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 <[email protected]> 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 > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev
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