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André Silva

You can't say that browser has a full SIP stack in it. It's false as SIP is only a signalling protocol and WebRTC is signalling agnostic. The only thing WebRTC provides to SIP is the SDP to send in INVITEs.

There are some projects and specifications to use Web sockets as a protocol to transport SIP, and some stack being migrated to Javascript. The application can use SIP, XMPP, or any other protocol over websockets or Http.


Hi André,

Thanks very much for this clarification! I got that important point wrong when I wrote the blog entry and have corrected the post accordingly.

Kind regards,

Aswath Rao

This is how I am using: RPi is running a webrtc app in a server. I am signed into it from a browser which maintains a websocket conn. I give an http URL to you. You visit the URL from a browser. RPi will authenticate you with OpenID/OAuth and if you are in my whitelist (it is a dynamic one), it will maintain a web socket conn with your browser and will inform me. If I respond we two can continue in our conversation.


I think G.722 is not part of the standard. G.711 and OPUS are.


Hi, according to the FAQ linked above "The currently supported voice codecs are G.711, G.722, iLBC, and iSAC".

Robert Syouta

Why incorporate the SIP stack into WebRTC in the first place? I may be missing a lot, however, it would seem to make more sense to not (duplicate) a SIP stack within WebRTC on the basis that it should be called as other applications stacks and might otherwise lead to conflicts in development.

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