Responding to some technical points first, but then after that I do see a future that isn't WebRTC. I don't think it matches where WebTransport+WebCodecs etc is going though.

> …but as a user, I would much rather wait an extra 200ms for my slow/expensive prompt to be accurate

This is the opposite of the feedback I get. Users want instant responses. If you have delay in generating responses/interruptions it kills the magic. You also don't want to send faster than real-time. If the user interrupts the model you just wasted a bunch of bandwidth sending 3 minutes of audio (but only played 10 seconds)

> TTS is faster than real-time

https://research.nvidia.com/labs/adlr/personaplex/ Voice AI for the latest/aspirational is moving away from what the author describes. It is trickled in/out at 20ms

> We really hope the user’s source IP/port never changes, because we broke that functionality.

That is supported. When new IP for ufrag comes in its supported

> It takes a minimum of 8* round trips (RTT)

That's wrong. https://datatracker.ietf.org/doc/draft-hancke-webrtc-sped/

> I’d just stream audio over WebSockets

You lose stuff like AEC. You also push complexity on clients. The simplicity of WebRTC (createOffer -> setRemoteDescription) is what lets people onboard easily. Lots of developers struggled with Realtime API + web sockets (lots of code and having to do stuff by hand)

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I think if I had my choice I would pick Offer/Answer model and then doing QUIC instead of DTLS+SCTP. Maybe do RTP over QUIC? I personally don't feel strongly about the protocol itself. I don't know how to ship code to multiple clients (and customers clients) with a much large code footprint.

OpenAI’s WebRTC problem