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

Asterisk Team (JIRA) noreply at issues.asterisk.org
Fri Dec 13 09:36:31 CST 2019


    [ https://issues.asterisk.org/jira/browse/ASTERISK-28659?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=249053#comment-249053 ] 

Asterisk Team commented on ASTERISK-28659:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

> 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_session
>    Affects Versions: 16.7.0
>            Reporter: nappsoft
>              Labels: webrtc
>
> 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 



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list