* Protocol questions surrounding call control for voice over IP
In the last newsletter we started discussing the protocol stack used for transporting voice over IP. Today we’ll move to the control for those conversations.
By the way, before we get all those e-mail messages telling us that Real Time Protocol is used for transport, we need to point out that RTP actually is used as YAP (Yet Another Protocol) inside UDP. So in a fashion similar to those Russian dolls that stack inside each other, you have your voice packet inside RTP inside UDP inside IP inside a Layer 2 frame (such as Ethernet) inside a physical layer.
So, we know what the transport of the voice conversation looks like. But for any advanced VoIP implementation, such as an IP-PBX, you also need call control, and this is usually provided by either H.323 or, increasingly, Session Initiation Protocol (SIP). However, a proprietary control protocol may also be used. And this protocol must fit into the protocol stack as described above.
The advantage to using TCP for transporting the control protocol is that you’re guaranteed the control information is delivered. On the other hand, you also could use UDP if the protocol is designed such that “best effort” delivery is acceptable.
As Tony Rybczynski of Nortel puts it, “H.323 uses TCP, with lots of session tear-downs and setups. SIP originally preferred UDP and said TCP was optional, but the last release of the spec reversed that.”
So it seems that a direction is set, but we’re just starting the journey. And that way appears to be a transition to SIP, with SIP running inside TCP.
For a more detailed description of how TCP and UDP fit into the overall VoIP communications scheme, you might want to check out Tony’s tutorial presentation, “Transport Protocols for Multimedia Networking,” at the link below.
In the next newsletter we’ll take a look at what some of the equipment manufacturers are saying about their support of SIP and TCP.
