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

[Jacek Konieczny (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25894?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=230151#comment-230151 ] 

Jacek Konieczny commented on ASTERISK-25894:
--------------------------------------------

I found the problem. It seems ASTERISK-25451 has not been really fixed.

The problem is {{ast_frdup()}} does not copy whole {{frame.subclass}} for voice, video and image frames, only the format is copied. For video frames, the subclass structure contains the {{.frame_ending}} flag used to put the RTP marker where it needs to be.

I have prepared the simplest patch to fix the problem: https://gerrit.asterisk.org/#/c/2538/, though this could be fixed in a more generic way (properly copy whole {{.subclass}} for every frame type) or in a more specific way (handle {{.frame_ending}} only for video frames).

> 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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