[asterisk-users] WebRTC signaling

[TTT]

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

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