In our last VoIP installment, we looked at the main reasons why SIP has become a widely adopted protocol, but we left details of the protocol’s inner workings fairly vague. This article will drill down into the way the Session Initiation Protocol (SIP) works, and it should serve as a good starting point for really learning SIP. If you haven’t already done so, you are encouraged to read the previous article, although it’s not a prerequisite. This introduction also covers the latest SIP extensions and changes, so it gives a complete view of the protocol’s current state, rather than just the basic, underlying RFC.
Session Initiation Protocol (SIP) is a VoIP signaling protocol. As its name suggests, it has everything to do with setting up sessions, which means it has the responsibility for starting a session after you dial a number (or double-click, in some cases). As such, SIP’s role also includes maintaining user registrations with a server, defining session routing, handling various error scenarios, and, of course, modifying and tearing down sessions.
We’ll present this introduction in two parts. In the first part, we’ll focus on the SIP foundation layers. These layers allow creating a network of SIP servers. In the next article, we will go through the way a phone communicates with the rest of the world using this server network, based on the same foundation layers.
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