Session Initiation Protocol (SIP) and Signaling System 7 (SS7) are the common protocols used to transmit voice over networks. How they work with VoIP….or not….opens the door to both concerns and opportunities.
Session Initiation Protocol (SIP) is a protocol developed by the IETF MMUSIC Working Group and a proposed standard for starting, modifying, and ending an interactive user session involving multimedia such as video, voice, instant messaging, online gaming. and virtual reality. SIP is a text-based signaling protocol similar to HTTP and SMTP, and is used to create, manage, and terminate sessions on an IP-based network. A session could be a simple two-way phone call, or it could be a collaborative multimedia conferencing session.
Entities that interact in a SIP scenario are called User Agents (UAs). User agents can operate in two ways:
o User Agent Client (UAC): Generates requests and sends them to the servers.
o User Agent Server (UAS): Receives requests, processes those requests and generates responses.
SIP works like this: callers and call recipients are identified by SIP addresses. When making a SIP call, the caller first locates the appropriate server and then sends a SIP request. The most common SIP operation is the invite. Instead of going directly to the intended recipient, a SIP request can be redirected or trigger a chain of new SIP requests by proxy servers. Users can register their location(s) with SIP servers.
Now… how is this different from the SS7 protocol?
Here is a simplified explanation:
Signaling System 7 (SS7) is an architecture to perform signaling in support of PSTN call setup, billing, routing and information exchange functions, while SIP is a protocol used to maintain sessions in VOIP.
SS7 is used to configure the vast majority of the world’s PSTN phone calls, while SIP is used on the IP network.
A bit more about the differences between SS7 and SIP.
SS7 uses a common channel to signal call setup and disconnection information for circuit switched services. It is common to have hundreds or thousands of voice circuits controlled by a pair of 64 kb/s signaling 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 signaling solution that does not use a separate signaling path, but instead relies on IP connectivity from the originator to a server and from there to the end point. It is used for packet-based communications and allows for many different types of calls, such as video, game interaction, etc., as well as voice.
As SIP rolls out with the rollout of next-generation networks, I’m sure we’ll see some very interesting network behavior, incalculable new glitches as we fix the bugs, and likely new fraud opportunities. These must be interesting times.