Hi all, i need to setup a server that handle VoIP connection in a way that always works in the most efficient way respect to latency and NAT traversal. FS provide very good NAT detection system but unfortunately when it fail, it just fail and does not have any kind of fallback mechanism. If FS do bypass-media and the client cannot communicate because the phone call is lost. If FS do proxy-media, the clients can always communicate (over UDP) but there is added latency. So we cannot rely only on FS NAT detection system and need also some client side system. Now, in theory the best approach would be not to touch FS but just use client-side system based on ICE/STUN with a TURN server-side component as a fallback. However there's no widely available TURN server, or at least TURN is not widely and well diffused just now. I am wondering whether it may be possible to make a setup where: - Client try to establish peer to peer connectivity with STUN/ICE (so FS is doing bypass-media not touching the SDP) - If they cannot work, will go in fallback to FS (but with proxy-media) instead than going via TURN So using the FS B2BUA proxy-media feature as alternative to TURN server, but letting clients to dynamically try to connect directly one each other by using the ICE methods that are dynamic. Does it seems something feasible? Fabio Pietrosanti