const char *s_host_family = json_string_value(stream_host_family);
+					int s_family = family;
+					if(s_host_family) {
+						if(!strcasecmp(s_host_family, "ipv4")) {
+							s_family = AF_INET;
+						} else if(!strcasecmp(s_host_family, "ipv6")) {
+							s_family = AF_INET6;
+						} else {
+							JANUS_LOG(LOG_ERR, "Unsupported protocol family (%s)\n", s_host_family);
+							error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+							g_snprintf(error_cause, 512, "Unsupported protocol family (%s)", s_host_family);
+							goto prepare_response;
+						}
+					}
+					memset(&hints, 0, sizeof(hints));
+					if(s_family != 0)
+						hints.ai_family = s_family;
+					start = NULL;
+					res = NULL;
+					if(getaddrinfo(s_host, NULL, s_family != 0 ? &hints : NULL, &res) == 0) {
+						start = res;
+						while(res != NULL) {
+							if(janus_network_address_from_sockaddr(res->ai_addr, &addr) == 0 &&
+									janus_network_address_to_string_buffer(&addr, &addr_buf) == 0) {
+								/* Resolved */
+								resolved_host = janus_network_address_string_from_buffer(&addr_buf);
+								freeaddrinfo(start);
+								start = NULL;
+								break;
+							}
+							res = res->ai_next;
+						}
+					}
+					if(resolved_host == NULL) {
+						if(start)
+							freeaddrinfo(start);
+						JANUS_LOG(LOG_ERR, "Could not resolve address (%s)...\n", s_host);
+						error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+						g_snprintf(error_cause, 512, "Could not resolve address (%s)...", s_host);
+						goto prepare_response;
+					}
+					/* Add the resolved address to the JSON object, so that we can use it later */
+					json_object_set_new(s, "host", json_string(resolved_host));
+				}
+			}
 		}
-		guint32 audio_handle = 0;
-		guint32 video_handle[3] = {0, 0, 0};
-		guint32 data_handle = 0;
-		if(audio_port > 0) {
-			audio_handle = janus_videoroom_rtp_forwarder_add_helper(publisher, host, audio_port, audio_rtcp_port, audio_pt, audio_ssrc,
-				FALSE, srtp_suite, srtp_crypto, 0, FALSE, FALSE);
-		}
-		if(video_port[0] > 0) {
-			video_handle[0] = janus_videoroom_rtp_forwarder_add_helper(publisher, host, video_port[0], video_rtcp_port, video_pt[0], video_ssrc[0],
-				simulcast, srtp_suite, srtp_crypto, 0, TRUE, FALSE);
-		}
-		if(video_port[1] > 0) {
-			video_handle[1] = janus_videoroom_rtp_forwarder_add_helper(publisher, host, video_port[1], 0, video_pt[1], video_ssrc[1],
-				FALSE, srtp_suite, srtp_crypto, 1, TRUE, FALSE);
+		/* We may need to SRTP-encrypt this stream */
+		int srtp_suite = 0;
+		const char *srtp_crypto = NULL;
+		json_t *s_suite = json_object_get(root, "srtp_suite");
+		json_t *s_crypto = json_object_get(root, "srtp_crypto");
+		if(s_suite && s_crypto) {
+			srtp_suite = json_integer_value(s_suite);
+			if(srtp_suite != 32 && srtp_suite != 80) {
+				JANUS_LOG(LOG_ERR, "Invalid SRTP suite (%d)\n", srtp_suite);
+				error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+				g_snprintf(error_cause, 512, "Invalid SRTP suite (%d)", srtp_suite);
+				goto prepare_response;
+			}
+			srtp_crypto = json_string_value(s_crypto);
 		}
-		if(video_port[2] > 0) {
-			video_handle[2] = janus_videoroom_rtp_forwarder_add_helper(publisher, host, video_port[2], 0, video_pt[2], video_ssrc[2],
-				FALSE, srtp_suite, srtp_crypto, 2, TRUE, FALSE);
+		/* Look for room and publisher */
+		guint64 room_id = json_integer_value(room);
+		guint64 publisher_id = json_integer_value(pub_id);
+		janus_mutex_lock(&rooms_mutex);
+		janus_videoroom *videoroom = NULL;
+		error_code = janus_videoroom_access_room(root, TRUE, FALSE, &videoroom, error_cause, sizeof(error_cause));
+		janus_mutex_unlock(&rooms_mutex);
+		if(error_code != 0)
+			goto prepare_response;
+		janus_refcount_increase(&videoroom->ref);
+		janus_mutex_lock(&videoroom->mutex);
+		janus_videoroom_publisher *publisher = g_hash_table_lookup(videoroom->participants, &publisher_id);
+		if(publisher == NULL) {
+			janus_mutex_unlock(&videoroom->mutex);
+			janus_refcount_decrease(&videoroom->ref);
+			JANUS_LOG(LOG_ERR, "No such publisher (%"SCNu64")\n", publisher_id);
+			error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+			g_snprintf(error_cause, 512, "No such feed (%"SCNu64")", publisher_id);
+			goto prepare_response;
 		}
-		if(data_port > 0) {
-			data_handle = janus_videoroom_rtp_forwarder_add_helper(publisher, host, data_port, 0, 0, 0, FALSE, 0, NULL, 0, FALSE, TRUE);
+		janus_refcount_increase(&publisher->ref);	/* This is just to handle the request for now */
+		janus_mutex_lock(&publisher->rtp_forwarders_mutex);
+		if(publisher->udp_sock <= 0) {
+			publisher->udp_sock = socket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP);
+			int v6only = 0;
+			if(publisher->udp_sock <= 0 ||
+					setsockopt(publisher->udp_sock, IPPROTO_IPV6, IPV6_V6ONLY, &v6only, sizeof(v6only)) != 0) {
+				janus_mutex_unlock(&publisher->rtp_forwarders_mutex);
+				janus_refcount_decrease(&publisher->ref);
+				janus_mutex_unlock(&videoroom->mutex);
+				janus_refcount_decrease(&videoroom->ref);
+				JANUS_LOG(LOG_ERR, "Could not open UDP socket for RTP stream for publisher (%"SCNu64")\n", publisher_id);
+				error_code = JANUS_VIDEOROOM_ERROR_UNKNOWN_ERROR;
+				g_snprintf(error_cause, 512, "Could not open UDP socket for RTP stream");
+				goto prepare_response;
+			}
 		}
-		janus_mutex_unlock(&videoroom->mutex);
+		/* Are we using the new approach, or the old deprecated one? */
 		response = json_object();
 		json_t *rtp_stream = json_object();
-		if(audio_handle > 0) {
-			json_object_set_new(rtp_stream, "audio_stream_id", json_integer(audio_handle));
-			json_object_set_new(rtp_stream, "audio", json_integer(audio_port));
-			/* Also notify event handlers */
-			if(notify_events && gateway->events_is_enabled()) {
-				json_t *info = json_object();
-				json_object_set_new(info, "event", json_string("rtp_forward"));
-				json_object_set_new(info, "room", json_integer(room_id));
-				json_object_set_new(info, "publisher_id", json_integer(publisher_id));
-				json_object_set_new(info, "media", json_string("audio"));
-				json_object_set_new(info, "stream_id", json_integer(audio_handle));
-				json_object_set_new(info, "host", json_string(host));
-				json_object_set_new(info, "port", json_integer(audio_port));
-				gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+		janus_videoroom_publisher_stream *stream = NULL;
+		json_t *new_forwarders = NULL;
+		if(streams != NULL) {
+			/* New approach: iterate on all objects, and create the related forwarder(s) */
+			new_forwarders = json_array();
+			size_t i = 0;
+			for(i=0; i<json_array_size(streams); i++) {
+				json_t *s = json_array_get(streams, i);
+				json_t *stream_mid = json_object_get(s, "mid");
+				const char *mid = json_string_value(stream_mid);
+				janus_mutex_lock(&publisher->streams_mutex);
+				stream = g_hash_table_lookup(publisher->streams_bymid, mid);
+				janus_mutex_unlock(&publisher->streams_mutex);
+				if(stream == NULL) {
+					/* FIXME Should we return an error instead? */
+					JANUS_LOG(LOG_WARN, "No such stream with mid '%s', skipping forwarder...\n", mid);
+					continue;
+				}
+				janus_videoroom_rtp_forwarder *f = NULL;
+				json_t *stream_host = json_object_get(s, "host");
+				host = json_string_value(stream_host) ? json_string_value(stream_host) : json_string_value(json_host);
+				json_t *stream_port = json_object_get(s, "port");
+				if(stream->type == JANUS_VIDEOROOM_MEDIA_DATA) {
+					/* We have all we need */
+					f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+						host, json_integer_value(stream_port), 0, 0, 0, FALSE, 0, NULL, 0, FALSE, TRUE);
+					if(f) {
+						json_t *rtpf = janus_videoroom_rtp_forwarder_summary(f);
+						json_array_append_new(new_forwarders, rtpf);
+						/* Also notify event handlers */
+						if(notify_events && gateway->events_is_enabled()) {
+							json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+							json_object_set_new(info, "event", json_string("rtp_forward"));
+							json_object_set_new(info, "room", json_integer(room_id));
+							json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+							gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+						}
+					}
+					continue;
+				}
+				/* If we got here, it's RTP media, check the other properties too */
+				json_t *stream_pt = json_object_get(root, "pt");
+				json_t *stream_ssrc = json_object_get(root, "ssrc");
+				json_t *stream_rtcp_port = json_object_get(s, "rtcp_port");
+				if(stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO) {
+					f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+						host, json_integer_value(stream_port), stream_rtcp_port ? json_integer_value(stream_rtcp_port) : -1,
+						json_integer_value(stream_pt), json_integer_value(stream_ssrc),
+						FALSE, srtp_suite, srtp_crypto, 0, FALSE, FALSE);
+					if(f) {
+						json_t *rtpf = janus_videoroom_rtp_forwarder_summary(f);
+						json_array_append_new(new_forwarders, rtpf);
+						/* Also notify event handlers */
+						if(notify_events && gateway->events_is_enabled()) {
+							json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+							json_object_set_new(info, "event", json_string("rtp_forward"));
+							json_object_set_new(info, "room", json_integer(room_id));
+							json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+							gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+						}
+					}
+				} else {
+					json_t *stream_simulcast = json_object_get(root, "simulcast");
+					f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+						host, json_integer_value(stream_port), stream_rtcp_port ? json_integer_value(stream_rtcp_port) : -1,
+						json_integer_value(stream_pt), json_integer_value(stream_ssrc),
+						json_is_true(stream_simulcast), srtp_suite, srtp_crypto, 0, TRUE, FALSE);
+					if(f) {
+						json_t *rtpf = janus_videoroom_rtp_forwarder_summary(f);
+						json_array_append_new(new_forwarders, rtpf);
+						/* Also notify event handlers */
+						if(notify_events && gateway->events_is_enabled()) {
+							json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+							json_object_set_new(info, "event", json_string("rtp_forward"));
+							json_object_set_new(info, "room", json_integer(room_id));
+							json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+							gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+						}
+					}
+					if(!json_is_true(stream_simulcast)) {
+						/* Check if there's simulcast substreams we need to relay */
+						stream_port = json_object_get(s, "port_2");
+						stream_pt = json_object_get(root, "pt_2");
+						stream_ssrc = json_object_get(root, "ssrc_2");
+						if(json_integer_value(stream_port) > 0) {
+							f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+								host, json_integer_value(stream_port), 0,
+								json_integer_value(stream_pt), json_integer_value(stream_ssrc),
+								FALSE, srtp_suite, srtp_crypto, 1, TRUE, FALSE);
+							if(f) {
+								json_t *rtpf = janus_videoroom_rtp_forwarder_summary(f);
+								json_array_append_new(new_forwarders, rtpf);
+								/* Also notify event handlers */
+								if(notify_events && gateway->events_is_enabled()) {
+									json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+									json_object_set_new(info, "event", json_string("rtp_forward"));
+									json_object_set_new(info, "room", json_integer(room_id));
+									json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+									gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+								}
+							}
+						}
+						stream_port = json_object_get(s, "port_3");
+						stream_pt = json_object_get(root, "pt_3");
+						stream_ssrc = json_object_get(root, "ssrc_3");
+						if(json_integer_value(stream_port) > 0) {
+							f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+								host, json_integer_value(stream_port), 0,
+								json_integer_value(stream_pt), json_integer_value(stream_ssrc),
+								FALSE, srtp_suite, srtp_crypto, 2, TRUE, FALSE);
+							if(f) {
+								json_t *rtpf = janus_videoroom_rtp_forwarder_summary(f);
+								json_array_append_new(new_forwarders, rtpf);
+								/* Also notify event handlers */
+								if(notify_events && gateway->events_is_enabled()) {
+									json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+									json_object_set_new(info, "event", json_string("rtp_forward"));
+									json_object_set_new(info, "room", json_integer(room_id));
+									json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+									gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+								}
+							}
+						}
+					}
+				}
 			}
-		}
-		if(video_handle[0] > 0 || video_handle[1] > 0 || video_handle[2] > 0) {
-			janus_videoroom_reqfir(publisher, "New RTP forward publisher");
-			/* Done */
-			if(video_handle[0] > 0) {
-				json_object_set_new(rtp_stream, "video_stream_id", json_integer(video_handle[0]));
-				json_object_set_new(rtp_stream, "video", json_integer(video_port[0]));
-				/* Also notify event handlers */
-				if(notify_events && gateway->events_is_enabled()) {
-					json_t *info = json_object();
-					json_object_set_new(info, "event", json_string("rtp_forward"));
-					json_object_set_new(info, "room", json_integer(room_id));
-					json_object_set_new(info, "publisher_id", json_integer(publisher_id));
-					json_object_set_new(info, "media", json_string("video"));
-					if(video_handle[1] > 0 || video_handle[2] > 0)
-						json_object_set_new(info, "video_substream", json_integer(0));
-					json_object_set_new(info, "stream_id", json_integer(video_handle[0]));
-					json_object_set_new(info, "host", json_string(host));
-					json_object_set_new(info, "port", json_integer(video_port[0]));
-					gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+		} else {
+			/* Old deprecated approach: return the legacy info as well */
+			int video_port[3] = {-1, -1, -1}, video_rtcp_port = -1, video_pt[3] = {0, 0, 0};
+			uint32_t video_ssrc[3] = {0, 0, 0};
+			int audio_port = -1, audio_rtcp_port = -1, audio_pt = 0;
+			uint32_t audio_ssrc = 0;
+			int data_port = -1;
+			/* There may be multiple target video ports (e.g., publisher simulcasting) */
+			json_t *vid_port = json_object_get(root, "video_port");
+			if(vid_port) {
+				video_port[0] = json_integer_value(vid_port);
+				json_t *pt = json_object_get(root, "video_pt");
+				if(pt)
+					video_pt[0] = json_integer_value(pt);
+				json_t *ssrc = json_object_get(root, "video_ssrc");
+				if(ssrc)
+					video_ssrc[0] = json_integer_value(ssrc);
+			}
+			vid_port = json_object_get(root, "video_port_2");
+			if(vid_port) {
+				video_port[1] = json_integer_value(vid_port);
+				json_t *pt = json_object_get(root, "video_pt_2");
+				if(pt)
+					video_pt[1] = json_integer_value(pt);
+				json_t *ssrc = json_object_get(root, "video_ssrc_2");
+				if(ssrc)
+					video_ssrc[1] = json_integer_value(ssrc);
+			}
+			vid_port = json_object_get(root, "video_port_3");
+			if(vid_port) {
+				video_port[2] = json_integer_value(vid_port);
+				json_t *pt = json_object_get(root, "video_pt_3");
+				if(pt)
+					video_pt[2] = json_integer_value(pt);
+				json_t *ssrc = json_object_get(root, "video_ssrc_3");
+				if(ssrc)
+					video_ssrc[2] = json_integer_value(ssrc);
+			}
+			json_t *vid_rtcp_port = json_object_get(root, "video_rtcp_port");
+			if(vid_rtcp_port)
+				video_rtcp_port = json_integer_value(vid_rtcp_port);
+			/* Audio target */
+			json_t *au_port = json_object_get(root, "audio_port");
+			if(au_port) {
+				audio_port = json_integer_value(au_port);
+				json_t *pt = json_object_get(root, "audio_pt");
+				if(pt)
+					audio_pt = json_integer_value(pt);
+				json_t *ssrc = json_object_get(root, "audio_ssrc");
+				if(ssrc)
+					audio_ssrc = json_integer_value(ssrc);
+			}
+			json_t *au_rtcp_port = json_object_get(root, "audio_rtcp_port");
+			if(au_rtcp_port)
+				audio_rtcp_port = json_integer_value(au_rtcp_port);
+			/* Data target */
+			json_t *d_port = json_object_get(root, "data_port");
+			if(d_port) {
+				data_port = json_integer_value(d_port);
+			}
+			/* Do we need to forward multiple simulcast streams to a single endpoint? */
+			gboolean simulcast = FALSE;
+			if(json_object_get(root, "simulcast") != NULL)
+				simulcast = json_is_true(json_object_get(root, "simulcast"));
+			if(simulcast) {
+				/* We do, disable the other video ports if they were requested */
+				video_port[1] = -1;
+				video_port[2] = -1;
+			}
+			/* Create all the forwarders we need */
+			janus_videoroom_rtp_forwarder *f = NULL;
+			guint32 audio_handle = 0;
+			guint32 video_handle[3] = {0, 0, 0};
+			guint32 data_handle = 0;
+			janus_mutex_lock(&publisher->streams_mutex);
+			if(audio_port > 0) {
+				/* FIXME Find the audio stream */
+				GList *temp = publisher->streams;
+				while(temp) {
+					stream = (janus_videoroom_publisher_stream *)temp->data;
+					if(stream && stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO) {
+						/* FIXME Found */
+						break;
+					}
+					stream = NULL;
+					temp = temp->next;
+				}
+				if(stream == NULL) {
+					JANUS_LOG(LOG_WARN, "Couldn't find any audio stream to forward, skipping...\n");
+				} else {
+					f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+						host, audio_port, audio_rtcp_port, audio_pt, audio_ssrc,
+						FALSE, srtp_suite, srtp_crypto, 0, FALSE, FALSE);
+					audio_handle = f ? f->stream_id : 0;
+					/* Also notify event handlers */
+					if(f != NULL && notify_events && gateway->events_is_enabled()) {
+						json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+						json_object_set_new(info, "event", json_string("rtp_forward"));
+						json_object_set_new(info, "room", json_integer(room_id));
+						json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+						gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+					}
 				}
 			}
-			if(video_handle[1] > 0) {
-				json_object_set_new(rtp_stream, "video_stream_id_2", json_integer(video_handle[1]));
-				json_object_set_new(rtp_stream, "video_2", json_integer(video_port[1]));
-				/* Also notify event handlers */
-				if(notify_events && gateway->events_is_enabled()) {
-					json_t *info = json_object();
-					json_object_set_new(info, "event", json_string("rtp_forward"));
-					json_object_set_new(info, "room", json_integer(room_id));
-					json_object_set_new(info, "publisher_id", json_integer(publisher_id));
-					json_object_set_new(info, "media", json_string("video"));
-					json_object_set_new(info, "video_substream", json_integer(1));
-					json_object_set_new(info, "stream_id", json_integer(video_handle[1]));
-					json_object_set_new(info, "host", json_string(host));
-					json_object_set_new(info, "port", json_integer(video_port[1]));
-					gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+			if(video_port[0] > 0 || video_port[1] > 0 || video_port[2] > 0) {
+				/* FIXME Find the video stream */
+				GList *temp = publisher->streams;
+				while(temp) {
+					stream = (janus_videoroom_publisher_stream *)temp->data;
+					if(stream && stream->type == JANUS_VIDEOROOM_MEDIA_VIDEO) {
+						/* FIXME Found */
+						break;
+					}
+					stream = NULL;
+					temp = temp->next;
+				}
+				if(stream == NULL) {
+					JANUS_LOG(LOG_WARN, "Couldn't find any video stream to forward, skipping...\n");
+				} else {
+					if(video_port[0] > 0) {
+						f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+							host, video_port[0], video_rtcp_port, video_pt[0], video_ssrc[0],
+							simulcast, srtp_suite, srtp_crypto, 0, TRUE, FALSE);
+						video_handle[0] = f ? f->stream_id : 0;
+						/* Also notify event handlers */
+						if(f != NULL && notify_events && gateway->events_is_enabled()) {
+							json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+							json_object_set_new(info, "event", json_string("rtp_forward"));
+							json_object_set_new(info, "room", json_integer(room_id));
+							json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+							gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+						}
+					}
+					if(video_port[1] > 0) {
+						f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+							host, video_port[1], 0, video_pt[1], video_ssrc[1],
+							FALSE, srtp_suite, srtp_crypto, 1, TRUE, FALSE);
+						video_handle[1] = f ? f->stream_id : 0;
+						/* Also notify event handlers */
+						if(f != NULL && notify_events && gateway->events_is_enabled()) {
+							json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+							json_object_set_new(info, "event", json_string("rtp_forward"));
+							json_object_set_new(info, "room", json_integer(room_id));
+							json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+							gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+						}
+					}
+					if(video_port[2] > 0) {
+						f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+							host, video_port[2], 0, video_pt[2], video_ssrc[2],
+							FALSE, srtp_suite, srtp_crypto, 2, TRUE, FALSE);
+						video_handle[2] = f ? f->stream_id : 0;
+						/* Also notify event handlers */
+						if(f != NULL && notify_events && gateway->events_is_enabled()) {
+							json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+							json_object_set_new(info, "event", json_string("rtp_forward"));
+							json_object_set_new(info, "room", json_integer(room_id));
+							json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+							gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+						}
+					}
+					janus_videoroom_reqpli(stream, "New RTP forward publisher");
 				}
 			}
-			if(video_handle[2] > 0) {
-				json_object_set_new(rtp_stream, "video_stream_id_3", json_integer(video_handle[2]));
-				json_object_set_new(rtp_stream, "video_3", json_integer(video_port[2]));
-				/* Also notify event handlers */
-				if(notify_events && gateway->events_is_enabled()) {
-					json_t *info = json_object();
-					json_object_set_new(info, "event", json_string("rtp_forward"));
-					json_object_set_new(info, "room", json_integer(room_id));
-					json_object_set_new(info, "publisher_id", json_integer(publisher_id));
-					json_object_set_new(info, "media", json_string("video"));
-					json_object_set_new(info, "video_substream", json_integer(2));
-					json_object_set_new(info, "stream_id", json_integer(video_handle[2]));
-					json_object_set_new(info, "host", json_string(host));
-					json_object_set_new(info, "port", json_integer(video_port[2]));
-					gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+			if(data_port > 0) {
+				/* FIXME Find the data stream */
+				GList *temp = publisher->streams;
+				while(temp) {
+					stream = (janus_videoroom_publisher_stream *)temp->data;
+					if(stream && stream->type == JANUS_VIDEOROOM_MEDIA_DATA) {
+						/* FIXME Found */
+						break;
+					}
+					stream = NULL;
+					temp = temp->next;
+				}
+				if(stream == NULL) {
+					JANUS_LOG(LOG_WARN, "Couldn't find any data stream to forward, skipping...\n");
+				} else {
+					f = janus_videoroom_rtp_forwarder_add_helper(publisher, stream,
+						host, data_port, 0, 0, 0, FALSE, 0, NULL, 0, FALSE, TRUE);
+					data_handle = f ? f->stream_id : 0;
+					/* Also notify event handlers */
+					if(f != NULL && notify_events && gateway->events_is_enabled()) {
+						json_t *info = janus_videoroom_rtp_forwarder_summary(f);
+						json_object_set_new(info, "event", json_string("rtp_forward"));
+						json_object_set_new(info, "room", json_integer(room_id));
+						json_object_set_new(info, "publisher_id", json_integer(publisher_id));
+						gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+					}
 				}
 			}
-		}
-		if(data_handle > 0) {
-			json_object_set_new(rtp_stream, "data_stream_id", json_integer(data_handle));
-			json_object_set_new(rtp_stream, "data", json_integer(data_port));
-			/* Also notify event handlers */
-			if(notify_events && gateway->events_is_enabled()) {
-				json_t *info = json_object();
-				json_object_set_new(info, "event", json_string("rtp_forward"));
-				json_object_set_new(info, "room", json_integer(room_id));
-				json_object_set_new(info, "publisher_id", json_integer(publisher_id));
-				json_object_set_new(info, "media", json_string("data"));
-				json_object_set_new(info, "stream_id", json_integer(data_handle));
-				json_object_set_new(info, "host", json_string(host));
-				json_object_set_new(info, "port", json_integer(data_port));
-				gateway->notify_event(&janus_videoroom_plugin, NULL, info);
+			janus_mutex_unlock(&publisher->streams_mutex);
+			if(audio_handle > 0) {
+				json_object_set_new(rtp_stream, "audio_stream_id", json_integer(audio_handle));
+				json_object_set_new(rtp_stream, "audio", json_integer(audio_port));
+			}
+			if(video_handle[0] > 0 || video_handle[1] > 0 || video_handle[2] > 0) {
+				/* Done */
+				if(video_handle[0] > 0) {
+					json_object_set_new(rtp_stream, "video_stream_id", json_integer(video_handle[0]));
+					json_object_set_new(rtp_stream, "video", json_integer(video_port[0]));
+				}
+				if(video_handle[1] > 0) {
+					json_object_set_new(rtp_stream, "video_stream_id_2", json_integer(video_handle[1]));
+					json_object_set_new(rtp_stream, "video_2", json_integer(video_port[1]));
+				}
+				if(video_handle[2] > 0) {
+					json_object_set_new(rtp_stream, "video_stream_id_3", json_integer(video_handle[2]));
+					json_object_set_new(rtp_stream, "video_3", json_integer(video_port[2]));
+				}
+			}
+			if(data_handle > 0) {
+				json_object_set_new(rtp_stream, "data_stream_id", json_integer(data_handle));
+				json_object_set_new(rtp_stream, "data", json_integer(data_port));
 			}
 		}
+		janus_mutex_unlock(&publisher->rtp_forwarders_mutex);
+		janus_mutex_unlock(&videoroom->mutex);
 		/* These two unrefs are related to the message handling */
 		janus_refcount_decrease(&publisher->ref);
 		janus_refcount_decrease(&videoroom->ref);
 		json_object_set_new(rtp_stream, "host", json_string(host));
+		if(new_forwarders != NULL)
+			json_object_set_new(rtp_stream, "forwarders", new_forwarders);
 		json_object_set_new(response, "publisher_id", json_integer(publisher_id));
 		json_object_set_new(response, "rtp_stream", rtp_stream);
 		json_object_set_new(response, "room", json_integer(room_id));
@@ -3676,20 +4696,32 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		}
 		janus_refcount_increase(&publisher->ref);	/* Just to handle the message now */
 		janus_mutex_lock(&publisher->rtp_forwarders_mutex);
-		if(!g_hash_table_remove(publisher->rtp_forwarders, GUINT_TO_POINTER(stream_id))) {
-			janus_mutex_unlock(&publisher->rtp_forwarders_mutex);
-			janus_refcount_decrease(&publisher->ref);
-			janus_mutex_unlock(&videoroom->mutex);
-			janus_refcount_decrease(&videoroom->ref);
-			JANUS_LOG(LOG_ERR, "No such stream (%"SCNu32")\n", stream_id);
-			error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
-			g_snprintf(error_cause, 512, "No such stream (%"SCNu32")", stream_id);
-			goto prepare_response;
+		/* FIXME Find the forwarder by iterating on all the streams */
+		gboolean found = FALSE;
+		GList *temp = publisher->streams;
+		while(temp) {
+			janus_videoroom_publisher_stream *stream = (janus_videoroom_publisher_stream *)temp->data;
+			janus_mutex_lock(&stream->rtp_forwarders_mutex);
+			if(g_hash_table_remove(stream->rtp_forwarders, GUINT_TO_POINTER(stream_id))) {
+				janus_mutex_unlock(&stream->rtp_forwarders_mutex);
+				/* Found, remove from global index too */
+				g_hash_table_remove(publisher->rtp_forwarders, GUINT_TO_POINTER(stream_id));
+				found = TRUE;
+				break;
+			}
+			janus_mutex_unlock(&stream->rtp_forwarders_mutex);
+			temp = temp->next;
 		}
 		janus_mutex_unlock(&publisher->rtp_forwarders_mutex);
 		janus_refcount_decrease(&publisher->ref);
 		janus_mutex_unlock(&videoroom->mutex);
 		janus_refcount_decrease(&videoroom->ref);
+		if(!found) {
+			JANUS_LOG(LOG_ERR, "No such stream (%"SCNu32")\n", stream_id);
+			error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+			g_snprintf(error_cause, 512, "No such stream (%"SCNu32")", stream_id);
+			goto prepare_response;
+		}
 		response = json_object();
 		json_object_set_new(response, "videoroom", json_string("stop_rtp_forward"));
 		json_object_set_new(response, "room", json_integer(room_id));
@@ -3867,9 +4899,6 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		}
 		participant->kicked = TRUE;
 		participant->session->started = FALSE;
-		participant->audio_active = FALSE;
-		participant->video_active = FALSE;
-		participant->data_active = FALSE;
 		/* Prepare an event for this */
 		json_t *kicked = json_object();
 		json_object_set_new(kicked, "videoroom", json_string("event"));
@@ -3889,9 +4918,6 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 				janus_videoroom_subscriber *subscriber = (janus_videoroom_subscriber *)s->data;
 				if(subscriber) {
 					subscriber->kicked = TRUE;
-					subscriber->audio = FALSE;
-					subscriber->video = FALSE;
-					subscriber->data = FALSE;
 					/* FIXME We should also close the PeerConnection, but we risk race conditions if we do it here,
 					 * so for now we mark the subscriber as kicked and prevent it from getting any media after this */
 				}
@@ -3939,10 +4965,22 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 			json_object_set_new(pl, "id", json_integer(p->user_id));
 			if(p->display)
 				json_object_set_new(pl, "display", json_string(p->display));
-			json_object_set_new(pl, "publisher", (p->sdp && p->session->started) ? json_true() : json_false());
-			if((p->sdp && p->session->started)) {
-				if(p->audio_level_extmap_id > 0)
-					json_object_set_new(pl, "talking", p->talking ? json_true() : json_false());
+			json_object_set_new(pl, "publisher", p->session->started ? json_true() : json_false());
+			/* FIXME To see if the participant is talking, we need to find the audio stream(s) */
+			if(p->session->started) {
+				gboolean found = FALSE, talking = FALSE;
+				GList *temp = p->streams;
+				while(temp) {
+					janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+					if(ps && ps->type == JANUS_VIDEOROOM_MEDIA_AUDIO &&
+							ps->audio_level_extmap_id > 0) {
+						found = TRUE;
+						talking |= ps->talking;
+					}
+					temp = temp->next;
+				}
+				if(found)
+					json_object_set_new(pl, "talking", talking ? json_true() : json_false());
 			}
 			json_array_append_new(list, pl);
 		}
@@ -3993,65 +5031,35 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		g_hash_table_iter_init(&iter, videoroom->participants);
 		while (!g_atomic_int_get(&videoroom->destroyed) && g_hash_table_iter_next(&iter, NULL, &value)) {
 			janus_videoroom_publisher *p = value;
-			janus_mutex_lock(&p->rtp_forwarders_mutex);
-			if(g_hash_table_size(p->rtp_forwarders) == 0) {
-				janus_mutex_unlock(&p->rtp_forwarders_mutex);
-				continue;
-			}
+			janus_videoroom_publisher_stream *ps = NULL;
 			json_t *pl = json_object();
 			json_object_set_new(pl, "publisher_id", json_integer(p->user_id));
 			if(p->display)
 				json_object_set_new(pl, "display", json_string(p->display));
 			json_t *flist = json_array();
-			GHashTableIter iter_f;
-			gpointer key_f, value_f;
-			g_hash_table_iter_init(&iter_f, p->rtp_forwarders);
-			while(g_hash_table_iter_next(&iter_f, &key_f, &value_f)) {
-				json_t *fl = json_object();
-				guint32 rpk = GPOINTER_TO_UINT(key_f);
-				janus_videoroom_rtp_forwarder *rpv = value_f;
-				char address[100];
-				if(rpv->serv_addr.sin_family == AF_INET) {
-					json_object_set_new(fl, "ip", json_string(
-						inet_ntop(AF_INET, &rpv->serv_addr.sin_addr, address, sizeof(address))));
-				} else {
-					json_object_set_new(fl, "ip", json_string(
-						inet_ntop(AF_INET6, &rpv->serv_addr6.sin6_addr, address, sizeof(address))));
-				}
-				if(rpv->is_data) {
-					json_object_set_new(fl, "data_stream_id", json_integer(rpk));
-					json_object_set_new(fl, "port", json_integer(ntohs(rpv->serv_addr.sin_port)));
-				} else if(rpv->is_video) {
-					json_object_set_new(fl, "video_stream_id", json_integer(rpk));
-					json_object_set_new(fl, "port", json_integer(ntohs(rpv->serv_addr.sin_port)));
-					if(rpv->local_rtcp_port > 0)
-						json_object_set_new(fl, "local_rtcp_port", json_integer(rpv->local_rtcp_port));
-					if(rpv->remote_rtcp_port > 0)
-						json_object_set_new(fl, "remote_rtcp_port", json_integer(rpv->remote_rtcp_port));
-					if(rpv->payload_type)
-						json_object_set_new(fl, "pt", json_integer(rpv->payload_type));
-					if(rpv->ssrc)
-						json_object_set_new(fl, "ssrc", json_integer(rpv->ssrc));
-					if(rpv->substream)
-						json_object_set_new(fl, "substream", json_integer(rpv->substream));
-				} else {
-					json_object_set_new(fl, "audio_stream_id", json_integer(rpk));
-					json_object_set_new(fl, "port", json_integer(ntohs(rpv->serv_addr.sin_port)));
-					if(rpv->local_rtcp_port > 0)
-						json_object_set_new(fl, "local_rtcp_port", json_integer(rpv->local_rtcp_port));
-					if(rpv->remote_rtcp_port > 0)
-						json_object_set_new(fl, "remote_rtcp_port", json_integer(rpv->remote_rtcp_port));
-					if(rpv->payload_type)
-						json_object_set_new(fl, "pt", json_integer(rpv->payload_type));
-					if(rpv->ssrc)
-						json_object_set_new(fl, "ssrc", json_integer(rpv->ssrc));
-				}
-				if(rpv->is_srtp)
-					json_object_set_new(fl, "srtp", json_true());
-				json_array_append_new(flist, fl);
+			/* FIXME We need to iterate on all media streams */
+			GList *temp = p->streams;
+			while(temp) {
+				ps = (janus_videoroom_publisher_stream *)temp->data;
+				janus_mutex_lock(&ps->rtp_forwarders_mutex);
+				if(g_hash_table_size(ps->rtp_forwarders) == 0) {
+					janus_mutex_unlock(&ps->rtp_forwarders_mutex);
+					temp = temp->next;
+					continue;
+				}
+				GHashTableIter iter_f;
+				gpointer key_f, value_f;
+				g_hash_table_iter_init(&iter_f, ps->rtp_forwarders);
+				while(g_hash_table_iter_next(&iter_f, &key_f, &value_f)) {
+					janus_videoroom_rtp_forwarder *rpv = value_f;
+					/* Return a different, media-agnostic, format */
+					json_t *fl = janus_videoroom_rtp_forwarder_summary(rpv);
+					json_array_append_new(flist, fl);
+				}
+				janus_mutex_unlock(&ps->rtp_forwarders_mutex);
+				json_object_set_new(pl, "forwarders", flist);
+				temp = temp->next;
 			}
-			janus_mutex_unlock(&p->rtp_forwarders_mutex);
-			json_object_set_new(pl, "rtp_forwarder", flist);
 			json_array_append_new(list, pl);
 		}
 		janus_mutex_unlock(&videoroom->mutex);
@@ -4059,7 +5067,7 @@ static json_t *janus_videoroom_process_synchronous_request(janus_videoroom_sessi
 		response = json_object();
 		json_object_set_new(response, "videoroom", json_string("forwarders"));
 		json_object_set_new(response, "room", json_integer(room_id));
-		json_object_set_new(response, "rtp_forwarders", list);
+		json_object_set_new(response, "publishers", list);
 		goto prepare_response;
 	} else {
 		/* Not a request we recognize, don't do anything */
@@ -4143,7 +5151,7 @@ struct janus_plugin_result *janus_videoroom_handle_message(janus_plugin_session
 	} else if(!strcasecmp(request_text, "join") || !strcasecmp(request_text, "joinandconfigure")
 			|| !strcasecmp(request_text, "configure") || !strcasecmp(request_text, "publish") || !strcasecmp(request_text, "unpublish")
 			|| !strcasecmp(request_text, "start") || !strcasecmp(request_text, "pause") || !strcasecmp(request_text, "switch")
-			|| !strcasecmp(request_text, "leave")) {
+			|| !strcasecmp(request_text, "subscribe") || !strcasecmp(request_text, "unsubscribe") || !strcasecmp(request_text, "leave")) {
 		/* These messages are handled asynchronously */
 
 		janus_videoroom_message *msg = g_malloc(sizeof(janus_videoroom_message));
@@ -4244,7 +5252,7 @@ void janus_videoroom_setup_media(janus_plugin_session *handle) {
 	session->started = TRUE;
 	if(session->participant) {
 		/* If this is a publisher, notify all subscribers about the fact they can
-		 * now subscribe; if this is a subscriber, instead, ask the publisher a FIR */
+		 * now subscribe; if this is a subscriber, instead, ask the publisher a PLI */
 		if(session->participant_type == janus_videoroom_p_type_publisher) {
 			janus_videoroom_publisher *participant = janus_videoroom_session_get_publisher(session);
 			/* Notify all other participants that there's a new boy in town */
@@ -4253,14 +5261,41 @@ void janus_videoroom_setup_media(janus_plugin_session *handle) {
 			json_object_set_new(pl, "id", json_integer(participant->user_id));
 			if(participant->display)
 				json_object_set_new(pl, "display", json_string(participant->display));
-			if(participant->audio)
-				json_object_set_new(pl, "audio_codec", json_string(janus_audiocodec_name(participant->acodec)));
-			if(participant->video)
-				json_object_set_new(pl, "video_codec", json_string(janus_videocodec_name(participant->vcodec)));
-			if(participant->ssrc[0] || participant->rid[0])
-				json_object_set_new(pl, "simulcast", json_true());
-			if(participant->audio_level_extmap_id > 0)
-				json_object_set_new(pl, "talking", participant->talking ? json_true() : json_false());
+			/* Add proper info on all the streams */
+			gboolean audio_added = FALSE, video_added = FALSE;
+			json_t *media = json_array();
+			GList *temp = participant->streams;
+			while(temp) {
+				janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+				json_t *info = json_object();
+				json_object_set_new(info, "type", json_string(janus_videoroom_media_str(ps->type)));
+				json_object_set_new(info, "mindex", json_integer(ps->mindex));
+				json_object_set_new(info, "mid", json_string(ps->mid));
+				if(ps->description)
+					json_object_set_new(info, "description", json_string(ps->description));
+				if(ps->type == JANUS_VIDEOROOM_MEDIA_AUDIO) {
+					json_object_set_new(info, "codec", json_string(janus_audiocodec_name(ps->acodec)));
+					/* FIXME For backwards compatibility, we need audio_codec in the global info */
+					if(!audio_added) {
+						audio_added = TRUE;
+						json_object_set_new(pl, "audio_codec", json_string(janus_audiocodec_name(ps->acodec)));
+					}
+				} else if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO) {
+					json_object_set_new(info, "codec", json_string(janus_videocodec_name(ps->vcodec)));
+					/* FIXME For backwards compatibility, we need video_codec in the global info */
+					if(!video_added) {
+						video_added = TRUE;
+						json_object_set_new(pl, "video_codec", json_string(janus_videocodec_name(ps->vcodec)));
+					}
+					if(ps->simulcast)
+						json_object_set_new(info, "simulcast", json_true());
+					if(ps->svc)
+						json_object_set_new(info, "svc", json_true());
+				}
+				json_array_append_new(media, info);
+				temp = temp->next;
+			}
+			json_object_set_new(pl, "streams", media);
 			json_array_append_new(list, pl);
 			json_t *pub = json_object();
 			json_object_set_new(pub, "videoroom", json_string("event"));
@@ -4283,18 +5318,25 @@ void janus_videoroom_setup_media(janus_plugin_session *handle) {
 			janus_refcount_decrease(&participant->ref);
 		} else if(session->participant_type == janus_videoroom_p_type_subscriber) {
 			janus_videoroom_subscriber *s = (janus_videoroom_subscriber *)session->participant;
-			if(s && s->feed) {
-				janus_videoroom_publisher *p = s->feed;
-				if(p && p->session) {
-					janus_videoroom_reqfir(p, "New subscriber available");
-					/* Also notify event handlers */
-					if(notify_events && gateway->events_is_enabled()) {
-						json_t *info = json_object();
-						json_object_set_new(info, "event", json_string("subscribed"));
-						json_object_set_new(info, "room", json_integer(p->room_id));
-						json_object_set_new(info, "feed", json_integer(p->user_id));
-						gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
+			if(s && s->streams) {
+				/* Send a PLI for all the video streams we subscribed to */
+				GList *temp = s->streams;
+				while(temp) {
+					janus_videoroom_subscriber_stream *ss = (janus_videoroom_subscriber_stream *)temp->data;
+					janus_videoroom_publisher_stream *ps = ss->publisher_streams ? ss->publisher_streams->data : NULL;
+					if(ps && ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO && ps->publisher && ps->publisher->session) {
+						janus_videoroom_reqpli(ps, "New subscriber available");
 					}
+					temp = temp->next;
+				}
+				/* Also notify event handlers */
+				if(notify_events && gateway->events_is_enabled()) {
+					json_t *info = json_object();
+					json_object_set_new(info, "event", json_string("subscribed"));
+					json_object_set_new(info, "room", json_integer(s->room_id));
+					/* TODO Fix the event to event handlers, we don't have a single feed anymore */
+					//~ json_object_set_new(info, "feed", json_integer(p->user_id));
+					gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
 				}
 			}
 		}
@@ -4302,7 +5344,7 @@ void janus_videoroom_setup_media(janus_plugin_session *handle) {
 	janus_mutex_unlock(&sessions_mutex);
 }
 
-void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized) || !gateway)
 		return;
 	janus_videoroom_session *session = (janus_videoroom_session *)handle->plugin_handle;
@@ -4311,42 +5353,54 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 	janus_videoroom_publisher *participant = janus_videoroom_session_get_publisher_nodebug(session);
 	if(participant == NULL)
 		return;
-	if(g_atomic_int_get(&participant->destroyed) || participant->kicked || participant->room == NULL) {
+	if(g_atomic_int_get(&participant->destroyed) || participant->kicked || !participant->streams || participant->room == NULL) {
 		janus_videoroom_publisher_dereference_nodebug(participant);
 		return;
 	}
 	janus_videoroom *videoroom = participant->room;
 
+	/* Find the stream this packet belongs to */
+	janus_mutex_lock(&participant->streams_mutex);
+	janus_videoroom_publisher_stream *ps = g_hash_table_lookup(participant->streams_byid, GINT_TO_POINTER(mindex));
+	janus_mutex_unlock(&participant->streams_mutex);
+	if(ps == NULL) {
+		/* No stream..? */
+		janus_videoroom_publisher_dereference_nodebug(participant);
+		return;
+	}
+
 	/* In case this is an audio packet and we're doing talk detection, check the audio level extension */
-	if(!video && videoroom->audiolevel_event && participant->audio_active) {
+	if(!video && videoroom->audiolevel_event && ps->active && ps->audio_level_extmap_id > 0) {
 		int level = 0;
-		if(janus_rtp_header_extension_parse_audio_level(buf, len, participant->audio_level_extmap_id, &level) == 0) {
-			participant->audio_dBov_sum += level;
-			participant->audio_active_packets++;
-			participant->audio_dBov_level = level;
-			if(participant->audio_active_packets > 0 && participant->audio_active_packets == videoroom->audio_active_packets) {
+		if(janus_rtp_header_extension_parse_audio_level(buf, len, ps->audio_level_extmap_id, &level) == 0) {
+			ps->audio_dBov_sum += level;
+			ps->audio_active_packets++;
+			ps->audio_dBov_level = level;
+			if(ps->audio_active_packets > 0 && ps->audio_active_packets == videoroom->audio_active_packets) {
 				gboolean notify_talk_event = FALSE;
-				float audio_dBov_avg = (float)participant->audio_dBov_sum/(float)participant->audio_active_packets;
+				float audio_dBov_avg = (float)ps->audio_dBov_sum/(float)ps->audio_active_packets;
 				if(audio_dBov_avg < videoroom->audio_level_average) {
 					/* Participant talking, should we notify all participants? */
-					if(!participant->talking)
+					if(!ps->talking)
 						notify_talk_event = TRUE;
-					participant->talking = TRUE;
+					ps->talking = TRUE;
 				} else {
 					/* Participant not talking anymore, should we notify all participants? */
-					if(participant->talking)
+					if(ps->talking)
 						notify_talk_event = TRUE;
-					participant->talking = FALSE;
+					ps->talking = FALSE;
 				}
-				participant->audio_active_packets = 0;
-				participant->audio_dBov_sum = 0;
+				ps->audio_active_packets = 0;
+				ps->audio_dBov_sum = 0;
 				/* Only notify in case of state changes */
 				if(notify_talk_event) {
 					janus_mutex_lock(&videoroom->mutex);
 					json_t *event = json_object();
-					json_object_set_new(event, "videoroom", json_string(participant->talking ? "talking" : "stopped-talking"));
+					json_object_set_new(event, "videoroom", json_string(ps->talking ? "talking" : "stopped-talking"));
 					json_object_set_new(event, "room", json_integer(videoroom->room_id));
 					json_object_set_new(event, "id", json_integer(participant->user_id));
+					json_object_set_new(event, "mindex", json_integer(ps->mindex));
+					json_object_set_new(event, "mid", json_string(ps->mid));
 					json_object_set_new(event, "audio-level-dBov-avg", json_real(audio_dBov_avg));
 					janus_videoroom_notify_participants(participant, event);
 					json_decref(event);
@@ -4354,9 +5408,11 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 					/* Also notify event handlers */
 					if(notify_events && gateway->events_is_enabled()) {
 						json_t *info = json_object();
-						json_object_set_new(info, "videoroom", json_string(participant->talking ? "talking" : "stopped-talking"));
+						json_object_set_new(info, "videoroom", json_string(ps->talking ? "talking" : "stopped-talking"));
 						json_object_set_new(info, "room", json_integer(videoroom->room_id));
 						json_object_set_new(info, "id", json_integer(participant->user_id));
+						json_object_set_new(event, "mindex", json_integer(ps->mindex));
+						json_object_set_new(event, "mid", json_string(ps->mid));
 						json_object_set_new(event, "audio-level-dBov-avg", json_real(audio_dBov_avg));
 						gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
 					}
@@ -4365,37 +5421,37 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 		}
 	}
 
-	if((!video && participant->audio_active) || (video && participant->video_active)) {
+	if(ps->active) {
 		janus_rtp_header *rtp = (janus_rtp_header *)buf;
 		int sc = video ? 0 : -1;
 		/* Check if we're simulcasting, and if so, keep track of the "layer" */
-		if(video && (participant->ssrc[0] != 0 || participant->rid[0] != NULL)) {
+		if(video && ps->simulcast) {
 			uint32_t ssrc = ntohl(rtp->ssrc);
-			if(ssrc == participant->ssrc[0])
+			if(ssrc == ps->vssrc[0])
 				sc = 0;
-			else if(ssrc == participant->ssrc[1])
+			else if(ssrc == ps->vssrc[1])
 				sc = 1;
-			else if(ssrc == participant->ssrc[2])
+			else if(ssrc == ps->vssrc[2])
 				sc = 2;
-			else if(participant->rid_extmap_id > 0) {
+			else if(ps->rid_extmap_id > 0) {
 				/* We may not know the SSRC yet, try the rid RTP extension */
 				char sdes_item[16];
-				if(janus_rtp_header_extension_parse_rid(buf, len, participant->rid_extmap_id, sdes_item, sizeof(sdes_item)) == 0) {
-					if(participant->rid[2] != NULL && !strcmp(participant->rid[2], sdes_item)) {
-						participant->ssrc[0] = ssrc;
+				if(janus_rtp_header_extension_parse_rid(buf, len, ps->rid_extmap_id, sdes_item, sizeof(sdes_item)) == 0) {
+					if(ps->rid[2] != NULL && !strcmp(ps->rid[2], sdes_item)) {
+						ps->vssrc[0] = ssrc;
 						sc = 0;
-					} else if(participant->rid[1] != NULL && !strcmp(participant->rid[1], sdes_item)) {
-						participant->ssrc[1] = ssrc;
+					} else if(ps->rid[1] != NULL && !strcmp(ps->rid[1], sdes_item)) {
+						ps->vssrc[1] = ssrc;
 						sc = 1;
-					} else if(participant->rid[0] != NULL && !strcmp(participant->rid[0], sdes_item)) {
-						participant->ssrc[2] = ssrc;
+					} else if(ps->rid[0] != NULL && !strcmp(ps->rid[0], sdes_item)) {
+						ps->vssrc[2] = ssrc;
 						sc = 2;
 					}
 				}
 			}
 		}
 		/* Forward RTP to the appropriate port for the rtp_forwarders associated with this publisher, if there are any */
-		janus_mutex_lock(&participant->rtp_forwarders_mutex);
+		janus_mutex_lock(&ps->rtp_forwarders_mutex);
 		if(participant->srtp_contexts && g_hash_table_size(participant->srtp_contexts) > 0) {
 			GHashTableIter iter;
 			gpointer value;
@@ -4407,7 +5463,7 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 		}
 		GHashTableIter iter;
 		gpointer value;
-		g_hash_table_iter_init(&iter, participant->rtp_forwarders);
+		g_hash_table_iter_init(&iter, ps->rtp_forwarders);
 		while(participant->udp_sock > 0 && g_hash_table_iter_next(&iter, NULL, &value)) {
 			janus_videoroom_rtp_forwarder *rtp_forward = (janus_videoroom_rtp_forwarder *)value;
 			if(rtp_forward->is_data || (video && !rtp_forward->is_video) || (!video && rtp_forward->is_video))
@@ -4423,9 +5479,9 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 			} else if(video && rtp_forward->simulcast) {
 				/* This is video and we're simulcasting, check if we need to forward this frame */
 				if(!janus_rtp_simulcasting_context_process_rtp(&rtp_forward->sim_context,
-						buf, len, participant->ssrc, participant->rid, participant->vcodec, &rtp_forward->context))
+						buf, len, ps->vssrc, ps->rid, ps->vcodec, &rtp_forward->context))
 					continue;
-				janus_rtp_header_update(rtp, &rtp_forward->context, TRUE, 4500);
+				janus_rtp_header_update(rtp, &rtp_forward->context, TRUE);
 				/* By default we use a fixed SSRC (it may be overwritten later) */
 				rtp->ssrc = htonl(participant->user_id & 0xffffffff);
 			}
@@ -4476,24 +5532,24 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 			rtp->timestamp = htonl(timestamp);
 			rtp->seq_number = htons(seq_number);
 		}
-		janus_mutex_unlock(&participant->rtp_forwarders_mutex);
+		janus_mutex_unlock(&ps->rtp_forwarders_mutex);
 		/* Set the payload type of the publisher */
-		rtp->type = video ? participant->video_pt : participant->audio_pt;
+		rtp->type = ps->pt;
 		/* Save the frame if we're recording */
-		if(!video || (participant->ssrc[0] == 0 && participant->rid[0] == NULL)) {
-			janus_recorder_save_frame(video ? participant->vrc : participant->arc, buf, len);
+		if(!video || !ps->simulcast) {
+			janus_recorder_save_frame(ps->rc, buf, len);
 		} else {
 			/* We're simulcasting, save the best video quality */
-			gboolean save = janus_rtp_simulcasting_context_process_rtp(&participant->rec_simctx,
-				buf, len, participant->ssrc, participant->rid, participant->vcodec, &participant->rec_ctx);
+			gboolean save = janus_rtp_simulcasting_context_process_rtp(&ps->rec_simctx,
+				buf, len, ps->vssrc, ps->rid, ps->vcodec, &ps->rec_ctx);
 			if(save) {
 				uint32_t seq_number = ntohs(rtp->seq_number);
 				uint32_t timestamp = ntohl(rtp->timestamp);
 				uint32_t ssrc = ntohl(rtp->ssrc);
-				janus_rtp_header_update(rtp, &participant->rec_ctx, TRUE, 4500);
+				janus_rtp_header_update(rtp, &ps->rec_ctx, TRUE);
 				/* We use a fixed SSRC for the whole recording */
 				rtp->ssrc = htonl(participant->user_id & 0xffffffff);
-				janus_recorder_save_frame(participant->vrc, buf, len);
+				janus_recorder_save_frame(ps->rc, buf, len);
 				/* Restore the header, as it will be needed by subscribers */
 				rtp->ssrc = htonl(ssrc);
 				rtp->timestamp = htonl(timestamp);
@@ -4502,11 +5558,12 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 		}
 		/* Done, relay it */
 		janus_videoroom_rtp_relay_packet packet;
+		packet.source = ps;
 		packet.data = rtp;
 		packet.length = len;
 		packet.is_video = video;
 		packet.svc = FALSE;
-		if(video && videoroom->do_svc) {
+		if(video && ps->svc) {
 			/* We're doing SVC: let's parse this packet to see which layers are there */
 			int plen = 0;
 			char *payload = janus_rtp_payload(buf, len, &plen);
@@ -4524,22 +5581,25 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 					packet.ubit = ubit;
 					packet.bbit = bbit;
 					packet.ebit = ebit;
+					/* Update the stream properties, if needed */
+					if(!ps->svc)
+						ps->svc = TRUE;
 				}
 			}
 		}
-		packet.ssrc[0] = (sc != -1 ? participant->ssrc[0] : 0);
-		packet.ssrc[1] = (sc != -1 ? participant->ssrc[1] : 0);
-		packet.ssrc[2] = (sc != -1 ? participant->ssrc[2] : 0);
+		packet.ssrc[0] = (sc != -1 ? ps->vssrc[0] : 0);
+		packet.ssrc[1] = (sc != -1 ? ps->vssrc[1] : 0);
+		packet.ssrc[2] = (sc != -1 ? ps->vssrc[2] : 0);
 		/* Backup the actual timestamp and sequence number set by the publisher, in case switching is involved */
 		packet.timestamp = ntohl(packet.data->timestamp);
 		packet.seq_number = ntohs(packet.data->seq_number);
 		/* Go: some viewers may decide to drop the packet, but that's up to them */
-		janus_mutex_lock_nodebug(&participant->subscribers_mutex);
-		g_slist_foreach(participant->subscribers, janus_videoroom_relay_rtp_packet, &packet);
-		janus_mutex_unlock_nodebug(&participant->subscribers_mutex);
+		janus_mutex_lock_nodebug(&ps->subscribers_mutex);
+		g_slist_foreach(ps->subscribers, janus_videoroom_relay_rtp_packet, &packet);
+		janus_mutex_unlock_nodebug(&ps->subscribers_mutex);
 
