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.