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Session Initiation Protocol (SIP) and Signaling System 7 (SS7) are the common protocols used for transmitting voice across networks. Just how they work with VoIP….or not….opens the door for both concerns and opportunity.

Session Initiation Protocol (SIP) is a protocol developed by IETF MMUSIC Working Group and proposed standard for initiating, modifying, and terminating an interactive user session that involves multimedia elements such as video, voice, instant messaging, online games, and virtual reality. SIP is a text based, signaling protocol similar to HTTP and SMTP , and its used to create, manage and terminate sessions in an IP based network. A session could be a simple two-way telephone call or it could be a collaborative multi-media conference session.

Entities interacting in a SIP scenario are called User Agents (UA) User Agents may operate in two fashions –

o User Agent Client (UAC): It generates requests and send those to servers.

o User Agent Server (UAS): It gets requests, processes those requests and generate responses.

SIP works as follows: Callers and callees are identified by SIP addresses. When making a SIP call, a caller first locates the appropriate server and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies. Users can register their location(s) with SIP servers.

Now…..how is this different than the SS7 protocol?

Here’s a simplied explanation:

Signaling System 7 (SS7) is architecture for performing signaling in support of the call-establishment, billing, routing, and information-exchange functions of the PSTN, whereas SIP is a protocol which is used for maintaining sessions in VOIP.

SS7 are used to set up the vast majority of the world’s PSTN telephone calls, where as SIP in used in IP network.

A little more on the differences between SS7 and SIP.

SS7 uses a common channel for signalling call setup and tear down information for circuit switched services. It is common to have hundreds or thousands of voice circuits controlled by a pair of 64 kb/s signalling links. SS7 was specifically designed for circuit switching although it has some very sophisticated additional call control and transaction control capabilities.

SIP is an IP based signalling solution which does not use a separate signalling path, but relies on the IP connectivity from the originator to a Server and thence to the terminating end. It is used for packet based communications and allows for many different call types such as video, gaming interaction etc as well as voice.

As SIP is implemented with the deployment of next generation networks I am certain we will see both some very interesting network behaviours, untold new technical issues as we iron the bugs out and probably new opportunities for fraud. They should be interesting times.

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