This is the first of two blog postings on this topic. The second posting is planned for Tuesday, April 28th.

Most early VoIP networks that use SIP Signaling use UDP as the transport layer for carrying signaling and payload. The advantage of using UDP as a transport layer protocol is that it is simple, has low overhead and is omnipresent. The disadvantage of UDP is that it is not a guaranteed delivery protocol.

While UDP is good enough for carrying the payload such as voice, its suitability for carrying Control Information is debatable.  Message loss has very limited impact on the quality of voice (hardly noticeable), whereas signaling message loss could have a significant impact on the quality of session setup (post-dial delay, dropped sessions, session setup time etc.), network resource utilization, and charging functions.

Use of UDP can also be justifiable at the edge of the network (User to Network boundary), where the advantage of simplicity and scalability outweighs any reliability requirements. However, as SIP becomes a mainstream signaling control protocol within carrier core networks, using SIP over UDP is questionable. Looking at the past, when carriers migrated SS7 signaling from TDM to IP, experience clearly indicates that SS7 over UDP (or TCP) was not a feasible option. The industry standardized on SS7 over SCTP implementation, which was developed as part of SIGTRAN group of protocols. The same was also true with BICC networks as mobile carriers started deploying R4 Mobile Switching Servers. Looking into the future, LTE networks and the IMS architecture have standardized on SCTP as the standard transport layer protocol for signaling.

What are the challenges of building a large scale SIP-based signaling network using SCTP as the foundation? Is it scalable? Have standard bodies addressed all the architectural issues related to building a large scale SIP signaling network over SCTP? I will be talking more about this next week, and meanwhile I would like to know what you think about this?