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

[Xiemin Chen]

Hi Joshua, set media port to 0 will cause a stopped track in webrtc client
side and cannot be reused anymore, refer to
https://github.com/w3c/webrtc-pc/issues/1975.
Is there another way to mark the track stopped for asterisk, for example,
add "a=inactive"?


Joshua C. Colp <jcolp at digium.com> 于2019年5月27日周一 下午7:04写道：

> On Sun, May 26, 2019, at 9:44 PM, Xiemin Chen wrote:
> > Is there a way to completely remove the stream or not?
>
> A stream is never removed from the SDP itself, it is only communicated
> that it is removed and then it can be reused later. Having not had to
> remove a stream from SDP in the land of WebRTC, I'm not sure how it is done
> or expressed in the SDP. Asterisk itself supports the SDP RFC defined
> method of port 0.
>
> --
> 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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