SIP protocol

What is?

SIP protocol

SIP protocol


Essentially, SIP (Session Initiation Protocol) is the protocol that dictates how we can make and receive calls, as well as receive and send information.

SIP is so closely related to VoIP (Voice Over Internet Protocol) that VoIP is sometimes confused with SIP. However, VoIP is not a protocol per se, but a term that is used to define the transport of voice information over an Internet protocol.

Despite the main functions and intentions of SIP, it does not encode the information of a telephone call nor carry the data. Instead, the role of SIP is simple: it starts and ends communication sessions. This goes for any type of application, from voice calls between two individuals to multi-party video conferencing. SIP is a protocol independent of the type of traffic that is not necessarily voice, video or even data - it can be anything.

Simply put, the main function of SIP is to create calls, group videoconferences, and other forms of interactive communication, as well as terminate these sessions once terminated by sending messages through endpoints that are called "SIP Addresses". These addresses can be linked using two methods:

    Physical SIP client, such as an IP phone;
    Software client (softphone) that must be installed on a computer or mobile device (tablet or smartphone).

Basically, communication not only does not involve a single protocol but also relies on a set of different protocols. These protocols are then built on top of each other across layers. This process is also known as a protocol stack. There are a multitude of different models for how protocols build on each other. The most common and well-known model is the OSI (Open Systems Interconnection) Reference Model. OSI has the following layers in order:

    Application (SIP, RTP, RTCP, etc.)
    Presentations
    Session
    transport
    Network
    Data link
    Physicist

As far as SIP is concerned, there are two types of layers involved.

 

Transport layer

The transport layer controls the speed, order and reliability of data exchange. This also includes data that is transmitted by voice calls. In order for data to be transported over the Internet, it has to be broken up into packets. This layer also regulates and manipulates the routes and ordering process of data packets as they are transmitted. The transport layer has 2 protocols:

     Transmission Control Protocol (TCP) - This is a program that is designed to transmit data packets while also retransmitting any data that might have been lost in the transmission process.
     User Datagram Protocol (UDP) - Unlike TCP, this program does not retransmit data that may be lost during the transmission process. However, it transmits the data through packets.

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Application layer

As the name suggests, this specifies the different interfaces and protocols for a more specific flow over the network connection that has been established. In this case, SIP is an application layer protocol. Essentially, SIP is the foundation of modern interactive communication devices, such as voice calls, video calls and other forms of communication, over the Internet.

Here are some other functions that SIP is capable of:

    Location and User Registration - Telephone line terminals will notify SIP proxies of their current locations.
    User Availability - SIP is used to find out if a person is available to "answer" a call so that a session can be initiated.
    User Capabilities - SIP is used by various endpoints as a means to negotiate media capabilities. An example would be if both parties agreed on a voice codec with bidirectional support.
    Session Management - SIP is also used to transfer calls, end calls, and even change call parameters while the session is still running. An example would be adding a third party to the conference call.

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