zrtp_patch_1.10.patch

diff --git a/pjsip/include/pjsua-lib/pjsua.h b/pjsip/include/pjsua-lib/pjsua.h
index 0a6cbdd..394c893 100644
--- a/pjsip/include/pjsua-lib/pjsua.h
+++ b/pjsip/include/pjsua-lib/pjsua.h
@@ -420,6 +420,18 @@ typedef struct pjsua_reg_info
 } pjsua_reg_info;
 
 
+typedef enum pjsua_create_media_transport_flag
+{
+   /**
+    * This flag indicates that the media transport must also close its
+    * "member" or "child" transport when pjmedia_transport_close() is
+    * called. If this flag is not specified, then the media transport
+    * must not call pjmedia_transport_close() of its member transport.
+    */
+   PJSUA_MED_TP_CLOSE_MEMBER = 1
+
+} pjsua_create_media_transport_flag;
+
 /**
  * This structure describes application callback to receive various event
  * notification from PJSUA-API. All of these callbacks are OPTIONAL, 
@@ -918,6 +930,31 @@ typedef struct pjsua_callback
     void (*on_ice_transport_error)(int index, pj_ice_strans_op op,
 				   pj_status_t status, void *param);
 
+    /**
+     * This callback can be used by application to implement custom media
+     * transport adapter for the call, or to replace the media transport
+     * with something completely new altogether.
+     *
+     * This callback is called when a new call is created. The library has
+     * created a media transport for the call, and it is provided as the
+     * \a base_tp argument of this callback. Upon returning, the callback
+     * must return an instance of media transport to be used by the call.
+     *
+     * @param call_id       Call ID
+     * @param media_idx     The media index in the SDP for which this media
+     *                      transport will be used.
+     * @param base_tp       The media transport which otherwise will be
+     *                      used by the call has this callback not been
+     *                      implemented.
+     * @param flags         Bitmask from pjsua_create_media_transport_flag.
+     *
+     * @return              The callback must return an instance of media
+     *                      transport to be used by the call.
+     */
+    pjmedia_transport* (*on_create_media_transport)(pjsua_call_id call_id,
+                                                 unsigned media_idx,
+                                                 pjmedia_transport *base_tp,
+                                                 unsigned flags);
 } pjsua_callback;
 
 
diff --git a/pjsip/src/pjsua-lib/pjsua_media.c b/pjsip/src/pjsua-lib/pjsua_media.c
index 47cf1e7..57f9f27 100644
--- a/pjsip/src/pjsua-lib/pjsua_media.c
+++ b/pjsip/src/pjsua-lib/pjsua_media.c
@@ -1174,7 +1174,8 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
 {
     pjsua_call *call = &pjsua_var.calls[call_id];
     pj_status_t status;
-
+    pjmedia_transport *app_tp = NULL;
+    
 #if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
     pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
     pjmedia_srtp_setting srtp_opt;
@@ -1232,7 +1233,8 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
 	call->med_tp = srtp;
     }
 #else
-    call->med_orig = call->med_tp;
+    if (!call->media_orig)
+        call->med_orig = call->med_tp;
     PJ_UNUSED_ARG(security_level);
 #endif
 
@@ -1258,6 +1260,48 @@ pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
 	call->audio_idx = find_audio_index(rem_sdp, srtp_active);
     }
 
+    /*
+     * Check if applicaton wnats a callback on media transport creation.
+     * 
+     * Similar to check to create SRTP above:
+     * Call the application's callback only on first channel_init, not on
+     * a re-invite that does not change the channel setup.
+     * 
+     * If SRTP was compiled pjsua always creates the SRTP transport. In this
+     * case check if the current transport is SRTP transport. If yes then
+     * call the callback.
+     * 
+     * If SRTP was not compiled then check if med_orig and med_tp are the same.
+     * If yes call the callback.
+     */
+    if (pjsua_var.ua_cfg.cb.on_create_media_transport)
+    {
+        pjsua_create_media_transport_flag flag = 0;
+
+#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+        if (call->med_tp == srtp) 
+        {
+
+            /* If the current transport is SRTP then the application's 
+             * transport shall close SRTP (SRTP itself will not close 
+             * it's slave transport, see above).
+             */
+            flag = PJSUA_MED_TP_CLOSE_MEMBER;
+#else
+        if (call->med_tp == call->med_orig)
+        {
+#endif
+            app_tp = pjsua_var.ua_cfg.cb.on_create_media_transport(call_id,
+                                                               call->audio_idx,
+                                                               call->med_tp,
+                                                               flag);
+            if (app_tp != NULL && app_tp != call->med_tp)
+            {
+                call->med_tp = app_tp;
+            }
+        }
+    }
+
     /* Reject offer if we couldn't find a good m=audio line in offer */
     if (call->audio_idx < 0) {
 	if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
-- 
1.7.3.4