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by aidenn0 1477 days ago
Okay it's funny because I had vaguely remembered something like that, but a "C-f rtp" on the wikipedia page for WebRTC didn't yield any hits, even though SIP (over websockets) was prominently mentioned, so I just figured the similarity in RTC and RTP had caused me to misremember.

I assume WebRTC includes STUN/TURN/ICE (negotiated over SIP?) then for traversing NATs? The last time I was really into networking was 2001-ish so that stuff was still around the corner, but I kept up with my reading for a few years after that. I also had some of these acronyms refreshed when setting up Jingle, which uses XMPP instead of SIP, but establishes an RTP connection much like traditional VOIP would use.
1 comments
jhugo 1477 days ago
WebRTC doesn't proscribe a signalling mechanism at all. SIP is sometimes used, Jingle (XMPP) sometimes, or sometimes it's just a custom protocol exchanging SDPs (or equivalent structures) over WebSockets or a REST API.

WebRTC itself is RTP (DTLS-SRTP), ICE (incl. STUN/TURN), codecs & related parameters, capture mechanisms, all bundled up into a Web API.

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