kixelated

Yeah that's a really good way of framing the argument, I wish I wrote that. The way robots listen/respond is bounded by compute, not time. Buffering audio isn't a great experience for humans but definitely works for robots.

Close, but that's a minimum latency. We want a maximum latency knob.

To clarify, I meant waiting an extra 200ms if the alternative was dropping part of the prompt. During periods of zero congestion, the latency would be the same.

HELLO MR SEAN,

1. Of course users want lower latency, but they also want fewer instances where the LLM "misheard" them. It would be amazing to run A/B experiments on the trade-off between latency vs quality, but WebRTC makes that knob difficult to turn.

2. I'm obviously not an TTS expert, but what benefit is there to trickling out the result? The silicon doesn't care how quickly the time number increments?

3. Yeah, sometimes the client is aware when their IP changes and can do an ICE renegotiation. But often they aren't aware, and normally would rely on the server detecting the change, but that's not possible with your LB setup. It's not a big deal, just unfortunate given how many hoops you have to jump through already.

4. Okay, that draft means 7 RTTs instead of 8 RTTs? Again some can be pipelined so the real number is a bit lower. But like the real issue is the mandatory signaling server which causes a double TLS handshake just in case P2P is being used.

5. Of course WebRTC is easier for a new developer because it's a black box conferencing app. But for a large company like OpenAI, that black box starts to cause problems that really could be fixed with lower level primitives.

I absolutely think you should mess around with RTP over QUIC and would love to help. If you're worried about code size, the browser (and one day the OS) provides the QUIC library. And if you switch to something closer to MoQ, QUIC handles fragmentation, retransmissions, congestion control, etc. Your application ends up being surprisingly small.

The main shortcoming with RoQ/MoQ is that we can't implement GCC because QUIC is congestion controlled (including datagrams). We're stuck with cubic/BBR when sending from the browser for now.

Hello Mr Author here. Apologies that my comment replies aren't as funny.

Every low-latency application has to decide the user experience trade-off between quality and latency. Congestion causes queuing (aka latency) and to avoid that, something needs to be skipped (lower quality).

The WebRTC latency vs. quality knob is fixed. It's great at minimizing latency, but suffers from a lack of flexibility. We still (try to) use WebRTC anyway, because like you implied, browser support has made it one of the only options.

Until now of course! WebTransport means you can achieve WebRTC-like behavior via a generic protocol. Choose how long you want to wait before dropping/resetting a stream, instead of that decision being made for you.

And yeah my point in the blog is that often the user wants streaming, but not dropping. Obviously you can stream audio input/output without WebRTC. The application should be able to decide when audio packets are lost forever... is it 50ms or 500ms or 5000ms? My argument is that voice AI shouldn't pick the 50ms option.

QUIC libraries work by looping over pending streams (in priority order) to determine which UDP packet to send next. If there's more stream data than available congestion control, the data will send there in the stream send buffer.

Either side can abort a stream if it's taking too long, clearing the send buffer and officially dropping the data. It's a lot more flexible than opaque UDP send buffers and random packet loss.

FEC would make the most sense at the QUIC level because random packet loss is primarily hop-by-hop. But I'm not aware of any serious efforts to do that. There's a lot of ideas out there, but TBH MoQ is too young to have the production usage required to evaluate a FEC scheme.

Yep, it's similar to multicast but L7.

But a huge difference is that there's a plan for congestion. We heavily rely on QUIC to drain network queues and prioritize/queue media based on importance. It's doable with multicast+unicast, but complicated.

Yeah for Safari support I'm using polyfills; it sucks.

- libav.js for AudioEncoder/AudioDecoder. - QMux over WebSockets for WebTransport.

Both are NPM packages if you want to use them. @kixelated/libavjs-webcodecs-polyfill and @moq/qmux

26.4 removes the need for both so there's hope!

Absolutely agree.

You can convert any push-based protocol into a pull-based one with a custom protocol to toggle sources on/off. But it's a non-standard solution, and soon enough you have to control the entire stack.

The goal of MoQ is to split WebRTC into 3-4 standard layers for reusability. You can use QUIC for networking, moq-lite/moq-transport for pub/sub, hang/msf for media, etc. Or don't! The composability depends on your use case.

And yeah lemme know if you want some help/advice on your QUIC-based solution. Join the discord and DM @kixelated.

Hey lewq, 40Mbps is an absolutely ridiculous bitrate. For context, Twitch maxes out around 8.5Mb/s for 1440p60. Your encoder was poorly configured, that's it. Also, it sounds like your mostly static content would greatly benefit from VBR; you could get the bitrate down to 1Mb/s or something for screen sharing.

And yeah, the usual approach is to adapt your bitrate to network conditions, but it's also common to modify the frame rate. There's actually no requirement for a fixed frame rate with video codecs. It also you could do the same "encode on demand" approach with a codec like H.264, provided you're okay with it being low FPS on high RTT connections (poor Australians).

Overall, using keyframes only is a very bad idea. It's how the low quality animated GIFs used to work before they were secretly replaced with video files. Video codecs are extremely efficient because of delta encoding.

But I totally agree with ditching WebRTC. WebSockets + WebCodecs is fine provided you have a plan for bufferbloat (ex. adaptive bitrate, ABR, GoP skipping).