Tuesday, February 24, 2009


Adobe's RTMFP is not P2P-VoIP as exemplified by Skype. On the other hand, RTMFP is closer to client-server SIP or H.323 where signaling happens via a server and media path can be end-to-end between the endpoints. When people refer to RTMFP as P2P, it is more like 'end-to-end media' similar to client-server SIP.

Why is RTMFP important? The previous Adobe protocol RTMP is strictly client-server even for media path. This gives poor quality for real-time media communication because media packets go from client to server, that too over TCP, and then are redistributed to the other client, again on TCP. End-to-end media based VoIP systems existed before Adobe implemented RTMP. I suppose the difficulty of NAT and firewall traversal and lack of interactive video communication requirement in Flash Player resulted in RTMP. Adobe corrected this mistake in the new protocol RTMFP which allows NAT and firewall traversal (to some extent) and allows end-to-end media path without going through the server. Although, the signaling is still going via the central server.

Once we understand this difference between P2P-VoIP and RTMFP, lets enumerate the differences between an RTMFP-based and a client-server SIP-based communication system.

1. RTMFP is a closed protocol, although Adobe recently opened up the previous RTMP. On the other hand, SIP is an open standard from IETF. This means anyone can implement SIP whereas only Adobe can implement RTMFP. That also means that a bug in the RTMFP protocol or its implementation is outside the scope of public review such as for security experts.

2. RTMFP is an integrated protocol that has support for signaling, encryption, media flow (flow control and congestion control), NAT traversal. Whereas SIP is just one piece of the puzzle, that is used in conjunction with RTP/RTCP, SDP, STUN, TURN, ICE, SRTP, etc. to build a complete system. In that regard there is more scope for interoperability problems in SIP systems. The SIP interoperability test (SIPit) events have helped in solving interoperability problems among current products for over a decade. (see next point on why RTMFP alone may not be sufficient?)

3. Based on the available documentation, RTMFP works on UDP. Whereas SIP can work on UDP as well as TCP. In an RTMFP application, the client should fall back to TCP-based RTMP if for some reason UDP is blocked for the client-server communication. This also means that the client will lose some of the benefits such as encryption available in RTMFP. There are other protocols RTMPS and RTMPE to facilitate security and encryption over TCP-based RTMP.

4. Although RTMFP works on UDP, it implements additional flow control and TCP-friendly congestion control. This helps media traffic deal with network congestion and slow receivers. On the other hand most existing SIP system do not implement such mechanisms in the media path. While this looks like an advantage in RTMFP, it turns out to be a problem because of the way it is implemented. In particular, the network components are disconnected from the media source components such as camera and microphone. The rate control mechanisms are implemented in network components which internally slow down the media traffic by delaying or dropping the UDP media packets. On the other hand the encode quality settings on camera and microphone components are unaffected. This results in packet drops due to congestion and hence choppy video or audio drop-outs. A good application built on top of RTMFP is supposed to get feedback from network components and adjust the encode quality parameters (framerate, bitrate, quality) in the camera and microphone components so that the packet drops are reduced. Thus, unless the application is smart enough to deal with this, the disconnected implementation of rate control and media source causes quality problems in RTMFP.

5. Both RTMFP and SIP can use media relays to workaround NATs and firewalls. However, RTMFP does not use a super-node architecture where some clients (Flash Player instances) act as relays, whereas (P2P) SIP can use existing client nodes to act as media relays. This means that when using RTMFP, the service provider must bear all the bandwidth cost of the relays, whereas in (P2P) SIP the cost can be distributed among the users because of the peer-to-peer nature. I analyze the cost due to NAT and firewall traversal in my next post.

1 comment:

Martin Supiot said...

So in 2013 what's the best protocol to use for a visio chat application?