> . You also push complexity on clients. The simplicity of WebRTC (createOffer -> setRemoteDescription) is what lets people onboard easily.

WebRTC is complex, even if it's a library (even if it's a library built into the browser they're already using). For a client/server voice interaction, I don't see why you would willingly use it. Ship voice samples over something else; maybe borrow some jitter buffer logic for playback.

My job currently involves voice and video conferencing and 1:1 calls, and WebRTC is so much complexity... it got our product going quickly, but when it does unreasonable things, it's a challenge to fix it; even though we fork it for our clients.

I could write an enormous rant about TURN [1]. But all of the webrtc protocol suite is designed for an internet that doesn't exist.

[1] Turn should allocate a rendesvous id rather than an ephemeral port when the turn client requests an allocation. Then their peer would connect to the turn server on the service port and request a connection to the rendesvous id, without needing the client to know the peer address and add a permission. It would require less communication to get to an end to end relayed connection. Advanced clusters could encode stuff in the id so the client and peer could each contact a turn server local to them and the servers could hook things up; less advanced clusters would need to share the turn server ip and service port(s) with the id.