From what I learned, for production use of WebRTC, a specialized signalling server (STUN/TURN) is needed to reliably coordinate communication among peers - in p

From what I learned, for production use of WebRTC, a specialized signalling server (STUN/TURN) is needed to reliably coordinate communication among peers - in particular, to deal with network address translators (NATs) and firewalls. Most open-source WebRTC implementations I've seen use a list of semi-public STUN/TURN servers, notably including Google's, which I believe are explicitly for dev/testing purposes. In addition to this "discovery layer" - and possibly handling reconnections - I imagine it's non-trivial to provide video/audio transport, compression, etc. I'd love it if it were simpler to use WebRTC, truly peer-to-peer, in the vein of how early Skype used to work. Unfortunately, from what I've learned so far, there are still technical hurdles to self-host such a service. https://en.wikipedia.org/wiki/STUN https://en.wikipedia.org/wiki/STUN https://en.wikipedia.org/wiki/Traversal_Using_Relays_around_NAT https://en.wikipedia.org/wiki/Traversal_Using_Relays_around_... https://en.wikipedia.org/wiki/Interactive_Connectivity_Establishment https://en.wikipedia.org/wiki/Interactive_Connectivity_Estab...