[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:all-tabpanel ]

Jacek Konieczny updated ASTERISK-25894:
---------------------------------------

    Description: 
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.

  was:
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.


> 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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