-		/* Check if we need to send any REMB, FIR or PLI back to this publisher */
-		if(video && participant->video_active) {
+		/* Check if we need to send any REMB or PLI back to this publisher */
+		if(video && ps->active) {
 			/* Did we send a REMB already, or is it time to send one? */
 			gboolean send_remb = FALSE;
 			if(participant->remb_latest == 0 && participant->remb_startup > 0) {
@@ -4560,12 +5620,12 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 				JANUS_LOG(LOG_VERB, "Sending REMB (%s, %"SCNu32")\n", participant->display, bitrate);
 				char rtcpbuf[24];
 				janus_rtcp_remb((char *)(&rtcpbuf), 24, bitrate);
-				gateway->relay_rtcp(handle, video, rtcpbuf, 24);
+				gateway->relay_rtcp(handle, -1, video, rtcpbuf, 24);
 				if(participant->remb_startup == 0)
 					participant->remb_latest = janus_get_monotonic_time();
 			}
-			/* Generate FIR/PLI too, if needed */
-			if(video && participant->video_active && (videoroom->fir_freq > 0)) {
+			/* Generate PLI too, if needed */
+			if(video && ps->active && (videoroom->pli_freq > 0)) {
 				/* We generate RTCP every tot seconds/frames */
 				gint64 now = janus_get_monotonic_time();
 				/* First check if this is a keyframe, though: if so, we reset the timer */
@@ -4573,19 +5633,19 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 				char *payload = janus_rtp_payload(buf, len, &plen);
 				if(payload == NULL)
 					return;
-				if(participant->vcodec == JANUS_VIDEOCODEC_VP8) {
+				if(ps->vcodec == JANUS_VIDEOCODEC_VP8) {
 					if(janus_vp8_is_keyframe(payload, plen))
-						participant->fir_latest = now;
-				} else if(participant->vcodec == JANUS_VIDEOCODEC_VP9) {
+						ps->pli_latest = now;
+				} else if(ps->vcodec == JANUS_VIDEOCODEC_VP9) {
 					if(janus_vp9_is_keyframe(payload, plen))
-						participant->fir_latest = now;
-				} else if(participant->vcodec == JANUS_VIDEOCODEC_H264) {
+						ps->pli_latest = now;
+				} else if(ps->vcodec == JANUS_VIDEOCODEC_H264) {
 					if(janus_h264_is_keyframe(payload, plen))
-						participant->fir_latest = now;
+						ps->pli_latest = now;
 				}
-				if((now-participant->fir_latest) >= ((gint64)videoroom->fir_freq*G_USEC_PER_SEC)) {
-					/* FIXME We send a FIR every tot seconds */
-					janus_videoroom_reqfir(participant, "Regular keyframe request");
+				if((now-ps->pli_latest) >= ((gint64)videoroom->pli_freq*G_USEC_PER_SEC)) {
+					/* We send a PLI every tot seconds */
+					janus_videoroom_reqpli(ps, "Regular keyframe request");
 				}
 			}
 		}
@@ -4593,7 +5653,7 @@ void janus_videoroom_incoming_rtp(janus_plugin_session *handle, int video, char
 	janus_videoroom_publisher_dereference_nodebug(participant);
 }
 
-void janus_videoroom_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_videoroom_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	janus_videoroom_session *session = (janus_videoroom_session *)handle->plugin_handle;
@@ -4608,35 +5668,22 @@ void janus_videoroom_incoming_rtcp(janus_plugin_session *handle, int video, char
 		janus_videoroom_subscriber *s = (janus_videoroom_subscriber *)session->participant;
 		if(s == NULL || g_atomic_int_get(&s->destroyed))
 			return;
-		if(!s->video)
-			return;	/* The only feedback we handle is video related anyway... */
-		if(janus_rtcp_has_fir(buf, len)) {
-			/* We got a FIR, forward it to the publisher */
-			if(s->feed) {
-				janus_videoroom_publisher *p = s->feed;
-				if(p && p->session) {
-					char rtcpbuf[20];
-					janus_rtcp_fir((char *)&rtcpbuf, 20, &p->fir_seq);
-					JANUS_LOG(LOG_VERB, "Got a FIR from a subscriber, forwarding it to %"SCNu64" (%s)\n", p->user_id, p->display ? p->display : "??");
-					gateway->relay_rtcp(p->session->handle, 1, rtcpbuf, 20);
-					/* Update the time of when we last sent a keyframe request */
-					p->fir_latest = janus_get_monotonic_time();
-				}
-			}
+		/* Find the stream this packet belongs to */
+		janus_mutex_lock(&s->streams_mutex);
+		janus_videoroom_subscriber_stream *ss = g_hash_table_lookup(s->streams_byid, GINT_TO_POINTER(mindex));
+		janus_mutex_unlock(&s->streams_mutex);
+		if(ss == NULL || ss->publisher_streams == NULL) {
+			/* No stream..? */
+			return;
 		}
-		if(janus_rtcp_has_pli(buf, len)) {
-			/* We got a PLI, forward it to the publisher */
-			if(s->feed) {
-				janus_videoroom_publisher *p = s->feed;
-				if(p && p->session) {
-					char rtcpbuf[12];
-					janus_rtcp_pli((char *)&rtcpbuf, 12);
-					JANUS_LOG(LOG_VERB, "Got a PLI from a subscriber, forwarding it to %"SCNu64" (%s)\n", p->user_id, p->display ? p->display : "??");
-					gateway->relay_rtcp(p->session->handle, 1, rtcpbuf, 12);
-					/* Update the time of when we last sent a keyframe request */
-					p->fir_latest = janus_get_monotonic_time();
-				}
-			}
+		janus_videoroom_publisher_stream *ps = ss->publisher_streams ? ss->publisher_streams->data : NULL;
+		if(ps->type != JANUS_VIDEOROOM_MEDIA_VIDEO)
+			return;		/* The only feedback we handle is video related anyway... */
+		if(janus_rtcp_has_fir(buf, len) || janus_rtcp_has_pli(buf, len)) {
+			/* We got a FIR or PLI, forward a PLI to the publisher */
+			janus_videoroom_publisher *p = ps->publisher;
+			if(p && p->session)
+				janus_videoroom_reqpli(ps, "PLI from subscriber");
 		}
 		uint32_t bitrate = janus_rtcp_get_remb(buf, len);
 		if(bitrate > 0) {
@@ -4656,16 +5703,26 @@ void janus_videoroom_incoming_data(janus_plugin_session *handle, char *label, ch
 	janus_videoroom_publisher *participant = janus_videoroom_session_get_publisher_nodebug(session);
 	if(participant == NULL)
 		return;
-	if(g_atomic_int_get(&participant->destroyed) || !participant->data_active || participant->kicked) {
+	if(g_atomic_int_get(&participant->destroyed) || participant->data_mindex < 0 || !participant->streams || participant->kicked) {
+		janus_videoroom_publisher_dereference_nodebug(participant);
+		return;
+	}
+	/* Find the stream this packet belongs to */
+	janus_mutex_lock(&participant->streams_mutex);
+	janus_videoroom_publisher_stream *ps = g_hash_table_lookup(participant->streams_byid, GINT_TO_POINTER(participant->data_mindex));
+	janus_mutex_unlock(&participant->streams_mutex);
+	if(ps == NULL || !ps->active) {
+		/* No or inactive stream..? */
 		janus_videoroom_publisher_dereference_nodebug(participant);
 		return;
 	}
+
 	/* Any forwarder involved? */
-	janus_mutex_lock(&participant->rtp_forwarders_mutex);
+	janus_mutex_lock(&ps->rtp_forwarders_mutex);
 	/* Forward RTP to the appropriate port for the rtp_forwarders associated with this publisher, if there are any */
 	GHashTableIter iter;
 	gpointer value;
-	g_hash_table_iter_init(&iter, participant->rtp_forwarders);
+	g_hash_table_iter_init(&iter, ps->rtp_forwarders);
 	while(participant->udp_sock > 0 && g_hash_table_iter_next(&iter, NULL, &value)) {
 		janus_videoroom_rtp_forwarder* rtp_forward = (janus_videoroom_rtp_forwarder*)value;
 		if(rtp_forward->is_data) {
@@ -4678,23 +5735,26 @@ void janus_videoroom_incoming_data(janus_plugin_session *handle, char *label, ch
 			}
 		}
 	}
-	janus_mutex_unlock(&participant->rtp_forwarders_mutex);
+	janus_mutex_unlock(&ps->rtp_forwarders_mutex);
 	/* Get a string out of the data */
 	char *text = g_malloc(len+1);
 	memcpy(text, buf, len);
 	*(text+len) = '\0';
 	JANUS_LOG(LOG_VERB, "Got a DataChannel message (%zu bytes) to forward: %s\n", strlen(text), text);
 	/* Save the message if we're recording */
-	janus_recorder_save_frame(participant->drc, text, strlen(text));
+	janus_recorder_save_frame(ps->rc, text, strlen(text));
 	/* Relay to all subscribers */
-	janus_mutex_lock_nodebug(&participant->subscribers_mutex);
-	g_slist_foreach(participant->subscribers, janus_videoroom_relay_data_packet, text);
-	janus_mutex_unlock_nodebug(&participant->subscribers_mutex);
+	janus_videoroom_data_relay_packet packet;
+	packet.source = ps;
+	packet.text = text;
+	janus_mutex_lock_nodebug(&ps->subscribers_mutex);
+	g_slist_foreach(ps->subscribers, janus_videoroom_relay_data_packet, &packet);
+	janus_mutex_unlock_nodebug(&ps->subscribers_mutex);
 	g_free(text);
 	janus_videoroom_publisher_dereference_nodebug(participant);
 }
 
-void janus_videoroom_slow_link(janus_plugin_session *handle, int uplink, int video) {
+void janus_videoroom_slow_link(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink) {
 	/* The core is informing us that our peer got too many NACKs, are we pushing media too hard? */
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized) || !gateway)
 		return;
@@ -4748,99 +5808,66 @@ void janus_videoroom_slow_link(janus_plugin_session *handle, int uplink, int vid
 	janus_refcount_decrease(&session->ref);
 }
 
-static void janus_videoroom_recorder_create(janus_videoroom_publisher *participant, gboolean audio, gboolean video, gboolean data) {
+static void janus_videoroom_recorder_create(janus_videoroom_publisher_stream *stream) {
 	char filename[255];
 	gint64 now = janus_get_real_time();
-	if(audio && participant->arc == NULL) {
-		memset(filename, 0, 255);
-		if(participant->recording_base) {
-			/* Use the filename and path we have been provided */
-			g_snprintf(filename, 255, "%s-audio", participant->recording_base);
-			participant->arc = janus_recorder_create(participant->room->rec_dir,
-				janus_audiocodec_name(participant->acodec), filename);
-			if(participant->arc == NULL) {
-				JANUS_LOG(LOG_ERR, "Couldn't open an audio recording file for this publisher!\n");
-			}
-		} else {
-			/* Build a filename */
-			g_snprintf(filename, 255, "videoroom-%"SCNu64"-user-%"SCNu64"-%"SCNi64"-audio",
-				participant->room_id, participant->user_id, now);
-			participant->arc = janus_recorder_create(participant->room->rec_dir,
-				janus_audiocodec_name(participant->acodec), filename);
-			if(participant->arc == NULL) {
-				JANUS_LOG(LOG_ERR, "Couldn't open an audio recording file for this publisher!\n");
-			}
+	if(stream->publisher && stream->rc == NULL) {
+		const char *type = NULL;
+		switch(stream->type) {
+			case JANUS_VIDEOROOM_MEDIA_AUDIO:
+				type = janus_audiocodec_name(stream->acodec);
+				break;
+			case JANUS_VIDEOROOM_MEDIA_VIDEO:
+				type = janus_videocodec_name(stream->vcodec);
+				break;
+			case JANUS_VIDEOROOM_MEDIA_DATA:
+				type = "text";
+				break;
+			default:
+				return;
 		}
-	}
-	if(video && participant->vrc == NULL) {
-		janus_rtp_switching_context_reset(&participant->rec_ctx);
-		janus_rtp_simulcasting_context_reset(&participant->rec_simctx);
-		participant->rec_simctx.substream_target = 2;
-		participant->rec_simctx.templayer_target = 2;
-		memset(filename, 0, 255);
-		if(participant->recording_base) {
-			/* Use the filename and path we have been provided */
-			g_snprintf(filename, 255, "%s-video", participant->recording_base);
-			participant->vrc = janus_recorder_create(participant->room->rec_dir,
-				janus_videocodec_name(participant->vcodec), filename);
-			if(participant->vrc == NULL) {
-				JANUS_LOG(LOG_ERR, "Couldn't open an video recording file for this publisher!\n");
-			}
-		} else {
-			/* Build a filename */
-			g_snprintf(filename, 255, "videoroom-%"SCNu64"-user-%"SCNu64"-%"SCNi64"-video",
-				participant->room_id, participant->user_id, now);
-			participant->vrc = janus_recorder_create(participant->room->rec_dir,
-				janus_videocodec_name(participant->vcodec), filename);
-			if(participant->vrc == NULL) {
-				JANUS_LOG(LOG_ERR, "Couldn't open an video recording file for this publisher!\n");
-			}
+		janus_rtp_switching_context_reset(&stream->rec_ctx);
+		janus_rtp_simulcasting_context_reset(&stream->rec_simctx);
+		if(stream->type == JANUS_VIDEOROOM_MEDIA_VIDEO) {
+			stream->rec_simctx.rid_ext_id = stream->rid_extmap_id;
+			stream->rec_simctx.substream_target = 2;
+			stream->rec_simctx.templayer_target = 2;
 		}
-	}
-	if(data && participant->drc == NULL) {
 		memset(filename, 0, 255);
-		if(participant->recording_base) {
+		if(stream->publisher->recording_base) {
 			/* Use the filename and path we have been provided */
-			g_snprintf(filename, 255, "%s-data", participant->recording_base);
-			participant->drc = janus_recorder_create(participant->room->rec_dir,
-				"text", filename);
-			if(participant->drc == NULL) {
-				JANUS_LOG(LOG_ERR, "Couldn't open an data recording file for this publisher!\n");
+			g_snprintf(filename, 255, "%s-%s-%d", stream->publisher->recording_base,
+				janus_videoroom_media_str(stream->type), stream->mindex);
+			stream->rc = janus_recorder_create(stream->publisher->room->rec_dir, type, filename);
+			if(stream->rc == NULL) {
+				JANUS_LOG(LOG_ERR, "Couldn't open a %s recording file for this publisher!\n", janus_videoroom_media_str(stream->type));
 			}
 		} else {
 			/* Build a filename */
-			g_snprintf(filename, 255, "videoroom-%"SCNu64"-user-%"SCNu64"-%"SCNi64"-data",
-				participant->room_id, participant->user_id, now);
-			participant->drc = janus_recorder_create(participant->room->rec_dir,
-				"text", filename);
-			if(participant->drc == NULL) {
-				JANUS_LOG(LOG_ERR, "Couldn't open an data recording file for this publisher!\n");
+			g_snprintf(filename, 255, "videoroom-%"SCNu64"-user-%"SCNu64"-%"SCNi64"-%s-%d",
+				stream->publisher->room_id, stream->publisher->user_id, now,
+				janus_videoroom_media_str(stream->type), stream->mindex);
+			stream->rc = janus_recorder_create(stream->publisher->room->rec_dir, type, filename);
+			if(stream->rc == NULL) {
+				JANUS_LOG(LOG_ERR, "Couldn't open an video recording file for this publisher!\n");
 			}
 		}
 	}
 }
 
 static void janus_videoroom_recorder_close(janus_videoroom_publisher *participant) {
-	if(participant->arc) {
-		janus_recorder *rc = participant->arc;
-		participant->arc = NULL;
-		janus_recorder_close(rc);
-		JANUS_LOG(LOG_INFO, "Closed audio recording %s\n", rc->filename ? rc->filename : "??");
-		janus_recorder_destroy(rc);
-	}
-	if(participant->vrc) {
-		janus_recorder *rc = participant->vrc;
-		participant->vrc = NULL;
-		janus_recorder_close(rc);
-		JANUS_LOG(LOG_INFO, "Closed video recording %s\n", rc->filename ? rc->filename : "??");
-		janus_recorder_destroy(rc);
-	}
-	if(participant->drc) {
-		janus_recorder *rc = participant->drc;
-		participant->drc = NULL;
-		janus_recorder_close(rc);
-		JANUS_LOG(LOG_INFO, "Closed data recording %s\n", rc->filename ? rc->filename : "??");
-		janus_recorder_destroy(rc);
+	GList *temp = participant->streams;
+	while(temp) {
+		janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+		if(ps->rc) {
+			janus_recorder *rc = ps->rc;
+			ps->rc = NULL;
+			janus_recorder_close(rc);
+			JANUS_LOG(LOG_INFO, "Closed %s recording %s\n", janus_videoroom_media_str(ps->type),
+				rc->filename ? rc->filename : "??");
+			janus_recorder_destroy(rc);
+		}
+		temp = temp->next;
 	}
 }
 
@@ -4851,37 +5878,6 @@ void janus_videoroom_hangup_media(janus_plugin_session *handle) {
 	janus_mutex_unlock(&sessions_mutex);
 }
 
