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Asterisk Tutorial 34 - Introducing SIP

Asterisk Tutorial 34 - Introducing SIP

Introducing SIP — What is SIP, How Does it Work?

It is time for a new topic in our Introducing Asterisk tutorial series, which means we are finally starting on Session Initiation Protocol (SIP). Over the previous 33 tutorials, we have done a lot in terms of configuring our Asterisk system and we have used the SIP protocol a lot, making internal calls etc, but we have never explained it, which is why it is now time for our “Introducing SIP” tutorial.

Why Now?

Firstly, a large number of our followers have asked us to and secondly, the aim of a functioning telephone system is to make and receive both internal and external calls, meaning that at some point we will need to integrate VoIP Trunks / Gateways into our system. Therefore, the answer really is that after all we have done in configuring our system, it is now time to make it “operation ready” and in order to do that, we will need SIP.

What is SIP?

In our Introducing SIP tutorial, we start by taking a look at the fundamentals of SIP. Understanding the protocol, what it is and how it works is essential for later on when we start integrating SIP providers, Gateways etc as understanding the concept of what is happening will be invaluable when it comes to debugging and error analysis should your integration not work out quite as smoothly as you had hoped.

So what is SIP? Simply put, Session Initiation Protocol is a communications protocol for starting (signalling) and managing multimedia (voice, video, data) sessions between IP end points and is most commonly used in Voice over IP telephony systems. Going into more detail, the SIP protocol consists of 3 main components:

  • Firstly, there is Session Initiation itself which as the name suggests is the starting (initiation) protocol for the communication session.
  • The next aspect of SIP is the Session Description Protocol (SDP) which provides a description of the session, for example which codecs are to be used, which port should be used and so on.

The final component of the SIP protocol is the payload which refers to the protocol for delivering content (voice, video, data), i.e. real-time transport protocol (RTP).

Mathias’ Top Tip

It is important to understand the session initiation and description take one route between your IP end points to start a communication session, however the payload can be transported via a completely different route as agreed upon in the session description.

This is important to understand in order to avoid perhaps the most common mistake when configuring and using SIP as your communication protocol, which is namely that the signalling is working correctly but the payload is not being received by your end points or is only being received by one of the end points (one way audio).


As you may have noticed during the tutorial, a lot of acronyms were being thrown around, so here is a break down of what they mean:

  • SIP — Session Initiation Protocol (see above)
  • SDP — Session Description Protocol (see above)
  • RTP / sRTP — (secure)Real-time Transport Protocol (see payload above)
  • UDP / TCP — User Datagram Protocol / Transmission Control Protocol. Both are transmission protocols for communicating data but are used for different types of data
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Final Word - We Upgrade Business Communications

A well-configured business phone system that oozes useful tools, delivers excellent audio quality and intuitive call flows will leave callers with a highly professional impression. The configuration options and application availability and how well they are utilized will have a decisive impact on the level of profressionalism when it comes to call management and therefore how your customers view your company. The switch to a pascom VoIP phone system solution provides the opportunity to not only upgrade your internal collaboration but also to redefine telephony in your company.

If you would like more information regarding pascom and our Asterisk based Software PBX, please visit our website or give the pascom team a call on +49 991 29691 0 to discuss your requirements and get started within minutes using our free hosted asterisk business VoIP phone system edition.

Until next time — Happy VoIPing!

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