WebRTC Signaling

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I’m looking at using Asterisk 20 with WebRTC clients (sipjs). I know the media runs over TCP, but what about the signaling?

I read something about signaling over UDP was proposed as part of a webrtc standard, but can’t find if that was ever ratified or if Asterisk can even use UDP for the signaling instead of TCP for the signaling.

Does encryption of the signaling (SIPS) change anything?

Thanks

Brian

One thought on - WebRTC Signaling

  • Media doesn’t generally go over TCP, it goes over UDP.

    WebRTC doesn’t define signaling. SIP is an option, and the browser provides websockets for its transport. It’s all in what the browser supports.