-static void janus_videoroom_hangup_subscriber(janus_videoroom_subscriber * s) {
-	/* Already hung up */
-	if (!s->feed) {
-		return;
-	}
-	/* Check if the owner needs to be cleaned up */
-	if(s->pvt_id > 0 && s->room != NULL) {
-		janus_mutex_lock(&s->room->mutex);
-		janus_videoroom_publisher *owner = g_hash_table_lookup(s->room->private_ids, GUINT_TO_POINTER(s->pvt_id));
-		if(owner != NULL) {
-			janus_mutex_lock(&owner->subscribers_mutex);
-			/* Note: we should refcount these subscription-publisher mappings as well */
-			owner->subscriptions = g_slist_remove(owner->subscriptions, s);
-			janus_mutex_unlock(&owner->subscribers_mutex);
-		}
-		janus_mutex_unlock(&s->room->mutex);
-	}
-	/* TODO: are we sure this is okay as other handlers use feed directly without synchronization */
-	if(s->feed)
-		g_clear_pointer(&s->feed, janus_videoroom_publisher_dereference_by_subscriber);
-	/* Only "leave" the room if we're closing the PeerConnection at this point */
-	if(s->close_pc) {
-		if(s->room)
-			g_clear_pointer(&s->room, janus_videoroom_room_dereference);
-		if(s->session)
-			gateway->close_pc(s->session->handle);
-		/* Remove the reference we added when "joining" the room */
-		janus_refcount_decrease(&s->ref);
-	}
-}
-
 static void janus_videoroom_hangup_media_internal(janus_plugin_session *handle) {
 	if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
@@ -4906,63 +5902,131 @@ static void janus_videoroom_hangup_media_internal(janus_plugin_session *handle)
 		janus_videoroom_recorder_close(participant);
 		janus_mutex_unlock(&participant->rec_mutex);
 		/* Use subscribers_mutex to protect fields used in janus_videoroom_incoming_rtp */
-		janus_mutex_lock(&participant->subscribers_mutex);
-		g_free(participant->sdp);
-		participant->sdp = NULL;
 		participant->firefox = FALSE;
-		participant->audio_active = FALSE;
-		participant->video_active = FALSE;
-		participant->data_active = FALSE;
-		participant->audio_active_packets = 0;
-		participant->audio_dBov_sum = 0;
-		participant->audio_dBov_level = 0;
-		participant->talking = FALSE;
 		participant->remb_startup = 4;
 		participant->remb_latest = 0;
-		participant->fir_latest = 0;
-		participant->fir_seq = 0;
-		int i=0;
-		for(i=0; i<3; i++) {
-			participant->ssrc[i] = 0;
-			g_free(participant->rid[i]);
-			participant->rid[i] = NULL;
-		}
-		GSList *subscribers = participant->subscribers;
-		participant->subscribers = NULL;
-		janus_mutex_unlock(&participant->subscribers_mutex);
-		/* Hangup all subscribers */
-		while(subscribers) {
-			janus_videoroom_subscriber *s = (janus_videoroom_subscriber *)subscribers->data;
-			subscribers = g_slist_remove(subscribers, s);
-			if(s) {
-				janus_videoroom_hangup_subscriber(s);
+		/* Get rid of streams */
+		janus_mutex_lock(&participant->streams_mutex);
+		GList *subscribers = NULL;
+		GList *temp = participant->streams;
+		while(temp) {
+			janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+			/* Close all subscriptions to this stream */
+			janus_mutex_lock(&ps->subscribers_mutex);
+			GSList *temp2 = ps->subscribers;
+			while(temp2) {
+				janus_videoroom_subscriber_stream *ss = (janus_videoroom_subscriber_stream *)temp2->data;
+				temp2 = temp2->next;
+				if(ss) {
+					/* Remove the subscription (turns the m-line to inactive) */
+					janus_videoroom_subscriber_stream_remove(ss, ps, FALSE);
+					/* Take note of the subscriber, so that we can send an updated offer */
+					if(ss->type != JANUS_VIDEOROOM_MEDIA_DATA && g_list_find(subscribers, ss->subscriber) == NULL) {
+						janus_refcount_increase(&ss->subscriber->ref);
+						subscribers = g_list_append(subscribers, ss->subscriber);
+					}
+				}
 			}
-		}
-		janus_videoroom_leave_or_unpublish(participant, FALSE, FALSE);
-		janus_refcount_decrease(&participant->ref);
-	} else if(session->participant_type == janus_videoroom_p_type_subscriber) {
-		/* Get rid of subscriber */
-		janus_videoroom_subscriber *subscriber = (janus_videoroom_subscriber *)session->participant;
-		if(subscriber) {
-			subscriber->paused = TRUE;
-			janus_videoroom_publisher *publisher = subscriber->feed;
-			/* It is safe to use feed as the only other place sets feed to NULL
-			   is in this function and accessing to this function is synchronized
-			   by sessions_mutex */
-			if(publisher != NULL) {
-				/* Also notify event handlers */
-				if(notify_events && gateway->events_is_enabled()) {
-					json_t *info = json_object();
-					json_object_set_new(info, "event", json_string("unsubscribed"));
-					json_object_set_new(info, "room", json_integer(publisher->room_id));
-					json_object_set_new(info, "feed", json_integer(publisher->user_id));
-					gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
+			g_slist_free(ps->subscribers);
+			ps->subscribers = NULL;
+			int i=0;
+			for(i=0; i<3; i++) {
+				ps->vssrc[i] = 0;
+				g_free(ps->rid[i]);
+				ps->rid[i] = NULL;
+			}
+			ps->rid_extmap_id = 0;
+			janus_mutex_unlock(&ps->subscribers_mutex);
+			temp = temp->next;
+		}
+		/* Any subscriber session to update? */
+		if(subscribers != NULL) {
+			temp = subscribers;
+			while(temp) {
+				janus_videoroom_subscriber *subscriber = (janus_videoroom_subscriber *)temp->data;
+				/* Send (or schedule) a new offer */
+				janus_mutex_lock(&subscriber->streams_mutex);
+				if(!g_atomic_int_get(&subscriber->answered)) {
+					/* We're still waiting for an answer to a previous offer, postpone this */
+					g_atomic_int_set(&subscriber->pending_offer, 1);
+					janus_mutex_unlock(&subscriber->streams_mutex);
+				} else {
+					json_t *event = json_object();
+					json_object_set_new(event, "videoroom", json_string("updated"));
+					json_object_set_new(event, "room", json_integer(subscriber->room_id));
+					json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+					json_object_set_new(event, "streams", media);
+					/* Generate a new offer */
+					json_t *jsep = janus_videoroom_subscriber_offer(subscriber);
+					janus_mutex_unlock(&subscriber->streams_mutex);
+					/* How long will the Janus core take to push the event? */
+					gint64 start = janus_get_monotonic_time();
+					int res = gateway->push_event(subscriber->session->handle, &janus_videoroom_plugin, NULL, event, jsep);
+					JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
+					json_decref(event);
+					json_decref(jsep);
+					/* Also notify event handlers */
+					if(notify_events && gateway->events_is_enabled()) {
+						json_t *info = json_object();
+						json_object_set_new(info, "event", json_string("updated"));
+						json_object_set_new(info, "room", json_integer(subscriber->room_id));
+						json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+						json_object_set_new(info, "streams", media);
+						json_object_set_new(info, "private_id", json_integer(subscriber->pvt_id));
+						gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
+					}
+				}
+				janus_refcount_decrease(&subscriber->ref);
+				temp = temp->next;
+			}
+		}
+		g_list_free(subscribers);
+		/* Free streams */
+		g_list_free(participant->streams);
+		participant->streams = NULL;
+		g_hash_table_remove_all(participant->streams_byid);
+		g_hash_table_remove_all(participant->streams_bymid);
+		janus_mutex_unlock(&participant->streams_mutex);
+		janus_videoroom_leave_or_unpublish(participant, FALSE, FALSE);
+		janus_refcount_decrease(&participant->ref);
+	} else if(session->participant_type == janus_videoroom_p_type_subscriber) {
+		/* Get rid of subscriber */
+		janus_videoroom_subscriber *subscriber = (janus_videoroom_subscriber *)session->participant;
+		if(subscriber) {
+			subscriber->paused = TRUE;
+			g_atomic_int_set(&subscriber->answered, 0);
+			g_atomic_int_set(&subscriber->pending_offer, 0);
+			g_atomic_int_set(&subscriber->pending_restart, 0);
+			/* TODO Get rid of streams */
+			janus_mutex_lock(&subscriber->streams_mutex);
+			GList *temp = subscriber->streams;
+			while(temp) {
+				janus_videoroom_subscriber_stream *s = (janus_videoroom_subscriber_stream *)temp->data;
+				GSList *list = s->publisher_streams;
+				while(list) {
+					janus_videoroom_publisher_stream *ps = list->data;
+					if(ps && ps->publisher != NULL) {
+						/* Also notify event handlers */
+						if(notify_events && gateway->events_is_enabled()) {
+							json_t *info = json_object();
+							json_object_set_new(info, "event", json_string("unsubscribed"));
+							json_object_set_new(info, "room", json_integer(ps->publisher->room_id));
+							json_object_set_new(info, "feed", json_integer(ps->publisher->user_id));
+							json_object_set_new(info, "mid", json_string(ps->mid));
+							gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
+						}
+					}
+					list = list->next;
 				}
-				janus_mutex_lock(&publisher->subscribers_mutex);
-				publisher->subscribers = g_slist_remove(publisher->subscribers, subscriber);
-				janus_mutex_unlock(&publisher->subscribers_mutex);
-				janus_videoroom_hangup_subscriber(subscriber);
+				janus_videoroom_subscriber_stream_remove(s, NULL, TRUE);
+				temp = temp->next;
 			}
+			/* TODO Free streams */
+			g_list_free(subscriber->streams);
+			subscriber->streams = NULL;
+			g_hash_table_remove_all(subscriber->streams_byid);
+			g_hash_table_remove_all(subscriber->streams_bymid);
+			janus_mutex_unlock(&subscriber->streams_mutex);
 		}
 		/* TODO Should we close the handle as well? */
 	}
@@ -5058,6 +6122,22 @@ static void *janus_videoroom_handler(void *data) {
 					janus_refcount_decrease(&videoroom->ref);
 					goto error;
 				}
+				json_t *descriptions = json_object_get(root, "descriptions");
+				if(descriptions != NULL && json_array_size(descriptions) > 0) {
+					size_t i = 0;
+					for(i=0; i<json_array_size(descriptions); i++) {
+						json_t *d = json_array_get(descriptions, i);
+						JANUS_VALIDATE_JSON_OBJECT(d, publish_desc_parameters,
+							error_code, error_cause, TRUE,
+							JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+						if(error_code != 0) {
+							janus_mutex_unlock(&videoroom->mutex);
+							janus_refcount_decrease(&videoroom->ref);
+							goto error;
+
+						}
+					}
+				}
 				/* A token might be required to join */
 				if(videoroom->check_allowed) {
 					json_t *token = json_object_get(root, "token");
@@ -5099,14 +6179,8 @@ static void *janus_videoroom_handler(void *data) {
 				}
 				JANUS_LOG(LOG_VERB, "  -- Publisher ID: %"SCNu64"\n", user_id);
 				/* Process the request */
-				json_t *audio = NULL, *video = NULL, *data = NULL,
-					*bitrate = NULL, *record = NULL, *recfile = NULL;
+				json_t *bitrate = NULL, *record = NULL, *recfile = NULL;
 				if(!strcasecmp(request_text, "joinandconfigure")) {
-					/* Also configure (or publish a new feed) audio/video/bitrate for this new publisher */
-					/* join_parameters were validated earlier. */
-					audio = json_object_get(root, "audio");
-					video = json_object_get(root, "video");
-					data = json_object_get(root, "data");
 					bitrate = json_object_get(root, "bitrate");
 					record = json_object_get(root, "record");
 					recfile = json_object_get(root, "filename");
@@ -5118,39 +6192,23 @@ static void *janus_videoroom_handler(void *data) {
 				videoroom = NULL;
 				publisher->user_id = user_id;
 				publisher->display = display_text ? g_strdup(display_text) : NULL;
-				publisher->sdp = NULL;		/* We'll deal with this later */
-				publisher->audio = FALSE;	/* We'll deal with this later */
-				publisher->video = FALSE;	/* We'll deal with this later */
-				publisher->data = FALSE;	/* We'll deal with this later */
-				publisher->acodec = JANUS_AUDIOCODEC_NONE;	/* We'll deal with this later */
-				publisher->vcodec = JANUS_VIDEOCODEC_NONE;	/* We'll deal with this later */
-				publisher->audio_active = TRUE;
-				publisher->video_active = TRUE;
-				publisher->data_active = TRUE;
 				publisher->recording_active = FALSE;
 				publisher->recording_base = NULL;
-				publisher->arc = NULL;
-				publisher->vrc = NULL;
-				publisher->drc = NULL;
-				janus_mutex_init(&publisher->rec_mutex);
 				publisher->firefox = FALSE;
 				publisher->bitrate = publisher->room->bitrate;
-				publisher->subscribers = NULL;
 				publisher->subscriptions = NULL;
+				publisher->streams_byid = g_hash_table_new_full(NULL, NULL,
+					NULL, (GDestroyNotify)janus_videoroom_publisher_stream_destroy);
+				publisher->streams_bymid = g_hash_table_new_full(g_str_hash, g_str_equal,
+					(GDestroyNotify)g_free, (GDestroyNotify)janus_videoroom_publisher_stream_unref);
+				janus_mutex_init(&publisher->streams_mutex);
 				janus_mutex_init(&publisher->subscribers_mutex);
-				publisher->audio_pt = -1;	/* We'll deal with this later */
-				publisher->video_pt = -1;	/* We'll deal with this later */
-				publisher->audio_level_extmap_id = 0;
-				publisher->video_orient_extmap_id = 0;
-				publisher->playout_delay_extmap_id = 0;
 				publisher->remb_startup = 4;
 				publisher->remb_latest = 0;
-				publisher->fir_latest = 0;
-				publisher->fir_seq = 0;
-				janus_mutex_init(&publisher->rtp_forwarders_mutex);
-				publisher->rtp_forwarders = g_hash_table_new_full(NULL, NULL, NULL, (GDestroyNotify)janus_videoroom_rtp_forwarder_destroy);
 				publisher->srtp_contexts = g_hash_table_new_full(g_str_hash, g_str_equal, NULL, (GDestroyNotify)janus_videoroom_srtp_context_free);
 				publisher->udp_sock = -1;
+				janus_mutex_init(&publisher->rtp_forwarders_mutex);
+				publisher->rtp_forwarders = g_hash_table_new(NULL, NULL);
 				/* Finally, generate a private ID: this is only needed in case the participant
 				 * wants to allow the plugin to know which subscriptions belong to them */
 				publisher->pvt_id = 0;
@@ -5165,18 +6223,6 @@ static void *janus_videoroom_handler(void *data) {
 				g_atomic_int_set(&publisher->destroyed, 0);
 				janus_refcount_init(&publisher->ref, janus_videoroom_publisher_free);
 				/* In case we also wanted to configure */
-				if(audio) {
-					publisher->audio_active = json_is_true(audio);
-					JANUS_LOG(LOG_VERB, "Setting audio property: %s (room %"SCNu64", user %"SCNu64")\n", publisher->audio_active ? "true" : "false", publisher->room_id, publisher->user_id);
-				}
-				if(video) {
-					publisher->video_active = json_is_true(video);
-					JANUS_LOG(LOG_VERB, "Setting video property: %s (room %"SCNu64", user %"SCNu64")\n", publisher->video_active ? "true" : "false", publisher->room_id, publisher->user_id);
-				}
-				if(data) {
-					publisher->data_active = json_is_true(data);
-					JANUS_LOG(LOG_VERB, "Setting data property: %s (room %"SCNu64", user %"SCNu64")\n", publisher->data_active ? "true" : "false", publisher->room_id, publisher->user_id);
-				}
 				if(bitrate) {
 					publisher->bitrate = json_integer_value(bitrate);
 					JANUS_LOG(LOG_VERB, "Setting video bitrate: %"SCNu32" (room %"SCNu64", user %"SCNu64")\n", publisher->bitrate, publisher->room_id, publisher->user_id);
@@ -5203,9 +6249,9 @@ static void *janus_videoroom_handler(void *data) {
 				janus_refcount_increase(&publisher->ref);
 				g_hash_table_insert(publisher->room->participants, janus_uint64_dup(publisher->user_id), publisher);
 				g_hash_table_iter_init(&iter, publisher->room->participants);
-				while (!g_atomic_int_get(&publisher->room->destroyed) && g_hash_table_iter_next(&iter, NULL, &value)) {
+				while(!g_atomic_int_get(&publisher->room->destroyed) && g_hash_table_iter_next(&iter, NULL, &value)) {
 					janus_videoroom_publisher *p = value;
-					if(p == publisher || !p->sdp || !p->session->started) {
+					if(p == publisher || !p->streams || !p->session->started) {
 						/* Check if we're also notifying normal joins and not just publishers */
 						if(p != publisher && publisher->room->notify_joining) {
 							json_t *al = json_object();
@@ -5220,14 +6266,63 @@ static void *janus_videoroom_handler(void *data) {
 					json_object_set_new(pl, "id", json_integer(p->user_id));
 					if(p->display)
 						json_object_set_new(pl, "display", json_string(p->display));
-					if(p->audio)
-						json_object_set_new(pl, "audio_codec", json_string(janus_audiocodec_name(p->acodec)));
-					if(p->video)
-						json_object_set_new(pl, "video_codec", json_string(janus_videocodec_name(p->vcodec)));
-					if(p->ssrc[0] || p->rid[0])
-						json_object_set_new(pl, "simulcast", json_true());
-					if(p->audio_level_extmap_id > 0)
-						json_object_set_new(pl, "talking", p->talking ? json_true() : json_false());
+					/* Add proper info on all the streams */
+					gboolean audio_added = FALSE, video_added = FALSE, talking_found = FALSE, talking = FALSE;
+					json_t *media = json_array();
+					GList *temp = p->streams;
+					while(temp) {
+						janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+						/* Are we updating the description? */
+						if(descriptions != NULL && json_array_size(descriptions) > 0) {
+							size_t i = 0;
+							for(i=0; i<json_array_size(descriptions); i++) {
+								json_t *d = json_array_get(descriptions, i);
+								const char *d_mid = json_string_value(json_object_get(d, "mid"));
+								const char *d_desc = json_string_value(json_object_get(d, "description"));
+								if(d_desc && d_mid && ps->mid && !strcasecmp(d_mid, ps->mid)) {
+									g_free(ps->description);
+									ps->description = g_strdup(d_desc);
+									break;
+								}
+							}
+						}
+						json_t *info = json_object();
+						json_object_set_new(info, "type", json_string(janus_videoroom_media_str(ps->type)));
+						json_object_set_new(info, "mindex", json_integer(ps->mindex));
+						json_object_set_new(info, "mid", json_string(ps->mid));
+						if(ps->description)
+							json_object_set_new(info, "description", json_string(ps->description));
+						if(ps->type == JANUS_VIDEOROOM_MEDIA_AUDIO) {
+							json_object_set_new(info, "codec", json_string(janus_audiocodec_name(ps->acodec)));
+							/* FIXME For backwards compatibility, we need audio_codec in the global info */
+							if(!audio_added) {
+								audio_added = TRUE;
+								json_object_set_new(pl, "audio_codec", json_string(janus_audiocodec_name(ps->acodec)));
+							}
+							if(ps->audio_level_extmap_id > 0) {
+								json_object_set_new(info, "talking", talking ? json_true() : json_false());
+								/* FIXME For backwards compatibility, we also need talking in the global info */
+								talking_found = TRUE;
+								talking |= ps->talking;
+							}
+						} else if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO) {
+							/* FIXME For backwards compatibility, we need video_codec in the global info */
+							json_object_set_new(info, "codec", json_string(janus_videocodec_name(ps->vcodec)));
+							if(!video_added) {
+								video_added = TRUE;
+								json_object_set_new(pl, "video_codec", json_string(janus_videocodec_name(ps->vcodec)));
+							}
+							if(ps->simulcast)
+								json_object_set_new(info, "simulcast", json_true());
+							if(ps->svc)
+								json_object_set_new(info, "svc", json_true());
+						}
+						json_array_append_new(media, info);
+						temp = temp->next;
+					}
+					json_object_set_new(pl, "streams", media);
+					if(talking_found)
+						json_object_set_new(pl, "talking", talking ? json_true() : json_false());
 					json_array_append_new(list, pl);
 				}
 				event = json_object();
@@ -5254,12 +6349,8 @@ static void *janus_videoroom_handler(void *data) {
 					gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
 				}
 				janus_mutex_unlock(&publisher->room->mutex);
-			} else if(!strcasecmp(ptype_text, "subscriber") || !strcasecmp(ptype_text, "listener")) {
+			} else if(!strcasecmp(ptype_text, "subscriber")) {
 				JANUS_LOG(LOG_VERB, "Configuring new subscriber\n");
-				gboolean legacy = !strcasecmp(ptype_text, "listener");
-				if(legacy) {
-					JANUS_LOG(LOG_WARN, "Subscriber is using the legacy 'listener' ptype\n");
-				}
 				/* This is a new subscriber */
 				JANUS_VALIDATE_JSON_OBJECT(root, subscriber_parameters,
 					error_code, error_cause, TRUE,
@@ -5268,186 +6359,350 @@ static void *janus_videoroom_handler(void *data) {
 					janus_mutex_unlock(&videoroom->mutex);
 					goto error;
 				}
-				json_t *feed = json_object_get(root, "feed");
-				guint64 feed_id = json_integer_value(feed);
+				/* Who does this subscription belong to? */
 				json_t *pvt = json_object_get(root, "private_id");
-				guint64 pvt_id = json_integer_value(pvt);
+				guint64 pvt_id = json_integer_value(pvt), feed_id = 0;
+				/* The new way of subscribing is specifying the streams we're interested in */
+				json_t *feeds = json_object_get(root, "streams");
+				gboolean legacy = FALSE;
+				if(feeds == NULL || json_array_size(feeds) == 0) {
+					/* For backwards compatibility, we still support the old "feed" property, which means
+					 * "subscribe to all the feeds from this publisher" (depending on offer_audio, etc.) */
+					json_t *feed = json_object_get(root, "feed");
+					feed_id = json_integer_value(feed);
+					if(feed_id == 0) {
+						JANUS_LOG(LOG_ERR, "At least one between 'streams' and 'feed' must be specified\n");
+						error_code = JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT;
+						g_snprintf(error_cause, 512, "At least one between 'streams' and 'feed' must be specified");
+						janus_mutex_unlock(&videoroom->mutex);
+						goto error;
+					}
+					/* Create a fake "streams" array and put the only feed there */
+					json_t *m = json_array();
+					json_t *s = json_object();
+					json_object_set_new(s, "feed", json_integer(feed_id));
+					json_array_append_new(m, s);
+					json_object_set_new(root, "streams", m);
+					feeds = json_object_get(root, "streams");
+					legacy = TRUE;
+				}
 				json_t *cpc = json_object_get(root, "close_pc");
 				gboolean close_pc  = cpc ? json_is_true(cpc) : TRUE;
-				json_t *audio = json_object_get(root, "audio");
-				json_t *video = json_object_get(root, "video");
-				json_t *data = json_object_get(root, "data");
-				json_t *offer_audio = json_object_get(root, "offer_audio");
-				json_t *offer_video = json_object_get(root, "offer_video");
-				json_t *offer_data = json_object_get(root, "offer_data");
-				json_t *spatial = json_object_get(root, "spatial_layer");
-				json_t *sc_substream = json_object_get(root, "substream");
-				if(json_integer_value(spatial) < 0 || json_integer_value(spatial) > 2 ||
-						json_integer_value(sc_substream) < 0 || json_integer_value(sc_substream) > 2) {
-					JANUS_LOG(LOG_ERR, "Invalid element (substream/spatial_layer should be 0, 1 or 2)\n");
-					error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
-					g_snprintf(error_cause, 512, "Invalid value (substream/spatial_layer should be 0, 1 or 2)");
-					janus_mutex_unlock(&videoroom->mutex);
-					goto error;
-				}
-				json_t *temporal = json_object_get(root, "temporal_layer");
-				json_t *sc_temporal = json_object_get(root, "temporal");
-				if(json_integer_value(temporal) < 0 || json_integer_value(temporal) > 2 ||
-						json_integer_value(sc_temporal) < 0 || json_integer_value(sc_temporal) > 2) {
-					JANUS_LOG(LOG_ERR, "Invalid element (temporal/temporal_layer should be 0, 1 or 2)\n");
-					error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
-					g_snprintf(error_cause, 512, "Invalid value (temporal/temporal_layer should be 0, 1 or 2)");
-					janus_mutex_unlock(&videoroom->mutex);
-					goto error;
-				}
-				janus_videoroom_publisher *owner = NULL;
-				janus_videoroom_publisher *publisher = g_hash_table_lookup(videoroom->participants, &feed_id);
-				if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || publisher->sdp == NULL) {
-					JANUS_LOG(LOG_ERR, "No such feed (%"SCNu64")\n", feed_id);
-					error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
-					g_snprintf(error_cause, 512, "No such feed (%"SCNu64")", feed_id);
-					janus_mutex_unlock(&videoroom->mutex);
-					goto error;
-				} else {
-					/* Increase the refcount before unlocking so that nobody can remove and free the publisher in the meantime. */
-					janus_refcount_increase(&publisher->ref);
-					janus_refcount_increase(&publisher->session->ref);
-					/* First of all, let's check if this room requires valid private_id values */
-					if(videoroom->require_pvtid) {
-						/* It does, let's make sure this subscription complies */
-						owner = g_hash_table_lookup(videoroom->private_ids, GUINT_TO_POINTER(pvt_id));
-						if(pvt_id == 0 || owner == NULL) {
-							JANUS_LOG(LOG_ERR, "Unauthorized (this room requires a valid private_id)\n");
-							error_code = JANUS_VIDEOROOM_ERROR_UNAUTHORIZED;
-							g_snprintf(error_cause, 512, "Unauthorized (this room requires a valid private_id)");
+				/* Make sure all the feeds we're subscribing to exist */
+				GList *publishers = NULL;
+				size_t i = 0;
+				for(i=0; i<json_array_size(feeds); i++) {
+					json_t *s = json_array_get(feeds, i);
+					JANUS_VALIDATE_JSON_OBJECT(s, subscriber_stream_parameters,
+						error_code, error_cause, TRUE,
+						JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+					if(error_code != 0) {
+						janus_mutex_unlock(&videoroom->mutex);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
+						}
+						goto error;
+					}
+					json_t *feed = json_object_get(s, "feed");
+					guint64 feed_id = json_integer_value(feed);
+					janus_videoroom_publisher *publisher = g_hash_table_lookup(videoroom->participants, &feed_id);
+					if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || !publisher->session->started) {
+						JANUS_LOG(LOG_ERR, "No such feed (%"SCNu64")\n", feed_id);
+						error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+						g_snprintf(error_cause, 512, "No such feed (%"SCNu64")", feed_id);
+						janus_mutex_unlock(&videoroom->mutex);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
+						}
+						goto error;
+					}
+					const char *mid = json_string_value(json_object_get(s, "mid"));
+					if(mid != NULL) {
+						/* Check the mid too */
+						janus_mutex_lock(&publisher->streams_mutex);
+						if(g_hash_table_lookup(publisher->streams_bymid, mid) == NULL) {
+							janus_mutex_unlock(&publisher->streams_mutex);
+							JANUS_LOG(LOG_ERR, "No such mid '%s' in feed (%"SCNu64")\n", mid, feed_id);
+							error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+							g_snprintf(error_cause, 512, "No such mid '%s' in feed (%"SCNu64")", mid, feed_id);
 							janus_mutex_unlock(&videoroom->mutex);
+							/* Unref publishers we may have taken note of so far */
+							while(publishers) {
+								publisher = (janus_videoroom_publisher *)publishers->data;
+								janus_refcount_decrease(&publisher->ref);
+								janus_refcount_decrease(&publisher->session->ref);
+								publishers = g_list_remove(publishers, publisher);
+							}
 							goto error;
 						}
-						janus_refcount_increase(&owner->ref);
-						janus_refcount_increase(&owner->session->ref);
+						janus_mutex_unlock(&publisher->streams_mutex);
 					}
-					janus_mutex_unlock(&videoroom->mutex);
-					janus_videoroom_subscriber *subscriber = g_malloc0(sizeof(janus_videoroom_subscriber));
-					subscriber->session = session;
-					subscriber->room_id = videoroom->room_id;
-					subscriber->room = videoroom;
-					videoroom = NULL;
-					subscriber->feed = publisher;
-					subscriber->pvt_id = pvt_id;
-					subscriber->close_pc = close_pc;
-					/* Initialize the subscriber context */
-					janus_rtp_switching_context_reset(&subscriber->context);
-					subscriber->audio_offered = offer_audio ? json_is_true(offer_audio) : TRUE;	/* True by default */
-					subscriber->video_offered = offer_video ? json_is_true(offer_video) : TRUE;	/* True by default */
-					subscriber->data_offered = offer_data ? json_is_true(offer_data) : TRUE;	/* True by default */
-					if((!publisher->audio || !subscriber->audio_offered) &&
-							(!publisher->video || !subscriber->video_offered) &&
-							(!publisher->data || !subscriber->data_offered)) {
-						g_free(subscriber);
-						if (owner) {
-							janus_refcount_decrease(&owner->session->ref);
-							janus_refcount_decrease(&owner->ref);
+					json_t *spatial = json_object_get(s, "spatial_layer");
+					json_t *sc_substream = json_object_get(s, "substream");
+					if(json_integer_value(spatial) < 0 || json_integer_value(spatial) > 2 ||
+							json_integer_value(sc_substream) < 0 || json_integer_value(sc_substream) > 2) {
+						JANUS_LOG(LOG_ERR, "Invalid element (substream/spatial_layer should be 0, 1 or 2)\n");
+						error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+						g_snprintf(error_cause, 512, "Invalid value (substream/spatial_layer should be 0, 1 or 2)");
+						janus_mutex_unlock(&videoroom->mutex);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
 						}
-						janus_refcount_decrease(&publisher->session->ref);
-						janus_refcount_decrease(&publisher->ref);
-						JANUS_LOG(LOG_ERR, "Can't offer an SDP with no audio, video or data\n");
-						error_code = JANUS_VIDEOROOM_ERROR_INVALID_SDP;
-						g_snprintf(error_cause, 512, "Can't offer an SDP with no audio, video or data");
 						goto error;
 					}
-					subscriber->audio = audio ? json_is_true(audio) : TRUE;	/* True by default */
-					if(!publisher->audio || !subscriber->audio_offered)
-						subscriber->audio = FALSE;	/* ... unless the publisher isn't sending any audio or we're skipping it */
-					subscriber->video = video ? json_is_true(video) : TRUE;	/* True by default */
-					if(!publisher->video || !subscriber->video_offered)
-						subscriber->video = FALSE;	/* ... unless the publisher isn't sending any video or we're skipping it */
-					subscriber->data = data ? json_is_true(data) : TRUE;	/* True by default */
-					if(!publisher->data || !subscriber->data_offered)
-						subscriber->data = FALSE;	/* ... unless the publisher isn't sending any data or we're skipping it */
-					subscriber->paused = TRUE;	/* We need an explicit start from the subscriber */
-					g_atomic_int_set(&subscriber->destroyed, 0);
-					janus_refcount_init(&subscriber->ref, janus_videoroom_subscriber_free);
-					janus_refcount_increase(&subscriber->ref);	/* The publisher references the new subscriber too */
-					/* Check if a simulcasting-related request is involved */
-					janus_rtp_simulcasting_context_reset(&subscriber->sim_context);
-					subscriber->sim_context.rid_ext_id = publisher->rid_extmap_id;
-					subscriber->sim_context.substream_target = sc_substream ? json_integer_value(sc_substream) : 2;
-					subscriber->sim_context.templayer_target = sc_temporal ? json_integer_value(sc_temporal) : 2;
-					janus_vp8_simulcast_context_reset(&subscriber->vp8_context);
-					/* Check if a VP9 SVC-related request is involved */
-					if(subscriber->room->do_svc) {
-						subscriber->spatial_layer = -1;
-						subscriber->target_spatial_layer = spatial ? json_integer_value(spatial) : 2;
-						subscriber->temporal_layer = -1;
-						subscriber->target_temporal_layer = temporal ? json_integer_value(temporal) : 2;
+					json_t *temporal = json_object_get(s, "temporal_layer");
+					json_t *sc_temporal = json_object_get(s, "temporal");
+					if(json_integer_value(temporal) < 0 || json_integer_value(temporal) > 2 ||
+							json_integer_value(sc_temporal) < 0 || json_integer_value(sc_temporal) > 2) {
+						JANUS_LOG(LOG_ERR, "Invalid element (temporal/temporal_layer should be 0, 1 or 2)\n");
+						error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+						g_snprintf(error_cause, 512, "Invalid value (temporal/temporal_layer should be 0, 1 or 2)");
+						janus_mutex_unlock(&videoroom->mutex);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
+						}
+						goto error;
 					}
-					session->participant = subscriber;
-					janus_mutex_lock(&publisher->subscribers_mutex);
-					publisher->subscribers = g_slist_append(publisher->subscribers, subscriber);
-					janus_mutex_unlock(&publisher->subscribers_mutex);
-					if(owner != NULL) {
-						/* Note: we should refcount these subscription-publisher mappings as well */
-						janus_mutex_lock(&owner->subscribers_mutex);
-						owner->subscriptions = g_slist_append(owner->subscriptions, subscriber);
-						janus_mutex_unlock(&owner->subscribers_mutex);
-						/* Done adding the subscription, owner is safe to be released */
-						janus_refcount_decrease(&owner->session->ref);
-						janus_refcount_decrease(&owner->ref);
+					/* Increase the refcount before unlocking so that nobody can remove and free the publisher in the meantime. */
+					janus_refcount_increase(&publisher->ref);
+					janus_refcount_increase(&publisher->session->ref);
+					publishers = g_list_append(publishers, publisher);
+				}
+				/* TODO These properties are only there for backwards compatibility */
+				json_t *offer_audio = json_object_get(root, "offer_audio");
+				json_t *offer_video = json_object_get(root, "offer_video");
+				json_t *offer_data = json_object_get(root, "offer_data");
+				janus_videoroom_publisher *owner = NULL;
+				/* Let's check if this room requires valid private_id values */
+				if(videoroom->require_pvtid) {
+					/* It does, let's make sure this subscription complies */
+					owner = g_hash_table_lookup(videoroom->private_ids, GUINT_TO_POINTER(pvt_id));
+					if(pvt_id == 0 || owner == NULL) {
+						JANUS_LOG(LOG_ERR, "Unauthorized (this room requires a valid private_id)\n");
+						error_code = JANUS_VIDEOROOM_ERROR_UNAUTHORIZED;
+						g_snprintf(error_cause, 512, "Unauthorized (this room requires a valid private_id)");
+						janus_mutex_unlock(&videoroom->mutex);
+						/* Unref publishers */
+						while(publishers) {
+							janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
+						}
+						goto error;
 					}
-					event = json_object();
-					json_object_set_new(event, "videoroom", json_string("attached"));
-					json_object_set_new(event, "room", json_integer(subscriber->room_id));
-					json_object_set_new(event, "id", json_integer(feed_id));
-					if(publisher->display)
-						json_object_set_new(event, "display", json_string(publisher->display));
-					if(legacy)
-						json_object_set_new(event, "warning", json_string("Deprecated use of 'listener' ptype, update to the new 'subscriber' ASAP"));
-					session->participant_type = janus_videoroom_p_type_subscriber;
-					JANUS_LOG(LOG_VERB, "Preparing JSON event as a reply\n");
-					/* Negotiate by sending the selected publisher SDP back */
-					janus_mutex_lock(&publisher->subscribers_mutex);
-					if(publisher->sdp != NULL) {
-						/* Check if there's something the original SDP has that we should remove */
-						janus_sdp *offer = janus_sdp_parse(publisher->sdp, NULL, 0);
-						subscriber->sdp = offer;
-						session->sdp_version = 1;
-						subscriber->sdp->o_version = session->sdp_version;
-						if((publisher->audio && !subscriber->audio_offered) ||
-								(publisher->video && !subscriber->video_offered) ||
-								(publisher->data && !subscriber->data_offered)) {
-							JANUS_LOG(LOG_VERB, "Munging SDP offer to adapt it to the subscriber's requirements\n");
-							if(publisher->audio && !subscriber->audio_offered)
-								janus_sdp_mline_remove(offer, JANUS_SDP_AUDIO);
-							if(publisher->video && !subscriber->video_offered)
-								janus_sdp_mline_remove(offer, JANUS_SDP_VIDEO);
-							if(publisher->data && !subscriber->data_offered)
-								janus_sdp_mline_remove(offer, JANUS_SDP_APPLICATION);
+					janus_refcount_increase(&owner->ref);
+					janus_refcount_increase(&owner->session->ref);
+				}
+				janus_mutex_unlock(&videoroom->mutex);
+				/* Allocate a new subscriber instance */
+				janus_videoroom_subscriber *subscriber = g_malloc0(sizeof(janus_videoroom_subscriber));
+				subscriber->session = session;
+				subscriber->room_id = videoroom->room_id;
+				subscriber->room = videoroom;
+				videoroom = NULL;
+				subscriber->pvt_id = pvt_id;
+				subscriber->close_pc = close_pc;
+				subscriber->paused = TRUE;	/* We need an explicit start from the stream */
+				subscriber->streams_byid = g_hash_table_new_full(NULL, NULL,
+					NULL, (GDestroyNotify)janus_videoroom_subscriber_stream_destroy);
+				subscriber->streams_bymid = g_hash_table_new_full(g_str_hash, g_str_equal,
+					(GDestroyNotify)g_free, (GDestroyNotify)janus_videoroom_subscriber_stream_unref);
+				janus_mutex_init(&subscriber->streams_mutex);
+				g_atomic_int_set(&subscriber->destroyed, 0);
+				janus_refcount_init(&subscriber->ref, janus_videoroom_subscriber_free);
+				/* FIXME backwards compatibility */
+				gboolean do_audio = offer_audio ? json_is_true(offer_audio) : TRUE;
+				gboolean do_video = offer_video ? json_is_true(offer_video) : TRUE;
+				gboolean do_data = offer_data ? json_is_true(offer_data) : TRUE;
+				/* Initialize the subscriber streams */
+				gboolean data_added = FALSE;
+				janus_videoroom_subscriber_stream *data_stream = NULL;
+				for(i=0; i<json_array_size(feeds); i++) {
+					json_t *s = json_array_get(feeds, i);
+					json_t *feed = json_object_get(s, "feed");
+					guint64 feed_id = json_integer_value(feed);
+					janus_videoroom_publisher *publisher = g_hash_table_lookup(subscriber->room->participants, &feed_id);
+					if(publisher == NULL) {
+						/* TODO We shouldn't let this happen... */
+						JANUS_LOG(LOG_WARN, "Skipping feed %"SCNu64"...\n", feed_id);
+						continue;
+					}
+					janus_mutex_lock(&publisher->streams_mutex);
+					const char *mid = json_string_value(json_object_get(s, "mid"));
+					json_t *spatial = json_object_get(s, "spatial_layer");
+					json_t *sc_substream = json_object_get(s, "substream");
+					json_t *temporal = json_object_get(s, "temporal_layer");
+					json_t *sc_temporal = json_object_get(s, "temporal");
+					if(mid) {
+						/* Subscribe to a specific mid */
+						janus_videoroom_publisher_stream *ps = g_hash_table_lookup(publisher->streams_bymid, mid);
+						if(ps == NULL) {
+							/* TODO We shouldn't let this happen either... */
+							JANUS_LOG(LOG_WARN, "Skipping mid %s in feed %"SCNu64"...\n", mid, feed_id);
+							janus_mutex_unlock(&publisher->streams_mutex);
+							continue;
 						}
-						char* sdp = janus_sdp_write(offer);
-						json_t *jsep = json_pack("{ssss}", "type", "offer", "sdp", sdp);
-						g_free(sdp);
-						janus_mutex_unlock(&publisher->subscribers_mutex);
-						/* How long will the Janus core take to push the event? */
-						g_atomic_int_set(&session->hangingup, 0);
-						gint64 start = janus_get_monotonic_time();
-						int res = gateway->push_event(msg->handle, &janus_videoroom_plugin, msg->transaction, event, jsep);
-						JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
-						json_decref(event);
-						json_decref(jsep);
-						janus_videoroom_message_free(msg);
-						/* Also notify event handlers */
-						if(notify_events && gateway->events_is_enabled()) {
-							json_t *info = json_object();
-							json_object_set_new(info, "event", json_string("subscribing"));
-							json_object_set_new(info, "room", json_integer(subscriber->room_id));
-							json_object_set_new(info, "feed", json_integer(feed_id));
-							json_object_set_new(info, "private_id", json_integer(pvt_id));
-							gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
+						if(ps->type == JANUS_VIDEOROOM_MEDIA_DATA && data_added) {
+							/* We already have a datachannel m-line, no need for others: just update the subscribers list */
+							janus_mutex_lock(&ps->subscribers_mutex);
+							if(g_slist_find(ps->subscribers, data_stream) == NULL && g_slist_find(data_stream->publisher_streams, ps) == NULL) {
+								ps->subscribers = g_slist_append(ps->subscribers, data_stream);
+								data_stream->publisher_streams = g_slist_append(data_stream->publisher_streams, ps);
+								/* The two streams reference each other */
+								janus_refcount_increase(&data_stream->ref);
+								janus_refcount_increase(&ps->ref);
+							}
+							janus_mutex_unlock(&ps->subscribers_mutex);
+							janus_mutex_unlock(&publisher->streams_mutex);
+							continue;
+						}
+						janus_videoroom_subscriber_stream *stream = janus_videoroom_subscriber_stream_add(subscriber,
+							ps, legacy, do_audio, do_video, do_data);
+						if(stream && ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO &&
+								(spatial || sc_substream || temporal || sc_temporal)) {
+							/* Override the default spatial/substream/temporal targets */
+							if(sc_substream)
+								stream->sim_context.substream_target = json_integer_value(sc_substream);
+							if(sc_temporal)
+								stream->sim_context.templayer_target = json_integer_value(sc_temporal);
+							if(spatial)
+								stream->target_spatial_layer = json_integer_value(spatial);
+							if(temporal)
+								stream->target_temporal_layer = json_integer_value(temporal);
+						}
+						if(ps->type == JANUS_VIDEOROOM_MEDIA_DATA) {
+							data_added = TRUE;
+							data_stream = stream;
+						}
+					} else {
+						/* Subscribe to all streams */
+						GList *temp = publisher->streams;
+						while(temp) {
+							janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+							if(ps->type == JANUS_VIDEOROOM_MEDIA_DATA && data_added) {
+								/* We already have a datachannel m-line, no need for others: just update the subscribers list */
+								janus_mutex_lock(&ps->subscribers_mutex);
+								if(g_slist_find(ps->subscribers, data_stream) == NULL && g_slist_find(data_stream->publisher_streams, ps) == NULL) {
+									ps->subscribers = g_slist_append(ps->subscribers, data_stream);
+									data_stream->publisher_streams = g_slist_append(data_stream->publisher_streams, ps);
+									/* The two streams reference each other */
+									janus_refcount_increase(&data_stream->ref);
+									janus_refcount_increase(&ps->ref);
+								}
+								janus_mutex_unlock(&ps->subscribers_mutex);
+								temp = temp->next;
+								continue;
+							}
+							janus_videoroom_subscriber_stream *stream = janus_videoroom_subscriber_stream_add(subscriber,
+								ps, legacy, do_audio, do_video, do_data);
+							if(stream && ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO &&
+									(spatial || sc_substream || temporal || sc_temporal)) {
+								/* Override the default spatial/substream/temporal targets */
+								if(sc_substream)
+									stream->sim_context.substream_target = json_integer_value(sc_substream);
+								if(sc_temporal)
+									stream->sim_context.templayer_target = json_integer_value(sc_temporal);
+								if(spatial)
+									stream->target_spatial_layer = json_integer_value(spatial);
+								if(temporal)
+									stream->target_temporal_layer = json_integer_value(temporal);
+							}
+							if(ps->type == JANUS_VIDEOROOM_MEDIA_DATA) {
+								data_added = TRUE;
+								data_stream = stream;
+							}
+							temp = temp->next;
 						}
-						continue;
 					}
-					janus_mutex_unlock(&publisher->subscribers_mutex);
+					janus_mutex_unlock(&publisher->streams_mutex);
+				}
+				/* Make sure we subscribed to at least something */
+				if(subscriber->streams == NULL) {
+					/* No subscription created? */
+					g_free(subscriber);
+					/* Unref publishers */
+					if(owner) {
+						janus_refcount_decrease(&owner->session->ref);
+						janus_refcount_decrease(&owner->ref);
+					}
+					while(publishers) {
+						janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+						janus_refcount_decrease(&publisher->ref);
+						janus_refcount_decrease(&publisher->session->ref);
+						publishers = g_list_remove(publishers, publisher);
+					}
+					JANUS_LOG(LOG_ERR, "Can't offer an SDP with no stream\n");
+					error_code = JANUS_VIDEOROOM_ERROR_INVALID_SDP;
+					g_snprintf(error_cause, 512, "Can't offer an SDP with no stream");
+					goto error;
+				}
+				session->participant = subscriber;
+				if(owner != NULL) {
+					/* Note: we should refcount these subscription-publisher mappings as well */
+					janus_mutex_lock(&owner->subscribers_mutex);
+					owner->subscriptions = g_slist_append(owner->subscriptions, subscriber);
+					janus_mutex_unlock(&owner->subscribers_mutex);
+					/* Done adding the subscription, owner is safe to be released */
+					janus_refcount_decrease(&owner->session->ref);
+					janus_refcount_decrease(&owner->ref);
+				}
+				event = json_object();
+				json_object_set_new(event, "videoroom", json_string("attached"));
+				json_object_set_new(event, "room", json_integer(subscriber->room_id));
+				/* If this is a legacy subscription, put the feed ID too */
+				if(legacy)
+					json_object_set_new(event, "id", json_integer(feed_id));
+				json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, legacy, event);
+				json_object_set_new(event, "streams", media);
+				session->participant_type = janus_videoroom_p_type_subscriber;
+				JANUS_LOG(LOG_VERB, "Preparing JSON event as a reply\n");
+				/* Negotiate by crafting a new SDP matching the subscriptions */
+				janus_mutex_lock(&subscriber->streams_mutex);
+				json_t *jsep = janus_videoroom_subscriber_offer(subscriber);
+				janus_mutex_unlock(&subscriber->streams_mutex);
+				/* How long will the Janus core take to push the event? */
+				g_atomic_int_set(&session->hangingup, 0);
+				gint64 start = janus_get_monotonic_time();
+				int res = gateway->push_event(msg->handle, &janus_videoroom_plugin, msg->transaction, event, jsep);
+				JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
+				json_decref(event);
+				json_decref(jsep);
+				janus_videoroom_message_free(msg);
+				/* Also notify event handlers */
+				if(notify_events && gateway->events_is_enabled()) {
+					json_t *info = json_object();
+					json_object_set_new(info, "event", json_string("subscribing"));
+					json_object_set_new(info, "room", json_integer(subscriber->room_id));
+					json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+					json_object_set_new(info, "streams", media);
+					json_object_set_new(info, "private_id", json_integer(pvt_id));
+					gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
 				}
+				/* Decrease the references we took before */
+				while(publishers) {
+					janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+					janus_refcount_decrease(&publisher->ref);
+					janus_refcount_decrease(&publisher->session->ref);
+					publishers = g_list_remove(publishers, publisher);
+				}
+				continue;
 			} else {
 				janus_mutex_unlock(&videoroom->mutex);
 				JANUS_LOG(LOG_ERR, "Invalid element (ptype)\n");
@@ -5478,7 +6733,7 @@ static void *janus_videoroom_handler(void *data) {
 				g_snprintf(error_cause, 512, "Already in as a publisher on this handle");
 				goto error;
 			} else if(!strcasecmp(request_text, "configure") || !strcasecmp(request_text, "publish")) {
-				if(!strcasecmp(request_text, "publish") && participant->sdp) {
+				if(!strcasecmp(request_text, "publish") && participant->session->started) {
 					janus_refcount_decrease(&participant->ref);
 					JANUS_LOG(LOG_ERR, "Can't publish, already published\n");
 					error_code = JANUS_VIDEOROOM_ERROR_ALREADY_PUBLISHED;
@@ -5500,91 +6755,130 @@ static void *janus_videoroom_handler(void *data) {
 					janus_refcount_decrease(&participant->ref);
 					goto error;
 				}
-				json_t *audio = json_object_get(root, "audio");
+				json_t *descriptions = json_object_get(root, "descriptions");
+				if(descriptions != NULL && json_array_size(descriptions) > 0) {
+					size_t i = 0;
+					for(i=0; i<json_array_size(descriptions); i++) {
+						json_t *d = json_array_get(descriptions, i);
+						JANUS_VALIDATE_JSON_OBJECT(d, publish_desc_parameters,
+							error_code, error_cause, TRUE,
+							JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+						if(error_code != 0) {
+							janus_mutex_unlock(&videoroom->mutex);
+							janus_refcount_decrease(&videoroom->ref);
+							goto error;
+
+						}
+					}
+				}
 				json_t *audiocodec = json_object_get(root, "audiocodec");
-				json_t *video = json_object_get(root, "video");
 				json_t *videocodec = json_object_get(root, "videocodec");
-				json_t *data = json_object_get(root, "data");
 				json_t *bitrate = json_object_get(root, "bitrate");
 				json_t *keyframe = json_object_get(root, "keyframe");
 				json_t *record = json_object_get(root, "record");
 				json_t *recfile = json_object_get(root, "filename");
 				json_t *display = json_object_get(root, "display");
 				json_t *update = json_object_get(root, "update");
-				if(audio) {
-					gboolean audio_active = json_is_true(audio);
-					if(session->started && audio_active && !participant->audio_active) {
-						/* Audio was just resumed, try resetting the RTP headers for viewers */
-						janus_mutex_lock(&participant->subscribers_mutex);
-						GSList *ps = participant->subscribers;
-						while(ps) {
-							janus_videoroom_subscriber *l = (janus_videoroom_subscriber *)ps->data;
-							if(l)
-								l->context.a_seq_reset = TRUE;
-							ps = ps->next;
+				/* Audio, video and data are deprecated properties */
+				json_t *audio = json_object_get(root, "audio");
+				json_t *video = json_object_get(root, "video");
+				json_t *data = json_object_get(root, "data");
+				/* Better to specify the 'send' property of a specific 'mid' */
+				const char *mid = json_string_value(json_object_get(root, "mid"));
+				json_t *send = json_object_get(root, "send");
+				/* A renegotiation may be taking place */
+				gboolean do_update = update ? json_is_true(update) : FALSE;
+				if(do_update && !sdp_update) {
+					JANUS_LOG(LOG_WARN, "Got an 'update' request, but no SDP update? Ignoring...\n");
+					do_update = FALSE;
+				}
+				/* Check if there's an SDP to take into account */
+				if(json_string_value(json_object_get(msg->jsep, "sdp"))) {
+					if(audiocodec && !sdp_update) {
+						/* The participant would like to use an audio codec in particular */
+						janus_audiocodec acodec = janus_audiocodec_from_name(json_string_value(audiocodec));
+						if(acodec == JANUS_AUDIOCODEC_NONE ||
+								(acodec != participant->room->acodec[0] &&
+								acodec != participant->room->acodec[1] &&
+								acodec != participant->room->acodec[2])) {
+							JANUS_LOG(LOG_ERR, "Participant asked for audio codec '%s', but it's not allowed (room %"SCNu64", user %"SCNu64")\n",
+								json_string_value(audiocodec), participant->room_id, participant->user_id);
+							janus_refcount_decrease(&participant->ref);
+							error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+							g_snprintf(error_cause, 512, "Audio codec unavailable in this room");
+							goto error;
 						}
-						janus_mutex_unlock(&participant->subscribers_mutex);
-					}
-					participant->audio_active = audio_active;
-					JANUS_LOG(LOG_VERB, "Setting audio property: %s (room %"SCNu64", user %"SCNu64")\n", participant->audio_active ? "true" : "false", participant->room_id, participant->user_id);
-				}
-				if(audiocodec && json_string_value(json_object_get(msg->jsep, "sdp")) != NULL) {
-					/* The participant would like to use an audio codec in particular */
-					janus_audiocodec acodec = janus_audiocodec_from_name(json_string_value(audiocodec));
-					if(acodec == JANUS_AUDIOCODEC_NONE ||
-							(acodec != participant->room->acodec[0] &&
-							acodec != participant->room->acodec[1] &&
-							acodec != participant->room->acodec[2])) {
-						JANUS_LOG(LOG_ERR, "Participant asked for audio codec '%s', but it's not allowed (room %"SCNu64", user %"SCNu64")\n",
+						JANUS_LOG(LOG_VERB, "Participant asked for audio codec '%s' (room %"SCNu64", user %"SCNu64")\n",
 							json_string_value(audiocodec), participant->room_id, participant->user_id);
-						janus_refcount_decrease(&participant->ref);
-						error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
-						g_snprintf(error_cause, 512, "Audio codec unavailable in this room");
-						goto error;
 					}
-					participant->acodec = acodec;
-					JANUS_LOG(LOG_VERB, "Participant asked for audio codec '%s' (room %"SCNu64", user %"SCNu64")\n",
-						json_string_value(audiocodec), participant->room_id, participant->user_id);
-				}
-				if(video) {
-					gboolean video_active = json_is_true(video);
-					if(session->started && video_active && !participant->video_active) {
-						/* Video was just resumed, try resetting the RTP headers for viewers */
-						janus_mutex_lock(&participant->subscribers_mutex);
-						GSList *ps = participant->subscribers;
-						while(ps) {
-							janus_videoroom_subscriber *l = (janus_videoroom_subscriber *)ps->data;
-							if(l)
-								l->context.v_seq_reset = TRUE;
-							ps = ps->next;
+					if(videocodec && !sdp_update) {
+						/* The participant would like to use a video codec in particular */
+						janus_videocodec vcodec = janus_videocodec_from_name(json_string_value(videocodec));
+						if(vcodec == JANUS_VIDEOCODEC_NONE ||
+								(vcodec != participant->room->vcodec[0] &&
+								vcodec != participant->room->vcodec[1] &&
+								vcodec != participant->room->vcodec[2])) {
+							JANUS_LOG(LOG_ERR, "Participant asked for video codec '%s', but it's not allowed (room %"SCNu64", user %"SCNu64")\n",
+								json_string_value(videocodec), participant->room_id, participant->user_id);
+							janus_refcount_decrease(&participant->ref);
+							error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+							g_snprintf(error_cause, 512, "Video codec unavailable in this room");
+							goto error;
 						}
-						janus_mutex_unlock(&participant->subscribers_mutex);
-					}
-					participant->video_active = video_active;
-					JANUS_LOG(LOG_VERB, "Setting video property: %s (room %"SCNu64", user %"SCNu64")\n", participant->video_active ? "true" : "false", participant->room_id, participant->user_id);
-				}
-				if(videocodec && json_string_value(json_object_get(msg->jsep, "sdp")) != NULL) {
-					/* The participant would like to use a video codec in particular */
-					janus_videocodec vcodec = janus_videocodec_from_name(json_string_value(videocodec));
-					if(vcodec == JANUS_VIDEOCODEC_NONE ||
-							(vcodec != participant->room->vcodec[0] &&
-							vcodec != participant->room->vcodec[1] &&
-							vcodec != participant->room->vcodec[2])) {
-						JANUS_LOG(LOG_ERR, "Participant asked for video codec '%s', but it's not allowed (room %"SCNu64", user %"SCNu64")\n",
+						JANUS_LOG(LOG_VERB, "Participant asked for video codec '%s' (room %"SCNu64", user %"SCNu64")\n",
 							json_string_value(videocodec), participant->room_id, participant->user_id);
-						janus_refcount_decrease(&participant->ref);
-						error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
-						g_snprintf(error_cause, 512, "Video codec unavailable in this room");
-						goto error;
 					}
-					participant->vcodec = vcodec;
-					JANUS_LOG(LOG_VERB, "Participant asked for video codec '%s' (room %"SCNu64", user %"SCNu64")\n",
-						json_string_value(videocodec), participant->room_id, participant->user_id);
 				}
-				if(data) {
-					gboolean data_active = json_is_true(data);
-					participant->data_active = data_active;
-					JANUS_LOG(LOG_VERB, "Setting data property: %s (room %"SCNu64", user %"SCNu64")\n", participant->data_active ? "true" : "false", participant->room_id, participant->user_id);
+				/* Update the audio/video/data flags, if set (and just configuring) */
+				if(audio || video || data || (mid && send)) {
+					janus_mutex_lock(&participant->streams_mutex);
+					GList *temp = participant->streams;
+					while(temp) {
+						janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+						gboolean mid_found = (mid && send && !strcasecmp(ps->mid, mid));
+						if(ps->type == JANUS_VIDEOROOM_MEDIA_AUDIO && (audio || mid_found)) {
+							gboolean audio_active = mid_found ? json_is_true(send) : json_is_true(audio);
+							if(!ps->active && audio_active) {
+								/* Audio was just resumed, try resetting the RTP headers for viewers */
+								janus_mutex_lock(&ps->subscribers_mutex);
+								GSList *slist = ps->subscribers;
+								while(slist) {
+									janus_videoroom_subscriber_stream *s = (janus_videoroom_subscriber_stream *)slist->data;
+									if(s)
+										s->context.seq_reset = TRUE;
+									slist = slist->next;
+								}
+								janus_mutex_unlock(&ps->subscribers_mutex);
+							}
+							ps->active = audio_active;
+							JANUS_LOG(LOG_VERB, "Setting audio property (%s): %s (room %"SCNu64", user %"SCNu64")\n",
+								ps->mid, ps->active ? "true" : "false", participant->room_id, participant->user_id);
+						} else if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO && (video || mid_found)) {
+							gboolean video_active = mid_found ? json_is_true(send) : json_is_true(video);
+							if(!ps->active && video_active) {
+								/* Video was just resumed, try resetting the RTP headers for viewers */
+								janus_mutex_lock(&participant->subscribers_mutex);
+								GSList *slist = ps->subscribers;
+								while(slist) {
+									janus_videoroom_subscriber_stream *s = (janus_videoroom_subscriber_stream *)slist->data;
+									if(s)
+										s->context.seq_reset = TRUE;
+									slist = slist->next;
+								}
+								janus_mutex_unlock(&participant->subscribers_mutex);
+							}
+							ps->active = video_active;
+							JANUS_LOG(LOG_VERB, "Setting video property (%s): %s (room %"SCNu64", user %"SCNu64")\n",
+								ps->mid, ps->active ? "true" : "false", participant->room_id, participant->user_id);
+						} else if(ps->type == JANUS_VIDEOROOM_MEDIA_DATA && (data || mid_found)) {
+							gboolean data_active = mid_found ? json_is_true(send) : json_is_true(data);
+							ps->active = data_active;
+							JANUS_LOG(LOG_VERB, "Setting data property (%s): %s (room %"SCNu64", user %"SCNu64")\n",
+								ps->mid, ps->active ? "true" : "false", participant->room_id, participant->user_id);
+						}
+						temp = temp->next;
+					}
+					janus_mutex_unlock(&participant->streams_mutex);
 				}
 				if(bitrate) {
 					participant->bitrate = json_integer_value(bitrate);
@@ -5594,11 +6888,17 @@ static void *janus_videoroom_handler(void *data) {
 						participant->remb_latest = janus_get_monotonic_time();
 					char rtcpbuf[24];
 					janus_rtcp_remb((char *)(&rtcpbuf), 24, participant->bitrate);
-					gateway->relay_rtcp(msg->handle, 1, rtcpbuf, 24);
+					gateway->relay_rtcp(msg->handle, -1, TRUE, rtcpbuf, 24);
 				}
 				if(keyframe && json_is_true(keyframe)) {
-					/* Send a FIR */
-					janus_videoroom_reqfir(participant, "Keyframe request");
+					/* FIXME Send a PLI on all video streams */
+					GList *temp = participant->streams;
+					while(temp) {
+						janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+						if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO)
+							janus_videoroom_reqpli(ps, "Keyframe request");
+						temp = temp->next;
+					}
 				}
 				janus_mutex_lock(&participant->rec_mutex);
 				gboolean prev_recording_active = participant->recording_active;
@@ -5616,15 +6916,17 @@ static void *janus_videoroom_handler(void *data) {
 					if(!participant->recording_active) {
 						/* Not recording (anymore?) */
 						janus_videoroom_recorder_close(participant);
-					} else if(participant->recording_active && participant->sdp) {
-						/* We've started recording, send a PLI/FIR and go on */
-						janus_videoroom_recorder_create(
-							participant, strstr(participant->sdp, "m=audio") != NULL,
-							strstr(participant->sdp, "m=video") != NULL,
-							strstr(participant->sdp, "m=application") != NULL);
-						if(strstr(participant->sdp, "m=video")) {
-							/* Send a FIR */
-							janus_videoroom_reqfir(participant, "Recording video");
+					} else if(participant->recording_active && participant->session->started) {
+						/* We've started recording, send a PLI and go on */
+						GList *temp = participant->streams;
+						while(temp) {
+							janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+							janus_videoroom_recorder_create(ps);
+							if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO) {
+								/* Send a PLI */
+								janus_videoroom_reqpli(ps, "Recording video");
+							}
+							temp = temp->next;
 						}
 					}
 				}
@@ -5634,21 +6936,19 @@ static void *janus_videoroom_handler(void *data) {
 					char *old_display = participant->display;
 					char *new_display = g_strdup(json_string_value(display));
 					participant->display = new_display;
-					g_free(old_display);
-					json_t *display_event = json_object();
-					json_object_set_new(display_event, "videoroom", json_string("event"));
-					json_object_set_new(display_event, "id", json_integer(participant->user_id));
-					json_object_set_new(display_event, "display", json_string(participant->display));
-					if(participant->room && !g_atomic_int_get(&participant->room->destroyed)) {
-						janus_videoroom_notify_participants(participant, display_event);
+					if(old_display != NULL) {
+						/* The display name changed, notify this */
+						json_t *display_event = json_object();
+						json_object_set_new(display_event, "videoroom", json_string("event"));
+						json_object_set_new(display_event, "id", json_integer(participant->user_id));
+						json_object_set_new(display_event, "display", json_string(participant->display));
+						if(participant->room && !g_atomic_int_get(&participant->room->destroyed)) {
+							janus_videoroom_notify_participants(participant, display_event);
+						}
+						json_decref(display_event);
 					}
+					g_free(old_display);
 					janus_mutex_unlock(&participant->room->mutex);
-					json_decref(display_event);
-				}
-				/* A renegotiation may be taking place */
-				gboolean do_update = update ? json_is_true(update) : FALSE;
-				if(do_update && !sdp_update) {
-					JANUS_LOG(LOG_WARN, "Got an 'update' request, but no SDP update? Ignoring...\n");
 				}
 				/* Done */
 				event = json_object();
@@ -5661,25 +6961,26 @@ static void *janus_videoroom_handler(void *data) {
 					json_object_set_new(info, "event", json_string("configured"));
 					json_object_set_new(info, "room", json_integer(participant->room_id));
 					json_object_set_new(info, "id", json_integer(participant->user_id));
-					json_object_set_new(info, "audio_active", participant->audio_active ? json_true() : json_false());
-					json_object_set_new(info, "video_active", participant->video_active ? json_true() : json_false());
-					json_object_set_new(info, "data_active", participant->data_active ? json_true() : json_false());
+						/* TODO Add info on all the streams, here */
+					//~ json_object_set_new(info, "audio_active", participant->audio_active ? json_true() : json_false());
+					//~ json_object_set_new(info, "video_active", participant->video_active ? json_true() : json_false());
+					//~ json_object_set_new(info, "data_active", participant->data_active ? json_true() : json_false());
 					json_object_set_new(info, "bitrate", json_integer(participant->bitrate));
-					if(participant->arc || participant->vrc || participant->drc) {
-						json_t *recording = json_object();
-						if(participant->arc && participant->arc->filename)
-							json_object_set_new(recording, "audio", json_string(participant->arc->filename));
-						if(participant->vrc && participant->vrc->filename)
-							json_object_set_new(recording, "video", json_string(participant->vrc->filename));
-						if(participant->drc && participant->drc->filename)
-							json_object_set_new(recording, "data", json_string(participant->drc->filename));
-						json_object_set_new(info, "recording", recording);
-					}
+					//~ if(participant->arc || participant->vrc || participant->drc) {
+						//~ json_t *recording = json_object();
+						//~ if(participant->arc && participant->arc->filename)
+							//~ json_object_set_new(recording, "audio", json_string(participant->arc->filename));
+						//~ if(participant->vrc && participant->vrc->filename)
+							//~ json_object_set_new(recording, "video", json_string(participant->vrc->filename));
+						//~ if(participant->drc && participant->drc->filename)
+							//~ json_object_set_new(recording, "data", json_string(participant->drc->filename));
+						//~ json_object_set_new(info, "recording", recording);
+					//~ }
 					gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
 				}
 			} else if(!strcasecmp(request_text, "unpublish")) {
 				/* This participant wants to unpublish */
-				if(!participant->sdp) {
+				if(!participant->session->started) {
 					janus_refcount_decrease(&participant->ref);
 					JANUS_LOG(LOG_ERR, "Can't unpublish, not published\n");
 					error_code = JANUS_VIDEOROOM_ERROR_NOT_PUBLISHED;
@@ -5703,9 +7004,6 @@ static void *janus_videoroom_handler(void *data) {
 				/* This publisher is leaving, tell everybody */
 				janus_videoroom_leave_or_unpublish(participant, TRUE, FALSE);
 				/* Done */
-				participant->audio_active = FALSE;
-				participant->video_active = FALSE;
-				participant->data_active = FALSE;
 				session->started = FALSE;
 				//~ session->destroy = TRUE;
 			} else {
@@ -5739,153 +7037,545 @@ static void *janus_videoroom_handler(void *data) {
 			} else if(!strcasecmp(request_text, "start")) {
 				/* Start/restart receiving the publisher streams */
 				if(subscriber->paused && msg->jsep == NULL) {
-					/* This is just resuming a paused stream, reset the RTP sequence numbers */
-					subscriber->context.a_seq_reset = TRUE;
-					subscriber->context.v_seq_reset = TRUE;
+					/* This is just resuming a paused subscription, reset the RTP sequence numbers on all streams */
+					GList *temp = subscriber->streams;
+					while(temp) {
+						janus_videoroom_subscriber_stream *stream = (janus_videoroom_subscriber_stream *)temp->data;
+						stream->context.seq_reset = TRUE;
+						temp = temp->next;
+					}
 				}
 				subscriber->paused = FALSE;
 				event = json_object();
 				json_object_set_new(event, "videoroom", json_string("event"));
 				json_object_set_new(event, "room", json_integer(subscriber->room_id));
 				json_object_set_new(event, "started", json_string("ok"));
-			} else if(!strcasecmp(request_text, "configure")) {
-				JANUS_VALIDATE_JSON_OBJECT(root, configure_parameters,
+			} else if(!strcasecmp(request_text, "subscribe")) {
+				/* Update a subscription by adding new streams */
+				JANUS_LOG(LOG_VERB, "Adding new subscriber streams\n");
+				JANUS_VALIDATE_JSON_OBJECT(root, subscriber_update_parameters,
 					error_code, error_cause, TRUE,
 					JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
 				if(error_code != 0)
 					goto error;
-				if(subscriber->kicked) {
-					JANUS_LOG(LOG_ERR, "Unauthorized, you have been kicked\n");
-					error_code = JANUS_VIDEOROOM_ERROR_UNAUTHORIZED;
-					g_snprintf(error_cause, 512, "Unauthorized, you have been kicked");
-					goto error;
-				}
-				json_t *audio = json_object_get(root, "audio");
-				json_t *video = json_object_get(root, "video");
-				json_t *data = json_object_get(root, "data");
-				json_t *restart = json_object_get(root, "restart");
-				json_t *update = json_object_get(root, "update");
-				json_t *spatial = json_object_get(root, "spatial_layer");
-				json_t *sc_substream = json_object_get(root, "substream");
-				if(json_integer_value(spatial) < 0 || json_integer_value(spatial) > 2 ||
-						json_integer_value(sc_substream) < 0 || json_integer_value(sc_substream) > 2) {
-					JANUS_LOG(LOG_ERR, "Invalid element (substream/spatial_layer should be 0, 1 or 2)\n");
-					error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
-					g_snprintf(error_cause, 512, "Invalid value (substream/spatial_layer should be 0, 1 or 2)");
-					goto error;
-				}
-				json_t *temporal = json_object_get(root, "temporal_layer");
-				json_t *sc_temporal = json_object_get(root, "temporal");
-				if(json_integer_value(temporal) < 0 || json_integer_value(temporal) > 2 ||
-						json_integer_value(sc_temporal) < 0 || json_integer_value(sc_temporal) > 2) {
-					JANUS_LOG(LOG_ERR, "Invalid element (temporal/temporal_layer should be 0, 1 or 2)\n");
+				json_t *feeds = json_object_get(root, "streams");
+				if(json_array_size(feeds) == 0) {
+					JANUS_LOG(LOG_ERR, "Empty subscription list\n");
 					error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
-					g_snprintf(error_cause, 512, "Invalid value (temporal/temporal_layer should be 0, 1 or 2)");
+					g_snprintf(error_cause, 512, "Empty subscription list");
 					goto error;
 				}
-				/* Update the audio/video/data flags, if set */
-				janus_videoroom_publisher *publisher = subscriber->feed;
-				if(publisher) {
-					if(audio && publisher->audio && subscriber->audio_offered) {
-						gboolean oldaudio = subscriber->audio;
-						gboolean newaudio = json_is_true(audio);
-						if(!oldaudio && newaudio) {
-							/* Audio just resumed, reset the RTP sequence numbers */
-							subscriber->context.a_seq_reset = TRUE;
+				/* Make sure all the feeds we're subscribing to exist */
+				GList *publishers = NULL;
+				size_t i = 0;
+				for(i=0; i<json_array_size(feeds); i++) {
+					json_t *s = json_array_get(feeds, i);
+					JANUS_VALIDATE_JSON_OBJECT(s, subscriber_stream_parameters,
+						error_code, error_cause, TRUE,
+						JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+					if(error_code != 0) {
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
 						}
-						subscriber->audio = newaudio;
+						goto error;
 					}
-					if(video && publisher->video && subscriber->video_offered) {
-						gboolean oldvideo = subscriber->video;
-						gboolean newvideo = json_is_true(video);
-						if(!oldvideo && newvideo) {
-							/* Video just resumed, reset the RTP sequence numbers */
-							subscriber->context.v_seq_reset = TRUE;
+					json_t *feed = json_object_get(s, "feed");
+					guint64 feed_id = json_integer_value(feed);
+					janus_mutex_lock(&subscriber->room->mutex);
+					janus_videoroom_publisher *publisher = g_hash_table_lookup(subscriber->room->participants, &feed_id);
+					janus_mutex_unlock(&subscriber->room->mutex);
+					if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || !publisher->session->started) {
+						JANUS_LOG(LOG_ERR, "No such feed (%"SCNu64")\n", feed_id);
+						error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+						g_snprintf(error_cause, 512, "No such feed (%"SCNu64")", feed_id);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
 						}
-						subscriber->video = newvideo;
-						if(subscriber->video) {
-							/* Send a FIR */
-							janus_videoroom_reqfir(publisher, "Restoring video for subscriber");
+						goto error;
+					}
+					const char *mid = json_string_value(json_object_get(s, "mid"));
+					if(mid != NULL) {
+						/* Check the mid too */
+						janus_mutex_lock(&publisher->streams_mutex);
+						if(g_hash_table_lookup(publisher->streams_bymid, mid) == NULL) {
+							janus_mutex_unlock(&publisher->streams_mutex);
+							JANUS_LOG(LOG_ERR, "No such mid '%s' in feed (%"SCNu64")\n", mid, feed_id);
+							error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+							g_snprintf(error_cause, 512, "No such mid '%s' in feed (%"SCNu64")", mid, feed_id);
+							/* Unref publishers we may have taken note of so far */
+							while(publishers) {
+								publisher = (janus_videoroom_publisher *)publishers->data;
+								janus_refcount_decrease(&publisher->ref);
+								janus_refcount_decrease(&publisher->session->ref);
+								publishers = g_list_remove(publishers, publisher);
+							}
+							goto error;
 						}
+						janus_mutex_unlock(&publisher->streams_mutex);
 					}
-					if(data && publisher->data && subscriber->data_offered)
-						subscriber->data = json_is_true(data);
-					/* Check if a simulcasting-related request is involved */
-					if(sc_substream && (publisher->ssrc[0] != 0 || publisher->rid[0] != NULL)) {
-						subscriber->sim_context.substream_target = json_integer_value(sc_substream);
-						JANUS_LOG(LOG_VERB, "Setting video SSRC to let through (simulcast): %"SCNu32" (index %d, was %d)\n",
-							publisher->ssrc[subscriber->sim_context.substream],
-							subscriber->sim_context.substream_target,
-							subscriber->sim_context.substream);
-						if(subscriber->sim_context.substream_target == subscriber->sim_context.substream) {
-							/* No need to do anything, we're already getting the right substream, so notify the user */
-							json_t *event = json_object();
-							json_object_set_new(event, "videoroom", json_string("event"));
-							json_object_set_new(event, "room", json_integer(subscriber->room_id));
-							json_object_set_new(event, "substream", json_integer(subscriber->sim_context.substream));
-							gateway->push_event(msg->handle, &janus_videoroom_plugin, NULL, event, NULL);
-							json_decref(event);
-						} else {
-							/* Send a FIR */
-							janus_videoroom_reqfir(publisher, "Simulcasting substream change");
+					json_t *spatial = json_object_get(s, "spatial_layer");
+					json_t *sc_substream = json_object_get(s, "substream");
+					if(json_integer_value(spatial) < 0 || json_integer_value(spatial) > 2 ||
+							json_integer_value(sc_substream) < 0 || json_integer_value(sc_substream) > 2) {
+						JANUS_LOG(LOG_ERR, "Invalid element (substream/spatial_layer should be 0, 1 or 2)\n");
+						error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+						g_snprintf(error_cause, 512, "Invalid value (substream/spatial_layer should be 0, 1 or 2)");
+						janus_mutex_unlock(&videoroom->mutex);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
 						}
+						goto error;
 					}
-					if(subscriber->feed && subscriber->feed->vcodec == JANUS_VIDEOCODEC_VP8 &&
-							sc_temporal && (publisher->ssrc[0] != 0 || publisher->rid[0] != NULL)) {
-						subscriber->sim_context.templayer_target = json_integer_value(sc_temporal);
-						JANUS_LOG(LOG_VERB, "Setting video temporal layer to let through (simulcast): %d (was %d)\n",
-							subscriber->sim_context.templayer_target, subscriber->sim_context.templayer);
-						if(subscriber->sim_context.templayer_target == subscriber->sim_context.templayer) {
-							/* No need to do anything, we're already getting the right temporal, so notify the user */
-							json_t *event = json_object();
-							json_object_set_new(event, "videoroom", json_string("event"));
-							json_object_set_new(event, "room", json_integer(subscriber->room_id));
-							json_object_set_new(event, "temporal", json_integer(subscriber->sim_context.templayer));
-							gateway->push_event(msg->handle, &janus_videoroom_plugin, NULL, event, NULL);
-							json_decref(event);
-						} else {
-							/* Send a FIR */
-							janus_videoroom_reqfir(publisher, "Simulcasting temporal layer change");
+					json_t *temporal = json_object_get(s, "temporal_layer");
+					json_t *sc_temporal = json_object_get(s, "temporal");
+					if(json_integer_value(temporal) < 0 || json_integer_value(temporal) > 2 ||
+							json_integer_value(sc_temporal) < 0 || json_integer_value(sc_temporal) > 2) {
+						JANUS_LOG(LOG_ERR, "Invalid element (temporal/temporal_layer should be 0, 1 or 2)\n");
+						error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+						g_snprintf(error_cause, 512, "Invalid value (temporal/temporal_layer should be 0, 1 or 2)");
+						janus_mutex_unlock(&videoroom->mutex);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
+						}
+						goto error;
+					}
+					/* Increase the refcount before unlocking so that nobody can remove and free the publisher in the meantime. */
+					janus_refcount_increase(&publisher->ref);
+					janus_refcount_increase(&publisher->session->ref);
+					publishers = g_list_append(publishers, publisher);
+				}
+				/* Update subscriptions, adding streams or replacing existing and inactive ones */
+				int changes = 0;
+				janus_mutex_lock(&subscriber->streams_mutex);
+				for(i=0; i<json_array_size(feeds); i++) {
+					json_t *s = json_array_get(feeds, i);
+					json_t *feed = json_object_get(s, "feed");
+					guint64 feed_id = json_integer_value(feed);
+					janus_mutex_lock(&subscriber->room->mutex);
+					janus_videoroom_publisher *publisher = g_hash_table_lookup(subscriber->room->participants, &feed_id);
+					janus_mutex_unlock(&subscriber->room->mutex);
+					if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || !publisher->session->started) {
+						JANUS_LOG(LOG_WARN, "Publisher '%"SCNu64"' not found, not subscribing...\n", feed_id);
+						continue;
+					}
+					/* Are we subscribing to this publisher as a whole or only to a single stream? */
+					const char *mid = json_string_value(json_object_get(s, "mid"));
+					json_t *spatial = json_object_get(s, "spatial_layer");
+					json_t *sc_substream = json_object_get(s, "substream");
+					json_t *temporal = json_object_get(s, "temporal_layer");
+					json_t *sc_temporal = json_object_get(s, "temporal");
+					if(mid != NULL) {
+						janus_mutex_lock(&publisher->streams_mutex);
+						janus_videoroom_publisher_stream *ps = g_hash_table_lookup(publisher->streams_bymid, mid);
+						janus_mutex_unlock(&publisher->streams_mutex);
+						if(ps == NULL) {
+							JANUS_LOG(LOG_WARN, "No mid '%s' in publisher '%"SCNu64"', not subscribing...\n", mid, feed_id);
+							continue;
+						}
+						janus_videoroom_subscriber_stream *stream = janus_videoroom_subscriber_stream_add_or_replace(subscriber, ps);
+						if(stream) {
+							changes++;
+							if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO &&
+									(spatial || sc_substream || temporal || sc_temporal)) {
+								/* Override the default spatial/substream/temporal targets */
+								if(sc_substream)
+									stream->sim_context.substream_target = json_integer_value(sc_substream);
+								if(sc_temporal)
+									stream->sim_context.templayer_target = json_integer_value(sc_temporal);
+								if(spatial)
+									stream->target_spatial_layer = json_integer_value(spatial);
+								if(temporal)
+									stream->target_temporal_layer = json_integer_value(temporal);
+							}
 						}
+					} else {
+						janus_mutex_lock(&publisher->streams_mutex);
+						GList *temp = publisher->streams;
+						while(temp) {
+							janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+							janus_videoroom_subscriber_stream *stream = janus_videoroom_subscriber_stream_add_or_replace(subscriber, ps);
+							if(stream) {
+								changes++;
+								if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO &&
+										(spatial || sc_substream || temporal || sc_temporal)) {
+									/* Override the default spatial/substream/temporal targets */
+									if(sc_substream)
+										stream->sim_context.substream_target = json_integer_value(sc_substream);
+									if(sc_temporal)
+										stream->sim_context.templayer_target = json_integer_value(sc_temporal);
+									if(spatial)
+										stream->target_spatial_layer = json_integer_value(spatial);
+									if(temporal)
+										stream->target_temporal_layer = json_integer_value(temporal);
+								}
+							}
+							temp = temp->next;
+						}
+						janus_mutex_unlock(&publisher->streams_mutex);
+					}
+				}
+				if(changes == 0) {
+					janus_mutex_unlock(&subscriber->streams_mutex);
+					/* Nothing changes, don't do anything */
+					JANUS_LOG(LOG_WARN, "No subscription done, skipping renegotiation\n");
+					janus_videoroom_message_free(msg);
+					/* Decrease the references we took before */
+					while(publishers) {
+						janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+						janus_refcount_decrease(&publisher->ref);
+						janus_refcount_decrease(&publisher->session->ref);
+						publishers = g_list_remove(publishers, publisher);
 					}
+					continue;
+				}
+				if(!g_atomic_int_get(&subscriber->answered)) {
+					/* We're still waiting for an answer to a previous offer, postpone this */
+					g_atomic_int_set(&subscriber->pending_offer, 1);
+					janus_mutex_unlock(&subscriber->streams_mutex);
+					JANUS_LOG(LOG_VERB, "Post-poning new offer, waiting for previous answer\n");
+					/* Decrease the references we took before */
+					while(publishers) {
+						janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+						janus_refcount_decrease(&publisher->ref);
+						janus_refcount_decrease(&publisher->session->ref);
+						publishers = g_list_remove(publishers, publisher);
+					}
+					janus_videoroom_message_free(msg);
+					continue;
+				}
+				event = json_object();
+				json_object_set_new(event, "videoroom", json_string("updated"));
+				json_object_set_new(event, "room", json_integer(subscriber->room_id));
+				json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+				json_object_set_new(event, "streams", media);
+				/* Generate a new offer */
+				json_t *jsep = janus_videoroom_subscriber_offer(subscriber);
+				janus_mutex_unlock(&subscriber->streams_mutex);
+				/* How long will the Janus core take to push the event? */
+				gint64 start = janus_get_monotonic_time();
+				int res = gateway->push_event(msg->handle, &janus_videoroom_plugin, msg->transaction, event, jsep);
+				JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
+				json_decref(event);
+				json_decref(jsep);
+				/* Also notify event handlers */
+				if(notify_events && gateway->events_is_enabled()) {
+					json_t *info = json_object();
+					json_object_set_new(info, "event", json_string("updated"));
+					json_object_set_new(info, "room", json_integer(subscriber->room_id));
+					json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+					json_object_set_new(info, "streams", media);
+					json_object_set_new(info, "private_id", json_integer(subscriber->pvt_id));
+					gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
+				}
+				/* Decrease the references we took before */
+				while(publishers) {
+					janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+					janus_refcount_decrease(&publisher->ref);
+					janus_refcount_decrease(&publisher->session->ref);
+					publishers = g_list_remove(publishers, publisher);
+				}
+				/* Done */
+				janus_videoroom_message_free(msg);
+				continue;
+			} else if(!strcasecmp(request_text, "unsubscribe")) {
+				/* TODO Update a subscription by removing existing streams */
+				JANUS_LOG(LOG_VERB, "Removing subscriber streams\n");
+				JANUS_VALIDATE_JSON_OBJECT(root, subscriber_update_parameters,
+					error_code, error_cause, TRUE,
+					JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+				if(error_code != 0)
+					goto error;
+				json_t *feeds = json_object_get(root, "streams");
+				if(json_array_size(feeds) == 0) {
+					JANUS_LOG(LOG_ERR, "Empty unsubscription list\n");
+					error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+					g_snprintf(error_cause, 512, "Empty unsubscription list");
+					goto error;
+				}
+				/* Validate the request first */
+				size_t i = 0;
+				for(i=0; i<json_array_size(feeds); i++) {
+					json_t *s = json_array_get(feeds, i);
+					JANUS_VALIDATE_JSON_OBJECT(s, subscriber_remove_parameters,
+						error_code, error_cause, TRUE,
+						JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+					if(error_code != 0)
+						goto error;
+				}
+				/* Now remove the specified subscriptions */
+				int changes = 0;
+				janus_mutex_lock(&subscriber->streams_mutex);
+				for(i=0; i<json_array_size(feeds); i++) {
+					json_t *s = json_array_get(feeds, i);
+					json_t *feed = json_object_get(s, "feed");
+					guint64 feed_id = json_integer_value(feed);
+					const char *sub_mid = json_string_value(json_object_get(s, "sub_mid"));
+					janus_videoroom_subscriber_stream *stream = NULL;
+					if(sub_mid) {
+						/* A specific subscription mid has been provided */
+						stream = g_hash_table_lookup(subscriber->streams_bymid, sub_mid);
+						if(stream == NULL) {
+							JANUS_LOG(LOG_WARN, "Subscriber stream with mid '%s' not found, not unsubscribing...\n", sub_mid);
+							continue;
+						}
+						janus_videoroom_subscriber_stream_remove(stream, NULL, TRUE);
+						changes++;
+					} else if(feed_id > 0) {
+						janus_mutex_lock(&subscriber->room->mutex);
+						janus_videoroom_publisher *publisher = g_hash_table_lookup(subscriber->room->participants, &feed_id);
+						janus_mutex_unlock(&subscriber->room->mutex);
+						if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || !publisher->session->started) {
+							JANUS_LOG(LOG_WARN, "Publisher '%"SCNu64"' not found, not unsubscribing...\n", feed_id);
+							continue;
+						}
+						/* Are we unsubscribing from the publisher as a whole or only a single stream? */
+						const char *mid = json_string_value(json_object_get(s, "mid"));
+						/* Iterate on all subscriptions, and remove those that don't match */
+						GList *temp = subscriber->streams;
+						while(temp) {
+							/* We need more fine grained mechanisms for changing streaming properties */
+							janus_videoroom_subscriber_stream *stream = (janus_videoroom_subscriber_stream *)temp->data;
+							janus_videoroom_publisher_stream *ps = NULL;
+							GSList *list = stream->publisher_streams;
+							while(list) {
+								ps = list->data;
+								if(ps == NULL || ps->publisher != publisher) {
+									/* Not the publisher we're interested in */
+									list = list->next;
+									continue;
+								}
+								if(mid && ps->mid && strcasecmp(ps->mid, mid)) {
+									/* Not the mid we're interested in */
+									list = list->next;
+									continue;
+								}
+								janus_videoroom_subscriber_stream_remove(stream, ps, TRUE);
+								if(stream->type != JANUS_VIDEOROOM_MEDIA_DATA)
+									changes++;
+								list = list->next;
+							}
+							temp = temp->next;
+						}
+					}
+				}
+				if(changes == 0) {
+					janus_mutex_unlock(&subscriber->streams_mutex);
+					/* Nothing changes, don't do anything */
+					JANUS_LOG(LOG_VERB, "No unsubscription done, skipping renegotiation\n");
+					janus_videoroom_message_free(msg);
+					continue;
 				}
-				if(subscriber->room->do_svc) {
-					/* Also check if the viewer is trying to configure a layer change */
-					if(spatial) {
-						int spatial_layer = json_integer_value(spatial);
-						if(spatial_layer > 1) {
-							JANUS_LOG(LOG_WARN, "Spatial layer higher than 1, it will be ignored if using EnabledByFlag_2SL3TL\n");
+				if(!g_atomic_int_get(&subscriber->answered)) {
+					/* We're still waiting for an answer to a previous offer, postpone this */
+					g_atomic_int_set(&subscriber->pending_offer, 1);
+					janus_mutex_unlock(&subscriber->streams_mutex);
+					JANUS_LOG(LOG_VERB, "Post-poning new offer, waiting for previous answer\n");
+					janus_videoroom_message_free(msg);
+					continue;
+				}
+				event = json_object();
+				json_object_set_new(event, "videoroom", json_string("updated"));
+				json_object_set_new(event, "room", json_integer(subscriber->room_id));
+				json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+				json_object_set_new(event, "streams", media);
+				/* Generate a new offer */
+				json_t *jsep = janus_videoroom_subscriber_offer(subscriber);
+				janus_mutex_unlock(&subscriber->streams_mutex);
+				/* How long will the Janus core take to push the event? */
+				gint64 start = janus_get_monotonic_time();
+				int res = gateway->push_event(msg->handle, &janus_videoroom_plugin, msg->transaction, event, jsep);
+				JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
+				json_decref(event);
+				json_decref(jsep);
+				/* Done */
+				janus_videoroom_message_free(msg);
+				continue;
+			} else if(!strcasecmp(request_text, "configure")) {
+				JANUS_VALIDATE_JSON_OBJECT(root, configure_parameters,
+					error_code, error_cause, TRUE,
+					JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+				if(error_code != 0)
+					goto error;
+				if(subscriber->kicked) {
+					JANUS_LOG(LOG_ERR, "Unauthorized, you have been kicked\n");
+					error_code = JANUS_VIDEOROOM_ERROR_UNAUTHORIZED;
+					g_snprintf(error_cause, 512, "Unauthorized, you have been kicked");
+					goto error;
+				}
+				/* Audio, video and data are deprecated properties */
+				json_t *audio = json_object_get(root, "audio");
+				json_t *video = json_object_get(root, "video");
+				json_t *data = json_object_get(root, "data");
+				/* Better to specify the 'send' property of a specific 'mid' */
+				const char *mid = json_string_value(json_object_get(root, "mid"));
+				json_t *send = json_object_get(root, "send");
+				json_t *restart = json_object_get(root, "restart");
+				json_t *update = json_object_get(root, "update");
+				json_t *spatial = json_object_get(root, "spatial_layer");
+				json_t *sc_substream = json_object_get(root, "substream");
+				if(json_integer_value(spatial) < 0 || json_integer_value(spatial) > 2 ||
+						json_integer_value(sc_substream) < 0 || json_integer_value(sc_substream) > 2) {
+					JANUS_LOG(LOG_ERR, "Invalid element (substream/spatial_layer should be 0, 1 or 2)\n");
+					error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+					g_snprintf(error_cause, 512, "Invalid value (substream/spatial_layer should be 0, 1 or 2)");
+					goto error;
+				}
+				json_t *temporal = json_object_get(root, "temporal_layer");
+				json_t *sc_temporal = json_object_get(root, "temporal");
+				if(json_integer_value(temporal) < 0 || json_integer_value(temporal) > 2 ||
+						json_integer_value(sc_temporal) < 0 || json_integer_value(sc_temporal) > 2) {
+					JANUS_LOG(LOG_ERR, "Invalid element (temporal/temporal_layer should be 0, 1 or 2)\n");
+					error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+					g_snprintf(error_cause, 512, "Invalid value (temporal/temporal_layer should be 0, 1 or 2)");
+					goto error;
+				}
+				/* Update the audio/video/data flags, if set */
+				janus_mutex_lock(&subscriber->streams_mutex);
+				GList *temp = subscriber->streams;
+				while(temp) {
+					/* We need more fine grained mechanisms for changing streaming properties */
+					janus_videoroom_subscriber_stream *stream = (janus_videoroom_subscriber_stream *)temp->data;
+					janus_videoroom_publisher_stream *ps = stream->publisher_streams ? stream->publisher_streams->data : NULL;
+					if(audio && stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO) {
+						gboolean oldaudio = stream->send;
+						gboolean newaudio = json_is_true(audio);
+						if(!oldaudio && newaudio) {
+							/* Audio just resumed, reset the RTP sequence numbers */
+							stream->context.seq_reset = TRUE;
 						}
-						if(spatial_layer == subscriber->spatial_layer) {
-							/* No need to do anything, we're already getting the right spatial layer, so notify the user */
-							json_t *event = json_object();
-							json_object_set_new(event, "videoroom", json_string("event"));
-							json_object_set_new(event, "room", json_integer(subscriber->room_id));
-							json_object_set_new(event, "spatial_layer", json_integer(subscriber->spatial_layer));
-							gateway->push_event(msg->handle, &janus_videoroom_plugin, NULL, event, NULL);
-							json_decref(event);
-						} else if(spatial_layer != subscriber->target_spatial_layer) {
-							/* Send a FIR to the new RTP forward publisher */
-							janus_videoroom_reqfir(publisher, "Need to downscale spatially");
+						stream->send = newaudio;
+					}
+					if(video && stream->type == JANUS_VIDEOROOM_MEDIA_VIDEO) {
+						gboolean oldvideo = stream->send;
+						gboolean newvideo = json_is_true(video);
+						if(!oldvideo && newvideo) {
+							/* Audio just resumed, reset the RTP sequence numbers */
+							stream->context.seq_reset = TRUE;
+						}
+						stream->send = newvideo;
+						if(newvideo) {
+							/* Send a PLI */
+							janus_videoroom_reqpli(ps, "Restoring video for subscriber");
 						}
-						subscriber->target_spatial_layer = spatial_layer;
 					}
-					if(temporal) {
-						int temporal_layer = json_integer_value(temporal);
-						if(temporal_layer > 2) {
-							JANUS_LOG(LOG_WARN, "Temporal layer higher than 2, will probably be ignored\n");
+					if(data && stream->type == JANUS_VIDEOROOM_MEDIA_DATA)
+						stream->send = json_is_true(data);
+					/* Let's also see if this is the right mid */
+					if(mid && strcasecmp(stream->mid, mid)) {
+						temp = temp->next;
+						continue;
+					}
+					if(send) {
+						gboolean oldsend = stream->send;
+						gboolean newsend = json_is_true(send);
+						if(!oldsend && newsend) {
+							/* Medium just resumed, reset the RTP sequence numbers */
+							stream->context.seq_reset = TRUE;
 						}
-						if(temporal_layer == subscriber->temporal_layer) {
-							/* No need to do anything, we're already getting the right temporal layer, so notify the user */
-							json_t *event = json_object();
-							json_object_set_new(event, "videoroom", json_string("event"));
-							json_object_set_new(event, "room", json_integer(subscriber->room_id));
-							json_object_set_new(event, "temporal_layer", json_integer(subscriber->temporal_layer));
-							gateway->push_event(msg->handle, &janus_videoroom_plugin, NULL, event, NULL);
-							json_decref(event);
+						stream->send = json_is_true(send);
+					}
+					/* Next properties are for video only */
+					if(stream->type != JANUS_VIDEOROOM_MEDIA_VIDEO) {
+						temp = temp->next;
+						continue;
+					}
+					/* Check if a simulcasting-related request is involved */
+					if(ps && ps->simulcast) {
+						if(sc_substream) {
+							stream->sim_context.substream_target = json_integer_value(sc_substream);
+							JANUS_LOG(LOG_VERB, "Setting video SSRC to let through (simulcast): %"SCNu32" (index %d, was %d)\n",
+								ps->vssrc[stream->sim_context.substream],
+								stream->sim_context.substream_target,
+								stream->sim_context.substream);
+							if(stream->sim_context.substream_target == stream->sim_context.substream) {
+								/* No need to do anything, we're already getting the right substream, so notify the user */
+								json_t *event = json_object();
+								json_object_set_new(event, "videoroom", json_string("event"));
+								json_object_set_new(event, "room", json_integer(subscriber->room_id));
+								json_object_set_new(event, "mid", json_string(stream->mid));
+								json_object_set_new(event, "substream", json_integer(stream->sim_context.substream));
+								gateway->push_event(msg->handle, &janus_videoroom_plugin, NULL, event, NULL);
+								json_decref(event);
+							} else {
+								/* Send a PLI */
+								janus_videoroom_reqpli(ps, "Simulcasting substream change");
+							}
+						}
+						if(ps->vcodec == JANUS_VIDEOCODEC_VP8 && ps->simulcast && sc_temporal) {
+							stream->sim_context.templayer_target = json_integer_value(sc_temporal);
+							JANUS_LOG(LOG_VERB, "Setting video temporal layer to let through (simulcast): %d (was %d)\n",
+								stream->sim_context.templayer_target, stream->sim_context.templayer);
+							if(stream->sim_context.templayer_target == stream->sim_context.templayer) {
+								/* No need to do anything, we're already getting the right temporal, so notify the user */
+								json_t *event = json_object();
+								json_object_set_new(event, "videoroom", json_string("event"));
+								json_object_set_new(event, "room", json_integer(subscriber->room_id));
+								json_object_set_new(event, "mid", json_string(stream->mid));
+								json_object_set_new(event, "temporal", json_integer(stream->sim_context.templayer));
+								gateway->push_event(msg->handle, &janus_videoroom_plugin, NULL, event, NULL);
+								json_decref(event);
+							} else {
+								/* Send a PLI */
+								janus_videoroom_reqpli(ps, "Simulcasting temporal layer change");
+							}
+						}
+					} else if(ps->svc) {
+						/* Also check if the viewer is trying to configure a layer change */
+						if(spatial) {
+							int spatial_layer = json_integer_value(spatial);
+							if(spatial_layer > 1) {
+								JANUS_LOG(LOG_WARN, "Spatial layer higher than 1, will probably be ignored\n");
+							}
+							if(spatial_layer == stream->spatial_layer) {
+								/* No need to do anything, we're already getting the right spatial layer, so notify the user */
+								json_t *event = json_object();
+								json_object_set_new(event, "videoroom", json_string("event"));
+								json_object_set_new(event, "room", json_integer(subscriber->room_id));
+								json_object_set_new(event, "mid", json_string(stream->mid));
+								json_object_set_new(event, "spatial_layer", json_integer(stream->spatial_layer));
+								gateway->push_event(msg->handle, &janus_videoroom_plugin, NULL, event, NULL);
+								json_decref(event);
+							} else if(spatial_layer != stream->target_spatial_layer) {
+								/* Send a PLI to the new RTP forward publisher */
+								janus_videoroom_reqpli(ps, "Need to downscale spatially");
+							}
+							stream->target_spatial_layer = spatial_layer;
+						}
+						if(temporal) {
+							int temporal_layer = json_integer_value(temporal);
+							if(temporal_layer > 2) {
+								JANUS_LOG(LOG_WARN, "Temporal layer higher than 2, will probably be ignored\n");
+							}
+							if(temporal_layer == stream->temporal_layer) {
+								/* No need to do anything, we're already getting the right temporal layer, so notify the user */
+								json_t *event = json_object();
+								json_object_set_new(event, "videoroom", json_string("event"));
+								json_object_set_new(event, "room", json_integer(subscriber->room_id));
+								json_object_set_new(event, "mid", json_string(stream->mid));
+								json_object_set_new(event, "temporal_layer", json_integer(stream->temporal_layer));
+								gateway->push_event(msg->handle, &janus_videoroom_plugin, NULL, event, NULL);
+								json_decref(event);
+							}
+							stream->target_temporal_layer = temporal_layer;
 						}
-						subscriber->target_temporal_layer = temporal_layer;
 					}
+					temp = temp->next;
 				}
 				event = json_object();
 				json_object_set_new(event, "videoroom", json_string("event"));
@@ -5896,97 +7586,30 @@ static void *janus_videoroom_handler(void *data) {
 				gboolean do_update = update ? json_is_true(update) : FALSE;
 				if(sdp_update || do_restart || do_update) {
 					/* Negotiate by sending the selected publisher SDP back, and/or force an ICE restart */
-					if(publisher->sdp != NULL) {
-						char temp_error[512];
-						JANUS_LOG(LOG_VERB, "Munging SDP offer (update) to adapt it to the subscriber's requirements\n");
-						janus_sdp *offer = janus_sdp_parse(publisher->sdp, temp_error, sizeof(temp_error));
-						if(publisher->audio && !subscriber->audio_offered)
-							janus_sdp_mline_remove(offer, JANUS_SDP_AUDIO);
-						if(publisher->video && !subscriber->video_offered)
-							janus_sdp_mline_remove(offer, JANUS_SDP_VIDEO);
-						if(publisher->data && !subscriber->data_offered)
-							janus_sdp_mline_remove(offer, JANUS_SDP_APPLICATION);
-						/* This is an update, check if we need to update */
-						janus_sdp_mtype mtype[3] = { JANUS_SDP_AUDIO, JANUS_SDP_VIDEO, JANUS_SDP_APPLICATION };
-						int i=0;
-						for(i=0; i<3; i++) {
-							janus_sdp_mline *m = janus_sdp_mline_find(subscriber->sdp, mtype[i]);
-							janus_sdp_mline *m_new = janus_sdp_mline_find(offer, mtype[i]);
-							if(m != NULL && m->port > 0 && m->direction != JANUS_SDP_INACTIVE) {
-								/* We have such an m-line and it's active, should it be changed? */
-								if(m_new == NULL || m_new->port == 0 || m_new->direction == JANUS_SDP_INACTIVE) {
-									/* Turn the m-line to inactive */
-									m->direction = JANUS_SDP_INACTIVE;
-								}
-							} else {
-								/* We don't have such an m-line or it's disabled, should it be added/enabled? */
-								if(m_new != NULL && m_new->port > 0 && m_new->direction != JANUS_SDP_INACTIVE) {
-									if(m != NULL) {
-										m->port = m_new->port;
-										m->direction = m_new->direction;
-									} else {
-										/* Add the new m-line */
-										m = janus_sdp_mline_create(m_new->type, m_new->port, m_new->proto, m_new->direction);
-										subscriber->sdp->m_lines = g_list_append(subscriber->sdp->m_lines, m);
-									}
-									/* Copy/replace the other properties */
-									m->c_ipv4 = m_new->c_ipv4;
-									if(m_new->c_addr && (m->c_addr == NULL || strcmp(m->c_addr, m_new->c_addr))) {
-										g_free(m->c_addr);
-										m->c_addr = g_strdup(m_new->c_addr);
-									}
-									if(m_new->b_name && (m->b_name == NULL || strcmp(m->b_name, m_new->b_name))) {
-										g_free(m->b_name);
-										m->b_name = g_strdup(m_new->b_name);
-									}
-									m->b_value = m_new->b_value;
-									g_list_free_full(m->fmts, (GDestroyNotify)g_free);
-									m->fmts = NULL;
-									GList *fmts = m_new->fmts;
-									while(fmts) {
-										char *fmt = (char *)fmts->data;
-										if(fmt)
-											m->fmts = g_list_append(m->fmts,g_strdup(fmt));
-										fmts = fmts->next;
-									}
-									g_list_free(m->ptypes);
-									m->ptypes = g_list_copy(m_new->ptypes);
-									g_list_free_full(m->attributes, (GDestroyNotify)janus_sdp_attribute_destroy);
-									m->attributes = NULL;
-									GList *attr = m_new->attributes;
-									while(attr) {
-										janus_sdp_attribute *a = (janus_sdp_attribute *)attr->data;
-										janus_sdp_attribute_add_to_mline(m,
-											janus_sdp_attribute_create(a->name, "%s", a->value));
-										attr = attr->next;
-									}
-								}
-							}
-						}
-						janus_sdp_destroy(offer);
-						session->sdp_version++;
-						subscriber->sdp->o_version = session->sdp_version;
-						char *newsdp = janus_sdp_write(subscriber->sdp);
-						JANUS_LOG(LOG_VERB, "Updating subscriber:\n%s\n", newsdp);
-						json_t *jsep = json_pack("{ssss}", "type", "offer", "sdp", newsdp);
-						if(do_restart)
-							json_object_set_new(jsep, "restart", json_true());
-						/* How long will the Janus core take to push the event? */
-						gint64 start = janus_get_monotonic_time();
-						int res = gateway->push_event(msg->handle, &janus_videoroom_plugin, msg->transaction, event, jsep);
-						JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
-						json_decref(event);
-						json_decref(jsep);
-						g_free(newsdp);
-						/* Any update in the media directions? */
-						subscriber->audio = publisher->audio && subscriber->audio_offered;
-						subscriber->video = publisher->video && subscriber->video_offered;
-						subscriber->data = publisher->data && subscriber->data_offered;
-						/* Done */
+					if(!g_atomic_int_get(&subscriber->answered)) {
+						/* We're still waiting for an answer to a previous offer, postpone this */
+						g_atomic_int_set(&subscriber->pending_offer, 1);
+						g_atomic_int_set(&subscriber->pending_restart, 1);
+						janus_mutex_unlock(&subscriber->streams_mutex);
+						JANUS_LOG(LOG_VERB, "Post-poning new ICE restart offer, waiting for previous answer\n");
 						janus_videoroom_message_free(msg);
 						continue;
 					}
+					json_t *jsep = janus_videoroom_subscriber_offer(subscriber);
+					janus_mutex_unlock(&subscriber->streams_mutex);
+					if(do_restart)
+						json_object_set_new(jsep, "restart", json_true());
+					/* How long will the Janus core take to push the event? */
+					gint64 start = janus_get_monotonic_time();
+					int res = gateway->push_event(msg->handle, &janus_videoroom_plugin, msg->transaction, event, jsep);
+					JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
+					json_decref(event);
+					json_decref(jsep);
+					/* Done */
+					janus_videoroom_message_free(msg);
+					continue;
 				}
+				janus_mutex_unlock(&subscriber->streams_mutex);
 			} else if(!strcasecmp(request_text, "pause")) {
 				/* Stop receiving the publisher streams for a while */
 				subscriber->paused = TRUE;
@@ -5996,17 +7619,10 @@ static void *janus_videoroom_handler(void *data) {
 				json_object_set_new(event, "paused", json_string("ok"));
 			} else if(!strcasecmp(request_text, "switch")) {
 				/* This subscriber wants to switch to a different publisher */
-				JANUS_VALIDATE_JSON_OBJECT(root, subscriber_parameters,
+				JANUS_VALIDATE_JSON_OBJECT(root, switch_parameters,
 					error_code, error_cause, TRUE,
 					JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
-				if(error_code != 0)
-					goto error;
-				json_t *feed = json_object_get(root, "feed");
-				guint64 feed_id = json_integer_value(feed);
-				json_t *audio = json_object_get(root, "audio");
-				json_t *video = json_object_get(root, "video");
-				json_t *data = json_object_get(root, "data");
-				if(!subscriber->room) {
+				if(!subscriber->room || g_atomic_int_get(&subscriber->room->destroyed)) {
 					JANUS_LOG(LOG_ERR, "Room Destroyed \n");
 					error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_ROOM;
 					g_snprintf(error_cause, 512, "No such room ");
@@ -6018,81 +7634,281 @@ static void *janus_videoroom_handler(void *data) {
 					g_snprintf(error_cause, 512, "No such room (%"SCNu64")", subscriber->room_id);
 					goto error;
 				}
-				janus_mutex_lock(&subscriber->room->mutex);
-				janus_videoroom_publisher *publisher = g_hash_table_lookup(subscriber->room->participants, &feed_id);
-				if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || publisher->sdp == NULL) {
-					JANUS_LOG(LOG_ERR, "No such feed (%"SCNu64")\n", feed_id);
-					error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
-					g_snprintf(error_cause, 512, "No such feed (%"SCNu64")", feed_id);
+				/* While the legacy way of switching by just providing a feed ID is
+				 * still supported (at least for now), it isn't flexible enough: the
+				 * new proper way of doing that is providing the list of changes that
+				 * need to be done, in terms of which stream to switch to, and which
+				 * subscription mid to attach it to. This allows for partial switches
+				 * (e.g., change the second video source to Bob's camera), while the
+				 * old approach simply forces a single publisher as the new source. */
+				json_t *feeds = json_object_get(root, "streams");
+				json_t *feed = json_object_get(root, "feed");
+				GList *publishers = NULL;
+				if(feeds == NULL || json_array_size(feeds) == 0) {
+					/* For backwards compatibility, we still support the old "feed" property, which means
+					 * "switch to all the feeds from this publisher" (much less sophisticated, though) */
+					guint64 feed_id = json_integer_value(feed);
+					if(feed_id == 0) {
+						JANUS_LOG(LOG_ERR, "At least one between 'streams' and 'feed' must be specified\n");
+						error_code = JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT;
+						g_snprintf(error_cause, 512, "At least one between 'streams' and 'feed' must be specified");
+						goto error;
+					}
+					janus_mutex_lock(&subscriber->room->mutex);
+					janus_videoroom_publisher *publisher = g_hash_table_lookup(subscriber->room->participants, &feed_id);
 					janus_mutex_unlock(&subscriber->room->mutex);
+					if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || !publisher->session->started) {
+						JANUS_LOG(LOG_ERR, "No such feed (%"SCNu64")\n", feed_id);
+						error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+						g_snprintf(error_cause, 512, "No such feed (%"SCNu64")", feed_id);
+						goto error;
+					}
+					/* Create a fake "streams" list out of this publisher */
+					feeds = json_array();
+					json_object_set_new(root, "streams", feeds);
+					janus_refcount_increase(&publisher->ref);
+					GList *temp = publisher->streams, *touched_already = NULL;
+					while(temp) {
+						janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+						/* Look for a subscriber stream compatible with this publisher stream */
+						janus_videoroom_subscriber_stream *stream = NULL;
+						GList *temp2 = subscriber->streams;
+						while(temp2) {
+							stream = (janus_videoroom_subscriber_stream *)temp->data;
+							if(stream->type == ps->type && !g_list_find(touched_already, stream) &&
+									((stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO && stream->acodec == ps->acodec) ||
+									(stream->type == JANUS_VIDEOROOM_MEDIA_VIDEO && stream->vcodec == ps->vcodec))) {
+								/* This streams looks right */
+								touched_already = g_list_append(touched_already, stream);
+								json_t *s = json_object();
+								json_object_set_new(s, "feed", json_integer(publisher->user_id));
+								json_object_set_new(s, "mid", json_string(ps->mid));
+								json_object_set_new(s, "sub_mid", json_string(stream->mid));
+								json_array_append_new(feeds, s);
+							} else {
+								JANUS_LOG(LOG_WARN, "Skipping %"SCNu64" stream '%s' legacy switch: no compliant subscriber stream\n",
+									publisher->user_id, ps->mid);
+							}
+							temp2 = temp2->next;
+						}
+						temp = temp->next;
+					}
+					g_list_free(touched_already);
+					janus_refcount_decrease(&publisher->ref);
+					/* Take note of the fact this is a legacy request */
+					JANUS_LOG(LOG_WARN, "Legacy 'switch' request: please start using the streams array instead\n");
+				}
+				/* If we got here, we have a feeds list: make sure we have everything we need */
+				if(json_array_size(feeds) == 0) {
+					JANUS_LOG(LOG_ERR, "Empty switch list\n");
+					error_code = JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT;
+					g_snprintf(error_cause, 512, "Empty switch list");
 					goto error;
 				}
-				janus_refcount_increase(&publisher->ref);
-				janus_refcount_increase(&publisher->session->ref);
-				janus_mutex_unlock(&subscriber->room->mutex);
+				/* Make sure all the feeds we're subscribing to exist */
+				size_t i = 0;
+				for(i=0; i<json_array_size(feeds); i++) {
+					json_t *s = json_array_get(feeds, i);
+					JANUS_VALIDATE_JSON_OBJECT(s, switch_update_parameters,
+						error_code, error_cause, TRUE,
+						JANUS_VIDEOROOM_ERROR_MISSING_ELEMENT, JANUS_VIDEOROOM_ERROR_INVALID_ELEMENT);
+					if(error_code != 0) {
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
+						}
+						goto error;
+					}
+					json_t *feed = json_object_get(s, "feed");
+					guint64 feed_id = json_integer_value(feed);
+					janus_mutex_lock(&subscriber->room->mutex);
+					janus_videoroom_publisher *publisher = g_hash_table_lookup(subscriber->room->participants, &feed_id);
+					janus_mutex_unlock(&subscriber->room->mutex);
+					if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || !publisher->session->started) {
+						JANUS_LOG(LOG_ERR, "No such feed (%"SCNu64")\n", feed_id);
+						error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+						g_snprintf(error_cause, 512, "No such feed (%"SCNu64")", feed_id);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
+						}
+						goto error;
+					}
+					const char *mid = json_string_value(json_object_get(s, "mid"));
+					/* Check the mid too */
+					janus_mutex_lock(&publisher->streams_mutex);
+					if(g_hash_table_lookup(publisher->streams_bymid, mid) == NULL) {
+						janus_mutex_unlock(&publisher->streams_mutex);
+						JANUS_LOG(LOG_ERR, "No such mid '%s' in feed (%"SCNu64")\n", mid, feed_id);
+						error_code = JANUS_VIDEOROOM_ERROR_NO_SUCH_FEED;
+						g_snprintf(error_cause, 512, "No such mid '%s' in feed (%"SCNu64")", mid, feed_id);
+						/* Unref publishers we may have taken note of so far */
+						while(publishers) {
+							publisher = (janus_videoroom_publisher *)publishers->data;
+							janus_refcount_decrease(&publisher->ref);
+							janus_refcount_decrease(&publisher->session->ref);
+							publishers = g_list_remove(publishers, publisher);
+						}
+						goto error;
+					}
+					janus_mutex_unlock(&publisher->streams_mutex);
+					/* Increase the refcount before unlocking so that nobody can remove and free the publisher in the meantime. */
+					janus_refcount_increase(&publisher->ref);
+					janus_refcount_increase(&publisher->session->ref);
+					publishers = g_list_append(publishers, publisher);
+				}
 				gboolean paused = subscriber->paused;
 				subscriber->paused = TRUE;
-				/* Unsubscribe from the previous publisher */
-				janus_videoroom_publisher *prev_feed = subscriber->feed;
-				if(prev_feed) {
-					/* ... but make sure the codecs are compliant first */
-					if(publisher->acodec != prev_feed->acodec || publisher->vcodec != prev_feed->vcodec) {
-						janus_refcount_decrease(&publisher->session->ref);
-						janus_refcount_decrease(&publisher->ref);
-						subscriber->paused = paused;
-						JANUS_LOG(LOG_ERR, "The two publishers are not using the same codecs, can't switch\n");
-						error_code = JANUS_VIDEOROOM_ERROR_INVALID_SDP;
-						g_snprintf(error_cause, 512, "The two publishers are not using the same codecs, can't switch");
-						goto error;
+				/* Switch to the new streams, unsubscribing from the ones we replace:
+				 * notice that no renegotiation happens, we just switch the sources */
+				int changes = 0;
+				gboolean update = FALSE;
+				janus_mutex_lock(&subscriber->streams_mutex);
+				for(i=0; i<json_array_size(feeds); i++) {
+					json_t *s = json_array_get(feeds, i);
+					/* Look for the specific subscription mid to update */
+					const char *sub_mid = json_string_value(json_object_get(s, "sub_mid"));
+					janus_videoroom_subscriber_stream *stream = g_hash_table_lookup(subscriber->streams_bymid, sub_mid);
+					if(stream == NULL) {
+						JANUS_LOG(LOG_WARN, "Subscriber stream with mid '%s' not found, not switching...\n", sub_mid);
+						continue;
 					}
-					/* Go on */
-					janus_mutex_lock(&prev_feed->subscribers_mutex);
-					prev_feed->subscribers = g_slist_remove(prev_feed->subscribers, subscriber);
-					janus_mutex_unlock(&prev_feed->subscribers_mutex);
-					janus_refcount_decrease(&prev_feed->session->ref);
-					g_clear_pointer(&subscriber->feed, janus_videoroom_publisher_dereference);
-				}
-				/* Subscribe to the new one */
-				subscriber->audio = audio ? json_is_true(audio) : TRUE;	/* True by default */
-				if(!publisher->audio)
-					subscriber->audio = FALSE;	/* ... unless the publisher isn't sending any audio */
-				subscriber->video = video ? json_is_true(video) : TRUE;	/* True by default */
-				if(!publisher->video)
-					subscriber->video = FALSE;	/* ... unless the publisher isn't sending any video */
-				subscriber->data = data ? json_is_true(data) : TRUE;	/* True by default */
-				if(!publisher->data)
-					subscriber->data = FALSE;	/* ... unless the publisher isn't sending any data */
-				if(subscriber->room && subscriber->room->do_svc) {
-					/* This subscriber belongs to a room where VP9 SVC has been enabled,
-					 * let's assume we're interested in all layers for the time being */
-					subscriber->spatial_layer = -1;
-					subscriber->target_spatial_layer = 2;		/* FIXME Chrome sends 0, 1 and 2 (if using EnabledByFlag_3SL3TL) */
-					subscriber->temporal_layer = -1;
-					subscriber->target_temporal_layer = 2;	/* FIXME Chrome sends 0, 1 and 2 */
-				}
-				janus_mutex_lock(&publisher->subscribers_mutex);
-				publisher->subscribers = g_slist_append(publisher->subscribers, subscriber);
-				janus_mutex_unlock(&publisher->subscribers_mutex);
-				subscriber->feed = publisher;
-				/* Send a FIR to the new publisher */
-				janus_videoroom_reqfir(publisher, "Switching existing subscriber to new publisher");
+					/* Look for the publisher stream to switch to */
+					json_t *feed = json_object_get(s, "feed");
+					guint64 feed_id = json_integer_value(feed);
+					const char *mid = json_string_value(json_object_get(s, "mid"));
+					janus_mutex_lock(&subscriber->room->mutex);
+					janus_videoroom_publisher *publisher = g_hash_table_lookup(subscriber->room->participants, &feed_id);
+					janus_mutex_unlock(&subscriber->room->mutex);
+					if(publisher == NULL || g_atomic_int_get(&publisher->destroyed) || !publisher->session->started) {
+						JANUS_LOG(LOG_WARN, "Publisher '%"SCNu64"' not found, not switching...\n", feed_id);
+						continue;
+					}
+					janus_mutex_lock(&publisher->streams_mutex);
+					janus_videoroom_publisher_stream *ps = g_hash_table_lookup(publisher->streams_bymid, mid);
+					janus_mutex_unlock(&publisher->streams_mutex);
+					if(ps == NULL || g_atomic_int_get(&ps->destroyed)) {
+						JANUS_LOG(LOG_WARN, "Publisher '%"SCNu64"' doesn't have any mid '%s', not switching...\n", feed_id, mid);
+						continue;
+					}
+					/* If this mapping already exists, do nothing */
+					if(g_slist_find(stream->publisher_streams, ps) != NULL) {
+						JANUS_LOG(LOG_WARN, "Publisher '%"SCNu64"'/'%s' is already feeding mid '%s', not switching...\n",
+							feed_id, mid, sub_mid);
+						continue;
+					}
+					/* If the streams are not of the same type, do nothing */
+					if(stream->type != ps->type) {
+						JANUS_LOG(LOG_WARN, "Publisher '%"SCNu64"'/'%s' is not the same type as subscription mid '%s', not switching...\n",
+							feed_id, mid, sub_mid);
+						continue;
+					}
+					/* If the streams are not using the same codec, do nothing */
+					if((stream->type == JANUS_VIDEOROOM_MEDIA_AUDIO && stream->acodec != ps->acodec) ||
+							(stream->type == JANUS_VIDEOROOM_MEDIA_VIDEO && stream->vcodec != ps->vcodec)) {
+						JANUS_LOG(LOG_WARN, "Publisher '%"SCNu64"'/'%s' is not using same codec as subscription mid '%s', not switching...\n",
+							feed_id, mid, sub_mid);
+						continue;
+					}
+					/* Unsubscribe the old stream and update it: we don't replace streams like we
+					 * do when doing new subscriptions, as that might change payload type, etc. */
+					changes++;
+					/* Unsubscribe from the previous source first */
+					janus_refcount_increase(&stream->ref);
+					gboolean unref = FALSE;
+					if(stream->publisher_streams == NULL) {
+						/* This stream was inactive, we'll need a renegotiation */
+						update = TRUE;
+					} else {
+						unref = TRUE;
+						janus_videoroom_publisher_stream *stream_ps = stream->publisher_streams->data;
+						janus_mutex_lock(&stream_ps->subscribers_mutex);
+						stream_ps->subscribers = g_slist_remove(stream_ps->subscribers, stream);
+						stream->publisher_streams = g_slist_remove(stream->publisher_streams, stream_ps);
+						janus_mutex_unlock(&stream_ps->subscribers_mutex);
+						janus_refcount_decrease(&stream_ps->ref);
+					}
+					/* Subscribe to the new one */
+					janus_mutex_lock(&ps->subscribers_mutex);
+					stream->publisher_streams = g_slist_append(stream->publisher_streams, ps);
+					ps->subscribers = g_slist_append(ps->subscribers, stream);
+					janus_refcount_increase(&ps->ref);
+					janus_refcount_increase(&stream->ref);
+					janus_mutex_unlock(&ps->subscribers_mutex);
+					janus_videoroom_reqpli(ps, "Subscriber switch");
+					if(unref)
+						janus_refcount_decrease(&stream->ref);
+					janus_refcount_decrease(&stream->ref);
+				}
+				janus_mutex_unlock(&subscriber->streams_mutex);
+				/* Decrease the references we took before */
+				while(publishers) {
+					janus_videoroom_publisher *publisher = (janus_videoroom_publisher *)publishers->data;
+					janus_refcount_decrease(&publisher->ref);
+					janus_refcount_decrease(&publisher->session->ref);
+					publishers = g_list_remove(publishers, publisher);
+				}
 				/* Done */
 				subscriber->paused = paused;
 				event = json_object();
 				json_object_set_new(event, "videoroom", json_string("event"));
 				json_object_set_new(event, "switched", json_string("ok"));
 				json_object_set_new(event, "room", json_integer(subscriber->room_id));
-				json_object_set_new(event, "id", json_integer(feed_id));
-				if(publisher->display)
-					json_object_set_new(event, "display", json_string(publisher->display));
+				json_object_set_new(event, "changes", json_integer(changes));
+				json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+				json_object_set_new(event, "streams", media);
 				/* Also notify event handlers */
 				if(notify_events && gateway->events_is_enabled()) {
 					json_t *info = json_object();
 					json_object_set_new(info, "event", json_string("switched"));
-					json_object_set_new(info, "room", json_integer(publisher->room_id));
-					json_object_set_new(info, "feed", json_integer(publisher->user_id));
+					json_object_set_new(info, "room", json_integer(subscriber->room_id));
+					json_object_set_new(event, "changes", json_integer(changes));
+					media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+					json_object_set_new(event, "streams", media);
 					gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
 				}
+				/* Check if we need a renegotiation as well */
+				if(update) {
+					/* We do */
+					janus_mutex_lock(&subscriber->streams_mutex);
+					if(!g_atomic_int_get(&subscriber->answered)) {
+						/* We're still waiting for an answer to a previous offer, postpone this */
+						g_atomic_int_set(&subscriber->pending_offer, 1);
+						janus_mutex_unlock(&subscriber->streams_mutex);
+						JANUS_LOG(LOG_VERB, "Post-poning new offer, waiting for previous answer\n");
+					} else {
+						json_t *revent = json_object();
+						json_object_set_new(revent, "videoroom", json_string("updated"));
+						json_object_set_new(revent, "room", json_integer(subscriber->room_id));
+						json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+						json_object_set_new(revent, "streams", media);
+						/* Generate a new offer */
+						json_t *jsep = janus_videoroom_subscriber_offer(subscriber);
+						janus_mutex_unlock(&subscriber->streams_mutex);
+						/* How long will the Janus core take to push the event? */
+						gint64 start = janus_get_monotonic_time();
+						int res = gateway->push_event(msg->handle, &janus_videoroom_plugin, msg->transaction, revent, jsep);
+						JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
+						json_decref(revent);
+						json_decref(jsep);
+						/* Also notify event handlers */
+						if(notify_events && gateway->events_is_enabled()) {
+							json_t *info = json_object();
+							json_object_set_new(info, "event", json_string("updated"));
+							json_object_set_new(info, "room", json_integer(subscriber->room_id));
+							json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+							json_object_set_new(info, "streams", media);
+							json_object_set_new(info, "private_id", json_integer(subscriber->pvt_id));
+							gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
+						}
+					}
+				}
 			} else if(!strcasecmp(request_text, "leave")) {
 				guint64 room_id = subscriber ? subscriber->room_id : 0;
 				/* Tell the core to tear down the PeerConnection, hangup_media will do the rest */
@@ -6145,7 +7961,58 @@ static void *janus_videoroom_handler(void *data) {
 				int ret = gateway->push_event(msg->handle, &janus_videoroom_plugin, msg->transaction, event, NULL);
 				JANUS_LOG(LOG_VERB, "  >> %d (%s)\n", ret, janus_get_api_error(ret));
 				json_decref(event);
+				/* Take note of the fact we got our answer */
+				janus_videoroom_subscriber *subscriber = (janus_videoroom_subscriber *)session->participant;
+				janus_mutex_lock(&subscriber->streams_mutex);
+				/* Mark all streams that were answered to as ready */
+				char error_str[512];
+				janus_sdp *answer = janus_sdp_parse(msg_sdp, error_str, sizeof(error_str));
+				GList *temp = answer->m_lines;
+				while(temp) {
+					janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+					if(m->direction != JANUS_SDP_INACTIVE) {
+						janus_videoroom_subscriber_stream *stream = g_hash_table_lookup(subscriber->streams_byid, GINT_TO_POINTER(m->index));
+						if(stream)
+							g_atomic_int_set(&stream->ready, 1);
+					}
+					temp = temp->next;
+				}
+				janus_sdp_destroy(answer);
 				janus_videoroom_message_free(msg);
+				/* Check if we have other pending offers to send for this subscriber */
+				if(g_atomic_int_compare_and_exchange(&subscriber->pending_offer, 1, 0)) {
+					JANUS_LOG(LOG_VERB, "Pending offer, sending it now\n");
+					event = json_object();
+					json_object_set_new(event, "videoroom", json_string("updated"));
+					json_object_set_new(event, "room", json_integer(subscriber->room_id));
+					json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+					json_object_set_new(event, "streams", media);
+					/* Generate a new offer */
+					json_t *jsep = janus_videoroom_subscriber_offer(subscriber);
+					/* Do we need an ICE restart as well? */
+					if(g_atomic_int_compare_and_exchange(&subscriber->pending_restart, 1, 0))
+						json_object_set_new(jsep, "restart", json_true());
+					janus_mutex_unlock(&subscriber->streams_mutex);
+					/* How long will the Janus core take to push the event? */
+					gint64 start = janus_get_monotonic_time();
+					int res = gateway->push_event(session->handle, &janus_videoroom_plugin, NULL, event, jsep);
+					JANUS_LOG(LOG_VERB, "  >> Pushing event: %d (took %"SCNu64" us)\n", res, janus_get_monotonic_time()-start);
+					json_decref(event);
+					json_decref(jsep);
+					/* Also notify event handlers */
+					if(notify_events && gateway->events_is_enabled()) {
+						json_t *info = json_object();
+						json_object_set_new(info, "event", json_string("updated"));
+						json_object_set_new(info, "room", json_integer(subscriber->room_id));
+						json_t *media = janus_videoroom_subscriber_streams_summary(subscriber, FALSE, NULL);
+						json_object_set_new(info, "streams", media);
+						json_object_set_new(info, "private_id", json_integer(subscriber->pvt_id));
+						gateway->notify_event(&janus_videoroom_plugin, session->handle, info);
+					}
+				} else {
+					g_atomic_int_set(&subscriber->answered, 1);
+					janus_mutex_unlock(&subscriber->streams_mutex);
+				}
 				continue;
 			} else {
 				/* TODO We don't support anything else right now... */
@@ -6179,14 +8046,11 @@ static void *janus_videoroom_handler(void *data) {
 				g_hash_table_iter_init(&iter, videoroom->participants);
 				while (!g_atomic_int_get(&videoroom->destroyed) && g_hash_table_iter_next(&iter, NULL, &value)) {
 					janus_videoroom_publisher *p = value;
-					if(p != participant && p->sdp)
+					if(p != participant && p->session->started)
 						count++;
 				}
 				janus_mutex_unlock(&videoroom->mutex);
 				if(count == videoroom->max_publishers) {
-					participant->audio_active = FALSE;
-					participant->video_active = FALSE;
-					participant->data_active = FALSE;
 					JANUS_LOG(LOG_ERR, "Maximum number of publishers (%d) already reached\n", videoroom->max_publishers);
 					error_code = JANUS_VIDEOROOM_ERROR_PUBLISHERS_FULL;
 					g_snprintf(error_cause, 512, "Maximum number of publishers (%d) already reached", videoroom->max_publishers);
@@ -6206,200 +8070,234 @@ static void *janus_videoroom_handler(void *data) {
 					g_snprintf(error_cause, 512, "Error parsing offer: %s", error_str);
 					goto error;
 				}
+				/* Prepare an answer, by iterating on all m-lines */
+				janus_sdp *answer = janus_sdp_generate_answer(offer);
+				json_t *media = json_array();
+				json_t *descriptions = json_object_get(root, "descriptions");
+				const char *audiocodec = NULL, *videocodec = NULL;
 				GList *temp = offer->m_lines;
 				while(temp) {
 					/* Which media are available? */
 					janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
-					if(m->type == JANUS_SDP_AUDIO && m->port > 0 &&
-							m->direction != JANUS_SDP_RECVONLY && m->direction != JANUS_SDP_INACTIVE) {
-						participant->audio = TRUE;
-					} else if(m->type == JANUS_SDP_VIDEO && m->port > 0 &&
-							m->direction != JANUS_SDP_RECVONLY && m->direction != JANUS_SDP_INACTIVE) {
-						participant->video = TRUE;
-					} else if(m->type == JANUS_SDP_APPLICATION && m->port > 0) {
-						participant->data = TRUE;
-					}
+					/* Initialize a new publisher stream */
+					janus_videoroom_publisher_stream *ps = g_malloc0(sizeof(janus_videoroom_publisher_stream));
+					ps->type = JANUS_VIDEOROOM_MEDIA_NONE;
+					if(m->type == JANUS_SDP_AUDIO)
+						ps->type = JANUS_VIDEOROOM_MEDIA_AUDIO;
+					else if(m->type == JANUS_SDP_VIDEO)
+						ps->type = JANUS_VIDEOROOM_MEDIA_VIDEO;
+					if(m->type == JANUS_SDP_APPLICATION)
+						ps->type = JANUS_VIDEOROOM_MEDIA_DATA;
+					ps->mindex = g_list_length(participant->streams);
+					ps->publisher = participant;
+					janus_refcount_increase(&participant->ref);	/* Add a reference to the publisher */
+					/* Initialize the stream */
+					ps->active = TRUE;
+					g_atomic_int_set(&ps->destroyed, 0);
+					janus_refcount_init(&ps->ref, janus_videoroom_publisher_stream_free);
+					janus_refcount_increase(&ps->ref);	/* This is for the mid-indexed hashtable */
+					janus_mutex_init(&ps->subscribers_mutex);
+					janus_mutex_init(&ps->rtp_forwarders_mutex);
+					ps->rtp_forwarders = g_hash_table_new_full(NULL, NULL, NULL, (GDestroyNotify)janus_videoroom_rtp_forwarder_destroy);
 					if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
 						/* Are the extmaps we care about there? */
 						GList *ma = m->attributes;
 						while(ma) {
 							janus_sdp_attribute *a = (janus_sdp_attribute *)ma->data;
-							if(a->value) {
-								if(videoroom->audiolevel_ext && m->type == JANUS_SDP_AUDIO && strstr(a->value, JANUS_RTP_EXTMAP_AUDIO_LEVEL)) {
-									participant->audio_level_extmap_id = atoi(a->value);
+							if(a->name && a->value) {
+								if(ps->mid == NULL && !strcasecmp(a->name, "mid")) {
+									ps->mid = g_strdup(a->value);
+								} else if(videoroom->audiolevel_ext && m->type == JANUS_SDP_AUDIO && strstr(a->value, JANUS_RTP_EXTMAP_AUDIO_LEVEL)) {
+									ps->audio_level_extmap_id = atoi(a->value);
 								} else if(videoroom->videoorient_ext && m->type == JANUS_SDP_VIDEO && strstr(a->value, JANUS_RTP_EXTMAP_VIDEO_ORIENTATION)) {
-									participant->video_orient_extmap_id = atoi(a->value);
+									ps->video_orient_extmap_id = atoi(a->value);
 								} else if(videoroom->playoutdelay_ext && m->type == JANUS_SDP_VIDEO && strstr(a->value, JANUS_RTP_EXTMAP_PLAYOUT_DELAY)) {
-									participant->playout_delay_extmap_id = atoi(a->value);
-								} else if(m->type == JANUS_SDP_AUDIO && !strcasecmp(a->name, "fmtp") && strstr(a->value, "useinbandfec=1")) {
-									participant->do_opusfec = videoroom->do_opusfec;
+									ps->playout_delay_extmap_id = atoi(a->value);
+								} else if(videoroom->do_opusfec && m->type == JANUS_SDP_AUDIO && !strcasecmp(a->name, "fmtp") && strstr(a->value, "useinbandfec=1")) {
+									ps->opusfec = TRUE;
 								}
 							}
 							ma = ma->next;
 						}
 					}
-					temp = temp->next;
-				}
-				/* Prepare an answer now: force the room codecs and recvonly on the Janus side */
-				JANUS_LOG(LOG_VERB, "The publisher %s going to send an audio stream\n", participant->audio ? "is" : "is NOT");
-				JANUS_LOG(LOG_VERB, "The publisher %s going to send a video stream\n", participant->video ? "is" : "is NOT");
-				JANUS_LOG(LOG_VERB, "The publisher %s going to open a data channel\n", participant->data ? "is" : "is NOT");
-				/* Check the codecs we can use, or the ones we should */
-				if(participant->acodec == JANUS_AUDIOCODEC_NONE) {
-					int i=0;
-					for(i=0; i<3; i++) {
-						if(videoroom->acodec[i] == JANUS_AUDIOCODEC_NONE)
-							continue;
-						if(janus_sdp_get_codec_pt(offer, janus_audiocodec_name(videoroom->acodec[i])) != -1) {
-							participant->acodec = videoroom->acodec[i];
-							break;
+					/* Check the codecs we can use, or the ones we should */
+					ps->acodec = JANUS_AUDIOCODEC_NONE;
+					ps->vcodec = JANUS_VIDEOCODEC_NONE;
+					ps->pt = -1;
+					if(m->type == JANUS_SDP_AUDIO) {
+						int i=0;
+						for(i=0; i<3; i++) {
+							if(videoroom->acodec[i] == JANUS_AUDIOCODEC_NONE)
+								continue;
+							if(janus_sdp_get_codec_pt(offer, m->index, janus_audiocodec_name(videoroom->acodec[i])) != -1) {
+								ps->acodec = videoroom->acodec[i];
+								ps->pt = janus_audiocodec_pt(ps->acodec);
+								break;
+							}
+						}
+					} else if(m->type == JANUS_SDP_VIDEO) {
+						int i=0;
+						for(i=0; i<3; i++) {
+							if(videoroom->vcodec[i] == JANUS_VIDEOCODEC_NONE)
+								continue;
+							if(janus_sdp_get_codec_pt(offer, m->index, janus_videocodec_name(videoroom->vcodec[i])) != -1) {
+								ps->vcodec = videoroom->vcodec[i];
+								ps->pt = janus_videocodec_pt(ps->vcodec);
+								break;
+							}
+						}
+						/* Check if simulcast is in place */
+						if(msg_simulcast != NULL && json_array_size(msg_simulcast) > 0 &&
+								(ps->vcodec == JANUS_VIDEOCODEC_VP8 || ps->vcodec == JANUS_VIDEOCODEC_H264)) {
+							size_t i = 0;
+							for(i=0; i<json_array_size(msg_simulcast); i++) {
+								json_t *s = json_array_get(msg_simulcast, i);
+								int mindex = json_integer_value(json_object_get(s, "mindex"));
+								if(mindex != ps->mindex)
+									continue;
+								JANUS_LOG(LOG_WARN, "Publisher stream is going to do simulcasting (#%d, %s)\n", ps->mindex, ps->mid);
+								ps->simulcast = TRUE;
+								janus_rtp_simulcasting_prepare(s,
+									&ps->rid_extmap_id,
+									&ps->framemarking_ext_id,
+									ps->vssrc, ps->rid);
+							}
 						}
 					}
-				}
-				JANUS_LOG(LOG_VERB, "The publisher is going to use the %s audio codec\n", janus_audiocodec_name(participant->acodec));
-				participant->audio_pt = janus_audiocodec_pt(participant->acodec);
-				if(participant->vcodec == JANUS_VIDEOCODEC_NONE) {
-					int i=0;
-					for(i=0; i<3; i++) {
-						if(videoroom->vcodec[i] == JANUS_VIDEOCODEC_NONE)
-							continue;
-						if(janus_sdp_get_codec_pt(offer, janus_videocodec_name(videoroom->vcodec[i])) != -1) {
-							participant->vcodec = videoroom->vcodec[i];
-							break;
+					/* Add a new m-line to the answer */
+					if(m->type == JANUS_SDP_AUDIO) {
+						janus_sdp_generate_answer_mline(offer, answer, m,
+							JANUS_SDP_OA_MLINE, JANUS_SDP_AUDIO,
+								JANUS_SDP_OA_DIRECTION, ps->acodec != JANUS_AUDIOCODEC_NONE ? JANUS_SDP_RECVONLY : JANUS_SDP_INACTIVE,
+								JANUS_SDP_OA_CODEC, janus_audiocodec_name(ps->acodec),
+								JANUS_SDP_OA_FMTP, ps->opusfec ? "useinbandfec=1" : NULL,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_RID,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_REPAIRED_RID,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->audiolevel_ext ? JANUS_RTP_EXTMAP_AUDIO_LEVEL : NULL,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->videoorient_ext ? JANUS_RTP_EXTMAP_VIDEO_ORIENTATION : NULL,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->playoutdelay_ext ? JANUS_RTP_EXTMAP_PLAYOUT_DELAY : NULL,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->transport_wide_cc_ext ? JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC : NULL,
+							JANUS_SDP_OA_DONE);
+						janus_sdp_mline *m_answer = janus_sdp_mline_find_by_index(answer, m->index);
+						if(m_answer != NULL) {
+							/* TODO Remove, this is just here for backwards compatibility */
+							if(audiocodec == NULL)
+								audiocodec = janus_audiocodec_name(ps->acodec);
+						}
+					} else if(m->type == JANUS_SDP_VIDEO) {
+						janus_sdp_generate_answer_mline(offer, answer, m,
+							JANUS_SDP_OA_MLINE, JANUS_SDP_VIDEO,
+								JANUS_SDP_OA_DIRECTION, ps->vcodec != JANUS_VIDEOCODEC_NONE ? JANUS_SDP_RECVONLY : JANUS_SDP_INACTIVE,
+								JANUS_SDP_OA_CODEC, janus_videocodec_name(ps->vcodec),
+								JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_RID,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_REPAIRED_RID,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_FRAME_MARKING,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->audiolevel_ext ? JANUS_RTP_EXTMAP_AUDIO_LEVEL : NULL,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->videoorient_ext ? JANUS_RTP_EXTMAP_VIDEO_ORIENTATION : NULL,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->playoutdelay_ext ? JANUS_RTP_EXTMAP_PLAYOUT_DELAY : NULL,
+								JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->transport_wide_cc_ext ? JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC : NULL,
+							JANUS_SDP_OA_DONE);
+						janus_sdp_mline *m_answer = janus_sdp_mline_find_by_index(answer, m->index);
+						if(m_answer != NULL) {
+							/* TODO Remove, this is just here for backwards compatibility */
+							if(videocodec == NULL)
+								videocodec = janus_videocodec_name(ps->vcodec);
+							/* Also add a bandwidth SDP attribute if we're capping the bitrate in the room */
+							if(videoroom->bitrate > 0 && videoroom->bitrate_cap) {
+								if(participant->firefox) {
+									/* Use TIAS (bps) instead of AS (kbps) for the b= attribute, as explained here:
+									 * https://github.com/meetecho/janus-gateway/issues/1277#issuecomment-397677746 */
+									m->b_name = g_strdup("TIAS");
+									m->b_value = videoroom->bitrate;
+								} else {
+									m->b_name = g_strdup("AS");
+									m->b_value = videoroom->bitrate/1000;
+								}
+							}
+						}
+					} else if(m->type == JANUS_SDP_APPLICATION) {
+						janus_sdp_generate_answer_mline(offer, answer, m,
+							JANUS_SDP_OA_MLINE, JANUS_SDP_APPLICATION,
+							JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
+							JANUS_SDP_OA_DONE);
+					}
+					/* Make sure we have a mid */
+					if(ps->mid == NULL) {
+						char mid[5];
+						g_snprintf(mid, sizeof(mid), "%d", ps->mindex);
+						ps->mid = g_strdup(mid);
+					}
+					/* Do we have a description as well? */
+					if(descriptions != NULL && json_array_size(descriptions) > 0) {
+						size_t i = 0;
+						for(i=0; i<json_array_size(descriptions); i++) {
+							json_t *d = json_array_get(descriptions, i);
+							const char *d_mid = json_string_value(json_object_get(d, "mid"));
+							const char *d_desc = json_string_value(json_object_get(d, "description"));
+							if(d_desc && d_mid && ps->mid && !strcasecmp(d_mid, ps->mid)) {
+								ps->description = g_strdup(d_desc);
+								break;
+							}
 						}
 					}
+					/* Add the stream to the list */
+					janus_mutex_lock(&participant->streams_mutex);
+					participant->streams = g_list_append(participant->streams, ps);
+					g_hash_table_insert(participant->streams_byid, GINT_TO_POINTER(ps->mindex), ps);
+					g_hash_table_insert(participant->streams_bymid, g_strdup(ps->mid), ps);
+					janus_mutex_unlock(&participant->streams_mutex);
+					temp = temp->next;
+					/* Add to the info we send back to the publisher */
+					json_t *info = json_object();
+					json_object_set_new(info, "type", json_string(janus_videoroom_media_str(ps->type)));
+					json_object_set_new(info, "mindex", json_integer(ps->mindex));
+					json_object_set_new(info, "mid", json_string(ps->mid));
+					if(ps->description)
+						json_object_set_new(info, "description", json_string(ps->description));
+					if(ps->type == JANUS_VIDEOROOM_MEDIA_AUDIO) {
+						json_object_set_new(info, "codec", json_string(janus_audiocodec_name(ps->acodec)));
+						if(ps->opusfec)
+							json_object_set_new(info, "opus-fec", json_true());
+					} else if(ps->type == JANUS_VIDEOROOM_MEDIA_VIDEO) {
+						json_object_set_new(info, "codec", json_string(janus_videocodec_name(ps->vcodec)));
+						if(ps->simulcast)
+							json_object_set_new(info, "simulcast", json_true());
+						if(ps->svc)
+							json_object_set_new(info, "svc", json_true());
+					}
+					json_array_append_new(media, info);
 				}
-				JANUS_LOG(LOG_VERB, "The publisher is going to use the %s video codec\n", janus_videocodec_name(participant->vcodec));
-				participant->video_pt = janus_videocodec_pt(participant->vcodec);
-				janus_sdp *answer = janus_sdp_generate_answer(offer,
-					JANUS_SDP_OA_AUDIO_CODEC, janus_audiocodec_name(participant->acodec),
-					JANUS_SDP_OA_AUDIO_DIRECTION, JANUS_SDP_RECVONLY,
-					JANUS_SDP_OA_AUDIO_FMTP, participant->do_opusfec ? "useinbandfec=1" : NULL,
-					JANUS_SDP_OA_VIDEO_CODEC, janus_videocodec_name(participant->vcodec),
-					JANUS_SDP_OA_VIDEO_DIRECTION, JANUS_SDP_RECVONLY,
-					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_MID,
-					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_RID,
-					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_REPAIRED_RID,
-					JANUS_SDP_OA_ACCEPT_EXTMAP, JANUS_RTP_EXTMAP_FRAME_MARKING,
-					JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->audiolevel_ext ? JANUS_RTP_EXTMAP_AUDIO_LEVEL : NULL,
-					JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->videoorient_ext ? JANUS_RTP_EXTMAP_VIDEO_ORIENTATION : NULL,
-					JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->playoutdelay_ext ? JANUS_RTP_EXTMAP_PLAYOUT_DELAY : NULL,
-					JANUS_SDP_OA_ACCEPT_EXTMAP, videoroom->transport_wide_cc_ext ? JANUS_RTP_EXTMAP_TRANSPORT_WIDE_CC : NULL,
-					JANUS_SDP_OA_DONE);
 				janus_sdp_destroy(offer);
 				/* Replace the session name */
 				g_free(answer->s_name);
 				char s_name[100];
 				g_snprintf(s_name, sizeof(s_name), "VideoRoom %"SCNu64, videoroom->room_id);
 				answer->s_name = g_strdup(s_name);
-				/* Which media are REALLY available? (some may have been rejected) */
-				participant->audio = FALSE;
-				participant->video = FALSE;
-				participant->data = FALSE;
-				temp = answer->m_lines;
-				while(temp) {
-					janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
-					if(m->type == JANUS_SDP_AUDIO && m->port > 0 && m->direction != JANUS_SDP_INACTIVE) {
-						participant->audio = TRUE;
-					} else if(m->type == JANUS_SDP_VIDEO && m->port > 0 && m->direction != JANUS_SDP_INACTIVE) {
-						participant->video = TRUE;
-					} else if(m->type == JANUS_SDP_APPLICATION && m->port > 0) {
-						participant->data = TRUE;
-					}
-					temp = temp->next;
-				}
-				JANUS_LOG(LOG_VERB, "Per the answer, the publisher %s going to send an audio stream\n", participant->audio ? "is" : "is NOT");
-				JANUS_LOG(LOG_VERB, "Per the answer, the publisher %s going to send a video stream\n", participant->video ? "is" : "is NOT");
-				JANUS_LOG(LOG_VERB, "Per the answer, the publisher %s going to open a data channel\n", participant->data ? "is" : "is NOT");
-				/* Update the event with info on the codecs that we'll be handling */
-				if(event) {
-					if(participant->audio)
-						json_object_set_new(event, "audio_codec", json_string(janus_audiocodec_name(participant->acodec)));
-					if(participant->video)
-						json_object_set_new(event, "video_codec", json_string(janus_videocodec_name(participant->vcodec)));
-				}
-				/* Also add a bandwidth SDP attribute if we're capping the bitrate in the room */
-				janus_sdp_mline *m = janus_sdp_mline_find(answer, JANUS_SDP_VIDEO);
-				if(m != NULL && videoroom->bitrate > 0 && videoroom->bitrate_cap) {
-					if(participant->firefox) {
-						/* Use TIAS (bps) instead of AS (kbps) for the b= attribute, as explained here:
-						 * https://github.com/meetecho/janus-gateway/issues/1277#issuecomment-397677746 */
-						m->b_name = g_strdup("TIAS");
-						m->b_value = videoroom->bitrate;
-					} else {
-						m->b_name = g_strdup("AS");
-						m->b_value = videoroom->bitrate/1000;
-					}
-				}
 				/* Generate an SDP string we can send back to the publisher */
 				char *answer_sdp = janus_sdp_write(answer);
-				/* Now turn the SDP into what we'll send subscribers, using the static payload types for making switching easier */
-				offer = janus_sdp_generate_offer(s_name, answer->c_addr,
-					JANUS_SDP_OA_AUDIO, participant->audio,
-					JANUS_SDP_OA_AUDIO_CODEC, janus_audiocodec_name(participant->acodec),
-					JANUS_SDP_OA_AUDIO_PT, janus_audiocodec_pt(participant->acodec),
-					JANUS_SDP_OA_AUDIO_DIRECTION, JANUS_SDP_SENDONLY,
-					JANUS_SDP_OA_AUDIO_FMTP, participant->do_opusfec ? "useinbandfec=1" : NULL,
-					JANUS_SDP_OA_VIDEO, participant->video,
-					JANUS_SDP_OA_VIDEO_CODEC, janus_videocodec_name(participant->vcodec),
-					JANUS_SDP_OA_VIDEO_PT, janus_videocodec_pt(participant->vcodec),
-					JANUS_SDP_OA_VIDEO_DIRECTION, JANUS_SDP_SENDONLY,
-					JANUS_SDP_OA_DATA, participant->data,
-					JANUS_SDP_OA_DONE);
-				/* Add the extmap attributes, if needed */
-				if(participant->audio_level_extmap_id > 0) {
-					janus_sdp_mline *m = janus_sdp_mline_find(offer, JANUS_SDP_AUDIO);
-					if(m != NULL) {
-						janus_sdp_attribute *a = janus_sdp_attribute_create("extmap",
-							"%d %s\r\n", participant->audio_level_extmap_id, JANUS_RTP_EXTMAP_AUDIO_LEVEL);
-						janus_sdp_attribute_add_to_mline(m, a);
-					}
-				}
-				if(participant->video_orient_extmap_id > 0) {
-					janus_sdp_mline *m = janus_sdp_mline_find(offer, JANUS_SDP_VIDEO);
-					if(m != NULL) {
-						janus_sdp_attribute *a = janus_sdp_attribute_create("extmap",
-							"%d %s\r\n", participant->video_orient_extmap_id, JANUS_RTP_EXTMAP_VIDEO_ORIENTATION);
-						janus_sdp_attribute_add_to_mline(m, a);
-					}
-				}
-				if(participant->playout_delay_extmap_id > 0) {
-					janus_sdp_mline *m = janus_sdp_mline_find(offer, JANUS_SDP_VIDEO);
-					if(m != NULL) {
-						janus_sdp_attribute *a = janus_sdp_attribute_create("extmap",
-							"%d %s\r\n", participant->playout_delay_extmap_id, JANUS_RTP_EXTMAP_PLAYOUT_DELAY);
-						janus_sdp_attribute_add_to_mline(m, a);
-					}
+				janus_sdp_destroy(answer);
+				/* For backwards compatibility, update the event with info on the codecs that we'll be handling
+				 * TODO This will make no sense in the future, as different streams may use different codecs */
+				if(event) {
+					if(audiocodec)
+						json_object_set_new(event, "audio_codec", json_string(audiocodec));
+					if(videocodec)
+						json_object_set_new(event, "video_codec", json_string(videocodec));
 				}
+				json_object_set_new(event, "streams", media);
 				/* Is this room recorded, or are we recording this publisher already? */
 				janus_mutex_lock(&participant->rec_mutex);
 				if(videoroom->record || participant->recording_active) {
-					janus_videoroom_recorder_create(participant, participant->audio, participant->video, participant->data);
-				}
-				janus_mutex_unlock(&participant->rec_mutex);
-				/* Generate an SDP string we can offer subscribers later on */
-				char *offer_sdp = janus_sdp_write(offer);
-				if(!sdp_update) {
-					/* Is simulcasting involved */
-					if(msg_simulcast && (participant->vcodec == JANUS_VIDEOCODEC_VP8 ||
-							participant->vcodec == JANUS_VIDEOCODEC_H264)) {
-						JANUS_LOG(LOG_VERB, "Publisher is going to do simulcasting\n");
-						janus_rtp_simulcasting_prepare(msg_simulcast,
-							&participant->rid_extmap_id,
-							&participant->framemarking_ext_id,
-							participant->ssrc, participant->rid);
-					} else {
-						/* No simulcasting involved */
-						int i=0;
-						for(i=0; i<3; i++) {
-							participant->ssrc[i] = 0;
-							g_free(participant->rid[i]);
-							participant->rid[i] = NULL;
-						}
+					GList *temp = participant->streams;
+					while(temp) {
+						janus_videoroom_publisher_stream *ps = (janus_videoroom_publisher_stream *)temp->data;
+						janus_videoroom_recorder_create(ps);
+						temp = temp->next;
 					}
 				}
-				janus_sdp_destroy(offer);
-				janus_sdp_destroy(answer);
+				janus_mutex_unlock(&participant->rec_mutex);
 				/* Send the answer back to the publisher */
 				JANUS_LOG(LOG_VERB, "Handling publisher: turned this into an '%s':\n%s\n", type, answer_sdp);
 				json_t *jsep = json_pack("{ssss}", "type", type, "sdp", answer_sdp);
@@ -6412,38 +8310,9 @@ static void *janus_videoroom_handler(void *data) {
 				/* Done */
 				if(res != JANUS_OK) {
 					/* TODO Failed to negotiate? We should remove this publisher */
-					g_free(offer_sdp);
 				} else {
-					/* Store the participant's SDP for interested subscribers */
-					g_free(participant->sdp);
-					participant->sdp = offer_sdp;
 					/* We'll wait for the setup_media event before actually telling subscribers */
 				}
-				/* Unless this is an update, in which case schedule a new offer for all viewers */
-				if(sdp_update) {
-					json_t *update = json_object();
-					json_object_set_new(update, "request", json_string("configure"));
-					json_object_set_new(update, "update", json_true());
-					janus_mutex_lock(&participant->subscribers_mutex);
-					GSList *s = participant->subscribers;
-					while(s) {
-						janus_videoroom_subscriber *subscriber = (janus_videoroom_subscriber *)s->data;
-						if(subscriber && subscriber->session && subscriber->session->handle) {
-							/* Enqueue the fake request: this will trigger a renegotiation */
-							janus_videoroom_message *msg = g_malloc(sizeof(janus_videoroom_message));
-							janus_refcount_increase(&subscriber->session->ref);
-							msg->handle = subscriber->session->handle;
-							msg->message = update;
-							msg->transaction = NULL;
-							msg->jsep = NULL;
-							json_incref(update);
-							g_async_queue_push(messages, msg);
-						}
-						s = s->next;
-					}
-					janus_mutex_unlock(&participant->subscribers_mutex);
-					json_decref(update);
-				}
 				json_decref(event);
 				json_decref(jsep);
 			}
@@ -6478,114 +8347,106 @@ static void janus_videoroom_relay_rtp_packet(gpointer data, gpointer user_data)
 		JANUS_LOG(LOG_ERR, "Invalid packet...\n");
 		return;
 	}
-	janus_videoroom_subscriber *subscriber = (janus_videoroom_subscriber *)data;
-	if(!subscriber || !subscriber->session) {
-		// JANUS_LOG(LOG_ERR, "Invalid session...\n");
+	janus_videoroom_subscriber_stream *stream = (janus_videoroom_subscriber_stream *)data;
+	if(!stream || !g_atomic_int_get(&stream->ready) || g_atomic_int_get(&stream->destroyed) ||
+			!stream->send || !stream->publisher_streams ||
+			!stream->subscriber || stream->subscriber->paused || stream->subscriber->kicked ||
+			!stream->subscriber->session || !stream->subscriber->session->handle || !stream->subscriber->session->started)
 		return;
-	}
-	if(subscriber->paused || subscriber->kicked) {
-		// JANUS_LOG(LOG_ERR, "This subscriber paused the stream...\n");
+	janus_videoroom_publisher_stream *ps = stream->publisher_streams->data;
+	if(ps != packet->source)
 		return;
-	}
+	janus_videoroom_subscriber *subscriber = stream->subscriber;
 	janus_videoroom_session *session = subscriber->session;
-	if(!session || !session->handle) {
-		// JANUS_LOG(LOG_ERR, "Invalid session...\n");
-		return;
-	}
-	if(!session->started) {
-		// JANUS_LOG(LOG_ERR, "Streaming not started yet for this session...\n");
-		return;
-	}
 
