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

[Xiemin Chen]

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.
Can you point out where's the actual SDP negotiation? I will check it too.
Thanks for your help.

Joshua C. Colp <jcolp at digium.com> 于2019年5月9日周四 上午12:40写道：

> On Wed, May 8, 2019, at 1:26 PM, Xiemin Chen wrote:
> > Please see the comments in the following codes, at the moment, A & B
> > both have one audio & video stream separately and B is starting to add
> > an extra sharing video:
>
> <snip>
>
> >
> > /* The stream count of B's active_media_state->topology is 3, one B's
> > audio, one B's video, one A's video with name softbridge_dest_PJSIP...
> >  The sdp->media_count is 3, one for audio, one for camera video, one
> > for sharing video
> >  Here B's sharing video is map to A's video with name
> > softbridge_dest_PJSIP... */
>
> So are you adding a stream or replacing a stream? If adding I would expect
> 4 streams in the SDP media count: Audio, camera video, A's video, and the
> new sharing video. Are you actually instead replacing/reusing an existing
> stream?
>
> <snip>
>
> >
> > /* If the code of B's new sharing video stream executes here, does it
> > need to call set_mid_and_bundle_group() &
> > set_remote_mslabel_and_stream_group()
> >  & handler->negotiate_incoming_sdp_stream for the new stream setup
> > instead of the handler->defer_incoming_sdp_stream()?
> >  */
>
> The purpose of this function is for determining if the SDP handling should
> be deferred until a later time, it's not for actually performing the SDP
> negotiation. That is done elsewhere. I don't believe this function should
> have any of the above done in it.
>
> --
> 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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