SIP (Session Initiation Protocol) is a signaling protocol 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. This makes possible to implement services like voice-enriched e-commerce, web page click-to-dial or Instant Messaging with buddy lists in an IP based environment. Don't worry if you don't know about these services. You don't need to know them before you learn about SIP.
SIP has been the choice for services related to Voice over IP (VoIP) in the recent past. It is a standard (RFC 3261) put forward by Internet Engineering Task Force (IETF). SIP is still growing and being modified to take into account all relevant features as the technology expands and evolves. But it should be noted that the job of SIP is limited to only the setup and control of sessions. The details of the data exchange within a session e.g. the encoding or codec related to an audio/video media is not controlled by SIP and is taken care of by other protocols. For an overview of the major SIP functions, click here.
This introduction is meant for beginners. This beginners' made easy tutorial is to give a brief introduction to SIP before one ventures into the long RFC documents. However, if you are a veteran please go through this short tutorial and suggest modifications.
Here on this site the aim is not to make you an expert of SIP based applications. I doubt whether any site can do that. You have to have hands on experience to muster the aspects related to Internet multimedia or IP telephony. Here I am proposing nothing new. The whole job is to initiate a newcomer with the facets of the Session Initiation protocol (SIP) so that a near 200 page RFC document does not intimidate you. However I strongly recommend that you go through the document of RFC 3261 once you have completed this tutorial.