 	/* Make sure there hasn't been a publisher switch by checking the SSRC */
 	if(packet->is_video) {
-		/* Check if this subscriber is subscribed to this medium */
-		if(!subscriber->video) {
-			/* Nope, don't relay */
-			return;
-		}
 		/* Check if there's any SVC info to take into account */
 		if(packet->svc) {
 			/* There is: check if this is a layer that can be dropped for this viewer
 			 * Note: Following core inspired by the excellent job done by Sergio Garcia Murillo here:
 			 * https://github.com/medooze/media-server/blob/master/src/vp9/VP9LayerSelector.cpp */
 			gboolean override_mark_bit = FALSE, has_marker_bit = packet->data->markerbit;
-			int temporal_layer = subscriber->temporal_layer;
-			if(subscriber->target_temporal_layer > subscriber->temporal_layer) {
+			int temporal_layer = stream->temporal_layer;
+			if(stream->target_temporal_layer > stream->temporal_layer) {
 				/* We need to upscale */
 				JANUS_LOG(LOG_HUGE, "We need to upscale temporally:\n");
-				if(packet->ubit && packet->bbit && packet->temporal_layer <= subscriber->target_temporal_layer) {
+				if(packet->ubit && packet->bbit && packet->temporal_layer <= stream->target_temporal_layer) {
 					JANUS_LOG(LOG_HUGE, "  -- Upscaling temporal layer: %u --> %u\n",
-						packet->temporal_layer, subscriber->target_temporal_layer);
-					subscriber->temporal_layer = packet->temporal_layer;
-					temporal_layer = subscriber->temporal_layer;
+						packet->temporal_layer, stream->target_temporal_layer);
+					stream->temporal_layer = packet->temporal_layer;
+					temporal_layer = stream->temporal_layer;
 					/* Notify the viewer */
 					json_t *event = json_object();
 					json_object_set_new(event, "videoroom", json_string("event"));
 					json_object_set_new(event, "room", json_integer(subscriber->room_id));
-					json_object_set_new(event, "temporal_layer", json_integer(subscriber->temporal_layer));
-					gateway->push_event(subscriber->session->handle, &janus_videoroom_plugin, NULL, event, NULL);
+					json_object_set_new(event, "mid", json_string(stream->mid));
+					json_object_set_new(event, "temporal_layer", json_integer(stream->temporal_layer));
+					gateway->push_event(session->handle, &janus_videoroom_plugin, NULL, event, NULL);
 					json_decref(event);
 				}
-			} else if(subscriber->target_temporal_layer < subscriber->temporal_layer) {
+			} else if(stream->target_temporal_layer < stream->temporal_layer) {
 				/* We need to downscale */
 				JANUS_LOG(LOG_HUGE, "We need to downscale temporally:\n");
 				if(packet->ebit) {
 					JANUS_LOG(LOG_HUGE, "  -- Downscaling temporal layer: %u --> %u\n",
-						subscriber->temporal_layer, subscriber->target_temporal_layer);
-					subscriber->temporal_layer = subscriber->target_temporal_layer;
+						stream->temporal_layer, stream->target_temporal_layer);
+					stream->temporal_layer = stream->target_temporal_layer;
 					/* Notify the viewer */
 					json_t *event = json_object();
 					json_object_set_new(event, "videoroom", json_string("event"));
 					json_object_set_new(event, "room", json_integer(subscriber->room_id));
-					json_object_set_new(event, "temporal_layer", json_integer(subscriber->temporal_layer));
-					gateway->push_event(subscriber->session->handle, &janus_videoroom_plugin, NULL, event, NULL);
+					json_object_set_new(event, "mid", json_string(stream->mid));
+					json_object_set_new(event, "temporal_layer", json_integer(stream->temporal_layer));
+					gateway->push_event(session->handle, &janus_videoroom_plugin, NULL, event, NULL);
 					json_decref(event);
 				}
 			}
 			if(temporal_layer < packet->temporal_layer) {
 				/* Drop the packet: update the context to make sure sequence number is increased normally later */
 				JANUS_LOG(LOG_HUGE, "Dropping packet (temporal layer %d < %d)\n", temporal_layer, packet->temporal_layer);
-				subscriber->context.v_base_seq++;
+				stream->context.base_seq++;
 				return;
 			}
-			int spatial_layer = subscriber->spatial_layer;
-			if(subscriber->target_spatial_layer > subscriber->spatial_layer) {
+			int spatial_layer = stream->spatial_layer;
+			if(stream->target_spatial_layer > stream->spatial_layer) {
 				JANUS_LOG(LOG_HUGE, "We need to upscale spatially:\n");
 				/* We need to upscale */
-				if(packet->pbit == 0 && packet->bbit && packet->spatial_layer == subscriber->spatial_layer+1) {
+				if(packet->pbit == 0 && packet->bbit && packet->spatial_layer == stream->spatial_layer+1) {
 					JANUS_LOG(LOG_HUGE, "  -- Upscaling spatial layer: %u --> %u\n",
-						packet->spatial_layer, subscriber->target_spatial_layer);
-					subscriber->spatial_layer = packet->spatial_layer;
-					spatial_layer = subscriber->spatial_layer;
+						packet->spatial_layer, stream->target_spatial_layer);
+					stream->spatial_layer = packet->spatial_layer;
+					spatial_layer = stream->spatial_layer;
 					/* Notify the viewer */
 					json_t *event = json_object();
 					json_object_set_new(event, "videoroom", json_string("event"));
 					json_object_set_new(event, "room", json_integer(subscriber->room_id));
-					json_object_set_new(event, "spatial_layer", json_integer(subscriber->spatial_layer));
-					gateway->push_event(subscriber->session->handle, &janus_videoroom_plugin, NULL, event, NULL);
+					json_object_set_new(event, "mid", json_string(stream->mid));
+					json_object_set_new(event, "spatial_layer", json_integer(stream->spatial_layer));
+					gateway->push_event(session->handle, &janus_videoroom_plugin, NULL, event, NULL);
 					json_decref(event);
 				}
-			} else if(subscriber->target_spatial_layer < subscriber->spatial_layer) {
+			} else if(stream->target_spatial_layer < stream->spatial_layer) {
 				/* We need to downscale */
 				JANUS_LOG(LOG_HUGE, "We need to downscale spatially:\n");
 				if(packet->ebit) {
 					JANUS_LOG(LOG_HUGE, "  -- Downscaling spatial layer: %u --> %u\n",
-						subscriber->spatial_layer, subscriber->target_spatial_layer);
-					subscriber->spatial_layer = subscriber->target_spatial_layer;
+						stream->spatial_layer, stream->target_spatial_layer);
+					stream->spatial_layer = stream->target_spatial_layer;
 					/* Notify the viewer */
 					json_t *event = json_object();
 					json_object_set_new(event, "videoroom", json_string("event"));
 					json_object_set_new(event, "room", json_integer(subscriber->room_id));
-					json_object_set_new(event, "spatial_layer", json_integer(subscriber->spatial_layer));
-					gateway->push_event(subscriber->session->handle, &janus_videoroom_plugin, NULL, event, NULL);
+					json_object_set_new(event, "mid", json_string(stream->mid));
+					json_object_set_new(event, "spatial_layer", json_integer(stream->spatial_layer));
+					gateway->push_event(session->handle, &janus_videoroom_plugin, NULL, event, NULL);
 					json_decref(event);
 				}
 			}
 			if(spatial_layer < packet->spatial_layer) {
 				/* Drop the packet: update the context to make sure sequence number is increased normally later */
 				JANUS_LOG(LOG_HUGE, "Dropping packet (spatial layer %d < %d)\n", spatial_layer, packet->spatial_layer);
-				subscriber->context.v_base_seq++;
+				stream->context.base_seq++;
 				return;
 			} else if(packet->ebit && spatial_layer == packet->spatial_layer) {
 				/* If we stop at layer 0, we need a marker bit now, as the one from layer 1 will not be received */
@@ -6596,12 +8457,12 @@ static void janus_videoroom_relay_rtp_packet(gpointer data, gpointer user_data)
 			JANUS_LOG(LOG_HUGE, "Sending packet (spatial=%d, temporal=%d)\n",
 				packet->spatial_layer, packet->temporal_layer);
 			/* Fix sequence number and timestamp (publisher switching may be involved) */
-			janus_rtp_header_update(packet->data, &subscriber->context, TRUE, 4500);
+			janus_rtp_header_update(packet->data, &stream->context, TRUE);
 			if(override_mark_bit && !has_marker_bit) {
 				packet->data->markerbit = 1;
 			}
 			if(gateway != NULL)
-				gateway->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+				gateway->relay_rtp(session->handle, ps->mindex, TRUE, (char *)packet->data, packet->length);
 			if(override_mark_bit && !has_marker_bit) {
 				packet->data->markerbit = 0;
 			}
@@ -6615,79 +8476,73 @@ static void janus_videoroom_relay_rtp_packet(gpointer data, gpointer user_data)
 			if(payload == NULL)
 				return;
 			/* Process this packet: don't relay if it's not the SSRC/layer we wanted to handle */
-			gboolean relay = janus_rtp_simulcasting_context_process_rtp(&subscriber->sim_context,
-				(char *)packet->data, packet->length, packet->ssrc, NULL, subscriber->feed->vcodec, &subscriber->context);
-			if(subscriber->sim_context.need_pli && subscriber->feed && subscriber->feed->session &&
-					subscriber->feed->session->handle) {
+			gboolean relay = janus_rtp_simulcasting_context_process_rtp(&stream->sim_context,
+				(char *)packet->data, packet->length, packet->ssrc, NULL, ps->vcodec, &stream->context);
+			if(stream->sim_context.need_pli && ps->publisher && ps->publisher->session &&
+					ps->publisher->session->handle) {
 				/* Send a PLI */
 				JANUS_LOG(LOG_VERB, "We need a PLI for the simulcast context\n");
-				char rtcpbuf[12];
-				memset(rtcpbuf, 0, 12);
-				janus_rtcp_pli((char *)&rtcpbuf, 12);
-				gateway->relay_rtcp(subscriber->feed->session->handle, 1, rtcpbuf, 12);
+				janus_videoroom_reqpli(ps, "Simulcast");
 			}
 			/* Do we need to drop this? */
 			if(!relay)
 				return;
 			/* Any event we should notify? */
-			if(subscriber->sim_context.changed_substream) {
+			if(stream->sim_context.changed_substream) {
 				/* Notify the user about the substream change */
 				json_t *event = json_object();
 				json_object_set_new(event, "videoroom", json_string("event"));
 				json_object_set_new(event, "room", json_integer(subscriber->room_id));
-				json_object_set_new(event, "substream", json_integer(subscriber->sim_context.substream));
-				gateway->push_event(subscriber->session->handle, &janus_videoroom_plugin, NULL, event, NULL);
+				json_object_set_new(event, "mid", json_string(stream->mid));
+				json_object_set_new(event, "substream", json_integer(stream->sim_context.substream));
+				gateway->push_event(session->handle, &janus_videoroom_plugin, NULL, event, NULL);
 				json_decref(event);
 			}
-			if(subscriber->sim_context.changed_temporal) {
+			if(stream->sim_context.changed_temporal) {
 				/* Notify the user about the temporal layer change */
 				json_t *event = json_object();
 				json_object_set_new(event, "videoroom", json_string("event"));
 				json_object_set_new(event, "room", json_integer(subscriber->room_id));
-				json_object_set_new(event, "temporal", json_integer(subscriber->sim_context.templayer));
-				gateway->push_event(subscriber->session->handle, &janus_videoroom_plugin, NULL, event, NULL);
+				json_object_set_new(event, "mid", json_string(stream->mid));
+				json_object_set_new(event, "temporal", json_integer(stream->sim_context.templayer));
+				gateway->push_event(session->handle, &janus_videoroom_plugin, NULL, event, NULL);
 				json_decref(event);
 			}
 			/* If we got here, update the RTP header and send the packet */
-			janus_rtp_header_update(packet->data, &subscriber->context, TRUE, 4500);
+			janus_rtp_header_update(packet->data, &stream->context, TRUE);
 			char vp8pd[6];
-			if(subscriber->feed && subscriber->feed->vcodec == JANUS_VIDEOCODEC_VP8) {
+			if(ps->vcodec == JANUS_VIDEOCODEC_VP8) {
 				/* For VP8, we save the original payload descriptor, to restore it after */
 				memcpy(vp8pd, payload, sizeof(vp8pd));
-				janus_vp8_simulcast_descriptor_update(payload, plen, &subscriber->vp8_context,
-					subscriber->sim_context.changed_substream);
+				janus_vp8_simulcast_descriptor_update(payload, plen, &stream->vp8_context,
+					stream->sim_context.changed_substream);
 			}
 			/* Send the packet */
 			if(gateway != NULL)
-				gateway->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+				gateway->relay_rtp(session->handle, stream->mindex, TRUE, (char *)packet->data, packet->length);
 			/* Restore the timestamp and sequence number to what the publisher set them to */
 			packet->data->timestamp = htonl(packet->timestamp);
 			packet->data->seq_number = htons(packet->seq_number);
-			if(subscriber->feed && subscriber->feed->vcodec == JANUS_VIDEOCODEC_VP8) {
+			if(ps->vcodec == JANUS_VIDEOCODEC_VP8) {
 				/* Restore the original payload descriptor as well, as it will be needed by the next viewer */
 				memcpy(payload, vp8pd, sizeof(vp8pd));
 			}
 		} else {
 			/* Fix sequence number and timestamp (publisher switching may be involved) */
-			janus_rtp_header_update(packet->data, &subscriber->context, TRUE, 4500);
+			janus_rtp_header_update(packet->data, &stream->context, TRUE);
 			/* Send the packet */
 			if(gateway != NULL)
-				gateway->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+				gateway->relay_rtp(session->handle, stream->mindex, FALSE, (char *)packet->data, packet->length);
 			/* Restore the timestamp and sequence number to what the publisher set them to */
 			packet->data->timestamp = htonl(packet->timestamp);
 			packet->data->seq_number = htons(packet->seq_number);
 		}
 	} else {
-		/* Check if this subscriber is subscribed to this medium */
-		if(!subscriber->audio) {
-			/* Nope, don't relay */
-			return;
-		}
 		/* Fix sequence number and timestamp (publisher switching may be involved) */
-		janus_rtp_header_update(packet->data, &subscriber->context, FALSE, 960);
+		janus_rtp_header_update(packet->data, &stream->context, FALSE);
 		/* Send the packet */
 		if(gateway != NULL)
-			gateway->relay_rtp(session->handle, packet->is_video, (char *)packet->data, packet->length);
+			gateway->relay_rtp(session->handle, stream->mindex, TRUE, (char *)packet->data, packet->length);
 		/* Restore the timestamp and sequence number to what the publisher set them to */
 		packet->data->timestamp = htonl(packet->timestamp);
 		packet->data->seq_number = htons(packet->seq_number);
@@ -6697,21 +8552,30 @@ static void janus_videoroom_relay_rtp_packet(gpointer data, gpointer user_data)
 }
 
