SIP (Session Initiation Protocol)

SIP stands for Session Initiation Protocol and is a worldwide established communication standard used in VoIP telephony. SIP controls how an internet call is established and how voice data is transferred during that call, as well as the termination of the connection.

A session is simply an exchange or transfer of data between two or more participants. Therefore SIP is the method in which an internet phone connects with another in order for a phone call to take place.

There are actually two phases that occur when you make an internet call via VoIP. The first phase is commonly known as the call setup and includes all of the information needed to get two telephones talking to each other. Once the call has been setup and the IP phones are talking to each other, the phones enter a data transfer phase and actually start transmitting data between each other. In the case of a phone call the data transfer is audio.

The audio is encrypted at one end and then decrypted at the other. The way in which it is encrypted is referred to as a Codec. The audio codecs must be the same on each phone in order for the audio to be heard. SIP is used by IP phones to negotiate media capabilities, mutually supported codecs and session attributes used by the calling and called party. SIP even tells the VoIP phone it should be ringing and is used to transfer calls, terminate calls and change call parameters in the middle of a session, for example adding another person to the conversation and creating a conference call.

In general we normally think of phones as having telephone numbers.  In SIP, an internet phone has a SIP URI, This is a type of user name or identifier and looks a little like an email address but is specifically for VoIP phones. The URI is unique will identify your phone on the network. Because phones generally have numeric keypads, an internet phone will translate the number or extension you dial into a SIP URI. For example if you want to dial 01234567890, SIP will translate this into a SIP URI, something similar to 01234567890@voip-int.co.uk.

SIP is used to establish Real Time Communications (RTC) such as audio and video conferencing as well as instant messaging, making it extremely versatile and appealing to businesses who are looking at the best way to manage their Unified Communications. It is also, therefore, very cost effective.

More often than not internet calls are made with the help of a SIP proxy. In the case of COM-TEL Telecom it is our PBX (Private Branch Exchange) server. As a COM-TEL Telecom customer, when you plug your VoIP phone in it will register on our server and effectively notify the server of its location. Therefore when someone calls your phone, SIP knows where it is and connects the call.

As previously mentioned the Session Initiation Protocol is used to establish the communication session. The actual transferring of the data is done by the TCP (Transmission Control Protocol) or the UDP (User Datagramm Protocol). The Session Initiation Protocol is text based and very similar to the Hypertext Transfer Protocol (HTTP) and therefore makes it very easy to integrate into electronic devices such as computers, laptops, tablets and mobile phones.

There is a lot of SIP based software available on the market today for most operating system platforms, such as Windows, Linux, Mac OS and Android. The software can be downloaded and installed and effectively turns your device into an internet ready telephone negating the need to buy a separate VoIP phone. Software like this is also referred to as Softphone software.