The SIP is a continuation of Session Initiation Protocol, ie protocol call setup that run on application layer. SIP text-based and very flexible and supports all services , and almost all current telecommunications devices already support SIP. This article describes what and how the SIP .

A. General.
Session Initiation Protocol (SIP) , namely the protocol issued by IETF (International Engineering Task Force) . In the IP and traditional telephone , always clearly distinguishable two- stage dialing voice. The first stage is "Call Setup" which covers all the details need to be two telephone devices can communicate . The next stage is the " data transfer " where call setup already formed. In the VoIP , SIP is call setup protocol which operates at the application layer.

SIP is very flexible and designed in general for setup real- time multimedia sessions between group participants. For example , in addition to telephone call simple , SIP can also be used for video conferencing set- ups and audio or instant messaging.

SIP is not only to handle call setup, but also has other functions the ability to support VoIP services .

Function	Information
Registration & identification of user location	End points ( IP Phones ) location to perform notification SIP proxies. SIP also determine the end point which will participate in the call .
User Availability	SIP is used by the end point to determine whether incoming calls answered or not.
User capabilities	SIP is used the end point negotiating with network capabilities , such as the use voice codecs.
Set -up Session	SIP notify to the the end point that the end point must ringingP. it is associated with features such as conference.
Session Management	SIP is used for transfer calls, decided calls, and changing parameters in the call center session.

One point in favor of SIP is the use a text-based model in request / response on HTTP . This will facilitate debug because messages easy to do reconstruction and easily viewed and analyzed ( for network manager) . Contrast with an H.323 , SIP is a protocol that is very simple , yet has enough features powerful for PBX telephone environment . SIP can run over IPv4 and IPv6 and can use TCP or UDP . Implementation in general use IPv4 and UDP .

B. Set-up Call on SIP
Although two SIP devices can talk directly, but generally uses the inter mediation system as a SIP Proxy. SIP proxy only participates in doing the set-up , once call set -up formed , the device can send voice traffic directly without involving the proxy again. SIP Proxies very helpful in sharing task and simplify the implementation . For example SIP phones want to make calls to other telephone with extension 4094 . Task SIP proxy is knowing what is really 4094 , if he telephones , extensionSome or the other telephone .

In telephony systemIn general, telephone having a numeric address . In the SIP URI in the SIP user can have (Uniform Resource Identifier) as an identity used by the SIP protocol. Because in general have a telephone terminal numeric keypads, then the terminal is responsible for performing the translation of numerical.

The diagram below illustrates a SIP dialog involving two participants and SIP Proxy Server. In this case message in SIP has been simplified to make it easier to see the flow traffic.



Figure 1 . Set -up Flow of SIP Signaling
 
Although the diagram shows that the proxies do not play again after call set -up formed, however this does not apply to all calls . be retained as a media proxy intermediaries to accommodate some mid -call features, such as service conference and accounting.

Another common SIP operation is Registration, this is related to how to identify the location of the user . registration capability is especially useful in environments where end terminal has no IP address static .

In SIP registration server can become one with proxy server or it could be a separate system . Registration user is also not limited in one location but may be from various locations , and proxy server user's responsibility to decide which one will " ring "when the call came. With the SIP user can choose one that will " ring "or all users.

Because SIP is used as call control, then features such as voice mail and auto response not part of the SIP protocol itself. SIP is very easily readable format to facilitate the implementation and debugging.

C. Migrating to SIP
While many companies want to adopt SIP , they are still considering many things , especially from the side cost savings and features to be developed next . In general there are three stages of the implementation of the SIP can be explained as follows :

1. Connecting to a SIP Proxy existing PBX
2. Migrating Users and terminals to SIP Proxy
3. Migration from connection to PSTN existing PBX to SIP

Step 1: The first step is to connect the SIP proxy with existing PBX systems as gateway to PSTN . To connect from the IP Network to PSTN , SIP proxy will routing whole calls from VoIP to PSTN calls . Links can connect using an ISDN PRI or T1 . First thing to do is setup a SIP proxy (SIP call manager) , after set-up went well , then stay connected to existing PBX with links ISDN , E1 or T1 .

Step 2 : The next stage is the migration existing users and services to SIP. Migrating all handset using either the SIP gateway and replace with handset which native SIP . Approach by combining the two is usually a good option . Next is a set-up SIP application server to accommodate the required features such as user voice mail, multi -party conferencing, IVR, follow me fine me and many more features are supported SIP.

Step 3 : Phase three is to do routing calls to the outside through an Internet connection . Many operators start to sell SIP based by offering services long distance with a cheaper price . That need to be anticipated is nature of the Internet network which unpredictable ( such as jitter and error rate) and of course security. As an initial stage of the SIP can be used as an alternative to make calls when outside the office or teleworkers.

D. The SIP terminology

SIP Proxy : Working to provide call routing and also as SIP gateway to existing PBX. SIP gateway can also be done by other devices .

SIP Gateway : also called analog telephone adapter that convert from analog to SIP calls . This is a solution where the device existing end point SIP can be used in the system.

SIP Phones : By using the end point which native SIP will benefit from the flexibility which the end point can be connected with a data network , anywhere. SIP phone also provides many features not provided by analog phone like multiple line Appearances, hold buttons, conferencing optionEtc. . Cost the end point still a constraint implementation terminal SIP Phone.

E. Conclusion

SIP is very flexible and is designed not only to general real-time voice but also for set-up a real- time multimedia sessions between participant group like conference. SIP has capabilities such as user registration and identification , user availability , user capabilities , set-up session and management session. SIP format is a text-based making it very easy to read and simplify the implementation and debugging. In the implementation stage is the approach commonly used in stages starting from the SIP proxy to connect the existing system , user migration and integration.