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My public instance runs on a world addressable host and doesn't have TURN. I have no problem using it from my phone and it is behind a CG NAT. No TURN/Relay ser
by Sean-Der 2y ago
My public instance runs on a world addressable host and doesn't have TURN. I have no problem using it from my phone and it is behind a CG NAT. No TURN/Relay servers are needed.
> I've never been able to use any WebRTC-enabled service for this reason, nor anyone else I know
Is this a browser/agent issue? You aren't able to use Google Meet or Zoom in the browser?
- ranger_danger 2y agoLive-streaming in one direction doesn't need a relay because the users aren't uploading... And google/zoom use their own relays.
- Sean-Der 2y agoGoogle doesn't use any STUN/TURN servers I just went and confirmed. The following is the RTCPeerConfiguration used. ``` https://meet.google.com/***-***-\*\ https://meet.google.com/***-***-\*\*, { iceServers: [], iceTransportPolicy: all, bundlePolicy: max-bundle, rtcpMuxPolicy: require, iceCandidatePoolSize: 0 }, ```
- ranger_danger 2y agoIt may not use it if you don't explicitly need it, there could be out-of-band detection that adds a TURN server or some other OOB relay if it's needed. I've also seen comments online that say Chrome in particular can support TCP for WebRTC which would negate the need for a relay (as normally only DTLS over UDP is used). But based on my understanding of how WebRTC and NAT works, using the typical UDP approach for bidirectional communication over symmetric or CGNAT absolutely should not work, barring some other method of NAT traversal such as a browser-based UPnP client or such.