fix caller and callee both simulcast or just one side simucast fail and fix the substream status show error on the webpage #2671

lucylu-star · 2021-05-21T14:00:05Z

fix caller and callee both simulcast fail.
support just callee simulcast ok.
fix the substream status show error on the webpage.

… fail

update

… fix the substream status show error on the webpage

lminiero · 2021-05-21T14:06:26Z

Simulcast cannot be enabled by the receiving party, as far as I know, because the simulcast envelope is created when the transceiver is created. Since the transceiver is created by the PeerConnection and not by the browser, in case of a callee, simulcast cannot be added there.

lucylu-star · 2021-05-24T03:15:27Z

As you said, callee cannot simulcast. I revert some.
But, now have the bug that is caller simulcast failed and substream level status shows error when videocall.
I think the new pull request fix the issue.

lucylu-star · 2021-05-24T03:18:07Z

Only modify janus_videocall.c

lminiero

Added some new notes inline. Thanks!

lminiero · 2021-05-24T09:11:35Z

plugins/janus_videocall.c

@@ -786,7 +786,7 @@ void janus_videocall_incoming_rtp(janus_plugin_session *handle, janus_plugin_rtp
 				json_t *result = json_object();
 				json_object_set_new(result, "event", json_string("simulcast"));
 				json_object_set_new(result, "videocodec", json_string(janus_videocodec_name(session->vcodec)));
-				json_object_set_new(result, "substream", json_integer(session->sim_context.substream));
+				json_object_set_new(result, "substream", json_integer(peer->sim_context.substream));


This is wrong. The substream returned here is the substream the user (so session) wants from the peer. Substream and temporal info are always receiver related.

disagree.
I think peer->sim_context.substream is right. you can check it seeing the substream status on videocall webpage. I have verified it. Using peer, the showing is ok. if using session, the status showing is abnormal.
Peer is the callee. The substream returned here is what substream the callee wants. It is showed on webpage.

Ah you're rght, sorry: this is an event we're sending to peer, not session as I thought 👍 The session->vcodec above threw me off.

lminiero · 2021-05-24T09:11:43Z

plugins/janus_videocall.c

@@ -798,7 +798,7 @@ void janus_videocall_incoming_rtp(janus_plugin_session *handle, janus_plugin_rtp
 				json_t *result = json_object();
 				json_object_set_new(result, "event", json_string("simulcast"));
 				json_object_set_new(result, "videocodec", json_string(janus_videocodec_name(session->vcodec)));
-				json_object_set_new(result, "temporal", json_integer(session->sim_context.templayer));
+				json_object_set_new(result, "temporal", json_integer(peer->sim_context.templayer));


lminiero · 2021-05-24T09:13:24Z

plugins/janus_videocall.c

-				session->ssrc[2] = json_integer_value(json_object_get(msg_simulcast, "ssrc-2"));
+		        int rid_ext_id = -1;
+				janus_rtp_simulcasting_prepare(msg_simulcast, &rid_ext_id, session->ssrc, session->rid);
+				session->sim_context.rid_ext_id = rid_ext_id;


As I pointed out, callees cannot do simulcast. It's likely the whole code above (the one already in the repo) is redundant and should be removed.

yes, I will remove it.

As soon as you remove tis part here, it's good to merge for me 👍

Note: I can get rid of the whole existing (and unneeded) simulcast check for callees myself later.

lminiero · 2021-05-24T09:14:32Z

plugins/janus_videocall.c

-						peer->ssrc[i] = 0;
-						g_free(peer->rid[0]);
-						peer->rid[0] = NULL;
-					}


This is the only fix that I think does belong here: in fact, it was incorrectly disabling the previously configured simulcast settings for the peer (the caller in this instance).

lucylu-star · 2021-05-25T02:50:18Z

just remove several unneeded lines. Why auto check failed?

lminiero · 2021-05-25T06:50:36Z

It looks like an issue pulling libnice from gitlab, so not caused by your patch. Hopefully it was a temporary error, I'll try to restart the automated build later.