 static void janus_videoroom_relay_data_packet(gpointer data, gpointer user_data) {
-	char *text = (char *)user_data;
-	janus_videoroom_subscriber *subscriber = (janus_videoroom_subscriber *)data;
-	if(!subscriber || !subscriber->session || !subscriber->data || subscriber->paused) {
+	janus_videoroom_data_relay_packet *packet = (janus_videoroom_data_relay_packet *)user_data;
+	if(!packet || !packet->text) {
+		JANUS_LOG(LOG_ERR, "Invalid packet...\n");
 		return;
 	}
-	janus_videoroom_session *session = subscriber->session;
-	if(!session || !session->handle) {
+	janus_videoroom_subscriber_stream *stream = (janus_videoroom_subscriber_stream *)data;
+	if(!stream || !stream->send || !stream->subscriber || stream->subscriber->paused || stream->subscriber->kicked ||
+			!stream->subscriber->session || !stream->subscriber->session->handle || !stream->subscriber->session->started)
 		return;
-	}
-	if(!session->started) {
+	janus_videoroom_subscriber *subscriber = stream->subscriber;
+	janus_videoroom_session *session = subscriber->session;
+
+	/* We use the publisher's user ID as the label for the data channel */
+	janus_videoroom_publisher_stream *ps = packet->source;
+	if(ps == NULL || ps->publisher == NULL)
 		return;
-	}
+	janus_videoroom_publisher *publisher = ps->publisher;
+	char label[64];
+	g_snprintf(label, sizeof(label), "%"SCNu64, publisher->user_id);
+
+	char *text = packet->text;
 	if(gateway != NULL && text != NULL) {
 		JANUS_LOG(LOG_VERB, "Forwarding DataChannel message (%zu bytes) to viewer: %s\n", strlen(text), text);
-		gateway->relay_data(session->handle, NULL, text, strlen(text));
+		gateway->relay_data(session->handle, label, text, strlen(text));
 	}
 	return;
 }
@@ -6724,10 +8588,10 @@ static void janus_videoroom_rtp_forwarder_rtcp_receive(janus_videoroom_rtp_forwa
 	int len = recvfrom(forward->rtcp_fd, buffer, sizeof(buffer), 0, (struct sockaddr *)&remote_addr, &addrlen);
 	if(len > 0 && janus_is_rtcp(buffer, len)) {
 		JANUS_LOG(LOG_HUGE, "Got %s RTCP packet: %d bytes\n", forward->is_video ? "video" : "audio", len);
-		/* We only handle incoming video PLIs or FIR at the moment */
+		/* We only handle incoming video PLI or FIR at the moment */
 		if(!janus_rtcp_has_fir(buffer, len) && !janus_rtcp_has_pli(buffer, len))
 			return;
-		janus_videoroom_reqfir((janus_videoroom_publisher *)forward->source, "RTCP from forwarder");
+		janus_videoroom_reqpli((janus_videoroom_publisher_stream *)forward->source, "RTCP from forwarder");
 	}
 }
 
diff --git a/plugins/janus_voicemail.c b/plugins/janus_voicemail.c
index 5ce38d7e5e..b1a6cf5843 100644
--- a/plugins/janus_voicemail.c
+++ b/plugins/janus_voicemail.c
@@ -115,8 +115,8 @@
 
 
 /* Plugin information */
-#define JANUS_VOICEMAIL_VERSION			7
-#define JANUS_VOICEMAIL_VERSION_STRING	"0.0.7"
+#define JANUS_VOICEMAIL_VERSION			8
+#define JANUS_VOICEMAIL_VERSION_STRING	"0.0.8"
 #define JANUS_VOICEMAIL_DESCRIPTION		"This is a plugin implementing a very simple VoiceMail service for Janus, recording Opus streams."
 #define JANUS_VOICEMAIL_NAME			"JANUS VoiceMail plugin"
 #define JANUS_VOICEMAIL_AUTHOR			"Meetecho s.r.l."
@@ -136,8 +136,8 @@ const char *janus_voicemail_get_package(void);
 void janus_voicemail_create_session(janus_plugin_session *handle, int *error);
 struct janus_plugin_result *janus_voicemail_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
 void janus_voicemail_setup_media(janus_plugin_session *handle);
-void janus_voicemail_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
-void janus_voicemail_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
+void janus_voicemail_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
+void janus_voicemail_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 void janus_voicemail_hangup_media(janus_plugin_session *handle);
 void janus_voicemail_destroy_session(janus_plugin_session *handle, int *error);
 json_t *janus_voicemail_query_session(janus_plugin_session *handle);
@@ -174,7 +174,7 @@ janus_plugin *create(void) {
 
 /* Parameter validation */
 static struct janus_json_parameter request_parameters[] = {
-	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
+	{"request", JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
 };
 
 /* Useful stuff */
@@ -362,6 +362,7 @@ int janus_voicemail_init(janus_callbacks *callback, const char *config_path) {
 	if(error != NULL) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the VoiceMail handler thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_VOICEMAIL_NAME);
@@ -499,8 +500,8 @@ json_t *janus_voicemail_query_session(janus_plugin_session *handle) {
 		json_object_set_new(info, "start_time", json_integer(session->start_time));
 		json_object_set_new(info, "filename", session->filename ? json_string(session->filename) : NULL);
 	}
-	json_object_set_new(info, "hangingup", json_integer(g_atomic_int_get(&session->hangingup)));
-	json_object_set_new(info, "destroyed", json_integer(g_atomic_int_get(&session->destroyed)));
+	json_object_set_new(info, "hangingup", g_atomic_int_get(&session->hangingup) ? json_true() : json_false());
+	json_object_set_new(info, "destroyed", g_atomic_int_get(&session->destroyed) ? json_true() : json_false());
 	janus_refcount_decrease(&session->ref);
 	return info;
 }
@@ -561,7 +562,7 @@ void janus_voicemail_setup_media(janus_plugin_session *handle) {
 	janus_refcount_decrease(&session->ref);
 }
 
-void janus_voicemail_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_voicemail_incoming_rtp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	janus_voicemail_session *session = (janus_voicemail_session *)handle->plugin_handle;
@@ -600,7 +601,7 @@ void janus_voicemail_incoming_rtp(janus_plugin_session *handle, int video, char
 	ogg_write(session);
 }
 
-void janus_voicemail_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
+void janus_voicemail_incoming_rtcp(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len) {
 	if(handle == NULL || g_atomic_int_get(&handle->stopped) || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
 		return;
 	/* FIXME Should we care? */
diff --git a/plugins/plugin.h b/plugins/plugin.h
index 9a5820fb6d..bb2a6f22f0 100644
--- a/plugins/plugin.h
+++ b/plugins/plugin.h
@@ -65,7 +65,7 @@ janus_plugin *create(void) {
  * - \c incoming_rtp(): a callback to notify you a peer has sent you a RTP packet;
  * - \c incoming_rtcp(): a callback to notify you a peer has sent you a RTCP message;
  * - \c incoming_data(): a callback to notify you a peer has sent you a message on a SCTP DataChannel;
- * - \c slow_link(): a callback to notify you a peer has sent a lot of NACKs recently, and the media path may be slow;
+ * - \c slow_link(): a callback to notify you Janus or the peer have lost packets recently, and the media path may be slow;
  * - \c hangup_media(): a callback to notify you the peer PeerConnection has been closed (e.g., after a DTLS alert);
  * - \c query_session(): this method is called by the core to get plugin-specific info on a session between you and a peer;
  * - \c destroy_session(): this method is called by the core to destroy a session between you and a peer.
@@ -288,20 +288,22 @@ struct janus_plugin {
 	void (* const setup_media)(janus_plugin_session *handle);
 	/*! \brief Method to handle an incoming RTP packet from a peer
 	 * @param[in] handle The plugin/gateway session used for this peer
+	 * @param[in] mindex Index of the source stream (relative to the SDP)
 	 * @param[in] video Whether this is an audio or a video frame
 	 * @param[in] buf The packet data (buffer)
 	 * @param[in] len The buffer lenght */
-	void (* const incoming_rtp)(janus_plugin_session *handle, int video, char *buf, int len);
+	void (* const incoming_rtp)(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 	/*! \brief Method to handle an incoming RTCP packet from a peer
 	 * @param[in] handle The plugin/gateway session used for this peer
+	 * @param[in] mindex Index of the source stream (relative to the SDP)
 	 * @param[in] video Whether this is related to an audio or a video stream
 	 * @param[in] buf The message data (buffer)
 	 * @param[in] len The buffer lenght */
-	void (* const incoming_rtcp)(janus_plugin_session *handle, int video, char *buf, int len);
+	void (* const incoming_rtcp)(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 	/*! \brief Method to handle incoming SCTP/DataChannel data from a peer (text only, for the moment)
 	 * \note We currently only support text data, binary data will follow... please also notice that
 	 * DataChannels send unterminated strings, so you'll have to terminate them with a \0 yourself to
-	 * use them.
+	 * use them. Notice that no mindex is needed here, as there can only be a single data channel stream.
 	 * @param[in] handle The plugin/gateway session used for this peer
 	 * @param[in] label The label of the data channel to use
 	 * @param[in] buf The message data (buffer)
@@ -323,7 +325,7 @@ struct janus_plugin {
 	 * @param[in] uplink Whether this is related to the uplink (Janus to peer)
 	 * or downlink (peer to Janus)
 	 * @param[in] video Whether this is related to an audio or a video stream */
-	void (* const slow_link)(janus_plugin_session *handle, int uplink, int video);
+	void (* const slow_link)(janus_plugin_session *handle, int mindex, gboolean video, gboolean uplink);
 	/*! \brief Callback to be notified about DTLS alerts from a peer (i.e., the PeerConnection is not valid any more)
 	 * @param[in] handle The plugin/gateway session used for this peer */
 	void (* const hangup_media)(janus_plugin_session *handle);
@@ -355,17 +357,20 @@ struct janus_callbacks {
 
 	/*! \brief Callback to relay RTP packets to a peer
 	 * @param[in] handle The plugin/gateway session used for this peer
+	 * @param[in] mindex Index of the target stream (relative to the SDP), -1 for the first audio/video stream
 	 * @param[in] video Whether this is an audio or a video frame
 	 * @param[in] buf The packet data (buffer)
 	 * @param[in] len The buffer lenght */
-	void (* const relay_rtp)(janus_plugin_session *handle, int video, char *buf, int len);
+	void (* const relay_rtp)(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 	/*! \brief Callback to relay RTCP messages to a peer
 	 * @param[in] handle The plugin/gateway session that will be used for this peer
+	 * @param[in] mindex Index of the target stream (relative to the SDP), -1 for the first audio/video stream
 	 * @param[in] video Whether this is related to an audio or a video stream
 	 * @param[in] buf The message data (buffer)
 	 * @param[in] len The buffer lenght */
-	void (* const relay_rtcp)(janus_plugin_session *handle, int video, char *buf, int len);
+	void (* const relay_rtcp)(janus_plugin_session *handle, int mindex, gboolean video, char *buf, int len);
 	/*! \brief Callback to relay SCTP/DataChannel messages to a peer
+	 * \note Notice that no mindex is needed here, as there can only be a single data channel stream.
 	 * @param[in] handle The plugin/gateway session that will be used for this peer
 	 * @param[in] label The label of the data channel to use
 	 * @param[in] buf The message data (buffer)
diff --git a/plugins/streams/test_gstreamer_1.sh b/plugins/streams/test_gstreamer1.sh
similarity index 100%
rename from plugins/streams/test_gstreamer_1.sh
rename to plugins/streams/test_gstreamer1.sh
diff --git a/plugins/streams/test_gstreamer1_multistream.sh b/plugins/streams/test_gstreamer1_multistream.sh
new file mode 100755
index 0000000000..7d00c86d30
--- /dev/null
+++ b/plugins/streams/test_gstreamer1_multistream.sh
@@ -0,0 +1,16 @@
+#!/bin/sh
+gst-launch-1.0 \
+  audiotestsrc ! \
+    audioresample ! audio/x-raw,channels=1,rate=16000 ! \
+    opusenc bitrate=20000 ! \
+      rtpopuspay ! udpsink host=127.0.0.1 port=5102 \
+  videotestsrc pattern=ball ! \
+    video/x-raw,width=320,height=240,framerate=15/1 ! \
+    videoscale ! videorate ! videoconvert ! timeoverlay ! \
+    vp8enc error-resilient=1 ! \
+      rtpvp8pay ! udpsink host=127.0.0.1 port=5104 \
+  videotestsrc ! \
+    video/x-raw,width=320,height=240,framerate=15/1 ! \
+    videoscale ! videorate ! videoconvert ! timeoverlay ! \
+    vp8enc error-resilient=1 ! \
+      rtpvp8pay ! udpsink host=127.0.0.1 port=5106
diff --git a/plugins/streams/test_gstreamer_multistream.sh b/plugins/streams/test_gstreamer_multistream.sh
new file mode 100755
index 0000000000..3cba4454ac
--- /dev/null
+++ b/plugins/streams/test_gstreamer_multistream.sh
@@ -0,0 +1,28 @@
+#!/bin/sh
+
+# Note: if you're on Ubuntu, good chances are your Gstreamer distribution
+# will not have a plugin for Opus installed out of the box. You can try
+# using the one available on Debian, e.g., installing the package you
+# can find here:
+#    http://packages.debian.org/sid/i386/gstreamer0.10-plugins-bad/download
+# For instance:
+#    wget http://ftp.it.debian.org/debian/pool/main/g/gst-plugins-bad0.10/gstreamer0.10-plugins-bad_0.10.23-7.1_i386.deb
+#    ar xv gstreamer0.10-plugins-bad_0.10.23-7.1_i386.deb
+#    tar xfv data.tar.xv
+#    cp usr/lib/i386-linux-gnu/gstreamer-0.10/libgstopus.so /usr/lib/i386-linux-gnu/gstreamer-0.10/
+
+gst-launch \
+	audiotestsrc ! \
+		audioresample ! audio/x-raw-int,channels=1,rate=16000 ! \
+		opusenc bitrate=20000 ! \
+			rtpopuspay ! udpsink host=127.0.0.1 port=5102 \
+	videotestsrc pattern=ball ! \
+		video/x-raw-rgb,width=320,height=240,framerate=15/1 ! \
+		videoscale ! videorate ! ffmpegcolorspace ! timeoverlay ! \
+		vp8enc bitrate=256000 speed=2 max-latency=1 error-resilient=true ! \
+			rtpvp8pay ! udpsink host=127.0.0.1 port=5104 \
+	videotestsrc ! \
+		video/x-raw-rgb,width=320,height=240,framerate=15/1 ! \
+		videoscale ! videorate ! ffmpegcolorspace ! timeoverlay ! \
+		vp8enc bitrate=256000 speed=2 max-latency=1 error-resilient=true ! \
+			rtpvp8pay ! udpsink host=127.0.0.1 port=5106
diff --git a/rtcp.c b/rtcp.c
index f3f46d2992..0ae0b972a6 100644
--- a/rtcp.c
+++ b/rtcp.c
@@ -852,7 +852,7 @@ int janus_rtcp_fix_report_data(char *packet, int len, uint32_t base_ts, uint32_t
 				uint32_t recv_ssrc = ntohl(sr->ssrc);
 				if (recv_ssrc != ssrc_expected) {
 					if(ssrc_expected != 0) {
-						JANUS_LOG(LOG_WARN,"Incoming RTCP SR SSRC (%"SCNu32") does not match the expected one (%"SCNu32") video=%d\n", recv_ssrc, ssrc_expected, video);
+						JANUS_LOG(LOG_VERB,"Incoming RTCP SR SSRC (%"SCNu32") does not match the expected one (%"SCNu32") video=%d\n", recv_ssrc, ssrc_expected, video);
 					}
 					return -3;
 				}
diff --git a/rtp.c b/rtp.c
index 5861541fb3..fe18a1137f 100644
--- a/rtp.c
+++ b/rtp.c
@@ -362,18 +362,18 @@ void janus_rtp_switching_context_reset(janus_rtp_switching_context *context) {
 
 int janus_rtp_skew_compensate_audio(janus_rtp_header *header, janus_rtp_switching_context *context, gint64 now) {
 	/* Reset values if a new ssrc has been detected */
-	if (context->a_new_ssrc) {
-		JANUS_LOG(LOG_VERB, "audio skew SSRC=%"SCNu32" resetting status\n", context->a_last_ssrc);
-		context->a_reference_time = now;
-		context->a_start_time = 0;
-		context->a_evaluating_start_time = 0;
-		context->a_start_ts = 0;
-		context->a_active_delay = 0;
-		context->a_prev_delay = 0;
-		context->a_seq_offset = 0;
-		context->a_ts_offset = 0;
-		context->a_target_ts = 0;
-		context->a_new_ssrc = FALSE;
+	if(context->new_ssrc) {
+		JANUS_LOG(LOG_VERB, "audio skew SSRC=%"SCNu32" resetting status\n", context->last_ssrc);
+		context->reference_time = now;
+		context->start_time = 0;
+		context->evaluating_start_time = 0;
+		context->start_ts = 0;
+		context->active_delay = 0;
+		context->prev_delay = 0;
+		context->seq_offset = 0;
+		context->ts_offset = 0;
+		context->target_ts = 0;
+		context->new_ssrc = FALSE;
 	}
 
 	/* N 	: a N sequence number jump has been performed */
@@ -382,84 +382,84 @@ int janus_rtp_skew_compensate_audio(janus_rtp_header *header, janus_rtp_switchin
 	int exit_status = 0;
 
 	/* Do not execute skew analysis in the first seconds */
-	if (now-context->a_reference_time < SKEW_DETECTION_WAIT_TIME_SECS/2 * G_USEC_PER_SEC) {
+	if(now-context->reference_time < SKEW_DETECTION_WAIT_TIME_SECS/2 * G_USEC_PER_SEC) {
 		return 0;
-	} else if (!context->a_start_time) {
-		JANUS_LOG(LOG_VERB, "audio skew SSRC=%"SCNu32" evaluation phase start\n", context->a_last_ssrc);
-		context->a_start_time = now;
-		context->a_evaluating_start_time = now;
-		context->a_start_ts = context->a_last_ts;
+	} else if(!context->start_time) {
+		JANUS_LOG(LOG_VERB, "audio skew SSRC=%"SCNu32" evaluation phase start\n", context->last_ssrc);
+		context->start_time = now;
+		context->evaluating_start_time = now;
+		context->start_ts = context->last_ts;
 	}
 
 	/* Skew analysis */
 	/* Are we waiting for a target timestamp? (a negative skew has been evaluated in a previous iteration) */
-	if (context->a_target_ts > 0 && (gint32)(context->a_target_ts - context->a_last_ts) > 0) {
-		context->a_seq_offset--;
+	if(context->target_ts > 0 && (gint32)(context->target_ts - context->last_ts) > 0) {
+		context->seq_offset--;
 		exit_status = -1;
 	} else {
-		context->a_target_ts = 0;
+		context->target_ts = 0;
 		/* Do not execute analysis for out of order packets or multi-packets frame */
-		if (context->a_last_seq == context->a_prev_seq + 1 && context->a_last_ts != context->a_prev_ts) {
+		if(context->last_seq == context->prev_seq + 1 && context->last_ts != context->prev_ts) {
 			/* Set the sample rate according to the header */
 			guint32 akhz = 48; /* 48khz for Opus */
 			if(header->type == 0 || header->type == 8 || header->type == 9)
 				akhz = 8;
 			/* Evaluate the local RTP timestamp according to the local clock */
-			guint32 expected_ts = ((now - context->a_start_time)*akhz)/1000 + context->a_start_ts;
+			guint32 expected_ts = ((now - context->start_time)*akhz)/1000 + context->start_ts;
 			/* Evaluate current delay */
-			gint32 delay_now = context->a_last_ts - expected_ts;
+			gint32 delay_now = context->last_ts - expected_ts;
 			/* Exponentially weighted moving average estimation */
-			gint32 delay_estimate = (63*context->a_prev_delay + delay_now)/64;
+			gint32 delay_estimate = (63*context->prev_delay + delay_now)/64;
 			/* Save previous delay for the next iteration*/
-			context->a_prev_delay = delay_estimate;
+			context->prev_delay = delay_estimate;
 			/* Evaluate the distance between active delay and current delay estimate */
-			gint32 offset = context->a_active_delay - delay_estimate;
-			JANUS_LOG(LOG_HUGE, "audio skew status SSRC=%"SCNu32" RECVD_TS=%"SCNu32" EXPTD_TS=%"SCNu32" OFFSET=%"SCNi32" TS_OFFSET=%"SCNi32" SEQ_OFFSET=%"SCNi16"\n", context->a_last_ssrc, context->a_last_ts, expected_ts, offset, context->a_ts_offset, context->a_seq_offset);
+			gint32 offset = context->active_delay - delay_estimate;
+			JANUS_LOG(LOG_HUGE, "audio skew status SSRC=%"SCNu32" RECVD_TS=%"SCNu32" EXPTD_TS=%"SCNu32" OFFSET=%"SCNi32" TS_OFFSET=%"SCNi32" SEQ_OFFSET=%"SCNi16"\n", context->last_ssrc, context->last_ts, expected_ts, offset, context->ts_offset, context->seq_offset);
 			gint32 skew_th = RTP_AUDIO_SKEW_TH_MS*akhz;
 			/* Evaluation phase */
-			if (context->a_evaluating_start_time > 0) {
+			if(context->evaluating_start_time > 0) {
 				/* Check if the offset has surpassed half the threshold during the evaluating phase */
-				if (now-context->a_evaluating_start_time <= SKEW_DETECTION_WAIT_TIME_SECS/2 * G_USEC_PER_SEC) {
-					if (abs(offset) <= skew_th/2) {
-						JANUS_LOG(LOG_HUGE, "audio skew SSRC=%"SCNu32" evaluation phase continue\n", context->a_last_ssrc);
+				if(now-context->evaluating_start_time <= SKEW_DETECTION_WAIT_TIME_SECS/2 * G_USEC_PER_SEC) {
+					if(abs(offset) <= skew_th/2) {
+						JANUS_LOG(LOG_HUGE, "audio skew SSRC=%"SCNu32" evaluation phase continue\n", context->last_ssrc);
 					} else {
-						JANUS_LOG(LOG_VERB, "audio skew SSRC=%"SCNu32" evaluation phase reset\n", context->a_last_ssrc);
-						context->a_start_time = now;
-						context->a_evaluating_start_time = now;
-						context->a_start_ts = context->a_last_ts;
+						JANUS_LOG(LOG_VERB, "audio skew SSRC=%"SCNu32" evaluation phase reset\n", context->last_ssrc);
+						context->start_time = now;
+						context->evaluating_start_time = now;
+						context->start_ts = context->last_ts;
 					}
 				} else {
-					JANUS_LOG(LOG_VERB, "audio skew SSRC=%"SCNu32" evaluation phase stop\n", context->a_last_ssrc);
-					context->a_evaluating_start_time = 0;
+					JANUS_LOG(LOG_VERB, "audio skew SSRC=%"SCNu32" evaluation phase stop\n", context->last_ssrc);
+					context->evaluating_start_time = 0;
 				}
 				return 0;
 			}
 			/* Check if the offset has surpassed the threshold */
-			if (offset >= skew_th) {
+			if(offset >= skew_th) {
 				/* The source is slowing down */
 				/* Update active delay */
-				context->a_active_delay = delay_estimate;
+				context->active_delay = delay_estimate;
 				/* Adjust ts offset */
-				context->a_ts_offset += skew_th;
+				context->ts_offset += skew_th;
 				/* Calculate last ts increase */
-				guint32 ts_incr = context->a_last_ts-context->a_prev_ts;
+				guint32 ts_incr = context->last_ts-context->prev_ts;
 				/* Evaluate sequence number jump */
 				guint16 jump = (skew_th+ts_incr-1)/ts_incr;
 				/* Adjust seq num offset */
-				context->a_seq_offset += jump;
+				context->seq_offset += jump;
 				exit_status = jump;
-			} else if (offset <= -skew_th) {
+			} else if(offset <= -skew_th) {
 				/* The source is speeding up*/
 				/* Update active delay */
-				context->a_active_delay = delay_estimate;
+				context->active_delay = delay_estimate;
 				/* Adjust ts offset */
-				context->a_ts_offset -= skew_th;
+				context->ts_offset -= skew_th;
 				/* Set target ts */
-				context->a_target_ts = context->a_last_ts + skew_th;
-				if (context->a_target_ts == 0)
-					context->a_target_ts = 1;
+				context->target_ts = context->last_ts + skew_th;
+				if(context->target_ts == 0)
+					context->target_ts = 1;
 				/* Adjust seq num offset */
-				context->a_seq_offset--;
+				context->seq_offset--;
 				exit_status = -1;
 			}
 		}
@@ -467,10 +467,10 @@ int janus_rtp_skew_compensate_audio(janus_rtp_header *header, janus_rtp_switchin
 
 	/* Skew compensation */
 	/* Fix header timestamp considering the active offset */
-	guint32 fixed_rtp_ts = context->a_last_ts + context->a_ts_offset;
+	guint32 fixed_rtp_ts = context->last_ts + context->ts_offset;
 	header->timestamp = htonl(fixed_rtp_ts);
 	/* Fix header sequence number considering the total offset */
-	guint16 fixed_rtp_seq = context->a_last_seq + context->a_seq_offset;
+	guint16 fixed_rtp_seq = context->last_seq + context->seq_offset;
 	header->seq_number = htons(fixed_rtp_seq);
 
 	return exit_status;
@@ -478,18 +478,18 @@ int janus_rtp_skew_compensate_audio(janus_rtp_header *header, janus_rtp_switchin
 
 int janus_rtp_skew_compensate_video(janus_rtp_header *header, janus_rtp_switching_context *context, gint64 now) {
 	/* Reset values if a new ssrc has been detected */
-	if (context->v_new_ssrc) {
-		JANUS_LOG(LOG_VERB, "video skew SSRC=%"SCNu32" resetting status\n", context->v_last_ssrc);
-		context->v_reference_time = now;
-		context->v_start_time = 0;
-		context->v_evaluating_start_time = 0;
-		context->v_start_ts = 0;
-		context->v_active_delay = 0;
-		context->v_prev_delay = 0;
-		context->v_seq_offset = 0;
-		context->v_ts_offset = 0;
-		context->v_target_ts = 0;
-		context->v_new_ssrc = FALSE;
+	if(context->new_ssrc) {
+		JANUS_LOG(LOG_VERB, "video skew SSRC=%"SCNu32" resetting status\n", context->last_ssrc);
+		context->reference_time = now;
+		context->start_time = 0;
+		context->evaluating_start_time = 0;
+		context->start_ts = 0;
+		context->active_delay = 0;
+		context->prev_delay = 0;
+		context->seq_offset = 0;
+		context->ts_offset = 0;
+		context->target_ts = 0;
+		context->new_ssrc = FALSE;
 	}
 
 	/* N 	: a N sequence numbers jump has been performed */
@@ -498,82 +498,82 @@ int janus_rtp_skew_compensate_video(janus_rtp_header *header, janus_rtp_switchin
 	int exit_status = 0;
 
 	/* Do not execute skew analysis in the first seconds */
-	if (now-context->v_reference_time < SKEW_DETECTION_WAIT_TIME_SECS/2 *G_USEC_PER_SEC) {
+	if(now-context->reference_time < SKEW_DETECTION_WAIT_TIME_SECS/2 *G_USEC_PER_SEC) {
 		return 0;
-	} else if (!context->v_start_time) {
-		JANUS_LOG(LOG_VERB, "video skew SSRC=%"SCNu32" evaluation phase start\n", context->v_last_ssrc);
-		context->v_start_time = now;
-		context->v_evaluating_start_time = now;
-		context->v_start_ts = context->v_last_ts;
+	} else if(!context->start_time) {
+		JANUS_LOG(LOG_VERB, "video skew SSRC=%"SCNu32" evaluation phase start\n", context->last_ssrc);
+		context->start_time = now;
+		context->evaluating_start_time = now;
+		context->start_ts = context->last_ts;
 	}
 
 	/* Skew analysis */
 	/* Are we waiting for a target timestamp? (a negative skew has been evaluated in a previous iteration) */
-	if (context->v_target_ts > 0 && (gint32)(context->v_target_ts - context->v_last_ts) > 0) {
-		context->v_seq_offset--;
+	if(context->target_ts > 0 && (gint32)(context->target_ts - context->last_ts) > 0) {
+		context->seq_offset--;
 		exit_status = -1;
 	} else {
-		context->v_target_ts = 0;
+		context->target_ts = 0;
 		/* Do not execute analysis for out of order packets or multi-packets frame */
-		if (context->v_last_seq == context->v_prev_seq + 1 && context->v_last_ts != context->v_prev_ts) {
+		if(context->last_seq == context->prev_seq + 1 && context->last_ts != context->prev_ts) {
 			/* Set the sample rate */
 			guint32 vkhz = 90; /* 90khz */
 			/* Evaluate the local RTP timestamp according to the local clock */
-			guint32 expected_ts = ((now - context->v_start_time)*vkhz)/1000 + context->v_start_ts;
+			guint32 expected_ts = ((now - context->start_time)*vkhz)/1000 + context->start_ts;
 			/* Evaluate current delay */
-			gint32 delay_now = context->v_last_ts - expected_ts;
+			gint32 delay_now = context->last_ts - expected_ts;
 			/* Exponentially weighted moving average estimation */
-			gint32 delay_estimate = (63*context->v_prev_delay + delay_now)/64;
+			gint32 delay_estimate = (63*context->prev_delay + delay_now)/64;
 			/* Save previous delay for the next iteration*/
-			context->v_prev_delay = delay_estimate;
+			context->prev_delay = delay_estimate;
 			/* Evaluate the distance between active delay and current delay estimate */
-			gint32 offset = context->v_active_delay - delay_estimate;
-			JANUS_LOG(LOG_HUGE, "video skew status SSRC=%"SCNu32" RECVD_TS=%"SCNu32" EXPTD_TS=%"SCNu32" OFFSET=%"SCNi32" TS_OFFSET=%"SCNi32" SEQ_OFFSET=%"SCNi16"\n", context->v_last_ssrc, context->v_last_ts, expected_ts, offset, context->v_ts_offset, context->v_seq_offset);
+			gint32 offset = context->active_delay - delay_estimate;
+			JANUS_LOG(LOG_HUGE, "video skew status SSRC=%"SCNu32" RECVD_TS=%"SCNu32" EXPTD_TS=%"SCNu32" OFFSET=%"SCNi32" TS_OFFSET=%"SCNi32" SEQ_OFFSET=%"SCNi16"\n", context->last_ssrc, context->last_ts, expected_ts, offset, context->ts_offset, context->seq_offset);
 			gint32 skew_th = RTP_VIDEO_SKEW_TH_MS*vkhz;
 			/* Evaluation phase */
-			if (context->v_evaluating_start_time > 0) {
+			if(context->evaluating_start_time > 0) {
 				/* Check if the offset has surpassed half the threshold during the evaluating phase */
-				if (now-context->v_evaluating_start_time <= SKEW_DETECTION_WAIT_TIME_SECS/2 * G_USEC_PER_SEC) {
-					if (abs(offset) <= skew_th/2) {
-						JANUS_LOG(LOG_HUGE, "video skew SSRC=%"SCNu32" evaluation phase continue\n", context->v_last_ssrc);
+				if(now-context->evaluating_start_time <= SKEW_DETECTION_WAIT_TIME_SECS/2 * G_USEC_PER_SEC) {
+					if(abs(offset) <= skew_th/2) {
+						JANUS_LOG(LOG_HUGE, "video skew SSRC=%"SCNu32" evaluation phase continue\n", context->last_ssrc);
 					} else {
-						JANUS_LOG(LOG_VERB, "video skew SSRC=%"SCNu32" evaluation phase reset\n", context->v_last_ssrc);
-						context->v_start_time = now;
-						context->v_evaluating_start_time = now;
-						context->v_start_ts = context->v_last_ts;
+						JANUS_LOG(LOG_VERB, "video skew SSRC=%"SCNu32" evaluation phase reset\n", context->last_ssrc);
+						context->start_time = now;
+						context->evaluating_start_time = now;
+						context->start_ts = context->last_ts;
 					}
 				} else {
-					JANUS_LOG(LOG_VERB, "video skew SSRC=%"SCNu32" evaluation phase stop\n", context->v_last_ssrc);
-					context->v_evaluating_start_time = 0;
+					JANUS_LOG(LOG_VERB, "video skew SSRC=%"SCNu32" evaluation phase stop\n", context->last_ssrc);
+					context->evaluating_start_time = 0;
 				}
 				return 0;
 			}
 			/* Check if the offset has surpassed the threshold */
-			if (offset >= skew_th) {
+			if(offset >= skew_th) {
 				/* The source is slowing down */
 				/* Update active delay */
-				context->v_active_delay = delay_estimate;
+				context->active_delay = delay_estimate;
 				/* Adjust ts offset */
-				context->v_ts_offset += skew_th;
+				context->ts_offset += skew_th;
 				/* Calculate last ts increase */
-				guint32 ts_incr = context->v_last_ts-context->v_prev_ts;
+				guint32 ts_incr = context->last_ts-context->prev_ts;
 				/* Evaluate sequence number jump */
 				guint16 jump = (skew_th+ts_incr-1)/ts_incr;
 				/* Adjust seq num offset */
-				context->v_seq_offset += jump;
+				context->seq_offset += jump;
 				exit_status = jump;
-			} else if (offset <= -skew_th) {
+			} else if(offset <= -skew_th) {
 				/* The source is speeding up*/
 				/* Update active delay */
-				context->v_active_delay = delay_estimate;
+				context->active_delay = delay_estimate;
 				/* Adjust ts offset */
-				context->v_ts_offset -= skew_th;
+				context->ts_offset -= skew_th;
 				/* Set target ts */
-				context->v_target_ts = context->v_last_ts + skew_th;
-				if (context->v_target_ts == 0)
-					context->v_target_ts = 1;
+				context->target_ts = context->last_ts + skew_th;
+				if(context->target_ts == 0)
+					context->target_ts = 1;
 				/* Adjust seq num offset */
-				context->v_seq_offset--;
+				context->seq_offset--;
 				exit_status = -1;
 			}
 		}
@@ -581,108 +581,64 @@ int janus_rtp_skew_compensate_video(janus_rtp_header *header, janus_rtp_switchin
 
 	/* Skew compensation */
 	/* Fix header timestamp considering the active offset */
-	guint32 fixed_rtp_ts = context->v_last_ts + context->v_ts_offset;
+	guint32 fixed_rtp_ts = context->last_ts + context->ts_offset;
 	header->timestamp = htonl(fixed_rtp_ts);
 	/* Fix header sequence number considering the total offset */
-	guint16 fixed_rtp_seq = context->v_last_seq + context->v_seq_offset;
+	guint16 fixed_rtp_seq = context->last_seq + context->seq_offset;
 	header->seq_number = htons(fixed_rtp_seq);
 
 	return exit_status;
 }
 
-void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step) {
+void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video) {
 	if(header == NULL || context == NULL)
 		return;
-	/* Note: while the step property is still there for compatibility reasons, to
-	 * keep the signature as it was before, it's ignored: whenever there's a switch
-	 * to take into account, we compute how much time passed between the last RTP
-	 * packet with the old SSRC and this new one, and prepare a timestamp accordingly */
 	uint32_t ssrc = ntohl(header->ssrc);
 	uint32_t timestamp = ntohl(header->timestamp);
 	uint16_t seq = ntohs(header->seq_number);
-	if(video) {
-		if(ssrc != context->v_last_ssrc) {
-			/* Video SSRC changed: update both sequence number and timestamp */
-			JANUS_LOG(LOG_VERB, "Video SSRC changed, %"SCNu32" --> %"SCNu32"\n",
-				context->v_last_ssrc, ssrc);
-			context->v_last_ssrc = ssrc;
-			context->v_base_ts_prev = context->v_last_ts;
-			context->v_base_ts = timestamp;
-			context->v_base_seq_prev = context->v_last_seq;
-			context->v_base_seq = seq;
-			/* How much time since the last video RTP packet? We compute an offset accordingly */
-			if(context->v_last_time > 0) {
-				gint64 time_diff = janus_get_monotonic_time() - context->v_last_time;
-				time_diff = (time_diff*90)/1000; 	/* We're assuming 90khz here */
-				if(time_diff == 0)
-					time_diff = 1;
-				context->v_base_ts_prev += (guint32)time_diff;
-				context->v_last_ts += (guint32)time_diff;
-				JANUS_LOG(LOG_VERB, "Computed offset for video RTP timestamp: %"SCNu32"\n", (guint32)time_diff);
-			}
-			/* Reset skew compensation data */
-			context->v_new_ssrc = TRUE;
-		}
-		if(context->v_seq_reset) {
-			/* Video sequence number was paused for a while: just update that */
-			context->v_seq_reset = FALSE;
-			context->v_base_seq_prev = context->v_last_seq;
-			context->v_base_seq = seq;
+	if(ssrc != context->last_ssrc) {
+		/* Audio SSRC changed: update both sequence number and timestamp */
+		JANUS_LOG(LOG_VERB, "Audio SSRC changed, %"SCNu32" --> %"SCNu32"\n",
+			context->last_ssrc, ssrc);
+		context->last_ssrc = ssrc;
+		context->base_ts_prev = context->last_ts;
+		context->base_ts = timestamp;
+		context->base_seq_prev = context->last_seq;
+		context->base_seq = seq;
+		/* How much time since the last audio RTP packet? We compute an offset accordingly */
+		if(context->last_time > 0) {
+			gint64 time_diff = janus_get_monotonic_time() - context->last_time;
+			/* We're assuming 90khz for video and 48khz for audio, here */
+			int khz = video ? 90 : 48;
+			if(!video && (header->type == 0 || header->type == 8 || header->type == 9))
+				khz = 8;	/* We're assuming 48khz for audio (Opus), unless it's G.711/G.722 (8khz) */
+			time_diff = (time_diff*khz)/1000;
+			if(time_diff == 0)
+				time_diff = 1;
+			context->base_ts_prev += (guint32)time_diff;
+			context->prev_ts += (guint32)time_diff;
+			context->last_ts += (guint32)time_diff;
+			JANUS_LOG(LOG_VERB, "Computed offset for audio RTP timestamp: %"SCNu32"\n", (guint32)time_diff);
 		}
-		/* Compute a coherent timestamp and sequence number */
-		context->v_prev_ts = context->v_last_ts;
-		context->v_last_ts = (timestamp-context->v_base_ts) + context->v_base_ts_prev;
-		context->v_prev_seq = context->v_last_seq;
-		context->v_last_seq = (seq-context->v_base_seq)+context->v_base_seq_prev+1;
-		/* Update the timestamp and sequence number in the RTP packet */
-		header->timestamp = htonl(context->v_last_ts);
-		header->seq_number = htons(context->v_last_seq);
-		/* Take note of when we last handled this RTP packet */
-		context->v_last_time = janus_get_monotonic_time();
-	} else {
-		if(ssrc != context->a_last_ssrc) {
-			/* Audio SSRC changed: update both sequence number and timestamp */
-			JANUS_LOG(LOG_VERB, "Audio SSRC changed, %"SCNu32" --> %"SCNu32"\n",
-				context->a_last_ssrc, ssrc);
-			context->a_last_ssrc = ssrc;
-			context->a_base_ts_prev = context->a_last_ts;
-			context->a_base_ts = timestamp;
-			context->a_base_seq_prev = context->a_last_seq;
-			context->a_base_seq = seq;
-			/* How much time since the last audio RTP packet? We compute an offset accordingly */
-			if(context->a_last_time > 0) {
-				gint64 time_diff = janus_get_monotonic_time() - context->a_last_time;
-				int akhz = 48;
-				if(header->type == 0 || header->type == 8 || header->type == 9)
-					akhz = 8;	/* We're assuming 48khz here (Opus), unless it's G.711/G.722 (8khz) */
-				time_diff = (time_diff*akhz)/1000;
-				if(time_diff == 0)
-					time_diff = 1;
-				context->a_base_ts_prev += (guint32)time_diff;
-				context->a_prev_ts += (guint32)time_diff;
-				context->a_last_ts += (guint32)time_diff;
-				JANUS_LOG(LOG_VERB, "Computed offset for audio RTP timestamp: %"SCNu32"\n", (guint32)time_diff);
-			}
-			/* Reset skew compensation data */
-			context->a_new_ssrc = TRUE;
-		}
-		if(context->a_seq_reset) {
-			/* Audio sequence number was paused for a while: just update that */
-			context->a_seq_reset = FALSE;
-			context->a_base_seq_prev = context->a_last_seq;
-			context->a_base_seq = seq;
-		}
-		/* Compute a coherent timestamp and sequence number */
-		context->a_prev_ts = context->a_last_ts;
-		context->a_last_ts = (timestamp-context->a_base_ts) + context->a_base_ts_prev;
-		context->a_prev_seq = context->a_last_seq;
-		context->a_last_seq = (seq-context->a_base_seq)+context->a_base_seq_prev+1;
-		/* Update the timestamp and sequence number in the RTP packet */
-		header->timestamp = htonl(context->a_last_ts);
-		header->seq_number = htons(context->a_last_seq);
-		/* Take note of when we last handled this RTP packet */
-		context->a_last_time = janus_get_monotonic_time();
-	}
+		/* Reset skew compensation data */
+		context->new_ssrc = TRUE;
+	}
+	if(context->seq_reset) {
+		/* Audio sequence number was paused for a while: just update that */
+		context->seq_reset = FALSE;
+		context->base_seq_prev = context->last_seq;
+		context->base_seq = seq;
+	}
+	/* Compute a coherent timestamp and sequence number */
+	context->prev_ts = context->last_ts;
+	context->last_ts = (timestamp-context->base_ts) + context->base_ts_prev;
+	context->prev_seq = context->last_seq;
+	context->last_seq = (seq-context->base_seq)+context->base_seq_prev+1;
+	/* Update the timestamp and sequence number in the RTP packet */
+	header->timestamp = htonl(context->last_ts);
+	header->seq_number = htons(context->last_seq);
+	/* Take note of when we last handled this RTP packet */
+	context->last_time = janus_get_monotonic_time();
 }
 
