4 ms·

How does WebRTC go together with HTTP/3 and QUIC, or is it unrelated at all?

Sean-Der 6y ago
Unrelated! (for now) WebRTC uses decades old protocols (RTP/RTCP) for media exchange and pre-dates QUIC. There is talk of using QUIC instead, but I have no idea how that is going!
lxgr 6y ago
Broadly speaking, WebRTC addresses P2P and does things like NAT traversal; HTTP/3 is "classic" server/client communications, with the server needing a publicly reachable hostname. That being said, nothing specifically prevents a server from being one of the peers in WebRTC, but the overhead can be significant, so it's not commonly used like that. I'm not aware of any way of doing something like WebSockets with datagram semantics (i.e. over UDP), but it seems like this is something that might be addressed once WebSockets can be used over HTTP/3.
simonpure 6y ago
There's the Web Transport proposal. It provides client/server streams similar to what you can do with data channels (ordered, unordered, multiple streams) and it's using QUIC - https://wicg.github.io/web-transport/ https://wicg.github.io/web-transport/
pthatcherg 6y ago
Hey, I used to work on this :). It's complicated :). Basically, WebRTC is a combination of a bunch of protocols: ICE, DTLS, SCTP, and RTP. You could theoretically reduce that to ICE + QUIC for p2p use cases and just QUIC for client/server use cases. For p2p, there is RTCQuicTransport (see https://developers.google.com/web/updates/2019/01/rtcquictransport-api https://developers.google.com/web/updates/2019/01/rtcquictra...) For client/server, there is QuicTranport (see https://web.dev/quictransport/ https://web.dev/quictransport/) Of course, you'll probably want to be able to encode and decode some audio and video as well to make that useful. For that, there is WebCodecs (see https://github.com/WICG/web-codecs https://github.com/WICG/web-codecs or https://www.chromestatus.com/feature/5669293909868544 https://www.chromestatus.com/feature/5669293909868544) For use cases like live streaming and cloud gaming, I did a presentation about the combination of WebTransport + WebCodecs: https://www.w3.org/2018/12/games-workshop/slides/21-webtransport-webcodecs.pdf https://www.w3.org/2018/12/games-workshop/slides/21-webtrans... And then there is the work happening in the IETF along these same lines: https://datatracker.ietf.org/wg/ript/documents/ https://datatracker.ietf.org/wg/ript/documents/
- TomMarius 6y ago
  Thank you and others for answering me! :-)

Unrelated! (for now) WebRTC uses decades old protocols (RTP/RTCP) for media exchange and pre-dates QUIC. There is talk of using QUIC instead, but I have no idea how that is going!

lxgr 6y ago

Broadly speaking, WebRTC addresses P2P and does things like NAT traversal; HTTP/3 is "classic" server/client communications, with the server needing a publicly reachable hostname. That being said, nothing specifically prevents a server from being one of the peers in WebRTC, but the overhead can be significant, so it's not commonly used like that. I'm not aware of any way of doing something like WebSockets with datagram semantics (i.e. over UDP), but it seems like this is something that might be addressed once WebSockets can be used over HTTP/3.

simonpure 6y ago

There's the Web Transport proposal. It provides client/server streams similar to what you can do with data channels (ordered, unordered, multiple streams) and it's using QUIC - https://wicg.github.io/web-transport/ https://wicg.github.io/web-transport/

pthatcherg 6y ago

Hey, I used to work on this :). It's complicated :). Basically, WebRTC is a combination of a bunch of protocols: ICE, DTLS, SCTP, and RTP. You could theoretically reduce that to ICE + QUIC for p2p use cases and just QUIC for client/server use cases. For p2p, there is RTCQuicTransport (see https://developers.google.com/web/updates/2019/01/rtcquictransport-api https://developers.google.com/web/updates/2019/01/rtcquictra...) For client/server, there is QuicTranport (see https://web.dev/quictransport/ https://web.dev/quictransport/) Of course, you'll probably want to be able to encode and decode some audio and video as well to make that useful. For that, there is WebCodecs (see https://github.com/WICG/web-codecs https://github.com/WICG/web-codecs or https://www.chromestatus.com/feature/5669293909868544 https://www.chromestatus.com/feature/5669293909868544) For use cases like live streaming and cloud gaming, I did a presentation about the combination of WebTransport + WebCodecs: https://www.w3.org/2018/12/games-workshop/slides/21-webtransport-webcodecs.pdf https://www.w3.org/2018/12/games-workshop/slides/21-webtrans... And then there is the work happening in the IETF along these same lines: https://datatracker.ietf.org/wg/ript/documents/ https://datatracker.ietf.org/wg/ript/documents/

TomMarius 6y ago
Thank you and others for answering me! :-)