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

[Joshua C. Colp (JIRA)]

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

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

    Component/s: Resources/res_pjsip_sdp_rtp

> 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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