 
@@ -1028,7 +984,7 @@ gboolean janus_rtp_simulcasting_context_process_rtp(janus_rtp_simulcasting_conte
 					tid, context->templayer);
 				/* We increase the base sequence number, or there will be gaps when delivering later */
 				if(sc)
-					sc->v_base_seq++;
+					sc->base_seq++;
 				return FALSE;
 			}
 		}
@@ -1049,7 +1005,7 @@ gboolean janus_rtp_simulcasting_context_process_rtp(janus_rtp_simulcasting_conte
 					tid, context->templayer);
 				/* We increase the base sequence number, or there will be gaps when delivering later */
 				if(sc)
-					sc->v_base_seq++;
+					sc->base_seq++;
 				return FALSE;
 			}
 		}
diff --git a/rtp.h b/rtp.h
index 0f4b445d3c..e66675597f 100644
--- a/rtp.h
+++ b/rtp.h
@@ -221,18 +221,12 @@ int janus_rtp_header_extension_replace_id(char *buf, int len, int id, int new_id
 
 /*! \brief RTP context, in order to make sure SSRC changes result in coherent seq/ts increases */
 typedef struct janus_rtp_switching_context {
-	uint32_t a_last_ssrc, a_last_ts, a_base_ts, a_base_ts_prev, a_prev_ts, a_target_ts, a_start_ts,
-			v_last_ssrc, v_last_ts, v_base_ts, v_base_ts_prev, v_prev_ts, v_target_ts, v_start_ts;
-	uint16_t a_last_seq, a_prev_seq, a_base_seq, a_base_seq_prev,
-			v_last_seq, v_prev_seq, v_base_seq, v_base_seq_prev;
-	gboolean a_seq_reset, a_new_ssrc,
-			v_seq_reset, v_new_ssrc;
-	gint16 a_seq_offset,
-			v_seq_offset;
-	gint32 a_prev_delay, a_active_delay, a_ts_offset,
-			v_prev_delay, v_active_delay, v_ts_offset;
-	gint64 a_last_time, a_reference_time, a_start_time, a_evaluating_start_time,
-			v_last_time, v_reference_time, v_start_time, v_evaluating_start_time;
+	uint32_t last_ssrc, last_ts, base_ts, base_ts_prev, prev_ts, target_ts, start_ts;
+	uint16_t last_seq, prev_seq, base_seq, base_seq_prev;
+	gboolean seq_reset, new_ssrc;
+	gint16 seq_offset;
+	gint32 prev_delay, active_delay, ts_offset;
+	gint64 last_time, reference_time, start_time, evaluating_start_time;
 } janus_rtp_switching_context;
 
 /*! \brief Set (or reset) the context fields to their default values
@@ -242,9 +236,8 @@ void janus_rtp_switching_context_reset(janus_rtp_switching_context *context);
 /*! \brief Use the context info to update the RTP header of a packet, if needed
  * @param[in] header The RTP header to update
  * @param[in] context The context to use as a reference
- * @param[in] video Whether this is an audio or a video packet
- * @param[in] step \b deprecated The expected timestamp step */
-void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video, int step);
+ * @param[in] video Whether this is an audio or a video packet */
+void janus_rtp_header_update(janus_rtp_header *header, janus_rtp_switching_context *context, gboolean video);
 
 #define RTP_AUDIO_SKEW_TH_MS 120
 #define RTP_VIDEO_SKEW_TH_MS 120
diff --git a/sctp.c b/sctp.c
index cc4a8dde75..0c6c1a842c 100644
--- a/sctp.c
+++ b/sctp.c
@@ -112,6 +112,7 @@ void janus_sctp_deinit(void) {
 static void janus_sctp_association_free(const janus_refcount *sctp_ref) {
 	janus_sctp_association *sctp = janus_refcount_containerof(sctp_ref, janus_sctp_association, ref);
 	/* This association can be destroyed, free all the resources */
+	janus_refcount_decrease(&sctp->handle->pc->ref);
 	janus_refcount_decrease(&sctp->handle->ref);
 	janus_refcount_decrease(&sctp->dtls->ref);
 #ifdef DEBUG_SCTP
@@ -124,7 +125,7 @@ static void janus_sctp_association_free(const janus_refcount *sctp_ref) {
 	sctp = NULL;
 }
 
-janus_sctp_association *janus_sctp_association_create(janus_dtls_srtp *dtls, janus_ice_handle *handle, uint16_t udp_port) {
+janus_sctp_association *janus_sctp_association_create(janus_dtls_srtp *dtls, janus_handle *handle, uint16_t udp_port) {
 	if(dtls == NULL || handle == NULL || udp_port == 0)
 		return NULL;
 
@@ -139,6 +140,7 @@ janus_sctp_association *janus_sctp_association_create(janus_dtls_srtp *dtls, jan
 	janus_refcount_increase(&dtls->ref);
 	sctp->handle = handle;
 	janus_refcount_increase(&handle->ref);
+	janus_refcount_increase(&handle->pc->ref);
 	sctp->handle_id = handle->handle_id;
 	sctp->local_port = 5000;	/* FIXME We always use this one */
 	sctp->remote_port = udp_port;
diff --git a/sctp.h b/sctp.h
index 4caf571276..0cf2203b64 100644
--- a/sctp.h
+++ b/sctp.h
@@ -75,7 +75,7 @@ void janus_sctp_deinit(void);
 #define DATA_CHANNEL_FLAGS_SEND_ACK 0x00000004
 
 struct janus_dtls_srtp;
-struct janus_ice_handle;
+struct janus_handle;
 
 typedef struct janus_sctp_channel {
 	/*! \brief SCTP channel ID */
@@ -100,7 +100,7 @@ typedef struct janus_sctp_association {
 	/*! \brief Pointer to the DTLS instance related to this SCTP association */
 	struct janus_dtls_srtp *dtls;
 	/*! \brief Pointer to the ICE handle related to this SCTP association */
-	struct janus_ice_handle *handle;
+	struct janus_handle *handle;
 	/*! \brief Identifier of the handle owning this SCTP association (for debugging purposes only) */
 	uint64_t handle_id;
 	/*! \brief Array of SCTP channels */
@@ -188,7 +188,7 @@ typedef struct janus_datachannel_ack {
  * \param[in] handle Pointer to the ICE handle that will send out SCTP messages.
  * \param[in] udp_port The port as negotiated in the sctpmap attribute (http://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-06)
  * \returns A janus_sctp_association instance if successful, NULL otherwise */
-janus_sctp_association *janus_sctp_association_create(struct janus_dtls_srtp *dtls, struct janus_ice_handle *handle, uint16_t udp_port);
+janus_sctp_association *janus_sctp_association_create(struct janus_dtls_srtp *dtls, struct janus_handle *handle, uint16_t udp_port);
 
 /*! \brief Destroy an existing SCTP association
  * \param[in] sctp The SCTP association to get rid of */
diff --git a/sdp-utils.c b/sdp-utils.c
index ce68a31064..843751facd 100644
--- a/sdp-utils.c
+++ b/sdp-utils.c
@@ -19,7 +19,7 @@
 #include "utils.h"
 #include "debug.h"
 
-#define JANUS_BUFSIZE	8192
+#define JANUS_BUFSIZE	16384
 
 /* Preferred codecs when negotiating audio/video, and number of supported codecs */
 const char *janus_preferred_audio_codecs[] = {
@@ -138,6 +138,19 @@ janus_sdp_mline *janus_sdp_mline_find(janus_sdp *sdp, janus_sdp_mtype type) {
 	return NULL;
 }
 
+janus_sdp_mline *janus_sdp_mline_find_by_index(janus_sdp *sdp, int index) {
+	if(sdp == NULL || index < 0)
+		return NULL;
+	GList *ml = sdp->m_lines;
+	while(ml) {
+		janus_sdp_mline *m = (janus_sdp_mline *)ml->data;
+		if(m->index == index)
+			return m;
+		ml = ml->next;
+	}
+	return NULL;
+}
+
 int janus_sdp_mline_remove(janus_sdp *sdp, janus_sdp_mtype type) {
 	if(sdp == NULL)
 		return -1;
@@ -242,6 +255,38 @@ const char *janus_sdp_mdirection_str(janus_sdp_mdirection direction) {
 	return NULL;
 }
 
+const char *janus_sdp_oa_type_str(janus_sdp_oa_type type) {
+	switch(type) {
+		case JANUS_SDP_OA_MLINE:
+			return "JANUS_SDP_OA_MLINE";
+		case JANUS_SDP_OA_ENABLED:
+			return "JANUS_SDP_OA_ENABLED";
+		case JANUS_SDP_OA_DIRECTION:
+			return "JANUS_SDP_OA_DIRECTION";
+		case JANUS_SDP_OA_MID:
+			return "JANUS_SDP_OA_MID";
+		case JANUS_SDP_OA_CODEC:
+			return "JANUS_SDP_OA_CODEC";
+		case JANUS_SDP_OA_PT:
+			return "JANUS_SDP_OA_PT";
+		case JANUS_SDP_OA_AUDIO_DTMF:
+			return "JANUS_SDP_OA_AUDIO_DTMF";
+		case JANUS_SDP_OA_FMTP:
+			return "JANUS_SDP_OA_FMTP";
+		case JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS:
+			return "JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS";
+		case JANUS_SDP_OA_VIDEO_H264_FMTP:
+			return "JANUS_SDP_OA_VIDEO_H264_FMTP";
+		case JANUS_SDP_OA_DATA_LEGACY:
+			return "JANUS_SDP_OA_DATA_LEGACY";
+		case JANUS_SDP_OA_DONE:
+			return "JANUS_SDP_OA_DONE";
+		default:
+			break;
+	}
+	return NULL;
+}
+
 janus_sdp *janus_sdp_parse(const char *sdp, char *error, size_t errlen) {
 	if(!sdp)
 		return NULL;
@@ -258,6 +303,7 @@ janus_sdp *janus_sdp_parse(const char *sdp, char *error, size_t errlen) {
 
 	gboolean success = TRUE;
 	janus_sdp_mline *mline = NULL;
+	int mlines = 0;
 
 	gchar **parts = g_strsplit(sdp, "\n", -1);
 	if(parts) {
@@ -425,6 +471,8 @@ janus_sdp *janus_sdp_parse(const char *sdp, char *error, size_t errlen) {
 							success = FALSE;
 							break;
 						}
+						m->index = mlines;
+						mlines++;
 						m->type = janus_sdp_parse_mtype(type);
 						m->type_str = g_strdup(type);
 						m->proto = g_strdup(proto);
@@ -600,12 +648,16 @@ janus_sdp *janus_sdp_parse(const char *sdp, char *error, size_t errlen) {
 	return imported;
 }
 
-int janus_sdp_remove_payload_type(janus_sdp *sdp, int pt) {
+int janus_sdp_remove_payload_type(janus_sdp *sdp, int index, int pt) {
 	if(!sdp || pt < 0)
 		return -1;
 	GList *ml = sdp->m_lines;
 	while(ml) {
 		janus_sdp_mline *m = (janus_sdp_mline *)ml->data;
+		if(index != -1 && index != m->index) {
+			ml = ml->next;
+			continue;
+		}
 		/* Remove any reference from the m-line */
 		m->ptypes = g_list_remove(m->ptypes, GINT_TO_POINTER(pt));
 		/* Also remove all attributes that reference the same payload type */
@@ -620,12 +672,14 @@ int janus_sdp_remove_payload_type(janus_sdp *sdp, int pt) {
 			}
 			ma = ma->next;
 		}
+		if(index != -1)
+			break;
 		ml = ml->next;
 	}
 	return 0;
 }
 
-int janus_sdp_get_codec_pt(janus_sdp *sdp, const char *codec) {
+int janus_sdp_get_codec_pt(janus_sdp *sdp, int index, const char *codec) {
 	if(sdp == NULL || codec == NULL)
 		return -1;
 	/* Check the format string (note that we only parse what browsers can negotiate) */
@@ -679,6 +733,10 @@ int janus_sdp_get_codec_pt(janus_sdp *sdp, const char *codec) {
 			ml = ml->next;
 			continue;
 		}
+		if(index != -1 && index != m->index) {
+			ml = ml->next;
+			continue;
+		}
 		/* Look in all rtpmap attributes */
 		GList *ma = m->attributes;
 		while(ma) {
@@ -690,12 +748,14 @@ int janus_sdp_get_codec_pt(janus_sdp *sdp, const char *codec) {
 			}
 			ma = ma->next;
 		}
+		if(index != -1)
+			break;
 		ml = ml->next;
 	}
 	return -1;
 }
 
-const char *janus_sdp_get_codec_name(janus_sdp *sdp, int pt) {
+const char *janus_sdp_get_codec_name(janus_sdp *sdp, int index, int pt) {
 	if(sdp == NULL || pt < 0)
 		return NULL;
 	if(pt == 0)
@@ -707,6 +767,10 @@ const char *janus_sdp_get_codec_name(janus_sdp *sdp, int pt) {
 	GList *ml = sdp->m_lines;
 	while(ml) {
 		janus_sdp_mline *m = (janus_sdp_mline *)ml->data;
+		if(index != -1 && index != m->index) {
+			ml = ml->next;
+			continue;
+		}
 		/* Look in all rtpmap attributes */
 		GList *ma = m->attributes;
 		while(ma) {
@@ -741,6 +805,8 @@ const char *janus_sdp_get_codec_name(janus_sdp *sdp, int pt) {
 			}
 			ma = ma->next;
 		}
+		if(index != -1)
+			break;
 		ml = ml->next;
 	}
 	return NULL;
@@ -886,80 +952,62 @@ char *janus_sdp_write(janus_sdp *imported) {
 	return sdp;
 }
 
-void janus_sdp_find_preferred_codecs(janus_sdp *sdp, const char **acodec, const char **vcodec) {
+void janus_sdp_find_preferred_codec(janus_sdp *sdp, janus_sdp_mtype type, int index, const char **codec) {
 	if(sdp == NULL)
 		return;
 	janus_refcount_increase(&sdp->ref);
-	gboolean audio = FALSE, video = FALSE;
+	gboolean found = FALSE;
 	GList *temp = sdp->m_lines;
 	while(temp) {
 		/* Which media are available? */
 		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
-		if(m->type == JANUS_SDP_AUDIO && m->port > 0 && m->direction != JANUS_SDP_INACTIVE) {
-			if(audio == FALSE) {
-				uint i=0;
-				for(i=0; i<janus_audio_codecs; i++) {
-					if(janus_sdp_get_codec_pt(sdp, janus_preferred_audio_codecs[i]) > 0) {
-						audio = TRUE;
-						if(acodec)
-							*acodec = janus_preferred_audio_codecs[i];
-						break;
-					}
-				}
-			}
-		} else if(m->type == JANUS_SDP_VIDEO && m->port > 0 && m->direction != JANUS_SDP_INACTIVE) {
-			if(video == FALSE) {
-				uint i=0;
-				for(i=0; i<janus_video_codecs; i++) {
-					if(janus_sdp_get_codec_pt(sdp, janus_preferred_video_codecs[i]) > 0) {
-						video = TRUE;
-						if(vcodec)
-							*vcodec = janus_preferred_video_codecs[i];
-						break;
-					}
+		if(index != -1 && index != m->index) {
+			temp = temp->next;
+			continue;
+		}
+		if(m->type == type && m->port > 0 && m->direction != JANUS_SDP_INACTIVE) {
+			uint i=0;
+			for(i=0; i<(type == JANUS_SDP_AUDIO ? janus_audio_codecs : janus_video_codecs); i++) {
+				if(janus_sdp_get_codec_pt(sdp, m->index,
+						type == JANUS_SDP_AUDIO ? janus_preferred_audio_codecs[i] : janus_preferred_video_codecs[i]) > 0) {
+					found = TRUE;
+					if(codec)
+						*codec = (type == JANUS_SDP_AUDIO ? janus_preferred_audio_codecs[i] : janus_preferred_video_codecs[i]);
+					break;
 				}
 			}
 		}
-		if(audio && video)
+		if(found || index != -1)
 			break;
 		temp = temp->next;
 	}
 	janus_refcount_decrease(&sdp->ref);
 }
 
-void janus_sdp_find_first_codecs(janus_sdp *sdp, const char **acodec, const char **vcodec) {
+void janus_sdp_find_first_codecs(janus_sdp *sdp, janus_sdp_mtype type, int index, const char **codec) {
 	if(sdp == NULL)
 		return;
 	janus_refcount_increase(&sdp->ref);
-	gboolean audio = FALSE, video = FALSE;
+	gboolean found = FALSE;
 	GList *temp = sdp->m_lines;
 	while(temp) {
 		/* Which media are available? */
 		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
-		if(m->type == JANUS_SDP_AUDIO && m->port > 0 && m->direction != JANUS_SDP_INACTIVE) {
-			if(audio == FALSE && m->ptypes) {
-				int pt = GPOINTER_TO_INT(m->ptypes->data);
-				const char *codec = janus_sdp_get_codec_name(sdp, pt);
-				codec = janus_sdp_match_preferred_codec(m->type, (char *)codec);
-				if(codec) {
-					audio = TRUE;
-					if(acodec)
-						*acodec = codec;
-				}
-			}
-		} else if(m->type == JANUS_SDP_VIDEO && m->port > 0 && m->direction != JANUS_SDP_INACTIVE) {
-			if(video == FALSE && m->ptypes) {
-				int pt = GPOINTER_TO_INT(m->ptypes->data);
-				const char *codec = janus_sdp_get_codec_name(sdp, pt);
-				codec = janus_sdp_match_preferred_codec(m->type, (char *)codec);
-				if(codec) {
-					video = TRUE;
-					if(vcodec)
-						*vcodec = codec;
-				}
+		if(index != -1 && index != m->index) {
+			temp = temp->next;
+			continue;
+		}
+		if(m->type == type && m->port > 0 && m->direction != JANUS_SDP_INACTIVE && m->ptypes) {
+			int pt = GPOINTER_TO_INT(m->ptypes->data);
+			const char *c = janus_sdp_get_codec_name(sdp, m->index, pt);
+			c = janus_sdp_match_preferred_codec(m->type, (char *)c);
+			if(c) {
+				found = TRUE;
+				if(codec)
+					*codec = c;
 			}
 		}
-		if(audio && video)
+		if(found || index != -1)
 			break;
 		temp = temp->next;
 	}
@@ -1006,39 +1054,124 @@ janus_sdp *janus_sdp_new(const char *name, const char *address) {
 
 janus_sdp *janus_sdp_generate_offer(const char *name, const char *address, ...) {
 	/* This method has a variable list of arguments, telling us what we should offer */
+	int property = -1;
 	va_list args;
 	va_start(args, address);
-	/* Let's see what we should do with the media */
-	gboolean do_audio = TRUE, do_video = TRUE, do_data = TRUE,
-		audio_dtmf = FALSE, video_rtcpfb = TRUE, h264_fmtp = TRUE,
-		data_legacy = TRUE;
-	const char *audio_codec = NULL, *video_codec = NULL, *audio_fmtp = NULL;
-	int audio_pt = 111, video_pt = 96;
-	janus_sdp_mdirection audio_dir = JANUS_SDP_SENDRECV, video_dir = JANUS_SDP_SENDRECV;
-	int property = va_arg(args, int);
+
+	/* Create a new janus_sdp object */
+	janus_sdp *offer = janus_sdp_new(name, address);
+
+	/* Iterate on all m-lines to add, if any */
+	gboolean new_mline = FALSE, mline_enabled = TRUE;
+	janus_sdp_mtype type = JANUS_SDP_OTHER;
+	gboolean audio_dtmf = FALSE, video_rtcpfb = TRUE, h264_fmtp = TRUE, data_legacy = FALSE;
+	int pt = -1;
+	const char *codec = NULL, *mid = NULL, *fmtp = NULL;
+	janus_sdp_mdirection mdir = JANUS_SDP_DEFAULT;
 	while(property != JANUS_SDP_OA_DONE) {
-		if(property == JANUS_SDP_OA_AUDIO) {
-			do_audio = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_VIDEO) {
-			do_video = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_DATA) {
-			do_data = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_AUDIO_DIRECTION) {
-			audio_dir = va_arg(args, janus_sdp_mdirection);
-		} else if(property == JANUS_SDP_OA_VIDEO_DIRECTION) {
-			video_dir = va_arg(args, janus_sdp_mdirection);
-		} else if(property == JANUS_SDP_OA_AUDIO_CODEC) {
-			audio_codec = va_arg(args, char *);
-		} else if(property == JANUS_SDP_OA_VIDEO_CODEC) {
-			video_codec = va_arg(args, char *);
-		} else if(property == JANUS_SDP_OA_AUDIO_PT) {
-			audio_pt = va_arg(args, int);
-		} else if(property == JANUS_SDP_OA_VIDEO_PT) {
-			video_pt = va_arg(args, int);
+		property = va_arg(args, int);
+		if(!new_mline && property != JANUS_SDP_OA_MLINE && property != JANUS_SDP_OA_DONE) {
+			/* The first attribute MUST be JANUS_SDP_OA_MLINE or JANUS_SDP_OA_DONE */
+			JANUS_LOG(LOG_ERR, "First attribute is not JANUS_SDP_OA_MLINE or JANUS_SDP_OA_DONE\n");
+			janus_sdp_destroy(offer);
+			va_end(args);
+			return NULL;
+		}
+		if(property == JANUS_SDP_OA_MLINE || property == JANUS_SDP_OA_DONE) {
+			/* A new m-line is starting or we're done, should we wrap the previous one? */
+			new_mline = TRUE;
+			if(mline_enabled) {
+				/* Create a new m-line with the data collected so far */
+				if(type == JANUS_SDP_AUDIO) {
+					if(janus_sdp_generate_offer_mline(offer,
+						JANUS_SDP_OA_MLINE, JANUS_SDP_AUDIO,
+						JANUS_SDP_OA_MID, mid,
+						JANUS_SDP_OA_PT, pt,
+						JANUS_SDP_OA_CODEC, codec,
+						JANUS_SDP_OA_DIRECTION, mdir,
+						JANUS_SDP_OA_FMTP, fmtp,
+						JANUS_SDP_OA_AUDIO_DTMF, audio_dtmf,
+						JANUS_SDP_OA_DONE
+					) < 0) {
+						janus_sdp_destroy(offer);
+						va_end(args);
+						return NULL;
+					}
+				} else if(type == JANUS_SDP_VIDEO) {
+					if(janus_sdp_generate_offer_mline(offer,
+						JANUS_SDP_OA_MLINE, JANUS_SDP_VIDEO,
+						JANUS_SDP_OA_MID, mid,
+						JANUS_SDP_OA_PT, pt,
+						JANUS_SDP_OA_CODEC, codec,
+						JANUS_SDP_OA_DIRECTION, mdir,
+						JANUS_SDP_OA_FMTP, fmtp,
+						JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS, video_rtcpfb,
+						JANUS_SDP_OA_VIDEO_H264_FMTP, h264_fmtp,
+						JANUS_SDP_OA_DONE
+					) < 0) {
+						janus_sdp_destroy(offer);
+						va_end(args);
+						return NULL;
+					}
+				} else if(type == JANUS_SDP_APPLICATION) {
+					if(janus_sdp_generate_offer_mline(offer,
+						JANUS_SDP_OA_MLINE, JANUS_SDP_APPLICATION,
+						JANUS_SDP_OA_MID, mid,
+						JANUS_SDP_OA_DATA_LEGACY, data_legacy,
+						JANUS_SDP_OA_DONE
+					) < 0) {
+						janus_sdp_destroy(offer);
+						va_end(args);
+						return NULL;
+					}
+				}
+			}
+			if(property != JANUS_SDP_OA_MLINE)
+				continue;
+			/* Now reset the properties */
+			audio_dtmf = FALSE;
+			video_rtcpfb = TRUE;
+			h264_fmtp = TRUE;
+			data_legacy = FALSE;
+			pt = -1;
+			codec = NULL;
+			mid = NULL;
+			fmtp = NULL;
+			mdir = JANUS_SDP_DEFAULT;
+			/* The value of JANUS_SDP_OA_MLINE MUST be the media we want to add */
+			type = va_arg(args, int);
+			if(type == JANUS_SDP_AUDIO) {
+				/* Audio */
+				pt = 111;
+				codec = "opus";
+			} else if(type == JANUS_SDP_VIDEO) {
+				/* Video */
+				pt = 96;
+				codec = "vp8";
+			} else if(type == JANUS_SDP_APPLICATION) {
+				/* Data */
+			} else {
+				/* Unsupported m-line type */
+				JANUS_LOG(LOG_ERR, "Invalid m-line type\n");
+				janus_sdp_destroy(offer);
+				va_end(args);
+				return NULL;
+			}
+			mline_enabled = TRUE;
+		} else if(property == JANUS_SDP_OA_ENABLED) {
+			mline_enabled = va_arg(args, gboolean);
+		} else if(property == JANUS_SDP_OA_DIRECTION) {
+			mdir = va_arg(args, janus_sdp_mdirection);
+		} else if(property == JANUS_SDP_OA_CODEC) {
+			codec = va_arg(args, char *);
+		} else if(property == JANUS_SDP_OA_MID) {
+			mid = va_arg(args, char *);
+		} else if(property == JANUS_SDP_OA_PT) {
+			pt = va_arg(args, int);
+		} else if(property == JANUS_SDP_OA_FMTP) {
+			fmtp = va_arg(args, char *);
 		} else if(property == JANUS_SDP_OA_AUDIO_DTMF) {
 			audio_dtmf = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_AUDIO_FMTP) {
-			audio_fmtp = va_arg(args, char *);
 		} else if(property == JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS) {
 			video_rtcpfb = va_arg(args, gboolean);
 		} else if(property == JANUS_SDP_OA_VIDEO_H264_FMTP) {
@@ -1046,87 +1179,162 @@ janus_sdp *janus_sdp_generate_offer(const char *name, const char *address, ...)
 		} else if(property == JANUS_SDP_OA_DATA_LEGACY) {
 			data_legacy = va_arg(args, gboolean);
 		} else {
-			JANUS_LOG(LOG_WARN, "Unknown property %d for preparing SDP answer, ignoring...\n", property);
+			JANUS_LOG(LOG_WARN, "Unknown property %d for preparing SDP offer, ignoring...\n", property);
 		}
-		property = va_arg(args, int);
 	}
-	if(audio_codec == NULL)
-		audio_codec = "opus";
-	const char *audio_rtpmap = janus_sdp_get_codec_rtpmap(audio_codec);
-	if(do_audio && audio_rtpmap == NULL) {
-		JANUS_LOG(LOG_ERR, "Unsupported audio codec '%s', can't prepare an offer\n", audio_codec);
-		va_end(args);
-		return NULL;
-	}
-	if(video_codec == NULL)
-		video_codec = "vp8";
-	const char *video_rtpmap = janus_sdp_get_codec_rtpmap(video_codec);
-	if(do_video && video_rtpmap == NULL) {
-		JANUS_LOG(LOG_ERR, "Unsupported video codec '%s', can't prepare an offer\n", video_codec);
+
+	/* Done */
+	va_end(args);
+
+	return offer;
+}
+
+int janus_sdp_generate_offer_mline(janus_sdp *offer, ...) {
+	if(offer == NULL)
+		return -1;
+
+	/* This method has a variable list of arguments, telling us what we should offer */
+	va_list args;
+	va_start(args, offer);
+
+	/* First of all, let's see what we should add */
+	janus_sdp_mtype type = JANUS_SDP_OTHER;
+	gboolean audio_dtmf = FALSE, video_rtcpfb = TRUE, h264_fmtp = TRUE, data_legacy = FALSE;
+	int pt = -1;
+	const char *codec = NULL, *mid = NULL, *rtpmap = NULL, *fmtp = NULL;
+	janus_sdp_mdirection mdir = JANUS_SDP_DEFAULT;
+	int property = va_arg(args, int);
+	if(property != JANUS_SDP_OA_MLINE) {
+		/* The first attribute MUST be JANUS_SDP_OA_MLINE */
+		JANUS_LOG(LOG_ERR, "First attribute is not JANUS_SDP_OA_MLINE\n");
 		va_end(args);
-		return NULL;
+		return -2;
 	}
+	type = va_arg(args, int);
+	if(type == JANUS_SDP_AUDIO) {
+		/* Audio */
+		pt = 111;
+		codec = "opus";
+	} else if(type == JANUS_SDP_VIDEO) {
+		/* Video */
+		pt = 96;
+		codec = "vp8";
+	} else if(type == JANUS_SDP_APPLICATION) {
+		/* Data */
 #ifndef HAVE_SCTP
-	do_data = FALSE;
+		va_end(args);
+		return -3;
 #endif
+	} else {
+		/* Unsupported m-line type */
+		JANUS_LOG(LOG_ERR, "Invalid m-line type\n");
+		janus_sdp_destroy(offer);
+		va_end(args);
+		return -4;
+	}
 
-	/* Create a new janus_sdp object */
-	janus_sdp *offer = janus_sdp_new(name, address);
-	/* Now add all the media we should */
-	if(do_audio) {
-		janus_sdp_mline *m = janus_sdp_mline_create(JANUS_SDP_AUDIO, 1, "UDP/TLS/RTP/SAVPF", audio_dir);
-		m->c_ipv4 = TRUE;
-		m->c_addr = g_strdup(offer->c_addr);
-		/* Add the selected audio codec */
-		m->ptypes = g_list_append(m->ptypes, GINT_TO_POINTER(audio_pt));
-		janus_sdp_attribute *a = janus_sdp_attribute_create("rtpmap", "%d %s", audio_pt, audio_rtpmap);
-		m->attributes = g_list_append(m->attributes, a);
-		/* Check if we need to add a payload type for DTMF tones (telephone-event/8000) */
-		if(audio_dtmf) {
-			/* We do */
-			int dtmf_pt = 126;
-			m->ptypes = g_list_append(m->ptypes, GINT_TO_POINTER(dtmf_pt));
-			janus_sdp_attribute *a = janus_sdp_attribute_create("rtpmap", "%d %s", dtmf_pt, janus_sdp_get_codec_rtpmap("dtmf"));
-			m->attributes = g_list_append(m->attributes, a);
-		}
-		/* Check if there's a custom fmtp line to add for audio */
-		if(audio_fmtp) {
-			janus_sdp_attribute *a = janus_sdp_attribute_create("fmtp", "%d %s", audio_pt, audio_fmtp);
-			m->attributes = g_list_append(m->attributes, a);
+	/* Let's see what we should do with the media to add */
+	property = va_arg(args, int);
+	while(property != JANUS_SDP_OA_DONE) {
+		if(property == JANUS_SDP_OA_DIRECTION) {
+			mdir = va_arg(args, janus_sdp_mdirection);
+		} else if(property == JANUS_SDP_OA_CODEC) {
+			codec = va_arg(args, char *);
+		} else if(property == JANUS_SDP_OA_MID) {
+			mid = va_arg(args, char *);
+		} else if(property == JANUS_SDP_OA_PT) {
+			pt = va_arg(args, int);
+		} else if(property == JANUS_SDP_OA_FMTP) {
+			fmtp = va_arg(args, char *);
+		} else if(type == JANUS_SDP_AUDIO && property == JANUS_SDP_OA_AUDIO_DTMF) {
+			audio_dtmf = va_arg(args, gboolean);
+		} else if(type == JANUS_SDP_VIDEO && property == JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS) {
+			video_rtcpfb = va_arg(args, gboolean);
+		} else if(type == JANUS_SDP_VIDEO && property == JANUS_SDP_OA_VIDEO_H264_FMTP) {
+			h264_fmtp = va_arg(args, gboolean);
+		} else if(type == JANUS_SDP_APPLICATION && property == JANUS_SDP_OA_DATA_LEGACY) {
+			data_legacy = va_arg(args, gboolean);
+		} else {
+			JANUS_LOG(LOG_WARN, "Unknown property %d for preparing SDP answer, ignoring...\n", property);
 		}
-		offer->m_lines = g_list_append(offer->m_lines, m);
+		property = va_arg(args, int);
 	}
-	if(do_video) {
-		janus_sdp_mline *m = janus_sdp_mline_create(JANUS_SDP_VIDEO, 1, "UDP/TLS/RTP/SAVPF", video_dir);
-		m->c_ipv4 = TRUE;
-		m->c_addr = g_strdup(offer->c_addr);
-		/* Add the selected video codec */
-		m->ptypes = g_list_append(m->ptypes, GINT_TO_POINTER(video_pt));
-		janus_sdp_attribute *a = janus_sdp_attribute_create("rtpmap", "%d %s", video_pt, video_rtpmap);
-		m->attributes = g_list_append(m->attributes, a);
-		if(!strcasecmp(video_codec, "h264") && h264_fmtp) {
-			/* If it's H.264 and we were asked to, add the default fmtp profile as well */
-			a = janus_sdp_attribute_create("fmtp", "%d profile-level-id=42e01f;packetization-mode=1", video_pt);
-			m->attributes = g_list_append(m->attributes, a);
+	/* Configure some defaults, if values weren't specified */
+	if(type == JANUS_SDP_AUDIO) {
+		if(codec == NULL)
+			codec = "opus";
+		rtpmap = janus_sdp_get_codec_rtpmap(codec);
+		if(rtpmap == NULL) {
+			JANUS_LOG(LOG_ERR, "Unsupported audio codec '%s', can't prepare an offer\n", codec);
+			va_end(args);
+			return -3;
 		}
-		if(video_rtcpfb) {
-			/* Add rtcp-fb attributes */
-			a = janus_sdp_attribute_create("rtcp-fb", "%d ccm fir", video_pt);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("rtcp-fb", "%d nack", video_pt);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("rtcp-fb", "%d nack pli", video_pt);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("rtcp-fb", "%d goog-remb", video_pt);
-			m->attributes = g_list_append(m->attributes, a);
+	} else if(type == JANUS_SDP_VIDEO) {
+		if(codec == NULL)
+			codec = "vp8";
+		rtpmap = janus_sdp_get_codec_rtpmap(codec);
+		if(rtpmap == NULL) {
+			JANUS_LOG(LOG_ERR, "Unsupported video codec '%s', can't prepare an offer\n", codec);
+			va_end(args);
+			return -4;
 		}
-		offer->m_lines = g_list_append(offer->m_lines, m);
 	}
-	if(do_data) {
-		janus_sdp_mline *m = janus_sdp_mline_create(JANUS_SDP_APPLICATION, 1,
-			data_legacy ? "DTLS/SCTP" : "UDP/DTLS/SCTP", JANUS_SDP_DEFAULT);
-		m->c_ipv4 = TRUE;
-		m->c_addr = g_strdup(offer->c_addr);
+
+	/* Create the m-line */
+	const char *transport = "UDP/TLS/RTP/SAVPF";
+	if(type == JANUS_SDP_APPLICATION)
+		transport = (data_legacy ? "DTLS/SCTP" : "UDP/DTLS/SCTP");
+	janus_sdp_mline *m = janus_sdp_mline_create(type, 9, transport, mdir);
+	m->index = g_list_length(offer->m_lines);
+	m->c_ipv4 = TRUE;
+	m->c_addr = g_strdup(offer->c_addr);
+	/* Any mid we should set? */
+	if(mid != NULL) {
+		janus_sdp_attribute *a = janus_sdp_attribute_create("mid", "%s", mid);
+		m->attributes = g_list_append(m->attributes, a);
+	}
+	if(type == JANUS_SDP_AUDIO || type == JANUS_SDP_VIDEO) {
+		/* Add the selected codec */
+		m->ptypes = g_list_append(m->ptypes, GINT_TO_POINTER(pt));
+		janus_sdp_attribute *a = janus_sdp_attribute_create("rtpmap", "%d %s", pt, rtpmap);
+		m->attributes = g_list_append(m->attributes, a);
+		if(type == JANUS_SDP_AUDIO) {
+			/* Check if we need to add a payload type for DTMF tones (telephone-event/8000) */
+			if(audio_dtmf) {
+				/* We do */
+				int dtmf_pt = 126;
+				m->ptypes = g_list_append(m->ptypes, GINT_TO_POINTER(dtmf_pt));
+				janus_sdp_attribute *a = janus_sdp_attribute_create("rtpmap", "%d %s", dtmf_pt, janus_sdp_get_codec_rtpmap("dtmf"));
+				m->attributes = g_list_append(m->attributes, a);
+			}
+			/* Check if there's a custom fmtp line to add for audio */
+			if(fmtp) {
+				janus_sdp_attribute *a = janus_sdp_attribute_create("fmtp", "%d %s", pt, fmtp);
+				m->attributes = g_list_append(m->attributes, a);
+			}
+		} else {
+			/* Check if there's a custom fmtp line to add for video */
+			if(fmtp) {
+				janus_sdp_attribute *a = janus_sdp_attribute_create("fmtp", "%d %s", pt, fmtp);
+				m->attributes = g_list_append(m->attributes, a);
+			} else if(!strcasecmp(codec, "h264") && h264_fmtp) {
+				/* If it's H.264 and we were asked to, add the default fmtp profile as well */
+				a = janus_sdp_attribute_create("fmtp", "%d profile-level-id=42e01f;packetization-mode=1", pt);
+				m->attributes = g_list_append(m->attributes, a);
+			}
+			if(video_rtcpfb) {
+				/* Add rtcp-fb attributes */
+				a = janus_sdp_attribute_create("rtcp-fb", "%d ccm fir", pt);
+				m->attributes = g_list_append(m->attributes, a);
+				a = janus_sdp_attribute_create("rtcp-fb", "%d nack", pt);
+				m->attributes = g_list_append(m->attributes, a);
+				a = janus_sdp_attribute_create("rtcp-fb", "%d nack pli", pt);
+				m->attributes = g_list_append(m->attributes, a);
+				a = janus_sdp_attribute_create("rtcp-fb", "%d goog-remb", pt);
+				m->attributes = g_list_append(m->attributes, a);
+			}
+		}
+	} else {
 		m->fmts = g_list_append(m->fmts, g_strdup(data_legacy ? "5000" : "webrtc-datachannel"));
 		/* Add an sctpmap attribute */
 		if(data_legacy) {
@@ -1136,70 +1344,24 @@ janus_sdp *janus_sdp_generate_offer(const char *name, const char *address, ...)
 			janus_sdp_attribute *aa = janus_sdp_attribute_create("sctp-port", "5000");
 			m->attributes = g_list_append(m->attributes, aa);
 		}
-		offer->m_lines = g_list_append(offer->m_lines, m);
 	}
+	offer->m_lines = g_list_append(offer->m_lines, m);
 
 	/* Done */
 	va_end(args);
 
-	return offer;
+	return 0;
 }
 
-janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
+janus_sdp *janus_sdp_generate_answer(janus_sdp *offer) {
 	if(offer == NULL)
 		return NULL;
 
 	janus_refcount_increase(&offer->ref);
-	/* This method has a variable list of arguments, telling us how we should respond */
-	va_list args;
-	va_start(args, offer);
-	/* Let's see what we should do with the media */
-	gboolean do_audio = TRUE, do_video = TRUE, do_data = TRUE,
-		audio_dtmf = FALSE, video_rtcpfb = TRUE, h264_fmtp = TRUE;
-	const char *audio_codec = NULL, *video_codec = NULL, *audio_fmtp = NULL;
-	GList *extmaps = NULL;
-	janus_sdp_mdirection audio_dir = JANUS_SDP_SENDRECV, video_dir = JANUS_SDP_SENDRECV;
-	int property = va_arg(args, int);
-	while(property != JANUS_SDP_OA_DONE) {
-		if(property == JANUS_SDP_OA_AUDIO) {
-			do_audio = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_VIDEO) {
-			do_video = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_DATA) {
-			do_data = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_AUDIO_DIRECTION) {
-			audio_dir = va_arg(args, janus_sdp_mdirection);
-		} else if(property == JANUS_SDP_OA_VIDEO_DIRECTION) {
-			video_dir = va_arg(args, janus_sdp_mdirection);
-		} else if(property == JANUS_SDP_OA_AUDIO_CODEC) {
-			audio_codec = va_arg(args, char *);
-		} else if(property == JANUS_SDP_OA_VIDEO_CODEC) {
-			video_codec = va_arg(args, char *);
-		} else if(property == JANUS_SDP_OA_AUDIO_DTMF) {
-			audio_dtmf = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_AUDIO_FMTP) {
-			audio_fmtp = va_arg(args, char *);
-		} else if(property == JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS) {
-			video_rtcpfb = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_VIDEO_H264_FMTP) {
-			h264_fmtp = va_arg(args, gboolean);
-		} else if(property == JANUS_SDP_OA_ACCEPT_EXTMAP) {
-			const char *extension = va_arg(args, char *);
-			if(extension != NULL)
-				extmaps = g_list_append(extmaps, (char *)extension);
-		} else {
-			JANUS_LOG(LOG_WARN, "Unknown property %d for preparing SDP answer, ignoring...\n", property);
-		}
-		property = va_arg(args, int);
-	}
-#ifndef HAVE_SCTP
-	do_data = FALSE;
-#endif
-
+	/* Create an SDP answer, and start by copying some of the headers */
 	janus_sdp *answer = g_malloc(sizeof(janus_sdp));
 	g_atomic_int_set(&answer->destroyed, 0);
 	janus_refcount_init(&answer->ref, janus_sdp_free);
-	/* Start by copying some of the headers */
 	answer->version = offer->version;
 	answer->o_name = g_strdup(offer->o_name ? offer->o_name : "-");
 	answer->o_sessid = offer->o_sessid;
@@ -1214,84 +1376,136 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 	answer->attributes = NULL;
 	answer->m_lines = NULL;
 
-	/* Now iterate on all media, and let's see what we should do */
-	int audio = 0, video = 0, data = 0;
+	/* Iterate on all m-lines to add, if any */
 	GList *temp = offer->m_lines;
 	while(temp) {
 		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
 		/* For each m-line we parse, we'll need a corresponding one in the answer */
 		janus_sdp_mline *am = g_malloc0(sizeof(janus_sdp_mline));
-		g_atomic_int_set(&am->destroyed, 0);
 		janus_refcount_init(&am->ref, janus_sdp_mline_free);
+		am->index = m->index;
 		am->type = m->type;
 		am->type_str = m->type_str ? g_strdup(m->type_str) : NULL;
 		am->proto = g_strdup(m->proto ? m->proto : "UDP/TLS/RTP/SAVPF");
-		am->port = m->port;
 		am->c_ipv4 = m->c_ipv4;
 		am->c_addr = g_strdup(am->c_addr ? am->c_addr : "127.0.0.1");
-		am->direction = JANUS_SDP_INACTIVE;	/* We'll change this later */
+		/* We reject the media line by default, but this can be changed later */
+		am->port = 0;
+		am->direction = JANUS_SDP_INACTIVE;
+		am->ptypes = g_list_append(am->ptypes, GINT_TO_POINTER(0));
 		/* Append to the list of m-lines in the answer */
 		answer->m_lines = g_list_append(answer->m_lines, am);
-		/* Let's see what this is */
-		if(m->type == JANUS_SDP_AUDIO) {
-			if(m->port > 0) {
-				audio++;
-			}
-			if(!do_audio || audio > 1) {
-				/* Reject */
-				if(audio > 1)
-					am->port = 0;
-				temp = temp->next;
-				continue;
-			}
-		} else if(m->type == JANUS_SDP_VIDEO && m->port > 0) {
-			if(m->port > 0) {
-				video++;
-			}
-			if(!do_video || video > 1) {
-				/* Reject */
-				if(video > 1)
-					am->port = 0;
-				temp = temp->next;
-				continue;
-			}
-		} else if(m->type == JANUS_SDP_APPLICATION && m->port > 0) {
-			if(m->port > 0) {
-				data++;
-			}
-			if(!do_data || data > 1) {
-				/* Reject */
-				am->port = 0;
-				temp = temp->next;
-				continue;
-			}
+		temp = temp->next;
+	}
+	janus_refcount_decrease(&offer->ref);
+
+	/* Done*/
+	return answer;
+}
+
+int janus_sdp_generate_answer_mline(janus_sdp *offer, janus_sdp *answer, janus_sdp_mline *offered, ...) {
+	if(answer == NULL || offered == NULL)
+		return -1;
+
+	/* This method has a variable list of arguments, telling us what we should offer */
+	va_list args;
+	va_start(args, offered);
+
+	/* First of all, let's see what we should add */
+	gboolean mline_enabled = TRUE;
+	janus_sdp_mtype type = JANUS_SDP_OTHER;
+	gboolean audio_dtmf = FALSE, video_rtcpfb = TRUE, h264_fmtp = TRUE;
+	const char *codec = NULL, *fmtp = NULL;
+	GList *extmaps = NULL;
+	janus_sdp_mdirection mdir = JANUS_SDP_DEFAULT;
+	int property = va_arg(args, int);
+	if(property != JANUS_SDP_OA_MLINE) {
+		/* The first attribute MUST be JANUS_SDP_OA_MLINE */
+		JANUS_LOG(LOG_ERR, "First attribute is not JANUS_SDP_OA_MLINE\n");
+		va_end(args);
+		return -2;
+	}
+	type = va_arg(args, int);
+	if(type != JANUS_SDP_AUDIO && type != JANUS_SDP_VIDEO && type != JANUS_SDP_APPLICATION) {
+		/* Unsupported m-line type */
+		JANUS_LOG(LOG_ERR, "Invalid m-line type\n");
+		va_end(args);
+		return -3;
+	}
+
+	/* Let's see what we should do with the media to add */
+	property = va_arg(args, int);
+	while(property != JANUS_SDP_OA_DONE) {
+		if(property == JANUS_SDP_OA_ENABLED) {
+			mline_enabled = va_arg(args, gboolean);
+		} else if(property == JANUS_SDP_OA_DIRECTION) {
+			mdir = va_arg(args, janus_sdp_mdirection);
+		} else if(property == JANUS_SDP_OA_CODEC) {
+			codec = va_arg(args, char *);
+		} else if(property == JANUS_SDP_OA_FMTP) {
+			fmtp = va_arg(args, char *);
+		} else if(type == JANUS_SDP_AUDIO && property == JANUS_SDP_OA_AUDIO_DTMF) {
+			audio_dtmf = va_arg(args, gboolean);
+		} else if(type == JANUS_SDP_VIDEO && property == JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS) {
+			video_rtcpfb = va_arg(args, gboolean);
+		} else if(type == JANUS_SDP_VIDEO && property == JANUS_SDP_OA_VIDEO_H264_FMTP) {
+			h264_fmtp = va_arg(args, gboolean);
+		} else if(property == JANUS_SDP_OA_ACCEPT_EXTMAP) {
+			const char *extension = va_arg(args, char *);
+			if(extension != NULL)
+				extmaps = g_list_append(extmaps, (char *)extension);
+		} else {
+			JANUS_LOG(LOG_WARN, "Unknown property %d for preparing SDP answer, ignoring...\n", property);
 		}
-		if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
-			janus_sdp_mdirection target_dir = m->type == JANUS_SDP_AUDIO ? audio_dir : video_dir;
+		property = va_arg(args, int);
+	}
+
+	/* Iterate on all m-lines to add, if any, to find the one with the same index */
+	GList *temp = answer->m_lines;
+	while(temp) {
+		janus_sdp_mline *am = (janus_sdp_mline *)temp->data;
+		if(am->index != offered->index) {
+			temp = temp->next;
+			continue;
+		}
+		/* When answering, m-lines are disabled by default */
+		am->direction = JANUS_SDP_INACTIVE;
+		g_list_free(am->ptypes);
+		am->ptypes = NULL;
+		g_list_free_full(am->fmts, (GDestroyNotify)g_free);
+		am->fmts = NULL;
+		g_list_free_full(am->attributes, (GDestroyNotify)janus_sdp_attribute_destroy);
+		am->attributes = NULL;
+		if(!mline_enabled) {
+			am->ptypes = g_list_append(am->ptypes, GINT_TO_POINTER(0));
+			break;
+		}
+		am->port = 9;
+		if(am->type == JANUS_SDP_AUDIO || am->type == JANUS_SDP_VIDEO) {
 			/* What is the direction we were offered? And how were we asked to react?
 			 * Adapt the direction in our answer accordingly */
-			switch(m->direction) {
+			switch(offered->direction) {
 				case JANUS_SDP_RECVONLY:
-					if(target_dir == JANUS_SDP_SENDRECV || target_dir == JANUS_SDP_SENDONLY) {
+					if(mdir == JANUS_SDP_SENDRECV || mdir == JANUS_SDP_DEFAULT || mdir == JANUS_SDP_SENDONLY) {
 						/* Peer is recvonly, we'll only send */
 						am->direction = JANUS_SDP_SENDONLY;
 					} else {
 						/* Peer is recvonly, but we're not ok to send, so reply with inactive */
 						JANUS_LOG(LOG_WARN, "%s offered as '%s', but we need '%s' for us: using 'inactive'\n",
-							m->type == JANUS_SDP_AUDIO ? "Audio" : "Video",
-							janus_sdp_mdirection_str(m->direction), janus_sdp_mdirection_str(target_dir));
+							am->type == JANUS_SDP_AUDIO ? "Audio" : "Video",
+							janus_sdp_mdirection_str(offered->direction), janus_sdp_mdirection_str(mdir));
 						am->direction = JANUS_SDP_INACTIVE;
 					}
 					break;
 				case JANUS_SDP_SENDONLY:
-					if(target_dir == JANUS_SDP_SENDRECV || target_dir == JANUS_SDP_RECVONLY) {
+					if(mdir == JANUS_SDP_SENDRECV || mdir == JANUS_SDP_DEFAULT || mdir == JANUS_SDP_RECVONLY) {
 						/* Peer is sendonly, we'll only receive */
 						am->direction = JANUS_SDP_RECVONLY;
 					} else {
 						/* Peer is sendonly, but we're not ok to receive, so reply with inactive */
 						JANUS_LOG(LOG_WARN, "%s offered as '%s', but we need '%s' for us: using 'inactive'\n",
-							m->type == JANUS_SDP_AUDIO ? "Audio" : "Video",
-							janus_sdp_mdirection_str(m->direction), janus_sdp_mdirection_str(target_dir));
+							offered->type == JANUS_SDP_AUDIO ? "Audio" : "Video",
+							janus_sdp_mdirection_str(offered->direction), janus_sdp_mdirection_str(mdir));
 						am->direction = JANUS_SDP_INACTIVE;
 					}
 					break;
@@ -1300,13 +1514,14 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 					am->direction = JANUS_SDP_INACTIVE;
 					break;
 				case JANUS_SDP_SENDRECV:
+				case JANUS_SDP_DEFAULT:
 				default:
 					/* The peer is fine with everything, so use our constraint */
-					am->direction = target_dir;
+					am->direction = mdir;
 					break;
 			}
 			/* Look for the right codec and stick to that */
-			const char *codec = m->type == JANUS_SDP_AUDIO ? audio_codec : video_codec;
+			int pt = -1;
 			if(codec == NULL) {
 				/* FIXME User didn't provide a codec to accept? Let's see if Opus (for audio)
 				 * of VP8 (for video) were negotiated: if so, use them, otherwise let's
@@ -1314,22 +1529,22 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 				 * Notice that if it's not a codec we understand, we reject the medium,
 				 * as browsers would reject it anyway. If you need more flexibility you'll
 				 * have to generate an answer yourself, rather than automatically... */
-				codec = m->type == JANUS_SDP_AUDIO ? "opus" : "vp8";
-				if(janus_sdp_get_codec_pt(offer, codec) < 0) {
+				codec = am->type == JANUS_SDP_AUDIO ? "opus" : "vp8";
+				if((pt = janus_sdp_get_codec_pt(offer, offered->index, codec)) < 0) {
 					/* We couldn't find our preferred codec, let's try something else */
-					if(m->type == JANUS_SDP_AUDIO) {
+					if(am->type == JANUS_SDP_AUDIO) {
 						/* Opus not found, maybe mu-law? */
 						codec = "pcmu";
-						if(janus_sdp_get_codec_pt(offer, codec) < 0) {
+						if((pt = janus_sdp_get_codec_pt(offer, offered->index, codec)) < 0) {
 							/* mu-law not found, maybe a-law? */
 							codec = "pcma";
-							if(janus_sdp_get_codec_pt(offer, codec) < 0) {
+							if((pt = janus_sdp_get_codec_pt(offer, offered->index, codec)) < 0) {
 								/* a-law not found, maybe G.722? */
 								codec = "g722";
-								if(janus_sdp_get_codec_pt(offer, codec) < 0) {
+								if((pt = janus_sdp_get_codec_pt(offer, offered->index, codec)) < 0) {
 									/* G.722 not found, maybe isac32? */
 									codec = "isac32";
-									if(janus_sdp_get_codec_pt(offer, codec) < 0) {
+									if((pt = janus_sdp_get_codec_pt(offer, offered->index, codec)) < 0) {
 										/* isac32 not found, maybe isac16? */
 										codec = "isac16";
 									}
@@ -1339,26 +1554,27 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 					} else {
 						/* VP8 not found, maybe VP9? */
 						codec = "vp9";
-						if(janus_sdp_get_codec_pt(offer, codec) < 0) {
+						if((pt = janus_sdp_get_codec_pt(offer, offered->index, codec)) < 0) {
 							/* VP9 not found either, maybe H.264? */
 							codec = "h264";
 						}
 					}
 				}
 			}
-			int pt = janus_sdp_get_codec_pt(offer, codec);
+			if(pt < 0)
+				pt = janus_sdp_get_codec_pt(offer, offered->index, codec);
 			if(pt < 0) {
 				/* Reject */
 				JANUS_LOG(LOG_WARN, "Couldn't find codec we needed (%s) in the offer, rejecting %s\n",
-					codec, m->type == JANUS_SDP_AUDIO ? "audio" : "video");
+					codec, am->type == JANUS_SDP_AUDIO ? "audio" : "video");
 				am->port = 0;
 				am->direction = JANUS_SDP_INACTIVE;
-				temp = temp->next;
-				continue;
+				am->ptypes = g_list_append(am->ptypes, GINT_TO_POINTER(0));
+				break;
 			}
 			am->ptypes = g_list_append(am->ptypes, GINT_TO_POINTER(pt));
 			/* Add the related attributes */
-			if(m->type == JANUS_SDP_AUDIO) {
+			if(am->type == JANUS_SDP_AUDIO) {
 				/* Add rtpmap attribute */
 				const char *codec_rtpmap = janus_sdp_get_codec_rtpmap(codec);
 				if(codec_rtpmap) {
@@ -1366,7 +1582,7 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 					am->attributes = g_list_append(am->attributes, a);
 					/* Check if we need to add a payload type for DTMF tones (telephone-event/8000) */
 					if(audio_dtmf) {
-						int dtmf_pt = janus_sdp_get_codec_pt(offer, "dtmf");
+						int dtmf_pt = janus_sdp_get_codec_pt(offer, am->index, "dtmf");
 						if(dtmf_pt >= 0) {
 							/* We do */
 							am->ptypes = g_list_append(am->ptypes, GINT_TO_POINTER(dtmf_pt));
@@ -1376,8 +1592,8 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 					}
 					/* Check if there's a custom fmtp line to add for audio
 					 * FIXME We should actually check if it matches the offer */
-					if(audio_fmtp) {
-						janus_sdp_attribute *a = janus_sdp_attribute_create("fmtp", "%d %s", pt, audio_fmtp);
+					if(fmtp) {
+						janus_sdp_attribute *a = janus_sdp_attribute_create("fmtp", "%d %s", pt, fmtp);
 						am->attributes = g_list_append(am->attributes, a);
 					}
 				}
@@ -1387,7 +1603,12 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 				if(codec_rtpmap) {
 					janus_sdp_attribute *a = janus_sdp_attribute_create("rtpmap", "%d %s", pt, codec_rtpmap);
 					am->attributes = g_list_append(am->attributes, a);
-					if(!strcasecmp(codec, "h264") && h264_fmtp) {
+					/* Check if there's a custom fmtp line to add for video
+					 * FIXME We should actually check if it matches the offer */
+					if(fmtp) {
+						janus_sdp_attribute *a = janus_sdp_attribute_create("fmtp", "%d %s", pt, fmtp);
+						am->attributes = g_list_append(am->attributes, a);
+					} else if(!strcasecmp(codec, "h264") && h264_fmtp) {
 						/* If it's H.264 and we were asked to, add the default fmtp profile as well */
 						a = janus_sdp_attribute_create("fmtp", "%d profile-level-id=42e01f;packetization-mode=1", pt);
 						am->attributes = g_list_append(am->attributes, a);
@@ -1410,7 +1631,7 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 			}
 			/* Add the extmap attributes, if needed */
 			if(extmaps != NULL) {
-				GList *ma = m->attributes;
+				GList *ma = offered->attributes;
 				while(ma) {
 					/* Iterate on all attributes, to see if there's an extension to accept */
 					janus_sdp_attribute *a = (janus_sdp_attribute *)ma->data;
@@ -1447,7 +1668,7 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 		} else {
 			/* This is for data, add formats and an sctpmap attribute */
 			am->direction = JANUS_SDP_DEFAULT;
-			GList *fmt = m->fmts;
+			GList *fmt = offered->fmts;
 			while(fmt) {
 				char *fmt_str = (char *)fmt->data;
 				if(fmt_str)
@@ -1455,13 +1676,13 @@ janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...) {
 				fmt = fmt->next;
 			}
 		}
-		temp = temp->next;
+		/* Nothing else we need to do */
+		break;
 	}
-	janus_refcount_decrease(&offer->ref);
 
