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Good work! I just want to point out that WebRTC requires servers for STUN and TURN.

If I may, we also developed an open source solution for WebRTC videoconferencing, livestreaming and even peer-to-peer broadcasting to unlimited numbers of people! Feel free to use it:

https://community.qbix.com/t/teleconferencing-and-live-broad...

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It does not require either of those.

NAT traversal is the first point on the feature list of the linked repo. So in context, it's important to point out.

While in theory it doesn't it reduces the possible connections quite a bit. Though STUN servers are plenty on the internet and free to use. There's even a service providing TURN (US geo though), but I have no idea who runs it and what their business model is, so I wouldn't rely on it.

Having used webrtc in offline environments (host device broadcast a local wifi), I can say it's a royal pain to use without ice/stun/turn servers. I had to manually mangle the sdp packets to make it work because browsers would randomly decide they needed ip privacy (from the device they connected to by ip...) where they'd translate ips to some_hash.local and try to find a turn server. I tried hosting a coturn server too, but browsers would sometimes ignore it since it was a local address. Mangling the sdp handshake is reliable but feels pretty jank.

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