[asterisk-bugs] [JIRA] (ASTERISK-28659) Video conferences over WebRTC do not work anymore if the caller offers only audio

Fri Dec 13 09:52:32 CST 2019

     [ https://issues.asterisk.org/jira/browse/ASTERISK-28659?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua C. Colp updated ASTERISK-28659:
--------------------------------------

    Comment: was deleted

(was: INVITE: 

v=0
o=mozilla...THIS_IS_SDPARTA-70.0.1 1021735662023755928 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 78:86:2F:CC:D9:41:DD:37:42:F6:A5:A3:B0:9B:C6:46:34:19:33:C6:9F:79:31:E6:89:3E:46:D4:6C:AB:13:F8
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 33841 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 77.3.11.234
a=candidate:0 1 UDP 2122252543 192.168.240.220 33841 typ host
a=candidate:2 1 UDP 2122187007 10.100.100.40 49571 typ host
a=candidate:4 1 TCP 2105524479 192.168.240.220 9 typ host tcptype active
a=candidate:5 1 TCP 2105458943 10.100.100.40 9 typ host tcptype active
a=candidate:0 2 UDP 2122252542 192.168.240.220 57631 typ host
a=candidate:2 2 UDP 2122187006 10.100.100.40 55332 typ host
a=candidate:4 2 TCP 2105524478 192.168.240.220 9 typ host tcptype active
a=candidate:5 2 TCP 2105458942 10.100.100.40 9 typ host tcptype active
a=candidate:1 1 UDP 1686052863 77.3.11.234 33841 typ srflx raddr 192.168.240.220 rport 33841
a=candidate:1 2 UDP 1686052862 77.3.11.234 57631 typ srflx raddr 192.168.240.220 rport 57631
a=sendrecv
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:736287e6dc848669068680a6b968cf84
a=ice-ufrag:c08b863a
a=mid:0
a=msid:{2ed8a359-57e6-4bcf-a942-d73e8e6441de} {f58df1bc-86c4-404f-b72c-d3d6a14ce346}
a=rtcp:57631 IN IP4 77.3.11.234
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:752054920 cname:{18014708-86bc-416f-88ac-875cf79ff3c7}


200 OK (Asterisk): 

v=0
o=- 3578153112 2 IN IP4 55.66.77.88
s=Asterisk
c=IN IP4 55.66.77.88
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 12040 UDP/TLS/RTP/SAVPF 9 8 101
a=connection:new
a=setup:active
a=fingerprint:SHA-256 07:A1:0F:4D:71:E0:B3:F5:2A:F8:A9:6E:40:CB:28:2E:D2:A3:04:59:9E:12:94:52:C1:10:63:5A:4B:09:9C:D6
a=ice-ufrag:1aae1dd34521dd7811af5dc20a6b101e
a=ice-pwd:2f59a086767dd8db4fed7f323f8bbdb3
a=candidate:H5396014d 1 UDP 2130706431 55.66.77.88 12040 typ host
a=candidate:Ha0a0a01 1 UDP 2130706431 10.10.10.1 12040 typ host
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
a=rtcp-mux
a=ssrc:170137883 cname:0062f826-a060-46b4-be9d-67851d0465ab
a=msid:ca4bcf00-603a-4b61-a151-153d44d1330a f04e9ed5-b8a2-452b-93cc-ef4b8dd4dd33
a=rtcp-fb:* transport-cc
a=mid:0
)

> Video conferences over WebRTC do not work anymore if the caller offers only audio
> ---------------------------------------------------------------------------------
>
>                 Key: ASTERISK-28659
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28659
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_sdp_rtp, Resources/res_pjsip_session
>    Affects Versions: 16.7.0
>            Reporter: nappsoft
>            Assignee: Unassigned
>              Labels: webrtc
>         Attachments: sdp.txt
>
>
> Our WebRTC implementation for video conferences (based on sip.js) was working as expected with Asterisk 16.6.2. We offer 3 modes: audio only (as sender, of course the sender can receive video streams...), audio+video and audio+screen.
> With Asterisk 16.7.0-rc1 the implementation stopped working if the sender only offers audio but no video stream. The only thing that differs in the SDP sent by Asterisk is the following line:
> a=group:BUNDLE 0
> which changed to:
> a=group:BUNDLE 0 video-1
> I was able to track down the changed behavior to the following commit: https://github.com/asterisk/asterisk/commit/6f0a69c51a25fa63405b991d7c20baa229eef96a
> res_pjsip_session: initialize pending's topology to endpoint's 


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