C videoconference Projects
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> despite WebRTC mostly being about client/client communication
This is actually kind of a misconception, though it’s an understandable one given that WebRTC is almost always pitched as a peer-to-peer protocol.
In practice, most people using WebRTC for video are sending their video to a server, not directly to another client. It’s pretty safe to assume that most people who use your app are going to need TURN, and at that point, you’re not really doing peer-to-peer at all, so you might as well just have your browser-based app talk to a server that’s pretending to be another browser.
These servers (called Selective Forwarding Units or SFUs) can operate like a TURN server in the case of a one-on-one call, but they can also multiplex everyone’s feeds in the case of a larger conference (peer-to-peer 5 person calls would require each participant to send 4 copies of their video) and often have extra features like the ability to record calls, transcode streams or convert to other protocols.
The one I’ve used a lot is called Janus[0], it’s open source and has good docs, I recommend people check it out if they’re interested in getting deeper into WebRTC or other video streaming tech.
[0] https://janus.conf.meetecho.com
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