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RTP is the fundamental media transport protocol in WebRTC (which is not a protocol, but rather a suite of protocols working together in a defined way). Basicall
by jhugo 4y ago
RTP is the fundamental media transport protocol in WebRTC (which is not a protocol, but rather a suite of protocols working together in a defined way). Basically all web videoconferencing uses RTP already.
- aidenn0 4y agoOkay 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.
- jhugo 4y agoWebRTC 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.