lminiero · 2021-05-25T09:09:10Z

Thanks, merging! 👍

commit 9c9d335 Author: Lionel Nicolas <lionelnicolas@users.noreply.github.com> Date: Thu Jun 10 03:04:43 2021 -0400 Fix streaming plugin mutex unlock when disabling mountpoint (meetecho#2690) commit 2d83e96 Author: Yurii Cherniavskyi <yurii.cherniavskyi@gmail.com> Date: Mon Jun 7 16:02:41 2021 +0300 Fix SIP plugin unhold request docs typo (meetecho#2688) commit 2cd0118 Author: August Black <augustblack@gmail.com> Date: Mon Jun 7 01:10:49 2021 -0700 minor adjustment to the audiobridge docs (meetecho#2687) commit de2117e Author: nicolasduteil <nduteil@freedev.org> Date: Tue Jun 1 11:26:29 2021 +0200 fix: [janus_sip] Fix "call_id" property in "missed_call" events (meetecho#2679) commit 9eeeb38 Author: Alessandro Toppi <atoppi@meetecho.com> Date: Mon May 31 15:57:41 2021 +0200 Fix status vector parsing for incoming twcc feedbacks (resolves meetecho#2677). commit 8a25f6e Merge: d3b39b9 394fb48 Author: Alessandro Toppi <atoppi@meetecho.com> Date: Fri May 28 13:29:54 2021 +0200 Merge pull request meetecho#2675 from kmeyerhofer/actions/fix GH Actions, fix variable name commit d3b39b9 Author: Lorenzo Miniero <lminiero@gmail.com> Date: Fri May 28 11:09:30 2021 +0200 Fixed race condition in VideoRoom commit 394fb48 Author: Kurt Meyerhofer <k@kcmr.io> Date: Thu May 27 14:52:08 2021 -0600 Fixes variable name. commit b45cd37 Author: Lorenzo Miniero <lminiero@gmail.com> Date: Thu May 27 18:31:55 2021 +0200 Clarify that libnice 0.1.18 is recommended commit 5757a37 Author: Lorenzo Miniero <lminiero@gmail.com> Date: Thu May 27 17:08:17 2021 +0200 Spatial audio support in AudioBridge via stereo mixing (meetecho#2446) commit 161fe7a Author: Luca Barbato <luca.barbato@gmail.com> Date: Thu May 27 15:29:01 2021 +0200 Cleanup avformat-based preprocessors (meetecho#2665) commit 7b010cd Author: lucylu-star <78361868+lucylu-star@users.noreply.github.com> Date: Tue May 25 17:09:40 2021 +0800 Fixed broken simulcast support in VideoCall plugin (meetecho#2671) commit 4ae44a4 Author: nicolasduteil <nduteil@freedev.org> Date: Mon May 24 17:57:34 2021 +0200 feat: support for custom call-id in subscribe request + add 'call_id' property to subscribe & notify related events (meetecho#2664) commit 4294f20 Author: Lorenzo Miniero <lminiero@gmail.com> Date: Mon May 24 11:02:48 2021 +0200 Fixed missing macro when using pthread mutexes (fixes meetecho#2666) commit f22ab0d Author: Lorenzo Miniero <lminiero@gmail.com> Date: Wed May 19 12:03:32 2021 +0200 Fixed warning commit b3f3f17 Author: Alessandro Toppi <atoppi@meetecho.com> Date: Tue May 18 12:10:47 2021 +0200 Remove duplicated flag for fuzzing coverage. commit 4a7560c Author: nu774 <honeycomb77@gmail.com> Date: Fri May 14 00:26:36 2021 +0900 janus-pp-rec: support HEVC AP(aggregation packet) (meetecho#2662) commit 5db4be2 Author: Lorenzo Miniero <lminiero@gmail.com> Date: Wed May 12 17:43:43 2021 +0200 Fixed out of bounds array access commit 69f56f4 Author: nicolasduteil <nduteil@freedev.org> Date: Tue May 11 14:36:22 2021 +0200 feat: support for SUBSCRIBE expiry (Expires header) in sip plugin (meetecho#2661) commit b047ccf Author: Lorenzo Miniero <lminiero@gmail.com> Date: Mon May 10 09:33:27 2021 +0200 Fixed types commit f8e8c5e Author: Chris Wiggins <chris@wiggins.nz> Date: Mon May 10 19:26:45 2021 +1200 RabbitMQ Transport Reconnect Logic (meetecho#2651) commit 280e8e4 Author: Lorenzo Miniero <lminiero@gmail.com> Date: Fri May 7 12:54:30 2021 +0200 Add per-participant recording options in AudioBridge to join API as well

lucylu-star and others added 15 commits April 8, 2021 21:29

test

c690ea9

fix deadlock when rtsp reconnect fail

6517f20

fix deadlock when destroying the mountpoint and rtsp reconnect always…

9064c25

… fail

replace space with tab

1045081

adjust since typo

d9303cb

del newline and brackets

65200da

remove the first check,only keep the second

8df2253

merge

ef8075b

just fix code

b2296a9

revert some removed by mistake

b30a52d

1

a2b331e

Merge branch 'meetecho-master'

a80b227

Merge pull request #2 from meetecho/master

b561737

update

Merge branch 'meetecho:master' into master

02bf013

fix caller and callee both simulcast or just one side simucast fail &…

8d56ba7

… fix the substream status show error on the webpage

revert

d652d37

lminiero reviewed May 24, 2021

View reviewed changes

lucylu-star and others added 3 commits May 24, 2021 18:58

Merge branch 'meetecho:master' into master

63f8d2f

Merge branch 'meetecho:master' into master

2f36ecf

remove unneed

3670132

lminiero merged commit 7b010cd into meetecho:master May 25, 2021

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fix caller and callee both simulcast or just one side simucast fail and fix the substream status show error on the webpage #2671

fix caller and callee both simulcast or just one side simucast fail and fix the substream status show error on the webpage #2671

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lminiero commented May 25, 2021

fix caller and callee both simulcast or just one side simucast fail and fix the substream status show error on the webpage #2671

fix caller and callee both simulcast or just one side simucast fail and fix the substream status show error on the webpage #2671

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lucylu-star commented May 21, 2021

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lucylu-star commented May 24, 2021

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lucylu-star commented May 25, 2021

lminiero commented May 25, 2021

lminiero commented May 25, 2021