WebRTC library capable of SIP signaling and P2P Video Conference

[apptes...@gmail.com]

My agenda is to have FreeSwitch as a signaling server for WebRTC client SIP.JS

I dint find any tutorial where i can use SIP.JS as a WebRTC library. It dont have option to create/join room.

Developing Server to Client calls works perfectly in FreeSwitch and SIP.JS. Please tell me how i can handle Peer to Peer Audio / Video Call and Conference using SIP.JS

[Eric Tamme]

You might want to checkout https://github.com/etamme/kwikykonf.  It is a peer to peer full mesh video conference built on SIP.js.

-Eric

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sip:eric....@onsip.com

[apptes...@gmail.com]

I have setup same thing, But i see that media is flowing through SIP server only (FreeSwitch) not Peer to Peer

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sip:eri...@onsip.com

[Eric Tamme]

Don't use a b2b UA (freeswitch), use a SIP proxy server.  WebRTC uses ICE and STUN to attempt to find the best media candidates.  If you have clients that are able to reach each other, ICE and STUN will negotiate peer to peer.  The demo link, https://video.uphreak.com/#github,  from the kwikykonf github page uses a SIP proxy server, and will route media peer to peer if the peers are able to reach each other.

-Eric

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sip:eric....@onsip.com

[Abdul Hakeem]

Did you try mod_verto and set the bypass_media option ?

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