The animated TECHTip tutorial available here.

IMS is the next-generation system for security such as AAA-Authentication, Authorization, Accounting messaging including ringtones and Push-to-Talk and QoS-Quality of Services and other services.

Some of the key functions of IMS are:

• AAA [Authentication, Authorization, Accounting] business and fraud prevention functions.

• PtP [Person-To-Person], PTT [Push-To-Talk], IM [Instant Messaging], CRBT [Color Ring Back Tones], Conferencing and streaming - “My” functions - calendar/tones/photo database
• QoS [Quality of Service] options - multimedia, video, streaming, content-driven services.
• Roaming - business interconnection functions.
• Rich Call, also known as MMS [MultiMedia Service] - integrated voice/data/video with parallel sharing (live and database) media.

Internet access or VXML [Voice eXtensible Markup Language] applications can be directed based on an exception-triggered instant conference such as a disaster. Because SIP is access independent - working with fixed domain (wireline - analog PSTN [Public Switched Telephone Network], ISDN [Integrated Services Digital Network], IP [Internet Protocol], DSL [Digital Subscriber Line], ATM [Asynchronous Transfer Mode] and others) and mobile domain (wireless - GSM [Global Service] for Mobile, CDMA [Code Davison Multiplex Access], WCDMA [Wideband CDMA], GPRS [General Packet Radio Service] and others), it has been adopted by the 3GPP [3rd Generation Partnership Project] and the Layer 5 Session Management Protocol and the basis for IMS [IP Multimedia] Subsystem.

SIP [Session Initiation Protocol] is the real-time communication protocol for VoIP [Voice over IP]. SIP has been expanded to support video and instant-messaging applications. SIP is designed to perform basic call-control tasks, such as session call set up and tear down and signaling for features such as call hold, caller ID, conferencing and call transferring. SIP Server Forks (select different paths) Invite(s) to multiple User Agents (telephones). Forks can be sequential or concurrent. Session established to first UA [User Agent] to respond with OK. Cancel sent to non-respondent (no answer) User Agents. Forks are useful in a call center environment or call hunting.

Call Deflection automatically redirects (forwards) a call from one called endpoint to another endpoint (such as a voice mailbox) when the called endpoint is busy or does not answer. Call deflection is one of several forms of call diversion (also known as call forwarding) defined under the H.450.3 specification. In call deflection, there are three communications points, referred to as the originating residential gateway, the deflecting gateway (also called an endpoint), and the deflected-to gateway (the final endpoint). Calls are deflected by a command called the reroute invoke. Proxy Server call flow is where Proxy Server sets up call. Proxy Server is an optional SIP component that handles routing of SIP signaling but does not initiate SIP messages. SIP call flow is provided with Redirect Server. Redirect Server an optional SIP component that does not route SIP messages. Redirect Server returns a redirect (change in routing such as Call Forwarding) to UA for direct routing (SIP is designed for end-to-end signaling without intervention by a server).

“Presence” is an all-encompassing term used to describe reachability control over how, where, when and by whom they can be contacted (reached). Presence covers any concept such as “buddy lists” (desired contacts) or the means (wireless/wireline), device (pager, cell, PDA, TV, etc.) or media (voice, data, music, multi-media) and yet-to-be-defined means of communication.

Here is an example of SIP used for instant messaging. The function of a Presence Server is to manage access, connections, directory (who’s who), billing and tracking and other management functions. However with SIP, the intelligence for call setup and features resides on the SIP device or user agent, such as an IP phone or a PC with voice or instant-messaging software. In contrast, traditional telephony or H.323-based telephony uses a model of intelligent, centralized phone switches with dumb phones with SS7-Signaling System 7 in PSTN telephone switching and H.323 or Media Gateway Control Protocol in IP telephony providing call control/routing. For more, go to sipforum.org.

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