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Improved congestion control for real-time use cases #676

@jan-ivar

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@jan-ivar

While WebTransport already offers tangible benefits over WebRTC like scaling, the model of using WebCodecs for encode/decode and WebTransport for secure delivery seems appealing for real-time conferencing as well, perhaps someday even replacing WebRTC.

Last year at TPAC 2024 we presented latency numbers around ~200 ms.

More recently, at today's WebRTC interim meeting proposing a third API for p2p real-time use cases, concerns were raised that ~200 ms isn't sufficient for real-time conferencing.

It seems most video conferencing today doesn't even rely on p2p, so if WebTransport's CC today is insufficient for real-time, this sounds like something we should consider improving.

I'm filing this issue to track this concern.

Note, WebTransport already supports the congestionControl option, so this may be more about tracking implementation progress toward that goal:

const wt = new WebTransport(url, {congestionControl: "low-latency"});

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