 	/* Done */
 	g_list_free(extmaps);
 	va_end(args);
 
-	return answer;
+	return 0;
 }
diff --git a/sdp-utils.h b/sdp-utils.h
index 34b79a250a..36b07aab6a 100644
--- a/sdp-utils.h
+++ b/sdp-utils.h
@@ -99,24 +99,26 @@ janus_sdp_mdirection janus_sdp_parse_mdirection(const char *direction);
  * @returns The direction as a string, if valid, or NULL otherwise */
 const char *janus_sdp_mdirection_str(janus_sdp_mdirection direction);
 
-/*! \brief Helper method to return the preferred audio and video codecs in an SDP offer or answer,
+/*! \brief Helper method to return the preferred audio or video codec in an SDP offer or answer,
  * (where by preferred we mean the codecs we prefer ourselves, and not the m-line SDP order)
  * as long as the m-line direction is not disabled (port=0 or direction=inactive) in the SDP
- * \note The acodec and vcodec arguments are input/output, and they'll be set to a static value
- * in janus_preferred_audio_codecs and janus_preferred_video_codecs, so don't free them.
+ * \note The codec arguments is input/output, and it'll be set to a static value
+ * in janus_preferred_audio_codecs or janus_preferred_video_codecs, so don't free it.
  * @param[in] sdp The Janus SDP object to parse
- * @param[out] acodec The audio codec that was found
- * @param[out] vcodec The video codec that was found */
-void janus_sdp_find_preferred_codecs(janus_sdp *sdp, const char **acodec, const char **vcodec);
+ * @param[in] type Whether we're looking at an audio or video codec
+ * @param[in] index The m-line to refer to (use -1 for the first m-line that matches)
+ * @param[out] codec The audio or video codec that was found */
+void janus_sdp_find_preferred_codec(janus_sdp *sdp, janus_sdp_mtype type, int index, const char **codec);
 /*! \brief Helper method to return the first audio and video codecs in an SDP offer or answer,
  * (no matter whether we personally prefer them ourselves or not)
  * as long as the m-line direction is not disabled (port=0 or direction=inactive) in the SDP
- * \note The acodec and vcodec arguments are input/output, and they'll be set to a static value
- * in janus_preferred_audio_codecs and janus_preferred_video_codecs, so don't free them.
+ * \note The codec arguments is input/output, and it'll be set to a static value
+ * in janus_preferred_audio_codecs and janus_preferred_video_codecs, so don't free it.
  * @param[in] sdp The Janus SDP object to parse
- * @param[out] acodec The audio codec that was found
- * @param[out] vcodec The video codec that was found */
-void janus_sdp_find_first_codecs(janus_sdp *sdp, const char **acodec, const char **vcodec);
+ * @param[in] type Whether we're looking at an audio or video codec
+ * @param[in] index The m-line to refer to (use -1 for the first m-line that matches)
+ * @param[out] codec The audio or video codec that was found */
+void janus_sdp_find_first_codecs(janus_sdp *sdp, janus_sdp_mtype type, int index, const char **codec);
 /*! \brief Helper method to match a codec to one of the preferred codecs
  * \note Don't free the returned value, as it's a constant value
  * @param[in] type The type of media to match
@@ -126,6 +128,8 @@ const char *janus_sdp_match_preferred_codec(janus_sdp_mtype type, char *codec);
 
 /*! \brief SDP m-line representation */
 typedef struct janus_sdp_mline {
+	/*! \brief Media index in the SDP */
+	int index;
 	/*! \brief Media type as a janus_sdp_mtype enumerator */
 	janus_sdp_mtype type;
 	/*! \brief Media type (string) */
@@ -170,17 +174,20 @@ janus_sdp_mline *janus_sdp_mline_create(janus_sdp_mtype type, guint16 port, cons
  * @param[in] mline The janus_sdp_mline instance to free */
 void janus_sdp_mline_destroy(janus_sdp_mline *mline);
 /*! \brief Helper method to get the janus_sdp_mline associated to a media type
- * @note This currently returns the first m-line of the specified type it finds: in
- * general, it shouldn't be an issue as we currently only support a single stream
- * of the same type per session anyway... this will need to be fixed in the future.
+ * @note This currently returns the first m-line of the specified type it finds: as
+ * such, it's only here for backwards compatibility with plugins not doing multistream.
  * @param[in] sdp The Janus SDP object to search
  * @param[in] type The type of media to search
  * @returns The janus_sdp_mline instance, if found, or NULL otherwise */
 janus_sdp_mline *janus_sdp_mline_find(janus_sdp *sdp, janus_sdp_mtype type);
+/*! \brief Helper method to get the janus_sdp_mline by its index
+ * @param[in] sdp The Janus SDP object to search
+ * @param[in] index The index of the m-line in the SDP
+ * @returns The janus_sdp_mline instance, if found, or NULL otherwise */
+janus_sdp_mline *janus_sdp_mline_find_by_index(janus_sdp *sdp, int index);
 /*! \brief Helper method to remove the janus_sdp_mline associated to a media type from the SDP
- * @note This currently removes the first m-line of the specified type it finds: in
- * general, it shouldn't be an issue as we currently only support a single stream
- * of the same type per session anyway... this will need to be fixed in the future.
+ * @note This currently removes the first m-line of the specified type it finds: as
+ * such, it's only here for backwards compatibility with plugins not doing multistream.
  * @param[in] sdp The Janus SDP object to modify
  * @param[in] type The type of media to remove
  * @returns 0 if successful, a negative integer otherwise */
@@ -222,12 +229,6 @@ int janus_sdp_attribute_add_to_mline(janus_sdp_mline *mline, janus_sdp_attribute
  * of errors, if provided the error string is filled with a reason  */
 janus_sdp *janus_sdp_parse(const char *sdp, char *error, size_t errlen);
 
-/*! \brief Helper method to quickly remove all traces (m-line, rtpmap, fmtp, etc.) of a payload type
- * @param[in] sdp The janus_sdp object to remove the payload type from
- * @param[in] pt The payload type to remove
- * @returns 0 in case of success, a negative integer otherwise */
-int janus_sdp_remove_payload_type(janus_sdp *sdp, int pt);
-
 /*! \brief Method to serialize a janus_sdp object to an SDP string
  * @param[in] sdp The janus_sdp object to serialize
  * @returns A pointer to a string with the serialized SDP, if successful, NULL otherwise */
@@ -246,97 +247,133 @@ janus_sdp *janus_sdp_new(const char *name, const char *address);
 void janus_sdp_destroy(janus_sdp *sdp);
 
 typedef enum janus_sdp_oa_type {
-/*! \brief When generating an offer or answer automatically, accept/reject audio if offered (depends on value that follows) */
-JANUS_SDP_OA_AUDIO = 1,
-/*! \brief When generating an offer or answer automatically, accept/reject video if offered (depends on value that follows) */
-JANUS_SDP_OA_VIDEO,
-/*! \brief When generating an offer or answer automatically, accept/reject datachannels if offered (depends on value that follows) */
-JANUS_SDP_OA_DATA,
-/*! \brief When generating an offer or answer automatically, use this direction for audio (depends on value that follows) */
-JANUS_SDP_OA_AUDIO_DIRECTION,
-/*! \brief When generating an offer or answer automatically, use this direction for video (depends on value that follows) */
-JANUS_SDP_OA_VIDEO_DIRECTION,
-/*! \brief When generating an offer or answer automatically, use this codec for audio (depends on value that follows) */
-JANUS_SDP_OA_AUDIO_CODEC,
-/*! \brief When generating an offer or answer automatically, use this codec for video (depends on value that follows) */
-JANUS_SDP_OA_VIDEO_CODEC,
-/*! \brief When generating an offer (this is ignored for answers), use this payload type for audio (depends on value that follows) */
-JANUS_SDP_OA_AUDIO_PT,
-/*! \brief When generating an offer (this is ignored for answers), use this payload type for video (depends on value that follows) */
-JANUS_SDP_OA_VIDEO_PT,
-/*! \brief When generating an offer or answer automatically, do or do not negotiate telephone events (FIXME telephone-event/8000 only) */
-JANUS_SDP_OA_AUDIO_DTMF,
-/*! \brief When generating an offer or answer automatically, add this custom fmtp string for audio */
-JANUS_SDP_OA_AUDIO_FMTP,
-/*! \brief When generating an offer or answer automatically, do or do not add the rtcpfb attributes we typically negotiate (fir, nack, pli, remb) */
-JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS,
-/*! \brief When generating an offer or answer automatically, do or do not add the default fmtp attribute for H.264 (profile-level-id=42e01f;packetization-mode=1) */
-JANUS_SDP_OA_VIDEO_H264_FMTP,
-/*! \brief When generating an offer (this is ignored for answers), use the old "DTLS/SCTP" instead of the new "UDP/DTLS/SCTP (default=TRUE for now, depends on what follows) */
-JANUS_SDP_OA_DATA_LEGACY,
-/*! \brief When generating an answer (this is ignored for offers), accept this extension (by default, we reject them all; can be used multiple times) */
-JANUS_SDP_OA_ACCEPT_EXTMAP,
-/*! \brief MUST be used as the last argument in janus_sdp_generate_offer and janus_sdp_generate_answer */
-JANUS_SDP_OA_DONE = 0
+	/*! \brief Add a new m-line of the specific kind (used as a separator for audio, video and data details passed to janus_sdp_generate_offer) */
+	JANUS_SDP_OA_MLINE = 1,
+	/*! \brief Whether we should enable a specific m-line when offering/answering (depends on what follows, true by default) */
+	JANUS_SDP_OA_ENABLED,
+	/*! \brief When generating an offer or answer automatically, use this direction for media (depends on value that follows, sendrecv by default) */
+	JANUS_SDP_OA_DIRECTION,
+	/*! \brief When generating an offer automatically, use this mid media (depends on value that follows, needs to be a string) */
+	JANUS_SDP_OA_MID,
+	/*! \brief When generating an offer or answer automatically, use this codec (depends on value that follows, opus/vp8 by default) */
+	JANUS_SDP_OA_CODEC,
+	/*! \brief When generating an offer (this is ignored for answers), use this payload type (depends on value that follows) */
+	JANUS_SDP_OA_PT,
+	/*! \brief When generating an offer or answer automatically, add this custom fmtp string */
+	JANUS_SDP_OA_FMTP,
+	/*! \brief When generating an offer or answer automatically, do or do not negotiate telephone events (FIXME telephone-event/8000 only, true by default) */
+	JANUS_SDP_OA_AUDIO_DTMF,
+	/*! \brief When generating an offer or answer automatically, do or do not add the rtcpfb attributes we typically negotiate (fir, nack, pli, remb; true by defaukt) */
+	JANUS_SDP_OA_VIDEO_RTCPFB_DEFAULTS,
+	/*! \brief When generating an offer or answer automatically, do or do not add the default fmtp attribute for H.264 (profile-level-id=42e01f;packetization-mode=1) */
+	JANUS_SDP_OA_VIDEO_H264_FMTP,
+	/*! \brief When generating an offer (this is ignored for answers), use the old "DTLS/SCTP" instead of the new "UDP/DTLS/SCTP (depends on what follows, false by default) */
+	JANUS_SDP_OA_DATA_LEGACY,
+	/*! \brief When generating an answer (this is ignored for offers), accept this extension (by default, we reject them all; can be used multiple times) */
+	JANUS_SDP_OA_ACCEPT_EXTMAP,
+	/*! \brief MUST be used as the last argument in janus_sdp_generate_offer, janus_sdp_generate_offer_mline and janus_sdp_generate_answer */
+	JANUS_SDP_OA_DONE = 0
 } janus_sdp_oa_type;
+const char *janus_sdp_oa_type_str(janus_sdp_oa_type type);
 
 /*! \brief Method to generate a janus_sdp offer, using variable arguments to dictate
  * what to negotiate (e.g., in terms of media to offer, directions, etc.). Variable
  * arguments are in the form of a sequence of name-value terminated by a JANUS_SDP_OA_DONE, e.g.:
  \verbatim
 	janus_sdp *offer = janus_sdp_generate_offer("My session", "127.0.0.1",
-		JANUS_SDP_OA_AUDIO, TRUE,
-		JANUS_SDP_OA_AUDIO_PT, 100,
-		JANUS_SDP_OA_AUDIO_DIRECTION, JANUS_SDP_SENDONLY,
-		JANUS_SDP_OA_AUDIO_CODEC, "opus",
-		JANUS_SDP_OA_VIDEO, FALSE,
-		JANUS_SDP_OA_DATA, FALSE,
+		JANUS_SDP_OA_MLINE, JANUS_SDP_AUDIO,
+			JANUS_SDP_OA_PT, 100,
+			JANUS_SDP_OA_DIRECTION, JANUS_SDP_SENDONLY,
+			JANUS_SDP_OA_CODEC, "opus",
+		JANUS_SDP_OA_MLINE, JANUS_SDP_VIDEO,
+			JANUS_SDP_OA_PT, 101,
+			JANUS_SDP_OA_DIRECTION, JANUS_SDP_RECVONLY,
+			JANUS_SDP_OA_CODEC, "vp8",
 		JANUS_SDP_OA_DONE);
  \endverbatim
- * to only offer a \c sendonly Opus audio stream being offered with 100 as
- * payload type, and avoid video and datachannels. Refer to the property names in
- * the header file for a complete list of how you can drive the offer.
- * The default, if not specified, is to offer everything, using Opus with pt=111
- * for audio, VP8 with pt=96 as video, and data channels, all as \c sendrecv.
+ * to offer a \c sendonly Opus audio stream being offered with 100 as
+ * payload type, and a \c recvonly VP8 video stream with 101 as payload type.
+ * Refer to the property names in the header file for a complete
+ * list of how you can drive the offer. Other media streams can be added,
+ * as long as you prefix/specify them with JANUS_SDP_OA_MLINE as done here.
+ * The default, if not specified, is to not offer anything, meaning it
+ * will be up to you to add m-lines subsequently.
  * @param[in] name The session name (if NULL, a default value will be set)
  * @param[in] address The IP to set in o= and c= fields (if NULL, a default value will be set)
  * @returns A pointer to a janus_sdp object, if successful, NULL otherwise */
 janus_sdp *janus_sdp_generate_offer(const char *name, const char *address, ...);
-/*! \brief Method to generate a janus_sdp answer to a provided janus_sdp offer, using variable arguments
- * to dictate how to respond (e.g., in terms of media to accept, reject, directions, etc.). Variable
- * arguments are in the form of a sequence of name-value terminated by a JANUS_SDP_OA_DONE, e.g.:
+/*! \brief Method to add a single m-line to a new offer, using the same
+ * variable arguments janus_sdp_generate_offer supports. This is useful
+ * whenever you need to create a new offer, but don't know in advance
+ * how many m-lines you'll need, or it would be hard to do programmatically
+ * in a single call to janus_sdp_generate_offer. The first argument
+ * MUST be JANUS_SDP_OA_MLINE, specifying the type of the media.
+ * \note In case case you add audio and don't specify anything else, the
+ * default is to use Opus and payload type 111. For video, the default
+ * is VP8 and payload type 96. The default media direction is \c sendrecv.
+ * @param[in] offer The Janus SDP offer to add the new m-line to
+ * @returns 0 if successful, a negative integer othwerwise */
+int janus_sdp_generate_offer_mline(janus_sdp *offer, ...);
+/*! \brief Method to generate a janus_sdp answer to a provided janus_sdp offer.
+ * Notice that this doesn't address the individual m-lines: it will just
+ * create an empty response, create the corresponding m-lines, but leave
+ * them all "rejected". To answer each m-line you'll have to iterate on
+ * the offer m-lines and call janus_sdp_generate_answer_mline instead, e.g.:
  \verbatim
-	janus_sdp *answer = janus_sdp_generate_answer(offer,
-		JANUS_SDP_OA_AUDIO, TRUE,
-		JANUS_SDP_OA_AUDIO_DIRECTION, JANUS_SDP_RECVONLY,
-		JANUS_SDP_OA_AUDIO_CODEC, "opus",
-		JANUS_SDP_OA_VIDEO, FALSE,
-		JANUS_SDP_OA_DATA, FALSE,
-		JANUS_SDP_OA_DONE);
+	janus_sdp *answer = janus_sdp_generate_answer(offer);
+	GList *temp = offer->m_lines;
+	while(temp) {
+		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+		janus_sdp_generate_answer_mline(offer, answer, m,
+			[..]
+			JANUS_SDP_OA_DONE);
+		temp = temp->next;
+	}
  \endverbatim
- * to only accept the audio stream being offered, but as \c recvonly, use Opus
- * and reject both video and datachannels. Refer to the property names in
- * the header file for a complete list of how you can drive the answer.
- * The default, if not specified, is to accept everything as \c sendrecv.
  * @param[in] offer The Janus SDP offer to respond to
  * @returns A pointer to a janus_sdp object, if successful, NULL otherwise */
-janus_sdp *janus_sdp_generate_answer(janus_sdp *offer, ...);
+janus_sdp *janus_sdp_generate_answer(janus_sdp *offer);
+/*! \brief Method to respond to a single m-line in an offer, using the same
+ * variable arguments janus_sdp_generate_offer_mline supports. The first
+ * argument MUST be JANUS_SDP_OA_MLINE, specifying the type of the media, e.g.:
+ \verbatim
+	janus_sdp_generate_answer_mline(offer, answer, offer_mline,
+		JANUS_SDP_OA_MLINE, JANUS_SDP_AUDIO,
+			JANUS_SDP_OA_CODEC, "opus",
+			JANUS_SDP_OA_DIRECTION, JANUS_SDP_RECVONLY,
+		JANUS_SDP_OA_DONE);
+ \endverbatim
+ * to respond to an offered m-line with recvonly audio and use Opus.
+ * @param[in] offer The Janus SDP offer
+ * @param[in] answer The Janus SDP answer to add the new m-line to
+ * @param[in] offered The Janus SDP m-line from the offer to respond to
+ * @returns 0 if successful, a negative integer othwerwise */
+int janus_sdp_generate_answer_mline(janus_sdp *offer, janus_sdp *answer, janus_sdp_mline *offered, ...);
 
-/*! \brief Helper to get the payload type associated to a specific codec
+/*! \brief Helper to get the payload type associated to a specific codec in an m-line
  * @param sdp The Janus SDP instance to process
+ * @param index The m-line to refer to (use -1 for the first m-line that matches)
  * @param codec The codec to find, as a string
  * @returns The payload type, if found, or -1 otherwise */
-int janus_sdp_get_codec_pt(janus_sdp *sdp, const char *codec);
+int janus_sdp_get_codec_pt(janus_sdp *sdp, int index, const char *codec);
 
-/*! \brief Helper to get the codec name associated to a specific payload type
+/*! \brief Helper to get the codec name associated to a specific payload type in an m-line
  * @param sdp The Janus SDP instance to process
+ * @param index The m-line to refer to (use -1 for the first m-line that matches)
  * @param pt The payload type to find
  * @returns The codec name, if found, or NULL otherwise */
-const char *janus_sdp_get_codec_name(janus_sdp *sdp, int pt);
+const char *janus_sdp_get_codec_name(janus_sdp *sdp, int index, int pt);
+
+/*! \brief Helper method to quickly remove all traces (m-line, rtpmap, fmtp, etc.) of a payload type
+ * @param[in] sdp The janus_sdp object to remove the payload type from
+ * @param[in] index The m-line to remove the payload type from (use -1 for the first m-line that matches)
+ * @param[in] pt The payload type to remove
+ * @returns 0 in case of success, a negative integer otherwise */
+int janus_sdp_remove_payload_type(janus_sdp *sdp, int index, int pt);
 
 /*! \brief Helper to get the rtpmap associated to a specific codec
  * @param codec The codec name, as a string (e.g., "opus")
- * @returns The rtpmap value, if found (e.g., "opus/48000/2"), or -1 otherwise */
+ * @returns The rtpmap value, if found (e.g., "opus/48000/2"), or NULL otherwise */
 const char *janus_sdp_get_codec_rtpmap(const char *codec);
 
 #endif
diff --git a/sdp.c b/sdp.c
index 75107a450e..2c4619827b 100644
--- a/sdp.c
+++ b/sdp.c
@@ -28,13 +28,12 @@
 
 
 /* Pre-parse SDP: is this SDP valid? how many audio/video lines? any features to take into account? */
-janus_sdp *janus_sdp_preparse(void *ice_handle, const char *jsep_sdp, char *error_str, size_t errlen,
-		int *audio, int *video, int *data) {
-	if(!ice_handle || !jsep_sdp || !audio || !video || !data) {
+janus_sdp *janus_sdp_preparse(void *ice_handle, const char *jsep_sdp,
+		int *audio, int *video, int *data, char *error_str, size_t errlen) {
+	if(!ice_handle || !jsep_sdp) {
 		JANUS_LOG(LOG_ERR, "  Can't preparse, invalid arguments\n");
 		return NULL;
 	}
-	janus_ice_handle *handle = (janus_ice_handle *)ice_handle;
 	janus_sdp *parsed_sdp = janus_sdp_parse(jsep_sdp, error_str, errlen);
 	if(!parsed_sdp) {
 		JANUS_LOG(LOG_ERR, "  Error parsing SDP? %s\n", error_str ? error_str : "(unknown reason)");
@@ -42,63 +41,34 @@ janus_sdp *janus_sdp_preparse(void *ice_handle, const char *jsep_sdp, char *erro
 		return NULL;
 	}
 	/* Look for m-lines */
+	int mlines = -1;
 	GList *temp = parsed_sdp->m_lines;
 	while(temp) {
+		mlines++;
 		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
-		if(m->type == JANUS_SDP_AUDIO && m->port > 0) {
-			*audio = *audio + 1;
-		} else if(m->type == JANUS_SDP_VIDEO && m->port > 0) {
-			*video = *video + 1;
-		}
-		/* Preparse the mid as well */
-		GList *tempA = m->attributes;
-		while(tempA) {
-			janus_sdp_attribute *a = (janus_sdp_attribute *)tempA->data;
-			if(a->name) {
-				if(!strcasecmp(a->name, "mid")) {
-					/* Found mid attribute */
-					if(m->type == JANUS_SDP_AUDIO && m->port > 0) {
-						JANUS_LOG(LOG_VERB, "[%"SCNu64"] Audio mid: %s\n", handle->handle_id, a->value);
-						if(handle->audio_mid == NULL)
-							handle->audio_mid = g_strdup(a->value);
-						if(handle->stream_mid == NULL)
-							handle->stream_mid = handle->audio_mid;
-					} else if(m->type == JANUS_SDP_VIDEO && m->port > 0) {
-						JANUS_LOG(LOG_VERB, "[%"SCNu64"] Video mid: %s\n", handle->handle_id, a->value);
-						if(handle->video_mid == NULL)
-							handle->video_mid = g_strdup(a->value);
-						if(handle->stream_mid == NULL)
-							handle->stream_mid = handle->video_mid;
-					}
-				}
-			}
-			tempA = tempA->next;
+		if(audio && m->type == JANUS_SDP_AUDIO) {
+			*audio = *audio+1;
+		} else if(video && m->type == JANUS_SDP_VIDEO) {
+			*video = *video+1;
+		} else if(data && m->type == JANUS_SDP_APPLICATION && strstr(m->proto, "DTLS/SCTP")) {
+			*data = *data+1;
 		}
 		temp = temp->next;
 	}
-#ifdef HAVE_SCTP
-	*data = (strstr(jsep_sdp, "DTLS/SCTP") && !strstr(jsep_sdp, " 0 DTLS/SCTP") &&
-		!strstr(jsep_sdp, " 0 UDP/DTLS/SCTP")) ? 1 : 0;	/* FIXME This is a really hacky way of checking... */
-#else
-	*data = 0;
-#endif
 
 	return parsed_sdp;
 }
 
-/* Parse SDP */
-int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update) {
+/* Parse remote SDP */
+int janus_sdp_process_remote(void *ice_handle, janus_sdp *remote_sdp, gboolean update) {
 	if(!ice_handle || !remote_sdp)
 		return -1;
-	janus_ice_handle *handle = (janus_ice_handle *)ice_handle;
-	janus_ice_stream *stream = handle->stream;
-	if(!stream)
+	janus_handle *handle = (janus_handle *)ice_handle;
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc)
 		return -1;
+	janus_handle_webrtc_medium *medium = NULL;
 	gchar *ruser = NULL, *rpass = NULL, *rhashing = NULL, *rfingerprint = NULL;
-	int audio = 0, video = 0;
-#ifdef HAVE_SCTP
-	int data = 0;
-#endif
 	gboolean rtx = FALSE;
 	/* Ok, let's start with global attributes */
 	GList *temp = remote_sdp->attributes;
@@ -134,143 +104,105 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 	while(temp) {
 		mlines++;
 		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
-		if(m->type == JANUS_SDP_AUDIO) {
-			if(handle->rtp_profile == NULL && m->proto != NULL)
-				handle->rtp_profile = g_strdup(m->proto);
-			audio++;
-			if(audio > 1) {
-				temp = temp->next;
-				continue;
-			}
-			if(m->port > 0) {
-				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Parsing audio candidates (stream=%d)...\n", handle->handle_id, stream->stream_id);
-				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO)) {
-					janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO);
-					stream->audio_ssrc = janus_random_uint32();	/* FIXME Should we look for conflicts? */
-					if(stream->audio_rtcp_ctx == NULL) {
-						stream->audio_rtcp_ctx = g_malloc0(sizeof(rtcp_context));
-						stream->audio_rtcp_ctx->tb = 48000;	/* May change later */
-					}
-				}
-				switch(m->direction) {
-					case JANUS_SDP_INACTIVE:
-					case JANUS_SDP_INVALID:
-						stream->audio_send = FALSE;
-						stream->audio_recv = FALSE;
-						break;
-					case JANUS_SDP_SENDONLY:
-						/* A sendonly peer means recvonly for Janus */
-						stream->audio_send = FALSE;
-						stream->audio_recv = TRUE;
-						break;
-					case JANUS_SDP_RECVONLY:
-						/* A recvonly peer means sendonly for Janus */
-						stream->audio_send = TRUE;
-						stream->audio_recv = FALSE;
-						break;
-					case JANUS_SDP_SENDRECV:
-					case JANUS_SDP_DEFAULT:
-					default:
-						stream->audio_send = TRUE;
-						stream->audio_recv = TRUE;
-						break;
-				}
-				if(m->ptypes != NULL) {
-					g_list_free(stream->audio_payload_types);
-					stream->audio_payload_types = g_list_copy(m->ptypes);
-				}
-			} else {
-				/* Audio rejected? */
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_AUDIO);
-				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Audio rejected by peer...\n", handle->handle_id);
-			}
-		} else if(m->type == JANUS_SDP_VIDEO) {
+		if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
+			/* Audio/Video */
 			if(handle->rtp_profile == NULL && m->proto != NULL)
 				handle->rtp_profile = g_strdup(m->proto);
-			video++;
-			if(video > 1) {
-				temp = temp->next;
-				continue;
+			/* Find the internal medium instance */
+			medium = g_hash_table_lookup(pc->media, GINT_TO_POINTER(m->index));
+			if(!medium) {
+				/* We don't have it, create one now */
+				medium = janus_handle_webrtc_medium_create(handle,
+					m->type == JANUS_SDP_VIDEO ? JANUS_MEDIA_VIDEO : JANUS_MEDIA_AUDIO);
 			}
 			if(m->port > 0) {
-				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Parsing video candidates (stream=%d)...\n", handle->handle_id, stream->stream_id);
-				if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO)) {
-					janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO);
-					stream->video_ssrc = janus_random_uint32();	/* FIXME Should we look for conflicts? */
-					if(stream->video_rtcp_ctx[0] == NULL) {
-						stream->video_rtcp_ctx[0] = g_malloc0(sizeof(rtcp_context));
-						stream->video_rtcp_ctx[0]->tb = 90000;	/* May change later */
-					}
-				}
+				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Parsing candidates (stream=%d)...\n", handle->handle_id, pc->stream_id);
 				switch(m->direction) {
 					case JANUS_SDP_INACTIVE:
 					case JANUS_SDP_INVALID:
-						stream->video_send = FALSE;
-						stream->video_recv = FALSE;
+						medium->send = FALSE;
+						medium->recv = FALSE;
 						break;
 					case JANUS_SDP_SENDONLY:
 						/* A sendonly peer means recvonly for Janus */
-						stream->video_send = FALSE;
-						stream->video_recv = TRUE;
+						medium->send = FALSE;
+						medium->recv = TRUE;
 						break;
 					case JANUS_SDP_RECVONLY:
 						/* A recvonly peer means sendonly for Janus */
-						stream->video_send = TRUE;
-						stream->video_recv = FALSE;
+						medium->send = TRUE;
+						medium->recv = FALSE;
 						break;
 					case JANUS_SDP_SENDRECV:
 					case JANUS_SDP_DEFAULT:
 					default:
-						stream->video_send = TRUE;
-						stream->video_recv = TRUE;
+						medium->send = TRUE;
+						medium->recv = TRUE;
 						break;
 				}
 				if(m->ptypes != NULL) {
-					g_list_free(stream->video_payload_types);
-					stream->video_payload_types = g_list_copy(m->ptypes);
+					g_list_free(medium->payload_types);
+					medium->payload_types = g_list_copy(m->ptypes);
 				}
 			} else {
-				/* Video rejected? */
-				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Video rejected by peer...\n", handle->handle_id);
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_HAS_VIDEO);
+				/* Medium rejected? */
+				medium->send = FALSE;
+				medium->recv = FALSE;
 			}
-#ifdef HAVE_SCTP
 		} else if(m->type == JANUS_SDP_APPLICATION) {
+			/* Find the internal medium instance */
+			medium = g_hash_table_lookup(pc->media, GINT_TO_POINTER(m->index));
+			if(!medium) {
+				/* We don't have it, create one now */
+				medium = janus_handle_webrtc_medium_create(handle, JANUS_MEDIA_DATA);
+			}
 			/* Is this SCTP for DataChannels? */
 			if(!strcasecmp(m->proto, "DTLS/SCTP") || !strcasecmp(m->proto, "UDP/DTLS/SCTP")) {
-				data++;
-				if(data > 1) {
-					temp = temp->next;
-					continue;
-				}
+#ifdef HAVE_SCTP
 				if(m->port > 0) {
 					/* Yep */
-					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Parsing SCTP candidates (stream=%d)...\n", handle->handle_id, stream->stream_id);
-					if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS)) {
-						janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS);
+					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Parsing SCTP candidates (stream=%d)...\n", handle->handle_id, pc->stream_id);
+					if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_DATA_CHANNELS)) {
+						janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_DATA_CHANNELS);
 					}
 					if(!strcasecmp(m->proto, "UDP/DTLS/SCTP")) {
-						janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+						janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
 					} else {
-						janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+						janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
 					}
+					medium->send = TRUE;
+					medium->recv = TRUE;
 				} else {
 					/* Data channels rejected? */
 					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Data channels rejected by peer...\n", handle->handle_id);
-					janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS);
-					janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+					janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+					medium->send = FALSE;
+					medium->recv = FALSE;
 				}
+#else
+				/* Data channels unsupported */
+				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Data channels unsupported...\n", handle->handle_id);
+				janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+				medium->send = FALSE;
+				medium->recv = FALSE;
+#endif
 			} else {
 				/* Unsupported data channels format. */
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Data channels format %s unsupported, skipping\n", handle->handle_id, m->proto);
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_DATA_CHANNELS);
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+				janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+				medium->send = FALSE;
+				medium->recv = FALSE;
 			}
-#endif
 		} else {
 			JANUS_LOG(LOG_WARN, "[%"SCNu64"] Skipping disabled/unsupported media line...\n", handle->handle_id);
+			medium = g_hash_table_lookup(pc->media, GINT_TO_POINTER(m->index));
+			if(!medium) {
+				/* We don't have it, create one now */
+				medium = janus_handle_webrtc_medium_create(handle, JANUS_MEDIA_UNKNOWN);
+			}
 		}
-		if(stream == NULL) {
+		if(medium == NULL) {
+			/* No medium? */
 			temp = temp->next;
 			continue;
 		}
@@ -281,27 +213,15 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 			if(a->name) {
 				if(!strcasecmp(a->name, "mid")) {
 					/* Found mid attribute */
-					if(m->type == JANUS_SDP_AUDIO && m->port > 0) {
-						JANUS_LOG(LOG_VERB, "[%"SCNu64"] Audio mid: %s\n", handle->handle_id, a->value);
-						if(handle->audio_mid == NULL)
-							handle->audio_mid = g_strdup(a->value);
-						if(handle->stream_mid == NULL)
-							handle->stream_mid = handle->audio_mid;
-					} else if(m->type == JANUS_SDP_VIDEO && m->port > 0) {
-						JANUS_LOG(LOG_VERB, "[%"SCNu64"] Video mid: %s\n", handle->handle_id, a->value);
-						if(handle->video_mid == NULL)
-							handle->video_mid = g_strdup(a->value);
-						if(handle->stream_mid == NULL)
-							handle->stream_mid = handle->video_mid;
-#ifdef HAVE_SCTP
-					} else if(m->type == JANUS_SDP_APPLICATION) {
-						JANUS_LOG(LOG_VERB, "[%"SCNu64"] Data Channel mid: %s\n", handle->handle_id, a->value);
-						if(handle->data_mid == NULL)
-							handle->data_mid = g_strdup(a->value);
-						if(handle->stream_mid == NULL)
-							handle->stream_mid = handle->data_mid;
-#endif
+					if(medium->mid == NULL) {
+						medium->mid = g_strdup(a->value);
+						if(!g_hash_table_lookup(pc->media_bymid, medium->mid)) {
+							g_hash_table_insert(pc->media_bymid, g_strdup(medium->mid), medium);
+							janus_refcount_increase(&medium->ref);
+						}
 					}
+					if(handle->pc_mid == NULL)
+						handle->pc_mid = g_strdup(a->value);
 				} else if(!strcasecmp(a->name, "fingerprint")) {
 					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Fingerprint (local) : %s\n", handle->handle_id, a->value);
 					if(strcasestr(a->value, "sha-256 ") == a->value) {
@@ -324,13 +244,13 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 					if(!update) {
 						if(!strcasecmp(a->value, "actpass") || !strcasecmp(a->value, "passive")) {
 							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Setting connect state (DTLS client)\n", handle->handle_id);
-							stream->dtls_role = JANUS_DTLS_ROLE_CLIENT;
+							pc->dtls_role = JANUS_DTLS_ROLE_CLIENT;
 						} else if(!strcasecmp(a->value, "active")) {
 							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Setting accept state (DTLS server)\n", handle->handle_id);
-							stream->dtls_role = JANUS_DTLS_ROLE_SERVER;
+							pc->dtls_role = JANUS_DTLS_ROLE_SERVER;
 						}
-						if(stream->component && stream->component->dtls)
-							stream->component->dtls->dtls_role = stream->dtls_role;
+						if(pc->dtls)
+							pc->dtls->dtls_role = pc->dtls_role;
 					}
 					/* TODO Handle holdconn... */
 				} else if(!strcasecmp(a->name, "ice-ufrag")) {
@@ -365,31 +285,31 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 				return -2;
 			}
 			/* If we received the ICE credentials for the first time, enforce them */
-			if(ruser && !stream->ruser && rpass && !stream->rpass) {
+			if(ruser && !pc->ruser && rpass && !pc->rpass) {
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"] Setting remote credentials...\n", handle->handle_id);
 				if(!nice_agent_set_remote_credentials(handle->agent, handle->stream_id, ruser, rpass)) {
 					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to set remote credentials!\n", handle->handle_id);
 				}
 			} else
 			/* If this is a renegotiation, check if this is an ICE restart */
-			if((ruser && stream->ruser && strcmp(ruser, stream->ruser)) ||
-					(rpass && stream->rpass && strcmp(rpass, stream->rpass))) {
+			if((ruser && pc->ruser && strcmp(ruser, pc->ruser)) ||
+					(rpass && pc->rpass && strcmp(rpass, pc->rpass))) {
 				JANUS_LOG(LOG_WARN, "[%"SCNu64"] ICE restart detected\n", handle->handle_id);
-				janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ALL_TRICKLES);
-				janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_ICE_RESTART);
+				janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ALL_TRICKLES);
+				janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_ICE_RESTART);
 			}
 			/* Store fingerprint and hashing */
 			if(janus_is_webrtc_encryption_enabled()) {
-				g_free(stream->remote_hashing);
-				stream->remote_hashing = g_strdup(rhashing);
-				g_free(stream->remote_fingerprint);
-				stream->remote_fingerprint = g_strdup(rfingerprint);
+				g_free(pc->remote_hashing);
+				pc->remote_hashing = g_strdup(rhashing);
+				g_free(pc->remote_fingerprint);
+				pc->remote_fingerprint = g_strdup(rfingerprint);
 			}
 			/* Store the ICE username and password for this stream */
-			g_free(stream->ruser);
-			stream->ruser = g_strdup(ruser);
-			g_free(stream->rpass);
-			stream->rpass = g_strdup(rpass);
+			g_free(pc->ruser);
+			pc->ruser = g_strdup(ruser);
+			g_free(pc->rpass);
+			pc->rpass = g_strdup(rpass);
 		}
 		/* Is simulcasting enabled, using rid? (we need to check this before parsing SSRCs) */
 		tempA = m->attributes;
@@ -402,37 +322,36 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 					JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to parse rid attribute...\n", handle->handle_id);
 				} else {
 					JANUS_LOG(LOG_VERB, "[%"SCNu64"] Parsed rid: %s\n", handle->handle_id, rid);
-					if(stream->rid[0] == NULL) {
-						stream->rid[0] = g_strdup(rid);
-					} else if(stream->rid[1] == NULL) {
-						stream->rid[1] = g_strdup(rid);
-					} else if(stream->rid[2] == NULL) {
-						stream->rid[2] = g_strdup(rid);
+					if(medium->rid[0] == NULL) {
+						medium->rid[0] = g_strdup(rid);
+					} else if(medium->rid[1] == NULL) {
+						medium->rid[1] = g_strdup(rid);
+					} else if(medium->rid[2] == NULL) {
+						medium->rid[2] = g_strdup(rid);
 					} else {
 						JANUS_LOG(LOG_WARN, "[%"SCNu64"] Too many RTP Stream IDs, ignoring '%s'...\n", handle->handle_id, rid);
 					}
 				}
 			} else if(a->name && !strcasecmp(a->name, "simulcast") && a->value) {
 				/* Firefox and Chrome signal simulcast support differently */
-				stream->legacy_rid = strstr(a->value, "rid=") ? TRUE : FALSE;
+				medium->legacy_rid = strstr(a->value, "rid=") ? TRUE : FALSE;
 			}
 			tempA = tempA->next;
 		}
 		/* Let's start figuring out the SSRCs, and any grouping that may be there */
-		stream->audio_ssrc_peer_new = 0;
-		stream->video_ssrc_peer_new[0] = 0;
-		stream->video_ssrc_peer_new[1] = 0;
-		stream->video_ssrc_peer_new[2] = 0;
-		stream->video_ssrc_peer_rtx_new[0] = 0;
-		stream->video_ssrc_peer_rtx_new[1] = 0;
-		stream->video_ssrc_peer_rtx_new[2] = 0;
+		medium->ssrc_peer_new[0] = 0;
+		medium->ssrc_peer_new[1] = 0;
+		medium->ssrc_peer_new[2] = 0;
+		medium->ssrc_peer_rtx_new[0] = 0;
+		medium->ssrc_peer_rtx_new[1] = 0;
+		medium->ssrc_peer_rtx_new[2] = 0;
 		/* Any SSRC SIM group? */
 		tempA = m->attributes;
 		while(tempA) {
 			janus_sdp_attribute *a = (janus_sdp_attribute *)tempA->data;
 			if(a->name && a->value) {
 				if(!strcasecmp(a->name, "ssrc-group") && strstr(a->value, "SIM")) {
-					int res = janus_sdp_parse_ssrc_group(stream, (const char *)a->value, m->type == JANUS_SDP_VIDEO);
+					int res = janus_sdp_parse_ssrc_group(medium, (const char *)a->value, m->type == JANUS_SDP_VIDEO);
 					if(res != 0) {
 						JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to parse SSRC SIM group attribute... (%d)\n", handle->handle_id, res);
 					}
@@ -446,7 +365,7 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 			janus_sdp_attribute *a = (janus_sdp_attribute *)tempA->data;
 			if(a->name && a->value) {
 				if(!strcasecmp(a->name, "ssrc-group") && strstr(a->value, "FID")) {
-					int res = janus_sdp_parse_ssrc_group(stream, (const char *)a->value, m->type == JANUS_SDP_VIDEO);
+					int res = janus_sdp_parse_ssrc_group(medium, (const char *)a->value, m->type == JANUS_SDP_VIDEO);
 					if(res != 0) {
 						JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to parse SSRC FID group attribute... (%d)\n", handle->handle_id, res);
 					}
@@ -460,7 +379,7 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 			janus_sdp_attribute *a = (janus_sdp_attribute *)tempA->data;
 			if(a->name && a->value) {
 				if(!strcasecmp(a->name, "ssrc")) {
-					int res = janus_sdp_parse_ssrc(stream, (const char *)a->value, m->type == JANUS_SDP_VIDEO);
+					int res = janus_sdp_parse_ssrc(medium, (const char *)a->value, m->type == JANUS_SDP_VIDEO);
 					if(res != 0) {
 						JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to parse SSRC attribute... (%d)\n", handle->handle_id, res);
 					}
@@ -483,20 +402,15 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 						JANUS_LOG(LOG_VERB, "[%"SCNu64"] This is a SCTP candidate but we're bundling on another stream, ignoring...\n", handle->handle_id);
 #endif
 					} else {
-						int res = janus_sdp_parse_candidate(stream, (const char *)a->value, 0);
+						int res = janus_sdp_parse_candidate(pc, (const char *)a->value, 0);
 						if(res != 0) {
 							JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to parse candidate... (%d)\n", handle->handle_id, res);
 						}
 					}
 				} else if(!strcasecmp(a->name, "rtcp-fb")) {
-					if(a->value && strstr(a->value, "nack") && stream->component) {
-						if(m->type == JANUS_SDP_AUDIO) {
-							/* Enable NACKs for audio */
-							stream->component->do_audio_nacks = TRUE;
-						} else if(m->type == JANUS_SDP_VIDEO) {
-							/* Enable NACKs for video */
-							stream->component->do_video_nacks = TRUE;
-						}
+					if(a->value && strstr(a->value, "nack") && medium) {
+						/* Enable NACKs */
+						medium->do_nacks = TRUE;
 					}
 				} else if(!strcasecmp(a->name, "fmtp")) {
 					if(a->value && strstr(a->value, "apt=")) {
@@ -507,10 +421,10 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 						} else {
 							if(janus_is_rfc4588_enabled()) {
 								rtx = TRUE;
-								janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX);
-								if(stream->rtx_payload_types == NULL)
-									stream->rtx_payload_types = g_hash_table_new(NULL, NULL);
-								g_hash_table_insert(stream->rtx_payload_types, GINT_TO_POINTER(ptype), GINT_TO_POINTER(rtx_ptype));
+								janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX);
+								if(medium->rtx_payload_types == NULL)
+									medium->rtx_payload_types = g_hash_table_new(NULL, NULL);
+								g_hash_table_insert(medium->rtx_payload_types, GINT_TO_POINTER(ptype), GINT_TO_POINTER(rtx_ptype));
 							}
 						}
 					}
@@ -525,75 +439,80 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 			tempA = tempA->next;
 		}
 		/* Any change in SSRCs we should be aware of? */
-		if(m->type == JANUS_SDP_AUDIO) {
-			if(stream->audio_ssrc_peer_new > 0) {
-				if(stream->audio_ssrc_peer > 0 && stream->audio_ssrc_peer != stream->audio_ssrc_peer_new) {
-					JANUS_LOG(LOG_INFO, "[%"SCNu64"] Audio SSRC changed: %"SCNu32" --> %"SCNu32"\n",
-						handle->handle_id, stream->audio_ssrc_peer, stream->audio_ssrc_peer_new);
-					/* FIXME Reset the RTCP context */
-					janus_ice_component *component = stream->component;
-					janus_mutex_lock(&component->mutex);
-					if(stream->audio_rtcp_ctx) {
-						memset(stream->audio_rtcp_ctx, 0, sizeof(*stream->audio_rtcp_ctx));
-						stream->audio_rtcp_ctx->tb = 48000;	/* May change later */
-					}
-					if(component->last_seqs_audio)
-						janus_seq_list_free(&component->last_seqs_audio);
-					janus_mutex_unlock(&component->mutex);
-				}
-				stream->audio_ssrc_peer = stream->audio_ssrc_peer_new;
-				stream->audio_ssrc_peer_new = 0;
-			}
-		} else if(m->type == JANUS_SDP_VIDEO) {
+		if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
 			int vindex = 0;
 			for(vindex=0; vindex<3; vindex++) {
-				if(stream->video_ssrc_peer_new[vindex] > 0) {
-					if(stream->video_ssrc_peer[vindex] > 0 && stream->video_ssrc_peer[vindex] != stream->video_ssrc_peer_new[vindex]) {
-						JANUS_LOG(LOG_INFO, "[%"SCNu64"] Video SSRC (#%d) changed: %"SCNu32" --> %"SCNu32"\n",
-							handle->handle_id, vindex, stream->video_ssrc_peer[vindex], stream->video_ssrc_peer_new[vindex]);
+				if(medium->ssrc_peer_new[vindex] > 0) {
+					if(medium->ssrc_peer[vindex] > 0 && medium->ssrc_peer[vindex] != medium->ssrc_peer_new[vindex]) {
+						JANUS_LOG(LOG_INFO, "[%"SCNu64"] %s SSRC (#%d) changed: %"SCNu32" --> %"SCNu32"\n",
+							handle->handle_id, m->type == JANUS_SDP_VIDEO ? "Video" : "Audio",
+							vindex, medium->ssrc_peer[vindex], medium->ssrc_peer_new[vindex]);
 						/* FIXME Reset the RTCP context */
-						janus_ice_component *component = stream->component;
-						if(component != NULL) {
-							janus_mutex_lock(&component->mutex);
-							if(stream->video_rtcp_ctx[vindex]) {
-								memset(stream->video_rtcp_ctx[vindex], 0, sizeof(*stream->video_rtcp_ctx[vindex]));
-								stream->video_rtcp_ctx[vindex]->tb = 90000;
+						if(medium != NULL) {
+							janus_mutex_lock(&medium->mutex);
+							if(medium->rtcp_ctx[vindex]) {
+								memset(medium->rtcp_ctx[vindex], 0, sizeof(*medium->rtcp_ctx[vindex]));
+								medium->rtcp_ctx[vindex]->tb = (m->type == JANUS_SDP_VIDEO ? 90000 : 48000);	/* May change later */
 							}
-							if(component->last_seqs_video[vindex])
-								janus_seq_list_free(&component->last_seqs_video[vindex]);
-							janus_mutex_unlock(&component->mutex);
+							if(medium->last_seqs[vindex])
+								janus_seq_list_free(&medium->last_seqs[vindex]);
+							janus_mutex_unlock(&medium->mutex);
 						}
 					}
-					stream->video_ssrc_peer[vindex] = stream->video_ssrc_peer_new[vindex];
-					stream->video_ssrc_peer_new[vindex] = 0;
+					medium->ssrc_peer[vindex] = medium->ssrc_peer_new[vindex];
+					medium->ssrc_peer_new[vindex] = 0;
+				}
+				if(!g_hash_table_lookup(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_peer[vindex]))) {
+					g_hash_table_insert(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_peer[vindex]), medium);
+					janus_refcount_increase(&medium->ref);
 				}
 				/* Do the same with the related rtx SSRC, if any */
-				if(stream->video_ssrc_peer_rtx_new[vindex] > 0) {
-					if(stream->video_ssrc_peer_rtx[vindex] > 0 && stream->video_ssrc_peer_rtx[vindex] != stream->video_ssrc_peer_rtx_new[vindex]) {
-						JANUS_LOG(LOG_INFO, "[%"SCNu64"] Video SSRC (#%d rtx) changed: %"SCNu32" --> %"SCNu32"\n",
-							handle->handle_id, vindex, stream->video_ssrc_peer_rtx[vindex], stream->video_ssrc_peer_rtx_new[vindex]);
+				if(medium->ssrc_peer_rtx_new[vindex] > 0) {
+					if(medium->ssrc_peer_rtx[vindex] > 0 && medium->ssrc_peer_rtx[vindex] != medium->ssrc_peer_rtx_new[vindex]) {
+						JANUS_LOG(LOG_INFO, "[%"SCNu64"] %s SSRC (#%d rtx) changed: %"SCNu32" --> %"SCNu32"\n",
+							handle->handle_id, m->type == JANUS_SDP_VIDEO ? "Video" : "Audio",
+							vindex, medium->ssrc_peer_rtx[vindex], medium->ssrc_peer_rtx_new[vindex]);
+					}
+					medium->ssrc_peer_rtx[vindex] = medium->ssrc_peer_rtx_new[vindex];
+					medium->ssrc_peer_rtx_new[vindex] = 0;
+					if(medium->ssrc_rtx == 0)
+						medium->ssrc_rtx = janus_random_uint32();	/* FIXME Should we look for conflicts? */
+					if(!g_hash_table_lookup(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_peer_rtx[vindex]))) {
+						g_hash_table_insert(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_peer_rtx[vindex]), medium);
+						janus_refcount_increase(&medium->ref);
+					}
+					if(!g_hash_table_lookup(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_rtx))) {
+						g_hash_table_insert(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_rtx), medium);
+						janus_refcount_increase(&medium->ref);
 					}
-					stream->video_ssrc_peer_rtx[vindex] = stream->video_ssrc_peer_rtx_new[vindex];
-					stream->video_ssrc_peer_rtx_new[vindex] = 0;
-					if(stream->video_ssrc_rtx == 0)
-						stream->video_ssrc_rtx = janus_random_uint32();	/* FIXME Should we look for conflicts? */
 				}
 			}
-			if(stream->video_ssrc_peer[1] && stream->video_rtcp_ctx[1] == NULL) {
-				stream->video_rtcp_ctx[1] = g_malloc0(sizeof(rtcp_context));
-				stream->video_rtcp_ctx[1]->tb = 90000;
-			}
-			if(stream->video_ssrc_peer[2] && stream->video_rtcp_ctx[2] == NULL) {
-				stream->video_rtcp_ctx[2] = g_malloc0(sizeof(rtcp_context));
-				stream->video_rtcp_ctx[2]->tb = 90000;
+			if(m->type == JANUS_SDP_VIDEO) {
+				if(medium->ssrc_peer[1] && medium->rtcp_ctx[1] == NULL) {
+					medium->rtcp_ctx[1] = g_malloc0(sizeof(rtcp_context));
+					medium->rtcp_ctx[1]->tb = 90000;
+				}
+				if(medium->ssrc_peer[2] && medium->rtcp_ctx[2] == NULL) {
+					medium->rtcp_ctx[2] = g_malloc0(sizeof(rtcp_context));
+					medium->rtcp_ctx[2]->tb = 90000;
+				}
 			}
 		}
 		temp = temp->next;
 	}
 	/* Disable RFC4588 if the peer didn't negotiate it */
 	if(!rtx) {
-		janus_flags_clear(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX);
-		stream->video_ssrc_rtx = 0;
+		janus_flags_clear(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX);
+		/* Iterate on all media */
+		janus_handle_webrtc_medium *medium = NULL;
+		uint mi=0;
+		for(mi=0; mi<g_hash_table_size(pc->media); mi++) {
+			medium = g_hash_table_lookup(pc->media, GUINT_TO_POINTER(mi));
+			if(medium) {
+				g_hash_table_remove(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_rtx));
+				medium->ssrc_rtx = 0;
+			}
+		}
 	}
 	/* Cleanup */
 	g_free(ruser);
@@ -604,14 +523,109 @@ int janus_sdp_process(void *ice_handle, janus_sdp *remote_sdp, gboolean update)
 	return 0;	/* FIXME Handle errors better */
 }
 
