SIP Trunking Explained

If you're new to SIP trunking, fear not. We've put together a quick guide explaining the basics of SIP trunking to get you started.

Andrew Bentley

26/11/2018

SIP trunking is one of those acronyms that you don’t really need to know what it stands for, just what it actually is. But, it will bug you if you don’t know. Session Initial Protocol. You don’t need to remember it, so don’t worry. What you will want to remember is what SIP trunking is and how you can use it to benefit your business.

What is SIP trunking?

Rather misleading, there are no physical trunks involved here. You may be used to ISDN trunks or PSTN lines. Essentially, a SIP trunk is a replacement for these. Picture removing the copper infrastructure and replacing it with an internet version. SIP is method – a protocol – of VoIP communication that allows users to make and receive calls over the internet. There you have it, SIP trunking.

Of course, there’s a whole lot more to it than that. We can get really technical and go into different types of SIP architecture but it won’t help you for now. So, let’s cover the basics.

How SIP works

Instead of calling from phone to phone over a copper telephone line, SIP uses addresses. You send a SIP message from one address to another. That message might contain a voice call or even a video call. Just like when you are sending an email to your colleague in another office, SIP is a data exchange online. The job of SIP itself is to set up the call session and terminate it when it’s over.

Sending information between addresses means you need some endpoints, somewhere for the information to be delivered to. Typical SIP endpoints include a physical desk phone, a softphone – like one you’d use in your Unified Comms stack – or a specialist SIP device that only works off SIP exchanges. This should make the transfer easier to picture in your mind.

Codecs

Before the information can be transferred over the internet, it must be encoded so that audio signals (speech) can be transferred as data. SIP is typically seen in G.711 and G729 codecs.

  • G.711 codec: commonly used for uncompressed voice. You will get better quality with G7.11 but that’s because it uses a little more bandwidth.
  • G.729 codec: commonly used for compressed voice. Quality is less than G.711 but doesn’t use up as much bandwidth; ideal for networks under pressure.

G.711 is always the preference when using SIP for business purposes. Where possible, you want the best audio quality. It is important to note that the reliability and performance of your SIP trunks is fully dependent on the quality of your network.

Network Impact

To get started on SIP, all you essentially need is some internet and a SIP trunk. SIP calls can be made over any form of internet – unless it has proactively been blocked.

The public internet is not as secure as a private internet connection like MPLS or SD-WAN.  However, if you ensure that your connection follows security best practice, you don’t need to worry.  As long as your underlying connectivity is secure end-to-end, the technology will follow suit. If your network is up to scratch, your telephony will also be secure. On the flip side, if your network drops regularly or is prone to security breaches, you’ll need to assess these issues before implementing SIP.

It can be a scary decision, moving from trusty ISDN to SIP for the first time. You’ve probably got loads of questions whirring around in your mind around security and reliability. Rightly so. This is your business and maybe even your personal reputation at stake.

But, don’t worry, we’ve been there and had those thoughts too. That’s why we’ve put together the most common SIP myths… and debunked them. Read our guide to the most common SIP myths here. 

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