Developed by the Internet Engineering Task Force (IETF), SIP (Session Initiation Protocol) is used to begin, modify, and terminate sessions between one or more users. A "session" is any interactive communication that travels across the Internet, whether it is voice, instant messaging, video, etc.

SIP is very similar to HTTP, which is the underlying protocol used by the World Wide Web. HTTP issues request-response messages to connect users with web pages. For example, when you type www.ind.alcatel.com into your web browser, HTTP sends a request to the server on which that web page is located, and the response, the transmitted web page, is returned. Similarly, a SIP request can be sent in the form of computername@ind.alcatel.com to request a connection. Both HTTP and SIP use URLs for addressing. A URL (universal resource locator) is a type of address that describes the location of information on the World Wide Web, e.g., www.ind.alcatel.com. With SIP, even phone numbers are converted to URLs. The similarity of SIP to HTTP not only makes it ideally suited for the IP environment, but it’s easy for HTTP experts to program it without learning a new language.

SIP works to establish a session by sending a request to talk across the Internet. When the request reaches the desired user, that user can respond in a variety of ways, for example, he can reply to the requester to continue communicating or send an automatic "busy" message. Since SIP sets up sessions for multiple types of communication, it uses the Internet standard multi-purpose Internet mail extensions (MIME) to describe the type of communication - this way the user can see what type of session he is being invited to. MIME also ensures that the information in a session is formatted correctly for Internet communication.

A remarkable feature of SIP is its ability to establish "presence," that is, to find not only where a user is physically located, but also if he is accepting communication, and what medium he is able to communicate with, e.g., voice, instant messaging, etc. SIP establishes presence by sending request-to-talk messages to a proxy server, which "forks" a single request to several of the intended user’s devices, for example, a desk phone, a home phone, and a PC. The first device to answer accepts the call. A close analogy is your home phone line; when someone calls you, every phone in the house rings at once until one is answered.

SIP integrates multi-media communications like voice seamlessly with web, email, and instant messaging. For example, you can just as easily transfer someone to a web site as you can transfer him/her to another phone extension. This is important, because with SIP, a single infrastructure can be used for voice, video, audio, instant messaging, and presence.

SIP promises to improve Internet telephony by integrating it with a new set of services and features, namely web, video, instant messaging, and more. "In essence, the value of VoIP and SIP comes not from integration at the network layer, (i.e., run your voice services on top of your data network), but at the services layer (i.e., combine your voice services with your data services)" (Rosenberg and Shocky, Computertelephony.com June 2000).