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

[Xiemin Chen]

A cannot see B's new video track,
A is not trigged a renegotiation progress after B's new video track was
added.

Joshua C. Colp <jcolp at digium.com> 于2019年5月10日周五 上午12:22写道：

> On Thu, May 9, 2019, at 1:15 PM, Xiemin Chen wrote:
> > 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?
>
> I'm not exactly clear on if A can see the new stream or not. Can you
> clarify?
>
> As well no, that field and type are used if an existing stream has
> changed. It's not used if a new stream is added.
>
> --
> 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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