[asterisk-bugs] [JIRA] (ASTERISK-25894) webrtc video broken due to missing marker bits in RTP streams

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25894?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230149#comment-230149 ] 

Asterisk Team commented on ASTERISK-25894:
------------------------------------------

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].

> webrtc video broken due to missing marker bits in RTP streams
> -------------------------------------------------------------
>
>                 Key: ASTERISK-25894
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25894
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.8.0
>            Reporter: Jacek Konieczny
>
> WebRTC Video calls to app_echo work for me, but calls to other PJSIP endpoints do not – the browser is not able to decode the video. Video sent by the browser is properly decoded on the other side (linphone).
> I found out, that my Firefox cannot decode video due to cleared 'Marker' bit in incoming RTP packets (marker should be set on the last packet of each frame).
> This seem similar to ASTERISK-25451, but seems to affect WebRTC/SRTP/DTLS media only.
> I will try to investigate this further.



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