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Session Initiation Protocol
   
 
How can a communications protocol transform your business? Consider the impact of IP communications: lower costs, powerful new applications, simplified management. These benefits are accelerating adoption of I-P Telephony on a worldwide basis.
 
Session Initiation Protocol, or “SIP”, builds on the benefits of I-P by adding intelligence to business processes and providing both users and enterprises with better control over their communications. SIP supports any form of real-time communication regardless of whether the content - is voice, video, instant messaging or a collaboration application.
 
And “SIP” delivers presence: a technology that lets users inform others of their status, their availability, and how they can be contacted -  before a communication session even begins.
 
Learn how “Sip” can make your people more productive, your business processes more intelligent and your customers more satisfied by putting  communications at the heart of your enterprise.
 
 
SIP - An Overview
   
SIP, or Session Initiation Protocol, is designed primarily to set-up, modify, and tear down interactive communication sessions. SIP is an agnostic protocol- it can support any type of communication session whether it is voice, video, or instant messaging.
 
Although SIP seems like a "new" protocol, it is actually based on many protocols widely used today across the Internet and in many enterprise applications.  If you use web browsers, then you already depend on a protocol very similar to SIP, called HTTP (Hyper-Text Transport Protocol).  SIP is modeled after HTTP, and in fact uses much of its syntax and semantics.  Both are text-encoded protocols that help promote interoperability and integration within an internet-centric architecture.
 
 
SIP is also similar to the e-mail you use everyday.  SIP uses URIs (Uniform Resource Indicators) for user addressing, in the same basic form as e-mail addresses:  user@domain  (i.e. jimsu@avaya.com).  URLs (Uniform Resource Locator) describe the location of resources or devices.

For SIP communications this can apply to any communication device, some examples of which are:
 
A phone:   tel: 408-555-1212 OR sip:408-555-1212@Avaya.com;user=phone
A fax:  fax:408-555-1212 OR sip:408-555-1212@Avaya.com;user=fax
An e-mail account:  mailto:jimsu@avaya.com
 
SIP supports both numerical (telephony) and web-based addressing, providing a critical bridge for communications between PSTN and Internet networks.  SIP even uses e-mail routing mechanisms.  Like e-mail, SIP resources can be easily found through a DNS query using the domain name in the address. 
 
Because of its internet-centric design, SIP communications can be easily integrated within internet and eCommerce-based applications, such as the integration of a SIP client within a web page using a SIP URL as a hyperlink within a web page.  Pressing this hyper-link could then initiate a communication session, such as an IP Telephony call.

What are the primary components of a SIP-Enabled Network?

SIP is designed for peer-to-peer communications, and uses a client-server (request/reply) model similar to HTTP.  A SIP client is a party that initiates a SIP request (i.e. wants to make a “call”).  A SIP server is an application that services SIP requests and sends responses (i.e. receives a “call”). During a session, a SIP endpoint, such as a phone, can serve as either a client or a server, depending on if it is initiating or responding to a request.

User agents are applications in SIP endpoints (such as a SIP phone) that interface between the user and the SIP network.  An agent can act as either a client or a server.  When making a call it acts as an User Agent Client (UAC), when receiving a call it acts as an User Agent Server (UAS).  A Back-to-Back User Agent (B2BUA) is an application that acts as an intermediary between two parties, but appears as an endpoint to both parties.  It serves as both an UAS/UAC simultaneously to process session requests.  An example of this is Avaya Communication Manager, which acts as a B2BUA that represents SIP endpoints to the PSTN network and PSTN endpoints to the SIP network.

SIP devices can communicate directly if they know each other’s URL, but in practice SIP servers are often used in the network to provide an infrastructure for routing, registration, and the authentication/authorization services.  The core SIP servers and their functions are summarized below.
  • Registrar Server:  The Registrar registers users when they come on-line and stores information on the users logical identity, and the associated device
    (identified by IP address or URL) or devices they will allow for communications.

  • Location Service:  A database that keeps track of users and the URL bindings that are “closer” to them.
    The location service gets its input from the registrar server and provides key information for the proxy and redirect servers.

  • Redirect Server:  Maps a SIP request destined for a user to the URL of the device “closest” to the user. 
    For example, if a call is destined for johnsmith@avaya.com and the user is on the road, the Redirect Server may reply to the caller’s user agent
    (or to the requesting proxy server) with the contact address of the user’s cell phone.

  • Proxy Server:  Services SIP requests by processing them and passing them along to other SIP servers. 
    A proxy server may act as both a server and a client, and can modify a SIP request before passing it along.
    A proxy is involved only in the set-up and teardown of communications. 
    Once a session is established, communications occur directly between the parties.

  • Presence Server:  Accepts, stores, and distributes presence information.  The presence server has two distinct sets of clients:
    • Presentities (producers of information) provide presence information to the server to be stored and distributed
    • Watchers (consumers of information) receive presence information from the server
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