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Comment by TD-Linux | Hacker News Reader

That's because you don't need to implement any of those things for a basic client-server architecture. Unfortunately it looks like the author saw the massive and complex webrtc.org implementation and gave up, rather than try writing their own minimal implementation, or borrow from other implementations like Janus (https://janus.conf.meetecho.com/textroomtest.html).

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There currently are only massive and complex implementations, because that's what the protocol requires. Even if you leave out the p2p bits you still need to get SCTP over UDP and DTLS support, which are both uncommon. The Janus thing also pulls in usrsctp and openssl for those. In Janus case there's even more things like libnice, which then requires you to use glib. All in all that's a massive set of C dependencies, which not everyone is comfortable pulling in. "Borrowing" from GPL software is also not what everybody is comfortable with. The webrtc.org reference implementation seems to contain half of chromes networking stack, which seems even worse.

I do also now prefer to use other languages than C++ to build servers, let it be Go, Java or C#. In all of those getting native webrtc data channel support is a giant effort, because there's neither sctp nor dtls support available. You could fallback on C libraries for this functionality, but that complicates the build process. Establishing HTTP, websocket or even HTTP/2 support is all easier than webrtc.

Yes; vaguely... there was some (string) message format that you could manually stitch together with the right IP and port, etc, and basically tell the browser's WebRTC implementation "other client is on this IP and port". I seem to have deleted this code unfortunately, the only thing I saved is this link:

https://github.com/cjb/serverless-webrtc/

EDIT:

Aha! I based my code on: https://github.com/js-platform/node-webrtc/blob/develop/exam...

Yup, the format is called SDP. It's an old, crufty, and mostly effective way to specify not only the port and IP, but also whether a TCP fallback is being used, video and audio formats, and other things you can mostly ignore when just using data channels. See the "answer" section of 5.2.3 of [1], for details of some of the other magic numbers (which your JS library should handle for you). You can also check out about:webrtc to see SDPs made by other websites, like the Janus demo page.

[1] https://tools.ietf.org/html/draft-ietf-rtcweb-sdp-03

You need to implement DTLS and SCTP for client->server, just not ICE. But client->server ICE is fairly trivial with ICE lite. It's DTLS+SCTP that are complex, and they don't go away client->server.

Replacing DTLS+SCTP with QUIC would be a good reduction in complexity, and we (the WebRTC team) are experimenting with doing so (https://cs.chromium.org/chromium/src/third_party/webrtc/p2p/...). It has a long way to go, but hopefully someday just having QUIC on your server will be enough.

(I work on WebRTC)

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