-int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trickle) {
-	if(ice_stream == NULL || candidate == NULL)
+/* Parse local SDP */
+int janus_sdp_process_local(void *ice_handle, janus_sdp *remote_sdp, gboolean update) {
+	if(!ice_handle || !remote_sdp)
+		return -1;
+	janus_handle *handle = (janus_handle *)ice_handle;
+	janus_handle_webrtc *pc = handle->pc;
+	if(!pc)
+		return -1;
+	janus_handle_webrtc_medium *medium = NULL;
+	/* We only go through m-lines to setup medium instances accordingly */
+	int mlines = -1;
+	GList *temp = remote_sdp->m_lines;
+	while(temp) {
+		mlines++;
+		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
+		/* Find the internal medium instance */
+		medium = g_hash_table_lookup(pc->media, GINT_TO_POINTER(m->index));
+		if(!medium) {
+			/* We don't have it, create one now */
+			if(m->type == JANUS_SDP_AUDIO)
+				medium = janus_handle_webrtc_medium_create(handle, JANUS_MEDIA_AUDIO);
+			else if(m->type == JANUS_SDP_VIDEO)
+				medium = janus_handle_webrtc_medium_create(handle, JANUS_MEDIA_VIDEO);
+			else if(m->type == JANUS_SDP_APPLICATION && strstr(m->proto, "DTLS/SCTP"))
+				medium = janus_handle_webrtc_medium_create(handle, JANUS_MEDIA_DATA);
+			else
+				medium = janus_handle_webrtc_medium_create(handle, JANUS_MEDIA_DATA);
+		}
+		/* Check if the offer contributed an mid */
+		GList *tempA = m->attributes;
+		while(tempA) {
+			janus_sdp_attribute *a = (janus_sdp_attribute *)tempA->data;
+			if(a->name) {
+				if(!strcasecmp(a->name, "mid")) {
+					/* Found mid attribute */
+					if(medium->mid == NULL) {
+						medium->mid = g_strdup(a->value);
+						if(!g_hash_table_lookup(pc->media_bymid, medium->mid)) {
+							g_hash_table_insert(pc->media_bymid, g_strdup(medium->mid), medium);
+							janus_refcount_increase(&medium->ref);
+						}
+					}
+					if(handle->pc_mid == NULL)
+						handle->pc_mid = g_strdup(a->value);
+				}
+			}
+			tempA = tempA->next;
+		}
+		if(medium->mid == NULL) {
+			/* No mid provided, generate one now */
+			char mid[5];
+			memset(mid, 0, sizeof(mid));
+			g_snprintf(mid, sizeof(mid), "%d", mlines);
+			medium->mid = g_strdup(mid);
+			if(!g_hash_table_lookup(pc->media_bymid, medium->mid)) {
+				g_hash_table_insert(pc->media_bymid, g_strdup(medium->mid), medium);
+				janus_refcount_increase(&medium->ref);
+			}
+		}
+		if(m->direction == JANUS_SDP_INACTIVE) {
+			/* FIXME Reset the local SSRCs and RTCP context */
+			if(medium->ssrc != 0)
+				g_hash_table_remove(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc));
+			medium->ssrc = 0;
+			if(medium->ssrc_rtx != 0)
+				g_hash_table_remove(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_rtx));
+			medium->ssrc_rtx = 0;
+			int vindex = 0;
+			for(vindex=0; vindex<3; vindex++) {
+				if(medium->rtcp_ctx[vindex]) {
+					int tb = medium->rtcp_ctx[vindex]->tb;
+					memset(medium->rtcp_ctx[vindex], 0, sizeof(janus_rtcp_context));
+					medium->rtcp_ctx[vindex]->tb = tb;
+				}
+			}
+		} else if(m->type != JANUS_SDP_APPLICATION) {
+			if(medium->ssrc == 0) {
+				medium->ssrc = janus_random_uint32();	/* FIXME Should we look for conflicts? */
+				if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX)) {
+					/* Create an SSRC for RFC4588 as well */
+					medium->ssrc_rtx = janus_random_uint32();	/* FIXME Should we look for conflicts? */
+				}
+				/* Update the SSRC-indexed map */
+				g_hash_table_insert(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc), medium);
+				janus_refcount_increase(&medium->ref);
+				if(medium->ssrc_rtx > 0) {
+					g_hash_table_insert(pc->media_byssrc, GINT_TO_POINTER(medium->ssrc_rtx), medium);
+					janus_refcount_increase(&medium->ref);
+				}
+			}
+		}
+		temp = temp->next;
+	}
+	return 0;	/* FIXME Handle errors better */
+}
+
+int janus_sdp_parse_candidate(void *webrtc, const char *candidate, int trickle) {
+	if(webrtc == NULL || candidate == NULL)
 		return -1;
-	janus_ice_stream *stream = (janus_ice_stream *)ice_stream;
-	janus_ice_handle *handle = stream->handle;
+	janus_handle_webrtc *pc = (janus_handle_webrtc *)webrtc;
+	janus_handle *handle = pc->handle;
 	if(handle == NULL)
 		return -2;
-	janus_ice_component *component = NULL;
 	if(strlen(candidate) == 0 || strstr(candidate, "end-of-candidates")) {
 		/* FIXME Should we do something with this? */
 		JANUS_LOG(LOG_VERB, "[%"SCNu64"] end-of-candidates received\n", handle->handle_id);
@@ -655,23 +669,22 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 			g_strlcpy(rip, janus_network_address_string_from_buffer(&addr_buf), sizeof(rip));
 		}
 		/* Add remote candidate */
-		component = stream->component;
-		if(component == NULL || rcomponent > 1) {
-			JANUS_LOG(LOG_VERB, "[%"SCNu64"]   -- Skipping component %d in stream %d (rtcp-muxing)\n", handle->handle_id, rcomponent, stream->stream_id);
+		if(rcomponent > 1) {
+			JANUS_LOG(LOG_VERB, "[%"SCNu64"]   -- Skipping component %d in stream %d (rtcp-muxing)\n", handle->handle_id, rcomponent, pc->stream_id);
 		} else {
 			//~ if(trickle) {
-				//~ if(component->dtls != NULL) {
+				//~ if(pc->dtls != NULL) {
 					//~ /* This component is already ready, ignore this further candidate */
 					//~ JANUS_LOG(LOG_VERB, "[%"SCNu64"]   -- Ignoring this candidate, the component is already ready\n", handle->handle_id);
 					//~ return 0;
 				//~ }
 			//~ }
-			component->component_id = rcomponent;
-			component->stream_id = stream->stream_id;
+			pc->component_id = rcomponent;
+			pc->stream_id = pc->stream_id;
 			NiceCandidate *c = NULL;
 			if(!strcasecmp(rtype, "host")) {
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"]  Adding remote candidate component:%d stream:%d type:host %s:%d\n",
-					handle->handle_id, rcomponent, stream->stream_id, rip, rport);
+					handle->handle_id, rcomponent, pc->stream_id, rip, rport);
 				/* Unless this is libnice >= 0.1.8, we only support UDP... */
 				if(!strcasecmp(rtransport, "udp")) {
 					c = nice_candidate_new(NICE_CANDIDATE_TYPE_HOST);
@@ -684,7 +697,7 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 				}
 			} else if(!strcasecmp(rtype, "srflx")) {
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"]  Adding remote candidate component:%d stream:%d type:srflx %s:%d --> %s:%d \n",
-					handle->handle_id, rcomponent, stream->stream_id,  rrelip, rrelport, rip, rport);
+					handle->handle_id, rcomponent, pc->stream_id,  rrelip, rrelport, rip, rport);
 				/* Unless this is libnice >= 0.1.8, we only support UDP... */
 				if(!strcasecmp(rtransport, "udp")) {
 					c = nice_candidate_new(NICE_CANDIDATE_TYPE_SERVER_REFLEXIVE);
@@ -697,7 +710,7 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 				}
 			} else if(!strcasecmp(rtype, "prflx")) {
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"]  Adding remote candidate component:%d stream:%d type:prflx %s:%d --> %s:%d\n",
-					handle->handle_id, rcomponent, stream->stream_id, rrelip, rrelport, rip, rport);
+					handle->handle_id, rcomponent, pc->stream_id, rrelip, rrelport, rip, rport);
 				/* Unless this is libnice >= 0.1.8, we only support UDP... */
 				if(!strcasecmp(rtransport, "udp")) {
 					c = nice_candidate_new(NICE_CANDIDATE_TYPE_PEER_REFLEXIVE);
@@ -710,7 +723,7 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 				}
 			} else if(!strcasecmp(rtype, "relay")) {
 				JANUS_LOG(LOG_VERB, "[%"SCNu64"]  Adding remote candidate component:%d stream:%d type:relay %s:%d --> %s:%d\n",
-					handle->handle_id, rcomponent, stream->stream_id, rrelip, rrelport, rip, rport);
+					handle->handle_id, rcomponent, pc->stream_id, rrelip, rrelport, rip, rport);
 				/* We only support UDP/TCP/TLS... */
 				if(strcasecmp(rtransport, "udp") && strcasecmp(rtransport, "tcp") && strcasecmp(rtransport, "tls")) {
 					JANUS_LOG(LOG_VERB, "[%"SCNu64"]    Skipping unsupported transport '%s' for media\n", handle->handle_id, rtransport);
@@ -720,11 +733,11 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 			} else {
 				/* FIXME What now? */
 				JANUS_LOG(LOG_ERR, "[%"SCNu64"]  Unknown remote candidate type:%s for component:%d stream:%d!\n",
-					handle->handle_id, rtype, rcomponent, stream->stream_id);
+					handle->handle_id, rtype, rcomponent, pc->stream_id);
 			}
 			if(c != NULL) {
 				c->component_id = rcomponent;
-				c->stream_id = stream->stream_id;
+				c->stream_id = pc->stream_id;
 #ifndef HAVE_LIBNICE_TCP
 				c->transport = NICE_CANDIDATE_TRANSPORT_UDP;
 #else
@@ -756,8 +769,8 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 				c->priority = rpriority;
 				nice_address_set_from_string(&c->addr, rip);
 				nice_address_set_port(&c->addr, rport);
-				c->username = g_strdup(stream->ruser);
-				c->password = g_strdup(stream->rpass);
+				c->username = g_strdup(pc->ruser);
+				c->password = g_strdup(pc->rpass);
 				if(c->type == NICE_CANDIDATE_TYPE_SERVER_REFLEXIVE || c->type == NICE_CANDIDATE_TYPE_PEER_REFLEXIVE) {
 					nice_address_set_from_string(&c->base_addr, rrelip);
 					nice_address_set_port(&c->base_addr, rrelport);
@@ -766,32 +779,32 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 					nice_address_set_from_string(&c->base_addr, rrelip);
 					nice_address_set_port(&c->base_addr, rrelport);
 				}
-				component->candidates = g_slist_append(component->candidates, c);
+				pc->candidates = g_slist_append(pc->candidates, c);
 				JANUS_LOG(LOG_HUGE, "[%"SCNu64"]    Candidate added to the list! (%u elements for %d/%d)\n", handle->handle_id,
-					g_slist_length(component->candidates), stream->stream_id, component->component_id);
+					g_slist_length(pc->candidates), pc->stream_id, pc->component_id);
 				/* Save for the summary, in case we need it */
-				component->remote_candidates = g_slist_append(component->remote_candidates, g_strdup(candidate));
+				pc->remote_candidates = g_slist_append(pc->remote_candidates, g_strdup(candidate));
 				/* Notify event handlers */
 				if(janus_events_is_enabled()) {
 					janus_session *session = (janus_session *)handle->session;
 					json_t *info = json_object();
 					json_object_set_new(info, "remote-candidate", json_string(candidate));
-					json_object_set_new(info, "stream_id", json_integer(stream->stream_id));
-					json_object_set_new(info, "component_id", json_integer(component->component_id));
+					json_object_set_new(info, "stream_id", json_integer(pc->stream_id));
+					json_object_set_new(info, "component_id", json_integer(pc->component_id));
 					janus_events_notify_handlers(JANUS_EVENT_TYPE_WEBRTC, session->session_id, handle->handle_id, handle->opaque_id, info);
 				}
 				/* See if we need to process this */
 				if(trickle) {
-					if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_START)) {
+					if(janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_START)) {
 						/* This is a trickle candidate and ICE has started, we should process it right away */
-						if(!component->process_started) {
+						if(!pc->process_started) {
 							/* Actually, ICE has JUST started for this component, take care of the candidates we've added so far */
 							JANUS_LOG(LOG_VERB, "[%"SCNu64"] ICE already started for this component, setting candidates we have up to now\n", handle->handle_id);
-							janus_ice_setup_remote_candidates(handle, component->stream_id, component->component_id);
+							janus_handle_setup_remote_candidates(handle, pc->stream_id, pc->component_id);
 						} else {
 							GSList *candidates = NULL;
 							candidates = g_slist_append(candidates, c);
-							if(nice_agent_set_remote_candidates(handle->agent, stream->stream_id, component->component_id, candidates) < 1) {
+							if(nice_agent_set_remote_candidates(handle->agent, pc->stream_id, pc->component_id, candidates) < 1) {
 								JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to add trickle candidate :-(\n", handle->handle_id);
 							} else {
 								JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Trickle candidate added!\n", handle->handle_id);
@@ -800,17 +813,17 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 						}
 					} else {
 						/* ICE hasn't started yet: to make sure we're not stuck, also check if we stopped processing the SDP */
-						if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_PROCESSING_OFFER)) {
-							janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_START);
+						if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_PROCESSING_OFFER)) {
+							janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_START);
 							/* This is a trickle candidate and ICE has started, we should process it right away */
-							if(!component->process_started) {
+							if(!pc->process_started) {
 								/* Actually, ICE has JUST started for this component, take care of the candidates we've added so far */
 								JANUS_LOG(LOG_VERB, "[%"SCNu64"] SDP processed but ICE not started yet for this component, setting candidates we have up to now\n", handle->handle_id);
-								janus_ice_setup_remote_candidates(handle, component->stream_id, component->component_id);
+								janus_handle_setup_remote_candidates(handle, pc->stream_id, pc->component_id);
 							} else {
 								GSList *candidates = NULL;
 								candidates = g_slist_append(candidates, c);
-								if(nice_agent_set_remote_candidates(handle->agent, stream->stream_id, component->component_id, candidates) < 1) {
+								if(nice_agent_set_remote_candidates(handle->agent, pc->stream_id, pc->component_id, candidates) < 1) {
 									JANUS_LOG(LOG_ERR, "[%"SCNu64"] Failed to add trickle candidate :-(\n", handle->handle_id);
 								} else {
 									JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Trickle candidate added!\n", handle->handle_id);
@@ -832,16 +845,17 @@ int janus_sdp_parse_candidate(void *ice_stream, const char *candidate, int trick
 	return 0;
 }
 
-int janus_sdp_parse_ssrc_group(void *ice_stream, const char *group_attr, int video) {
-	if(ice_stream == NULL || group_attr == NULL)
+int janus_sdp_parse_ssrc_group(void *m, const char *group_attr, int video) {
+	if(m == NULL || group_attr == NULL)
 		return -1;
-	janus_ice_stream *stream = (janus_ice_stream *)ice_stream;
-	janus_ice_handle *handle = stream->handle;
+	janus_handle_webrtc_medium *medium = (janus_handle_webrtc_medium *)m;
+	janus_handle_webrtc *pc = medium->pc;
+	janus_handle *handle = pc->handle;
 	if(handle == NULL)
 		return -2;
 	if(!video)
 		return -3;
-	if(stream->rid[0] != NULL) {
+	if(medium->rid[0] != NULL) {
 		/* Simulcasting is rid-based, don't parse SSRCs for now */
 		return 0;
 	}
@@ -859,23 +873,23 @@ int janus_sdp_parse_ssrc_group(void *ice_stream, const char *group_attr, int vid
 				switch(i) {
 					case 1:
 						first_ssrc = ssrc;
-						if(stream->video_ssrc_peer_new[0] == ssrc || stream->video_ssrc_peer_new[1] == ssrc
-								|| stream->video_ssrc_peer_new[2] == ssrc) {
+						if(medium->ssrc_peer_new[0] == ssrc || medium->ssrc_peer_new[1] == ssrc
+								|| medium->ssrc_peer_new[2] == ssrc) {
 							JANUS_LOG(LOG_HUGE, "[%"SCNu64"] Already parsed this SSRC: %"SCNu64" (%s group)\n",
 								handle->handle_id, ssrc, (fid ? "FID" : (sim ? "SIM" : "??")));
 						} else {
-							if(stream->video_ssrc_peer_new[0] == 0) {
-								stream->video_ssrc_peer_new[0] = ssrc;
-								JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC: %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_new[0]);
+							if(medium->ssrc_peer_new[0] == 0) {
+								medium->ssrc_peer_new[0] = ssrc;
+								JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC: %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_new[0]);
 							} else {
 								/* We already have a video SSRC: check if rid is involved, and we'll keep track of this for simulcasting */
-								if(stream->rid[0]) {
-									if(stream->video_ssrc_peer_new[1] == 0) {
-										stream->video_ssrc_peer_new[1] = ssrc;
-										JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-1): %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_new[1]);
-									} else if(stream->video_ssrc_peer_new[2] == 0) {
-										stream->video_ssrc_peer_new[2] = ssrc;
-										JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-2): %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_new[2]);
+								if(medium->rid[0]) {
+									if(medium->ssrc_peer_new[1] == 0) {
+										medium->ssrc_peer_new[1] = ssrc;
+										JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-1): %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_new[1]);
+									} else if(medium->ssrc_peer_new[2] == 0) {
+										medium->ssrc_peer_new[2] = ssrc;
+										JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-2): %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_new[2]);
 									} else {
 										JANUS_LOG(LOG_WARN, "[%"SCNu64"] Don't know what to do with video SSRC: %"SCNu64"\n", handle->handle_id, ssrc);
 									}
@@ -885,21 +899,21 @@ int janus_sdp_parse_ssrc_group(void *ice_stream, const char *group_attr, int vid
 						break;
 					case 2:
 						if(fid) {
-							if(stream->video_ssrc_peer_new[0] == first_ssrc && stream->video_ssrc_peer_rtx_new[0] == 0) {
-								stream->video_ssrc_peer_rtx_new[0] = ssrc;
-								JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (rtx): %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_rtx_new[0]);
-							} else if(stream->video_ssrc_peer_new[1] == first_ssrc && stream->video_ssrc_peer_rtx_new[1] == 0) {
-								stream->video_ssrc_peer_rtx_new[1] = ssrc;
-								JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-1 rtx): %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_rtx_new[1]);
-							} else if(stream->video_ssrc_peer_new[2] == first_ssrc && stream->video_ssrc_peer_rtx_new[2] == 0) {
-								stream->video_ssrc_peer_rtx_new[2] = ssrc;
-								JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-2 rtx): %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_rtx_new[2]);
+							if(medium->ssrc_peer_new[0] == first_ssrc && medium->ssrc_peer_rtx_new[0] == 0) {
+								medium->ssrc_peer_rtx_new[0] = ssrc;
+								JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (rtx): %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_rtx_new[0]);
+							} else if(medium->ssrc_peer_new[1] == first_ssrc && medium->ssrc_peer_rtx_new[1] == 0) {
+								medium->ssrc_peer_rtx_new[1] = ssrc;
+								JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-1 rtx): %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_rtx_new[1]);
+							} else if(medium->ssrc_peer_new[2] == first_ssrc && medium->ssrc_peer_rtx_new[2] == 0) {
+								medium->ssrc_peer_rtx_new[2] = ssrc;
+								JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-2 rtx): %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_rtx_new[2]);
 							} else {
 								JANUS_LOG(LOG_WARN, "[%"SCNu64"] Don't know what to do with rtx SSRC: %"SCNu64"\n", handle->handle_id, ssrc);
 							}
 						} else if(sim) {
-							stream->video_ssrc_peer_new[1] = ssrc;
-							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-1): %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_new[1]);
+							medium->ssrc_peer_new[1] = ssrc;
+							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-1): %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_new[1]);
 						} else {
 							JANUS_LOG(LOG_WARN, "[%"SCNu64"] Don't know what to do with SSRC: %"SCNu64"\n", handle->handle_id, ssrc);
 						}
@@ -908,8 +922,8 @@ int janus_sdp_parse_ssrc_group(void *ice_stream, const char *group_attr, int vid
 						if(fid) {
 							JANUS_LOG(LOG_WARN, "[%"SCNu64"] Found one too many retransmission SSRC (rtx): %"SCNu64"\n", handle->handle_id, ssrc);
 						} else if(sim) {
-							stream->video_ssrc_peer_new[2] = ssrc;
-							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-2): %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_new[2]);
+							medium->ssrc_peer_new[2] = ssrc;
+							JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC (sim-2): %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_new[2]);
 						} else {
 							JANUS_LOG(LOG_WARN, "[%"SCNu64"] Don't know what to do with SSRC: %"SCNu64"\n", handle->handle_id, ssrc);
 						}
@@ -927,34 +941,29 @@ int janus_sdp_parse_ssrc_group(void *ice_stream, const char *group_attr, int vid
 	return 0;
 }
 
-int janus_sdp_parse_ssrc(void *ice_stream, const char *ssrc_attr, int video) {
-	if(ice_stream == NULL || ssrc_attr == NULL)
+int janus_sdp_parse_ssrc(void *m, const char *ssrc_attr, int video) {
+	if(m == NULL || ssrc_attr == NULL)
 		return -1;
-	janus_ice_stream *stream = (janus_ice_stream *)ice_stream;
-	janus_ice_handle *handle = stream->handle;
+	janus_handle_webrtc_medium *medium = (janus_handle_webrtc_medium *)m;
+	janus_handle_webrtc *pc = medium->pc;
+	janus_handle *handle = pc->handle;
 	if(handle == NULL)
 		return -2;
 	guint64 ssrc = g_ascii_strtoull(ssrc_attr, NULL, 0);
 	if(ssrc == 0 || ssrc > G_MAXUINT32)
 		return -3;
-	if(video) {
-		if(stream->rid[0] != NULL) {
-			/* Simulcasting is rid-based, only keep track of a single SSRC for fallback */
-			if(stream->video_ssrc_peer_temp == 0) {
-				stream->video_ssrc_peer_temp = ssrc;
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Peer video fallback SSRC: %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_temp);
-			}
-			return 0;
-		}
-		if(stream->video_ssrc_peer_new[0] == 0) {
-			stream->video_ssrc_peer_new[0] = ssrc;
-			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer video SSRC: %"SCNu32"\n", handle->handle_id, stream->video_ssrc_peer_new[0]);
-		}
-	} else {
-		if(stream->audio_ssrc_peer_new == 0) {
-			stream->audio_ssrc_peer_new = ssrc;
-			JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer audio SSRC: %"SCNu32"\n", handle->handle_id, stream->audio_ssrc_peer_new);
+	if(medium->rid[0] != NULL) {
+		/* Simulcasting is rid-based, only keep track of a single SSRC for fallback */
+		if(medium->ssrc_peer_temp == 0) {
+			medium->ssrc_peer_temp = ssrc;
+			JANUS_LOG(LOG_WARN, "[%"SCNu64"] Peer video fallback SSRC: %"SCNu32"\n", handle->handle_id, medium->ssrc_peer_temp);
 		}
+		return 0;
+	}
+	if(medium->ssrc_peer_new[0] == 0) {
+		medium->ssrc_peer_new[0] = ssrc;
+		JANUS_LOG(LOG_VERB, "[%"SCNu64"] Peer %s SSRC: %"SCNu32"\n",
+			handle->handle_id, video ? "video" : "audio", medium->ssrc_peer_new[0]);
 	}
 	return 0;
 }
@@ -995,14 +1004,14 @@ int janus_sdp_anonymize(janus_sdp *anon) {
 		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
 		if(m->type == JANUS_SDP_AUDIO && m->port > 0) {
 			audio++;
-			m->port = audio == 1 ? 9 : 0;
+			m->port = 9;
 		} else if(m->type == JANUS_SDP_VIDEO && m->port > 0) {
 			video++;
-			m->port = video == 1 ? 9 : 0;
+			m->port = 9;
 		} else if(m->type == JANUS_SDP_APPLICATION && m->port > 0) {
 			if(m->proto != NULL && (!strcasecmp(m->proto, "DTLS/SCTP") || !strcasecmp(m->proto, "UDP/DTLS/SCTP"))) {
 				data++;
-				m->port = data == 1 ? 9 : 0;
+				m->port = (data == 1 ? 9 : 0);
 			} else {
 				m->port = 0;
 			}
@@ -1082,7 +1091,7 @@ int janus_sdp_anonymize(janus_sdp *anon) {
 			tempA = purged_ptypes;
 			while(tempA) {
 				int ptype = GPOINTER_TO_INT(tempA->data);
-				janus_sdp_remove_payload_type(anon, ptype);
+				janus_sdp_remove_payload_type(anon, m->index, ptype);
 				tempA = tempA->next;
 			}
 			g_list_free(purged_ptypes);
@@ -1101,10 +1110,11 @@ int janus_sdp_anonymize(janus_sdp *anon) {
 char *janus_sdp_merge(void *ice_handle, janus_sdp *anon, gboolean offer) {
 	if(ice_handle == NULL || anon == NULL)
 		return NULL;
-	janus_ice_handle *handle = (janus_ice_handle *)ice_handle;
-	janus_ice_stream *stream = handle->stream;
-	if(stream == NULL)
+	janus_handle *handle = (janus_handle *)ice_handle;
+	janus_handle_webrtc *pc = handle->pc;
+	if(pc == NULL)
 		return NULL;
+	janus_handle_webrtc_medium *medium = NULL;
 	char *rtp_profile = handle->rtp_profile ? handle->rtp_profile : (char *)"UDP/TLS/RTP/SAVPF";
 	if(!janus_is_webrtc_encryption_enabled())
 		rtp_profile = (char *)"RTP/AVPF";
@@ -1127,50 +1137,38 @@ char *janus_sdp_merge(void *ice_handle, janus_sdp *anon, gboolean offer) {
 	g_free(anon->c_addr);
 	anon->c_addr = NULL;
 	/* bundle: add new global attribute */
-	char buffer[2048], buffer_part[512];
+	char buffer[JANUS_BUFSIZE], buffer_part[512];
 	buffer[0] = '\0';
 	buffer_part[0] = '\0';
 	g_snprintf(buffer, sizeof(buffer), "BUNDLE");
 	/* Iterate on available media */
-	int audio = 0;
-	int video = 0;
 #ifdef HAVE_SCTP
 	int data = 0;
 #endif
 	GList *temp = anon->m_lines;
 	while(temp) {
 		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
-		if(m->type == JANUS_SDP_AUDIO) {
-			audio++;
-			if(audio == 1) {
-				g_snprintf(buffer_part, sizeof(buffer_part),
-					" %s", handle->audio_mid ? handle->audio_mid : "audio");
-				g_strlcat(buffer, buffer_part, JANUS_BUFSIZE);
-			}
-		} else if(m->type == JANUS_SDP_VIDEO) {
-			video++;
-			if(video == 1) {
-				g_snprintf(buffer_part, sizeof(buffer_part),
-					" %s", handle->video_mid ? handle->video_mid : "video");
-				g_strlcat(buffer, buffer_part, JANUS_BUFSIZE);
-			}
-#ifdef HAVE_SCTP
-		} else if(m->type == JANUS_SDP_APPLICATION) {
-			if(m->proto && (!strcasecmp(m->proto, "DTLS/SCTP") || !strcasecmp(m->proto, "UDP/DTLS/SCTP")))
-				data++;
-			if(data == 1) {
-				g_snprintf(buffer_part, sizeof(buffer_part),
-					" %s", handle->data_mid ? handle->data_mid : "data");
-				g_strlcat(buffer, buffer_part, JANUS_BUFSIZE);
-			}
-#endif
+		/* Find the internal medium instance */
+		medium = g_hash_table_lookup(pc->media, GINT_TO_POINTER(m->index));
+		if(!medium) {
+			/* TODO We don't have it, which should never happen! */
 		}
+		g_snprintf(buffer_part, sizeof(buffer_part), " %s", medium->mid);
+		g_strlcat(buffer, buffer_part, JANUS_BUFSIZE);
 		temp = temp->next;
 	}
 	/* Global attributes: start with group */
 	GList *first = anon->attributes;
 	janus_sdp_attribute *a = janus_sdp_attribute_create("group", "%s", buffer);
 	anon->attributes = g_list_insert_before(anon->attributes, first, a);
+	/* Advertise trickle support */
+	a = janus_sdp_attribute_create("ice-options", "trickle");
+	anon->attributes = g_list_insert_before(anon->attributes, first, a);
+	if(janus_is_webrtc_encryption_enabled()) {
+		/* We put the fingerprint in the global attributes */
+		a = janus_sdp_attribute_create("fingerprint", "sha-256 %s", janus_dtls_get_local_fingerprint());
+		anon->attributes = g_list_insert_before(anon->attributes, first, a);
+	}
 	/* msid-semantic: add new global attribute */
 	a = janus_sdp_attribute_create("msid-semantic", " WMS janus");
 	anon->attributes = g_list_insert_before(anon->attributes, first, a);
@@ -1181,8 +1179,6 @@ char *janus_sdp_merge(void *ice_handle, janus_sdp *anon, gboolean offer) {
 		anon->attributes = g_list_insert_before(anon->attributes, first, a);
 	}
 	/* Media lines now */
-	audio = 0;
-	video = 0;
 #ifdef HAVE_SCTP
 	data = 0;
 #endif
@@ -1190,6 +1186,11 @@ char *janus_sdp_merge(void *ice_handle, janus_sdp *anon, gboolean offer) {
 	while(temp) {
 		janus_sdp_mline *m = (janus_sdp_mline *)temp->data;
 		first = m->attributes;
+		/* Find the internal medium instance */
+		medium = g_hash_table_lookup(pc->media, GINT_TO_POINTER(m->index));
+		if(!medium) {
+			/* TODO We don't have it, which should never happen! */
+		}
 		/* Overwrite RTP profile for audio and video */
 		if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
 			g_free(m->proto);
@@ -1200,94 +1201,54 @@ char *janus_sdp_merge(void *ice_handle, janus_sdp *anon, gboolean offer) {
 		m->c_ipv4 = ipv4;
 		m->c_addr = g_strdup(janus_get_public_ip());
 		/* Check if we need to refuse the media or not */
-		if(m->type == JANUS_SDP_AUDIO) {
-			audio++;
-			/* Audio */
-			if(audio > 1) {
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Skipping audio line (we have one already)\n", handle->handle_id);
-				m->port = 0;
-			}
+		if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
+			/* Audio/Video */
 			if(m->port == 0) {
 				m->direction = JANUS_SDP_INACTIVE;
-				stream->audio_ssrc = 0;
-			}
-			if(audio == 1) {
-				switch(m->direction) {
-					case JANUS_SDP_INACTIVE:
-						stream->audio_send = FALSE;
-						stream->audio_recv = FALSE;
-						break;
-					case JANUS_SDP_SENDONLY:
-						stream->audio_send = TRUE;
-						stream->audio_recv = FALSE;
-						break;
-					case JANUS_SDP_RECVONLY:
-						stream->audio_send = FALSE;
-						stream->audio_recv = TRUE;
-						break;
-					case JANUS_SDP_SENDRECV:
-					case JANUS_SDP_DEFAULT:
-					default:
-						stream->audio_send = TRUE;
-						stream->audio_recv = TRUE;
-						break;
-				}
+				medium->ssrc = 0;
 			}
-		} else if(m->type == JANUS_SDP_VIDEO) {
-			video++;
-			/* Video */
-			if(video > 1) {
-				JANUS_LOG(LOG_WARN, "[%"SCNu64"] Skipping video line (we have one already)\n", handle->handle_id);
-				m->port = 0;
-			}
-			if(m->port == 0) {
-				m->direction = JANUS_SDP_INACTIVE;
-				stream->video_ssrc = 0;
+			switch(m->direction) {
+				case JANUS_SDP_INACTIVE:
+					medium->send = FALSE;
+					medium->recv = FALSE;
+					break;
+				case JANUS_SDP_SENDONLY:
+					medium->send = TRUE;
+					medium->recv = FALSE;
+					break;
+				case JANUS_SDP_RECVONLY:
+					medium->send = FALSE;
+					medium->recv = TRUE;
+					break;
+				case JANUS_SDP_SENDRECV:
+				case JANUS_SDP_DEFAULT:
+				default:
+					medium->send = TRUE;
+					medium->recv = TRUE;
+					break;
 			}
-			if(video == 1) {
-				switch(m->direction) {
-					case JANUS_SDP_INACTIVE:
-						stream->video_send = FALSE;
-						stream->video_recv = FALSE;
-						break;
-					case JANUS_SDP_SENDONLY:
-						stream->video_send = TRUE;
-						stream->video_recv = FALSE;
-						break;
-					case JANUS_SDP_RECVONLY:
-						stream->video_send = FALSE;
-						stream->video_recv = TRUE;
-						break;
-					case JANUS_SDP_SENDRECV:
-					case JANUS_SDP_DEFAULT:
-					default:
-						stream->video_send = TRUE;
-						stream->video_recv = TRUE;
-						break;
-				}
-				if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX)) {
-					/* Add RFC4588 stuff */
-					if(stream->rtx_payload_types && g_hash_table_size(stream->rtx_payload_types) > 0) {
-						janus_sdp_attribute *a = NULL;
-						GList *ptypes = g_list_copy(m->ptypes), *tempP = ptypes;
-						while(tempP) {
-							int ptype = GPOINTER_TO_INT(tempP->data);
-							int rtx_ptype = GPOINTER_TO_INT(g_hash_table_lookup(stream->rtx_payload_types, GINT_TO_POINTER(ptype)));
-							if(rtx_ptype > 0) {
-								m->ptypes = g_list_append(m->ptypes, GINT_TO_POINTER(rtx_ptype));
-								a = janus_sdp_attribute_create("rtpmap", "%d rtx/90000", rtx_ptype);
-								m->attributes = g_list_append(m->attributes, a);
-								a = janus_sdp_attribute_create("fmtp", "%d apt=%d", rtx_ptype, ptype);
-								m->attributes = g_list_append(m->attributes, a);
-							}
-							tempP = tempP->next;
+			if(medium->do_nacks && janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX)) {
+				/* Add RFC4588 stuff */
+				if(medium->rtx_payload_types && g_hash_table_size(medium->rtx_payload_types) > 0) {
+					janus_sdp_attribute *a = NULL;
+					GList *ptypes = g_list_copy(m->ptypes), *tempP = ptypes;
+					while(tempP) {
+						int ptype = GPOINTER_TO_INT(tempP->data);
+						int rtx_ptype = GPOINTER_TO_INT(g_hash_table_lookup(medium->rtx_payload_types, GINT_TO_POINTER(ptype)));
+						if(rtx_ptype > 0) {
+							m->ptypes = g_list_append(m->ptypes, GINT_TO_POINTER(rtx_ptype));
+							a = janus_sdp_attribute_create("rtpmap", "%d rtx/90000", rtx_ptype);
+							m->attributes = g_list_append(m->attributes, a);
+							a = janus_sdp_attribute_create("fmtp", "%d apt=%d", rtx_ptype, ptype);
+							m->attributes = g_list_append(m->attributes, a);
 						}
-						g_list_free(ptypes);
+						tempP = tempP->next;
 					}
+					g_list_free(ptypes);
 				}
 			}
-#ifdef HAVE_SCTP
 		} else if(m->type == JANUS_SDP_APPLICATION) {
+#ifdef HAVE_SCTP
 			/* Is this SCTP for DataChannels? */
 			if(m->port > 0 && (!strcasecmp(m->proto, "DTLS/SCTP") || !strcasecmp(m->proto, "UDP/DTLS/SCTP"))) {
 				/* Yep */
@@ -1306,6 +1267,12 @@ char *janus_sdp_merge(void *ice_handle, janus_sdp *anon, gboolean offer) {
 				temp = temp->next;
 				continue;
 			}
+#else
+			JANUS_LOG(LOG_WARN, "[%"SCNu64"] Skipping unsupported application media line...\n", handle->handle_id);
+			m->port = 0;
+			m->direction = JANUS_SDP_INACTIVE;
+			temp = temp->next;
+			continue;
 #endif
 		} else {
 			JANUS_LOG(LOG_WARN, "[%"SCNu64"] Skipping disabled/unsupported media line...\n", handle->handle_id);
@@ -1314,109 +1281,78 @@ char *janus_sdp_merge(void *ice_handle, janus_sdp *anon, gboolean offer) {
 			temp = temp->next;
 			continue;
 		}
-		/* a=mid:(audio|video|data) */
-		if(m->type == JANUS_SDP_AUDIO && audio == 1) {
-			a = janus_sdp_attribute_create("mid", "%s", handle->audio_mid);
+		/* a=mid: */
+		if(medium->mid) {
+			a = janus_sdp_attribute_create("mid", "%s", medium->mid);
 			m->attributes = g_list_insert_before(m->attributes, first, a);
-		} else if(m->type == JANUS_SDP_VIDEO && video == 1) {
-			a = janus_sdp_attribute_create("mid", "%s", handle->video_mid);
-			m->attributes = g_list_insert_before(m->attributes, first, a);
-#ifdef HAVE_SCTP
-		} else if(m->type == JANUS_SDP_APPLICATION && data == 1) {
+		}
+		if(m->type == JANUS_SDP_APPLICATION) {
 			if(!strcasecmp(m->proto, "UDP/DTLS/SCTP"))
-				janus_flags_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
-			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_NEW_DATACHAN_SDP)) {
+				janus_flags_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP);
+			if(!janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_NEW_DATACHAN_SDP)) {
 				a = janus_sdp_attribute_create("sctpmap", "5000 webrtc-datachannel 16");
 				m->attributes = g_list_insert_before(m->attributes, first, a);
 			} else {
 				a = janus_sdp_attribute_create("sctp-port", "5000");
 				m->attributes = g_list_insert_before(m->attributes, first, a);
 			}
-			a = janus_sdp_attribute_create("mid", "%s", handle->data_mid);
-			m->attributes = g_list_insert_before(m->attributes, first, a);
-#endif
-		}
-		if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
+		} else if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
 			a = janus_sdp_attribute_create("rtcp-mux", NULL);
 			m->attributes = g_list_insert_before(m->attributes, first, a);
 		}
 		/* ICE ufrag and pwd, DTLS fingerprint setup and connection a= */
 		gchar *ufrag = NULL;
 		gchar *password = NULL;
-		nice_agent_get_local_credentials(handle->agent, stream->stream_id, &ufrag, &password);
+		nice_agent_get_local_credentials(handle->agent, pc->stream_id, &ufrag, &password);
 		a = janus_sdp_attribute_create("ice-ufrag", "%s", ufrag);
 		m->attributes = g_list_insert_before(m->attributes, first, a);
 		a = janus_sdp_attribute_create("ice-pwd", "%s", password);
 		m->attributes = g_list_insert_before(m->attributes, first, a);
 		g_free(ufrag);
 		g_free(password);
-		a = janus_sdp_attribute_create("ice-options", "trickle");
-		m->attributes = g_list_insert_before(m->attributes, first, a);
 		if(janus_is_webrtc_encryption_enabled()) {
-			a = janus_sdp_attribute_create("fingerprint", "sha-256 %s", janus_dtls_get_local_fingerprint());
-			m->attributes = g_list_insert_before(m->attributes, first, a);
-			a = janus_sdp_attribute_create("setup", "%s", janus_get_dtls_srtp_role(offer ? JANUS_DTLS_ROLE_ACTPASS : stream->dtls_role));
+			a = janus_sdp_attribute_create("setup", "%s", janus_get_dtls_srtp_role(offer ? JANUS_DTLS_ROLE_ACTPASS : pc->dtls_role));
 			m->attributes = g_list_insert_before(m->attributes, first, a);
 		}
 		/* Add last attributes, rtcp and ssrc (msid) */
-		if(m->type == JANUS_SDP_VIDEO && janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX) &&
+		if(medium->ssrc_rtx > 0 && m->type == JANUS_SDP_VIDEO && janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX) &&
 				(m->direction == JANUS_SDP_DEFAULT || m->direction == JANUS_SDP_SENDRECV || m->direction == JANUS_SDP_SENDONLY)) {
 			/* Add FID group to negotiate the RFC4588 stuff */
-			a = janus_sdp_attribute_create("ssrc-group", "FID %"SCNu32" %"SCNu32, stream->video_ssrc, stream->video_ssrc_rtx);
+			a = janus_sdp_attribute_create("ssrc-group", "FID %"SCNu32" %"SCNu32, medium->ssrc, medium->ssrc_rtx);
 			m->attributes = g_list_append(m->attributes, a);
 		}
-		if(m->type == JANUS_SDP_AUDIO) {
-			a = janus_sdp_attribute_create("msid", "janus janusa0");
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("ssrc", "%"SCNu32" cname:janus", stream->audio_ssrc);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("ssrc", "%"SCNu32" msid:janus janusa0", stream->audio_ssrc);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("ssrc", "%"SCNu32" mslabel:janus", stream->audio_ssrc);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("ssrc", "%"SCNu32" label:janusa0", stream->audio_ssrc);
-			m->attributes = g_list_append(m->attributes, a);
-		} else if(m->type == JANUS_SDP_VIDEO) {
-			a = janus_sdp_attribute_create("msid", "janus janusv0");
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("ssrc", "%"SCNu32" cname:janus", stream->video_ssrc);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("ssrc", "%"SCNu32" msid:janus janusv0", stream->video_ssrc);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("ssrc", "%"SCNu32" mslabel:janus", stream->video_ssrc);
-			m->attributes = g_list_append(m->attributes, a);
-			a = janus_sdp_attribute_create("ssrc", "%"SCNu32" label:janusv0", stream->video_ssrc);
+		/* TODO Get rid of this ugly code */
+		if(m->type == JANUS_SDP_AUDIO || m->type == JANUS_SDP_VIDEO) {
+			a = janus_sdp_attribute_create("msid", "janus janus%s", medium->mid);
 			m->attributes = g_list_append(m->attributes, a);
-			if(janus_flags_is_set(&handle->webrtc_flags, JANUS_ICE_HANDLE_WEBRTC_RFC4588_RTX)) {
-				/* Add rtx SSRC group to negotiate the RFC4588 stuff */
-				a = janus_sdp_attribute_create("ssrc", "%"SCNu32" cname:janus", stream->video_ssrc_rtx);
-				m->attributes = g_list_append(m->attributes, a);
-				a = janus_sdp_attribute_create("ssrc", "%"SCNu32" msid:janus janusv0", stream->video_ssrc_rtx);
-				m->attributes = g_list_append(m->attributes, a);
-				a = janus_sdp_attribute_create("ssrc", "%"SCNu32" mslabel:janus", stream->video_ssrc_rtx);
-				m->attributes = g_list_append(m->attributes, a);
-				a = janus_sdp_attribute_create("ssrc", "%"SCNu32" label:janusv0", stream->video_ssrc_rtx);
+			if(medium->ssrc > 0) {
+				a = janus_sdp_attribute_create("ssrc", "%"SCNu32" cname:janus", medium->ssrc);
 				m->attributes = g_list_append(m->attributes, a);
+				if(m->type == JANUS_SDP_VIDEO && janus_flags_is_set(&handle->webrtc_flags, JANUS_HANDLE_WEBRTC_RFC4588_RTX)) {
+					/* Add rtx SSRC group to negotiate the RFC4588 stuff */
+					a = janus_sdp_attribute_create("ssrc", "%"SCNu32" cname:janus", medium->ssrc_rtx);
+					m->attributes = g_list_append(m->attributes, a);
+				}
 			}
 		}
 		/* FIXME If the peer is Firefox and is negotiating simulcasting, add the rid attributes */
-		if(m->type == JANUS_SDP_VIDEO && stream->rid[0] != NULL) {
+		if(m->type == JANUS_SDP_VIDEO && medium->rid[0] != NULL) {
 			char rids[50];
 			rids[0] = '\0';
 			int i=0;
 			for(i=0; i<3; i++) {
-				if(stream->rid[i] == NULL)
+				if(medium->rid[i] == NULL)
 					continue;
-				a = janus_sdp_attribute_create("rid", "%s recv", stream->rid[i]);
+				a = janus_sdp_attribute_create("rid", "%s recv", medium->rid[i]);
 				m->attributes = g_list_append(m->attributes, a);
 				if(strlen(rids) == 0) {
-					g_strlcat(rids, stream->rid[i], sizeof(rids));
+					g_strlcat(rids, medium->rid[i], sizeof(rids));
 				} else {
 					g_strlcat(rids, ";", sizeof(rids));
-					g_strlcat(rids, stream->rid[i], sizeof(rids));
+					g_strlcat(rids, medium->rid[i], sizeof(rids));
 				}
 			}
-			if(stream->legacy_rid) {
+			if(medium->legacy_rid) {
 				a = janus_sdp_attribute_create("simulcast", " recv rid=%s", rids);
 			} else {
 				a = janus_sdp_attribute_create("simulcast", " recv %s", rids);
@@ -1425,7 +1361,7 @@ char *janus_sdp_merge(void *ice_handle, janus_sdp *anon, gboolean offer) {
 		}
 		if(!janus_ice_is_full_trickle_enabled()) {
 			/* And now the candidates (but only if we're half-trickling) */
-			janus_ice_candidates_to_sdp(handle, m, stream->stream_id, 1);
+			janus_handle_candidates_to_sdp(handle, m, pc->stream_id, 1);
 			/* Since we're half-trickling, we need to notify the peer that these are all the
 			 * candidates we have for this media stream, via an end-of-candidates attribute:
 			 * https://tools.ietf.org/html/draft-ietf-mmusic-trickle-ice-02#section-4.1 */
diff --git a/sdp.h b/sdp.h
index cca5775936..a29fb2a2ef 100644
--- a/sdp.h
+++ b/sdp.h
@@ -33,53 +33,56 @@
 /** @name Janus SDP helper methods
  */
 ///@{
-/*! \brief Method to pre-parse a session description
- * \details This method is only used to quickly check how many audio and video lines are in an SDP, and to generate a Janus SDP instance
+/*! \brief Method to pre-parse a session description to generate a Janus SDP instance
  * @param[in] handle Opaque pointer to the ICE handle this session description will modify
  * @param[in] jsep_sdp The SDP that the browser peer originated
+ * @param[in,out] audio The number of audio m-lines in the SDP
+ * @param[in,out] video The number of video m-lines in the SDP
+ * @param[in,out] data The number of data m-lines in the SDP
  * @param[in,out] error_str Buffer to receive a reason for an error, if any
  * @param[in] errlen The length of the error buffer
- * @param[out] audio The number of audio m-lines
- * @param[out] video The number of video m-lines
- * @param[out] data The number of SCTP m-lines
  * @returns The Janus SDP object in case of success, NULL in case the SDP is invalid */
-janus_sdp *janus_sdp_preparse(void *handle, const char *jsep_sdp, char *error_str, size_t errlen, int *audio, int *video, int *data);
+janus_sdp *janus_sdp_preparse(void *handle, const char *jsep_sdp,
+	int *audio, int *video, int *data, char *error_str, size_t errlen);
 
-/*! \brief Method to process a parsed session description
+/*! \brief Method to process a remote parsed session description
  * \details This method will process a session description coming from a peer, and set up the ICE candidates accordingly
- * \note While this method can handle SDP updates, renegotiations are currently
- * limited to updates to the media direction of existing media streams
- * (e.g., sendrecv to recvonly) and ICE restarts. Adding/removing streams
- * and supporting multiple streams in the same PeerConnection are still WIP.
  * @param[in] handle Opaque pointer to the ICE handle this session description will modify
  * @param[in] sdp The Janus SDP object to process
  * @param[in] update Whether this SDP is an update to an existing session or not
  * @returns 0 in case of success, -1 in case of an error */
-int janus_sdp_process(void *handle, janus_sdp *sdp, gboolean update);
+int janus_sdp_process_remote(void *handle, janus_sdp *sdp, gboolean update);
+
+/*! \brief Method to process a local parsed session description
+ * \details This method will process a session description coming from a plugin, and set up media accordingly
+ * @param[in] handle Opaque pointer to the ICE handle this session description will modify
+ * @param[in] sdp The Janus SDP object to process
+ * @param[in] update Whether this SDP is an update to an existing session or not
+ * @returns 0 in case of success, -1 in case of an error */
+int janus_sdp_process_local(void *handle, janus_sdp *sdp, gboolean update);
 
 /*! \brief Method to parse a single candidate
  * \details This method will parse a single remote candidate provided by a peer, whether it is trickling or not
- * @param[in] stream Opaque pointer to the ICE stream this candidate refers to
+ * @param[in] webrtc Opaque pointer to the WebRTC PeerConnection this candidate refers to
  * @param[in] candidate The remote candidate to process
  * @param[in] trickle Whether this is a trickle candidate, or coming from the SDP
  * @returns 0 in case of success, a non-zero integer in case of an error */
-int janus_sdp_parse_candidate(void *stream, const char *candidate, int trickle);
+int janus_sdp_parse_candidate(void *webrtc, const char *candidate, int trickle);
 
 /*! \brief Method to parse a SSRC group attribute
  * \details This method will parse a SSRC group attribute, and set the parsed values for the peer
- * @param[in] stream Opaque pointer to the ICE stream this candidate refers to
+ * @param[in] medium Opaque pointer to the medium this candidate refers to
  * @param[in] group_attr The SSRC group attribute value to parse
- * @param[in] video Whether this is video-related or not
  * @returns 0 in case of success, a non-zero integer in case of an error */
-int janus_sdp_parse_ssrc_group(void *stream, const char *group_attr, int video);
+int janus_sdp_parse_ssrc_group(void *medium, const char *group_attr, int video);
 
 /*! \brief Method to parse a SSRC attribute
  * \details This method will parse a SSRC attribute, and set it for the peer
- * @param[in] stream Opaque pointer to the ICE stream this candidate refers to
+ * @param[in] medium Opaque pointer to the medium this candidate refers to
  * @param[in] ssrc_attr The SSRC attribute value to parse
  * @param[in] video Whether this is a video SSRC or not
  * @returns 0 in case of success, a non-zero integer in case of an error */
-int janus_sdp_parse_ssrc(void *stream, const char *ssrc_attr, int video);
+int janus_sdp_parse_ssrc(void *medium, const char *ssrc_attr, int video);
 
 /*! \brief Method to strip/anonymize a session description
  * @param[in,out] sdp The Janus SDP description object to strip/anonymize
diff --git a/transports/janus_nanomsg.c b/transports/janus_nanomsg.c
index cbad5eea40..00086eead7 100644
--- a/transports/janus_nanomsg.c
+++ b/transports/janus_nanomsg.c
@@ -311,6 +311,7 @@ int janus_nanomsg_init(janus_transport_callbacks *callback, const char *config_p
 	if(!nanomsg_thread) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Nanomsg thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 
diff --git a/transports/janus_pfunix.c b/transports/janus_pfunix.c
index d648a3aa42..71290582a1 100644
--- a/transports/janus_pfunix.c
+++ b/transports/janus_pfunix.c
@@ -32,7 +32,7 @@
 #include <sys/socket.h>
 #include <poll.h>
 #include <sys/un.h>
- 
+
 #ifdef  HAVE_LIBSYSTEMD
 #include "systemd/sd-daemon.h"
 #endif /* HAVE_LIBSYSTEMD */
@@ -341,6 +341,7 @@ int janus_pfunix_init(janus_transport_callbacks *callback, const char *config_pa
 	if(!pfunix_thread) {
 		g_atomic_int_set(&initialized, 0);
 		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the Unix Sockets thread...\n", error->code, error->message ? error->message : "??");
+		g_error_free(error);
 		return -1;
 	}
 
diff --git a/transports/janus_rabbitmq.c b/transports/janus_rabbitmq.c
index b7110e8d54..4cf4572c73 100644
--- a/transports/janus_rabbitmq.c
+++ b/transports/janus_rabbitmq.c
@@ -481,6 +481,7 @@ int janus_rabbitmq_init(janus_transport_callbacks *callback, const char *config_
 		if(error != NULL) {
 			/* Something went wrong... */
 			JANUS_LOG(LOG_FATAL, "Got error %d (%s) trying to launch the RabbitMQ incoming thread...\n", error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			janus_transport_session_destroy(rmq_session);
 			g_free(rmq_client);
 			janus_config_destroy(config);
@@ -490,6 +491,7 @@ int janus_rabbitmq_init(janus_transport_callbacks *callback, const char *config_
 		if(error != NULL) {
 			/* Something went wrong... */
 			JANUS_LOG(LOG_FATAL, "Got error %d (%s) trying to launch the RabbitMQ outgoing thread...\n", error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			janus_transport_session_destroy(rmq_session);
 			g_free(rmq_client);
 			janus_config_destroy(config);
diff --git a/transports/janus_websockets.c b/transports/janus_websockets.c
index 1d2147a821..33a82d23fc 100644
--- a/transports/janus_websockets.c
+++ b/transports/janus_websockets.c
@@ -786,9 +786,10 @@ int janus_websockets_init(janus_transport_callbacks *callback, const char *confi
 	/* Start the WebSocket service thread */
 	if(ws_janus_api_enabled || ws_admin_api_enabled) {
 		ws_thread = g_thread_try_new("ws thread", &janus_websockets_thread, wsc, &error);
-		if(!ws_thread) {
+		if(error != NULL) {
 			g_atomic_int_set(&initialized, 0);
 			JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the WebSockets thread...\n", error->code, error->message ? error->message : "??");
+			g_error_free(error);
 			return -1;
 		}
 	}
diff --git a/utils.h b/utils.h
index 4d52faf5db..53286d0567 100644
--- a/utils.h
+++ b/utils.h
@@ -19,6 +19,7 @@
 #define JANUS_JSON_STRING			JSON_STRING
 #define JANUS_JSON_INTEGER			JSON_INTEGER
 #define JANUS_JSON_OBJECT			JSON_OBJECT
+#define JANUS_JSON_ARRAY			JSON_ARRAY
 /* Use JANUS_JSON_BOOL instead of the non-existing JSON_BOOLEAN */
 #define JANUS_JSON_BOOL				JSON_TRUE
 #define JANUS_JSON_PARAM_REQUIRED	1
@@ -134,7 +135,7 @@ int janus_pidfile_create(const char *file);
 int janus_pidfile_remove(void);
 
 /*! \brief Creates a string describing the JSON type and constraint
- * @param jtype The JSON type, e.g., JSON_STRING
+ * @param jtype The JSON type, e.g., JANUS_JSON_STRING
  * @param flags Indicates constraints for the described type
  * @param[out] type_name The type description, e.g., "a positive integer"; required size is 19 characters
  * @returns 0 if successful, a negative integer otherwise */
@@ -142,7 +143,7 @@ void janus_get_json_type_name(int jtype, unsigned int flags, char *type_name);
 
 /*! \brief Checks whether the JSON value matches the type and constraint
  * @param val The JSON value to be checked
- * @param jtype The JSON type, e.g., JSON_STRING
+ * @param jtype The JSON type, e.g., JANUS_JSON_STRING
  * @param flags Indicates constraints for the described type
  * @returns TRUE if the value is valid */
 gboolean janus_json_is_valid(json_t *val, json_type jtype, unsigned int flags);
@@ -211,7 +212,7 @@ gboolean janus_json_is_valid(json_t *val, json_type jtype, unsigned int flags);
 	do { \
 		if (secret) { \
 			static struct janus_json_parameter secret_parameters[] = { \
-				{member, JSON_STRING, JANUS_JSON_PARAM_REQUIRED} \
+				{member, JANUS_JSON_STRING, JANUS_JSON_PARAM_REQUIRED} \
 			}; \
 			JANUS_VALIDATE_JSON_OBJECT(obj, secret_parameters, error_code, error_cause, TRUE, missing_code, invalid_code); \
 			if(error_code == 0 && !janus_strcmp_const_time((secret), json_string_value(json_object_get(obj, member)))) { \