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

Joshua C. Colp jcolp at digium.com
Wed May 8 12:41:35 CDT 2019

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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