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What is SIP, An Overview

Friday, April 24, 2009 ≡ Mario Stocco

Session Initiation Protocol, or SIP for short, is often defined as a smarter, lighter way of achieving the goals that the H.323 protocol was intended to solve. H.323 evolved from the telecommunications sector and from those origins, it turned into an overly verbose protocol that placed undue overhead on data networks. SIP received its start from within the Internet Engineering Task Force and, as a result, is much more efficient and web-centric in nature.

SIP is formally described as a control protocol for creating, modifying and terminating sessions with one or more participants. Sessions typically being IP network telephone calls or multimedia conferences. Participants in a session may communicate in a multicast fashion or via an association of uni-cast relations. SIP supports session descriptions that allow participants to agree on a set of compatible media types and also supports user mobility by proxying and redirecting requests to the user's current location.

In more user-friendly terms, SIP provides a standardized language for disparate devices or applications to agree on how to communicate and interact with each other.

Four Key SIP Elements

SIP User Agents - these are the end points, residing in desktop SIP telephones, PCs, PDAs, 2.5/3G wireless handsets, or SIP gateways, which initiate and answer calls and are responsible for call features such as transfer, conference and hold.

SIP Proxy and Redirect Servers - these reside in the network and provide the necessary infrastructure for name resolution and user location. SIP Proxy Servers perform routing functions, directing requests to the correct end point user agent, possibly via a chain of other proxies in the path. SIP Proxy Servers may also perform a number of related functions such "forking" to attempt to contact multiple user locations simultaneously - or act as platforms for specific applications such as call filtering. Redirect servers perform a similar function of user location, redirecting the calling side to a different end point rather than doing the routing first hand. While some of these servers deal specifically with SIP, others - such as location and ENUM servers (effectively directories that translate between IP addresses and telephone numbers) provide support functions to the primary SIP servers in the network.

SIP Registration Service - this provides a means for a particular device to register to use a SIP address. SIP addresses use "URLs" based on the same addressing scheme used in the web and similar in form to an email address (example: sip://info@advantia.ca). The SIP address provides a single address of record for the user that delivers a one number service for all communications applications. Users can dynamically register the devices through which they may be contacted for all types of applications. As a result, people will no longer have to hand out multiple contact addresses as the system will automatically handle the distribution of all types of calls appropriately through the proxy and redirect servers.

SIP Event and Presence Servers - these allow the effective sharing of information about and between users and/or applications. Presence and events work in the same manner as found in closed Instant Messaging networks; SIP provides a standardized manner for this idea to work.

SIP was originally designed to operate as an end-to-end, device-to-device protocol, rather than a network-centric one. The development of a framework to manage events means that SIP has evolved to work more like a web-based system. Applications such as conferencing can be created and made available to the network through SIP, with the SIP event framework allowing services to be managed and shared effectively between users. The presence and buddy list applications outlined above show how this event- based information sharing can increase the effectiveness of applications.

Want to leverage SIP for your business communications? Click here, Advantia VoIP Systems is ready to help you.

About this blog

With nearly a decade of in-the-trenches computer telephony experience, Mario Stocco writes this weblog to articulate his thoughts on topics like VoIP, Open Source and life in general.

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