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We use a WS or TCP connection for the signalling processs to initiate or terminate a session (sip messages).
Once the session is connected WebRTC transmits the actual media (audio/video) over UDP.
Why would you want to use UDP for the signalling step? (Genuine question)
Also do you realise what UDP is and how much is needed to set it up, packet ordering, error checking, reliability timeouts, flow control, security etc etc
Can you add a new feature for TransportType? I need Transport type UDP.
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