Vendors on SIP and TCP

Today we’ll conclude (for the time being) our discussion on how best to transport control information for voice over IP. We asked some of the manufacturers of VoIP gear to talk about their support for Session Initiation Protocol (SIP) and how they transport the control protocol.

Speaking for Nortel, Tony Rybczynski commented that Nortel has done interoperability testing for SIP using UDP. He sees this as having been sufficient since SIP has application timers. However, he also notes that Nortel supports TCP and expects future work in SIP on TCP. UNIStim, Nortel’s skinny protocol for IP phones, uses UDP and depends on application timers for reliable transmission.

Avaya will start supporting SIP this fall, offering the option of transporting SIP over UDP, TCP or Transport Layer Security. An Avaya spokesperson agreed that TCP is a better protocol than UDP for transporting SIP signaling.

Cisco’s Craig Cotton confirmed that its CallManager uses TCP for the call control traffic, and Simon Gwatkin from Mitel likewise confirmed that his company’s products use TCP for call control.

As these and other manufacturers of VoIP gear migrate to transporting SIP over TCP, there is a strong need for test equipment that supports TCP, especially to stress-test both for throughput and security. Additionally, not all TCP is the same, so all modes of TCP should be tested. At least one company helping to address this challenge, Radcom, has recently added SIP-over-TCP capabilities to its test gear, testing TCP’s persistent, port reuse and non-persistent modes.

Note: In regards to the previous newsletter, “TCP vs. UDP for VoIP control,” you can get a more detailed description of how TCP and UDP fit into the overall VoIP communications scheme by checking out Tony’s tutorial presentation, “Transport Protocols for Multimedia Networking” (https://www.webtorials.com/main/eduweb/voice/index.shtml).