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- Hacker News
- There're tons of ways to fine-tune WebRTC that it wouldn't corrupt audio in poor network - it has all of the controls to smoothly trade-off latency vs quality. Not just NACKs - FEC, disable PLC/Acceleration/Deceleration, larger JB (tons of parameters) etc.
Most of the glitches I heard with OpenAI's Voice were not WebRTC related - but rather, to my ear, they sounded more like realtime issues with their inference - which is a very different component to optimize.
by yalok - > 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.
by fidotron - > But nope, WebRTC has no buffering and renders based on arrival time. Like seriously, timestamps are just suggestions. It’s even more annoying when video enters the picture.
I felt that comment my bones. Why would anyone possibly have the need to know actual presentation timestamp and how that corresponds to actual realtime? Evidently, no one working on WebRTC has had to synchronise data streams from varying sources before with millisecond accuracy.
I was doing a demo for a video stabilisation using a webcam and IMU module in the browser. It turns out the latency between video->rtc->browser and sensor->websocket->browser are wildly different and not constant. The obvious solution would be to send UTC timestamps for the sensors data and synchronise in browser. Not possible, the video has no UTC timestamp reference. When you have control of both sides of the WebRTC pipe, you can do fun things like send the UTC timestamp of the start of the stream, but this won’t solve browser jitter. It worked well enough for a POC but the entire solution had to be reengineered.
by fps-hero - 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.
by r2vcap - 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".by awkii
- I didn't make it all the way through the post, but I have to say I think he fundamentally understands the purpose of WebRTC. He calls himself an expert, and yeah he's written SFU's in go and rust and different companies ... but his technical credentials do not mean he's correct.
Maybe it's a comprehension issue on my end, but he seems to associate things like stun and dtls as related, compounding issues (particularly in round trip time), but they are really orthogonal.
Also, he spends too much time talking about how you can't resend packets, and reiterates that point by stating they tried really hard (at discord?). That's where he lost the plot, imo.
The RTC in WebRTC is about real time communication. Humans will naturally prefer the auditory experience of an occasional dropped packet, vs backed up audio or audio that plays at an uneven rate. To clarify, I'm talking about human speech here.
If you want to tolerate packet loss, use a protocol based on tcp instead of udp. But you know what happens when you send audio over poor network conditions with tcp? There will be pauses on the receiving end as it waits for the next correct packet. Let's say the delay is multiple seconds. What should the receiving end do when packets start flowing again? Plays the clogged audio at a natural clock? Attempt to play the audio back at a higher rate to "catch up" with any other channels? People, humans, do not generally prefer that experience.
Forget about WebRTC for a minute, but instead think about tcp vs udp for voice. Voip has been based on udp since the 90's for a reason.
by thutch76 - I have a lot of experience in this area (and some patent applications). For Alexa, the device established a connection back to the server and then kept that open, sending basically HTTP2/SPDY/Something like it over the wire after it detected the wake word. This allowed the STT start processing before you finish talking, so there is only a small delay in processing the last few chunks of your utterance.
The answer came back over the same connection.
In the case of OpenAI, they can't exactly keep a persistent connection open like Alexa does, but they can use HTTP2 from the phone and both iOS and Android will pretty much take care of that connection magically.
The author is absolutely right, a real time protocol isn't necessary. It's more important to get all the data. The user won't even notice a delay until you get over 500ms. Especially in the age of mobile phones, where most people are used to their real time human to human communications to have a delay.
(If you work at OpenAI or Anthropic, give me a shout, I'm happy to get into more details with you)
by jedberg - 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.
by Sean-Der