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What is SIP?

Though not too many people know about or use it, there is a hardware/software combination that lets you use the Internet to make phone calls. In fact, in many cases, you can make free voice calls anywhere in the world. That’s known as Internet telephony.

And that’s where a special term comes into play—”SIP.” Short for Session Initiation Protocol, SIP is an IP telephony signaling protocol for Voice over Internet Protocol (VoIP) calls. SIP is responsible for connecting, monitoring, and disconnecting VoIP sessions. Sometimes instant messaging (IM) uses a modified version of SIP.

SIP can establish sessions for real-time sessions such as online gaming, teleconferencing, and video conferencing. All of this is transmitted over networks connected by IP addresses and IP protocols.

In simple terms, SIP makes it possible for your Internet Service Provider to integrate basic IP phone capabilities with Web, email, and even online chat. If you’re keeping track of technical points, SIP is part of the application layer of the TCP/IP protocol layer or stack.

Take a cool SIP.

It gets even better because in addition to user authentication, redirect and registration services, the SIP server supports traditional telephony features such as personal mobility, time-of-day routing, and call forwarding, based on the geographical location of the person being called.

SIP is mostly used to start and end VoIP phone calls. Originally, it was designed to mimic the call setup and signaling characteristics of a traditional telephone network by using the Internet Protocol infrastructure. A typical SIP session, technology-wise, involves a caller (client) requesting a voice connection with a SIP server from his computer. After the call is sent, the SIP server sends a response back to the caller indicating whether or not a voice connection is possible. Callers are identified by their SIP address, which is similar to an email address.

SIP relies on a peer-to-peer setup (computer to computer) that uses network protocols for advanced call processing and call management functions. These two endpoints for communication (caller and receiver) are referred to as the user-agent client and the user-agent server. A proxy server can be used as an intermediary to transfer the connection request from the client to the SIP server. SIP proxy servers can provide advanced call-processing functions including security, authentication, and call routing. Another protocol called RTP (Real-time Transport) carries the voice or video content at the TCP/IP application layer between SIP endpoints.

Instant messaging.

As mentioned before, instant messaging applications are SIP clients that can be used to send out voice and video messages free of charge. Microsoft MSN Messenger and Apple iChat are examples. The popularity of instant messaging led to the development of refined instant messaging protocols based on the SIP standard.

SIP has some limitations as a phone system, mainly around emergency calling and law enforcement interception. Unlike regular 911 calls, with SIP there is no way to pinpoint the location of a caller, and it’s also hard for law enforcement to monitor and intercept SIP-based phone calls. But despite these limitations, SIP is growing in popularity due to its ease of use and management, low cost, and how easy it is to expand in scope.

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