SIP Communication Systems

Are you getting bored with a lot of information being fed to you about the Voice over Internet Protocol (VOIP)? Are you looking for some spice from the regular tidbits that you are getting? Or a new flavor at least… SIP VOIP is the answer to your problem!

With the continuous dedication of the computer networks in providing their clients with the most innovative products that they can utilize for everyday use, they do not just stop in discovering the VOIP, rather they persist on going beyond their innovations.

This is now the so called "SIP VOIP"…

SIP stands for Session Initiation Protocol which is an application-layer control protocol that can establish, modify, and terminate multimedia sessions such as Internet telephony calls.

SIP can also invite participants to already existing sessions, such as multicast conferences. Media can be added to (and removed from) an existing session. SIP transparently supports name mapping and redirection services, which supports personal mobility – users can maintain a single externally visible identifier regardless of their network location.

SIP supports the following five facets of establishing and terminating multimedia communications:

* User location: determination of the end system to be used for communication;

* User availability: determination of the willingness of the called party to engage in communications;

* User capabilities: determination of the media and media parameters to be used;

* Session setup: "ringing", establishment of session parameters at both called and calling party;

* Session management: including transfer and termination of sessions, modifying session parameters, and invoking services.

Bear in mind that SIP is a component that can be used with other IETF protocols to build a complete multimedia architecture. It primarily includes the Real-time Transport Protocol (RTP) utilized for transporting real-time data and providing network feedback.

There is also the Real-Time streaming protocol (RTSP) that is used for controlling delivery of streaming media. The Media Gateway Control Protocol (MEGACO) responsible for controlling gateways to the Public Switched Telephone Network (PSTN). Lastly is the Session Description Protocol (SDP ) useful for describing multimedia sessions.

For that reason, SIP should be used in conjunction with other protocols in order to provide complete services to the users. However, the basic functionality and operation of SIP does not depend on any of these protocols.

In order to guarantee your security SIP also provides a suite of safety services to include some are the denial-of-service prevention, authentication (verification can either be this way or vice versa-both user to user and proxy to user), integrity protection, and encryption and privacy services.

When ask about its function.. It works with both IPv4 and IPv6 for starters. For Internet telephony sessions, SIP provides many convincing points:

Callers and callees are identified by SIP addresses. When making a SIP call, a caller first locates the appropriate server and then sends a SIP request. The most common SIP operation is the invitation. Instead of directly reaching the intended callee, a SIP request may be redirected or may trigger a chain of new SIP requests by proxies.

Users can register their location(s) with SIP servers. SIP addresses (URL) can be embedded in Web pages and therefore can be integrated as part of powerful implementations such as Click to talk.

Actually, there are still other features that it present but as of now that is the most noted and usable for the client to understand.

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