[asterisk-dev] WebRTC SFU: support add video track dynamically

[Xiemin Chen]

I use webrtc's AddTransceiver() interface to create a standalone video
track, now the stream count is correct and the server's answer sdp is right
too.

However after B's new track is negotiated, A can see B's new video track. I
check the function handle_negotiated_sdp() and found that
the session_media->changed is never set so
the AST_CONTROL_STREAM_TOPOLOGY_SOURCE_CHANGED cannot be triggered. Is that
matter?

Sincerely
Xiemin




Joshua C. Colp <jcolp at digium.com> 于2019年5月9日周四 上午1:42写道：

> On Wed, May 8, 2019, at 2:11 PM, Xiemin Chen wrote:
> > I add a stream instead of replace stream, I think the problem is caused
> > by the a=sendrecv created by WebRTC client, I will get deeper in to
> > this.
>
> If it's not adding a stream but replacing like it seems to be, then I
> could certainly see things not working as bridge_softmix also does not
> expect to use a stream for both sending and receiving.
>
> > Can you point out where's the actual SDP negotiation? I will check it
> > too. Thanks for your help.
>
> Within Asterisk the handle_incoming_sdp function is what does the
> negotiation and handle_negotiated_sdp_session_media applies the result of
> that negotiation. The SDP negotiator itself otherwise is within PJSIP
> itself, specifically pjmedia/src/pjmedia/sdp_neg.c
>
> --
> Joshua C. Colp
> Digium - A Sangoma Company | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> Check us out at: www.digium.com & www.asterisk.org
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