Reapplying local SDP with different SSRC for video halts video packet transmission

[Dennis]

If I take my local SDP for a running session and then modify the SSRC values for audio and video and then re-apply the offer/answer, audio packets continue to transmit over the network using the new SSRC value but video packets halt altogether.  It can take a minute or more for video to resume using the new SSRC.  

For example, if my original SSRC values are '1' for video and '2' for audio, and I reapply the SDP using '3' for video and '4' for audio, the audio packets immediately transmit using the new  '4' SSRC value, but the video RTP packets completely halt ('1' packets halt) for a minute or so before resuming transmission with '3'.

Also, chrome://webrtc-internals/ "lies" in this situation too.  If you examine the 'bytesSent' and 'packetsSent' for '3' & '4' after the switch, both are increasing in value but a wireshark capture proves that only '4' (audio) is being transmitted on the wire.

Continuing to toggle this has the same results (e.g. eventually after '3' starts, if I toggle back to '1' and '2', the audio packets '2' immediately start but video packets on the network are gone for a few minutes before resuming at '1', again chrome://webrtc-internals/ "lies" showing that '1' is outputting on the network when it is actually not proven by wireshark.

Any known reason why this occurs or should I create a new issue in the webrtc bug issue tracker?

[Vikas]

Hi Dennis,

Thanks for the feedback, i can recreate it. But in my case, i never
get video but i do hear audio, also webrtc-internals tells that video
packets flowing on that ssrc. We will look into it further. Please
feel free to file an issue in the tracker.

/Vikas

[Dennis]

Ok, thanks!  created issue 1817 http://code.google.com/p/webrtc/issues/detail?id=1817

[Justin Uberti]

This sounds like it could be an encryption issue. Can you attach a Chrome log from the sending side into the bug?

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[Justin Uberti]

Never mind, I see that Vikas created a log.