Probably because WebTransport is the lesser known alternative to WebRTC.

This is frustratingly one-sided writing. Yeah, WebRTC has limitations, but relying on a standard buys you a lot of correctness and reduces long-term engineering cost. The fact that WebRTC is complicated does not mean it is wrong; it means real-time media over the public internet is complicated.

Also, networking is inherently stateful. NAT traversal, jitter buffers, congestion control, packet loss, codec state, encryption, and session routing do not disappear because you put audio over TCP or WebSocket. Pretending otherwise is not architectural clarity. It is just moving the complexity somewhere less visible.

  Like 6 years ago I wrote a WebRTC SFU at Twitch.
  Originally we used Pion (Go) just like OpenAI,
  but forked after benchmarking revealed that it was too slow.
  I ended up rewriting every protocol, because of course I did!

  Just a year ago, I was at Discord and I rewrote the WebRTC SFU in Rust.
  Because of course I did! You’re probably noticing a trend.

  Fun Fact: WebRTC consists of ~45 RFCs dating back to the early 2000s.
  And some de-facto standards that are technically drafts (ex. TWCC, REMB).
  Not a fun fact when you have to implement them all.

  You should consider me a Certified WebRTC Expert.
  Which is why I never, never want to use WebRTC again.

This poor soul. There are few protocols I hate implementing more than WebRTC. Getting a simple client going means you need to quickly acclimate to SDP, TURN/STUN, ice-candidates, offers, peer-to-peer protocols, and the complex handshake that is implemented from scratch each time. I can't imagine re-writing the whole trenchcoat of protocols and unintended "best-practices".

Yet another victim of IPv4, and you still find countless detractors of IPv6 on every thread where it's mentioned.

> WebRTC is designed to degrade and drop my prompt during poor network conditions

You want real time that's what you are going to deal with. If you don't want real time and instead imagine everything as STT -> Prompt -> TTS then maybe you shouldn't even be sending audio on the wire at all.

I haven't really experienced disconnections while using ChatGPT. Gemini is the frustrating part. Simply backgrounding the app (and the web version too) and resuming it causes the response or the conversation with an assigned ID to disappear. Haha.

this misses a few key things but hits on many others

webrtc is a bad protocol, without a doubt. I do like websockets as an easy alternative, but you do need to reinvent decent portions of webrtc as a result

I like the idea of MoQ but it's not widely used. probably worth experimenting with, especially as video enters the chat

> and then a GPU pretends to talk to you via text-to-speech

OpenAI is speech-to-speech, there is no TTS in voice mode

> It takes a minimum of 8* round trips (RTT) to establish a WebRTC connection

signalling can be done long ahead of time, though I don't see this mentioned in the OpenAI blog. I also saw some new webrtc extensions that should reduce setup time further

ultimately though, it comes down to

> It’s not like LLMs are particularly responsive anyway

I expect to see a shift in how S2S models work to be lower latency like the new voice API models that OpenAI announced

to be fair, the new models were released the day after this MoQ blog was published

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.