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I am trying to develop a live video chat app using flash and rtmfp protocol but I have doubts as to
how rtmfp guarantees to connect peers especially when the peers are located in different networks.
RTMFP relies on a central rendezvous server to “introduce” 2 clients that are in different networks and potentially behind firewalls. To navigate through firewalls it uses NAT traversal techniques which essentially amounts to:
There is no guarantee that any 2 clients networks/firewalls are compatible for RTMFP P2P connections (you can diagnose with the RTMFP connection tester), that’s why Adobe provides fallback solutions through their LiveCycle Collaboration Service or Adobe Flash Media Server to a central relay service (basically all network traffic would pass through a Server that both clients can publicly access) if a direct P2P connection can’t be established.
Adobe’s RTMFP is their proprietary bundling of Peer 2 Peer network rendevous service, as well as providing some other higher level P2P network mesh features.
If you’re interested in a more open standard P2P protocol you should look into WebRTC. Essentially WebRTC is the same concept of allowing clients to connect P2P over UDP but geared towards being adopted as a web browser standard, and can also be implemented on a variety of native devices (e.g. Android, iPhone etc) under the hood it uses standard NAT and firewall traversal technology using STUN, ICE, TURN, RTP-over-TCP and support for proxies. I believe WebRTC is a standardization of some of the work done in the libjingle P2P Google Talk library.