A codec is an integral part of how a VoIP telephone system works. The codec you use will affect how well your telephone system works for your business. However, many people don’t know what a codec is, despite how important one is to the functionality of a VoIP system.
What is a VoIP codec? A codec is a piece of software or hardware that encodes and decodes signals. In the context of VoIP, the codec compresses and converts the audio signals into a digital form for transmission. The codec then converts the signal again so the other user hears the audio signals.
In this blog post, I’ll talk through how codecs actually work. I’ll also explain which codecs can be used with VoIP and how you can decide which to use. Read on to find out all you need to know about codecs in the context of VoIP.
How Does A Codec Work?
So we know that codecs convert audio signals for transmission while compressing them. However, when you are on the phone, it sounds like a constant stream of conversation; it doesn’t sound like the audio signals have gone through compression and conversion. So how does a VoIP codec work?
The codec samples the audio signal thousands of times per second which is why, when you’re having a conversation on the phone, the conversation seems to be a constant stream without any interruptions.
Different codecs sample the audio a different number of times. For example, a G.729A samples the audio signal at 8,000 times per second. While that may sound like a lot, some codecs sample the audio signal at a higher rate. For example, the G.711 codec which samples audio at a rate of 64,000 times per second.
So the codec samples the audio signal thousands of times per second. Then, each sample of audio is then digitised and compressed ahead of transmission. Once the digital signals reach other other user’s telephone system, the digitised data is decompressed and converted into audio signals again. This entire process is so fast that you can’t tell it’s happening. To the human ear, the gaps in the audio are so small that they’re unnoticeable.
Why Is Choosing The Right Codec So Important?
In most cases, the end user won’t have any part in deciding which codec is used for their telephone system. The VoIP provider chooses which codec is used and that’s that. However, making sure that your system uses the right codec is so important. Here’s why.
If your VoIP telephone system is using a codec that isn’t suited to your requirements, you may find that the voice quality is very poor on your system. Why? It usually comes down to bandwidth. Different codecs require different bandwidth. The codecs that sample at a higher rate require more bandwidth.
The codec your system uses is only really a problem if you don’t have a good connection with decent bandwidth. Each VoIP call requires a certain amount of dedicated bandwidth. Without dedicated bandwidth, you will experience issues such as calls dropping and poor voice quality.
If you’re using one of the more bandwidth-hungry codecs, you will struggle if you don’t have the dedicated bandwidth that you need. The table below shows the bandwidth that is needed by each codec.
Bandwidth Required By Popular VoIP Codecs
|Silk||6 - 40Kbps|
As you can see, some codecs require much more bandwidth than others. The G.711 requires over double the bandwidth needed by the G.729 codec. This could pose an issue if your system is using the G.711 codec but you can’t guarantee you have the bandwidth you need.
The bandwidth required above doubles with each concurrent call. For example, if you are using the G.711 and need to make 5 concurrent calls, you will need at least 500Kbps upstream and downstream. With all the other processes consuming bandwidth in your office, you may struggle with a poor connection such as ADSL.
If you find that voice quality is low and calls dropping is a regular occurrence, it could be because you don’t have enough dedicated bandwidth. This could be resolved by changing the codec that your system uses or upgrading to a connection with more bandwidth.
Which Codec Should You Use?
If you don’t have much bandwidth available, you might be wondering which codec you should use. Or even if you have a good connection, you might wonder which codec is the best for your system.
When deciding between codecs, there are several important considerations. One of the most important considerations is how much bandwidth you have available. If you don’t have enough bandwidth available, the decision is essentially already made for you; you will have to use one of the codecs that utilises heavy compression to save bandwidth. On the other hand, if you have plenty of bandwidth, you can go for a bandwidth-hungry codec that guarantees higher voice quality on calls.
If top voice quality is a must for your business, you must use a codec that has a high sample rate and doesn’t compress the data as much. For example, the G.711. If clear voice calls are an absolute must, you can’t justify using one of the codecs that significantly compresses the audio signals.
Another important consideration is whether you’re willing to pay for a licensed codec or not. While there are plenty of free codecs available to use, there are licensed codecs available that improve your call quality. Some licensed codecs are developed specifically with certain systems and devices in mind. For example, the G.722.1, licensed from Polycom. This codec is ideal for use with Polycom devices and will offer a better call quality.
So which codec is right for your business? How can you choose between all the codecs that are available? It’s hard to say without knowing your business, but let’s have a look at some popular codecs. Hopefully, this brief overview will help you to choose which codec is right for you.
Popular VoIP Codecs
There are a wide selection of VoIP codecs to choose from. That’s why deciding on the right one for your business can be tough. Here are some of the most popular VoIP codecs and why they might be right for your business.
The G.711 is a widely used codec. It is often believed to offer voice quality on par with a good PSTN call. This codec is ideal if you have dedicated bandwidth. It needs at least 100Kbps downstream and upstream, but this is a minimum. Ideally, you would have more bandwidth available to ensure call quality.
The G.711 is still one of the best codecs around, despite it also being one of the oldest. You can’t go wrong with G.711 when considering voice quality providing you have the bandwidth needed.
The G.729 is another widely used VoIP codec. It uses much less bandwidth than the G.711, but this often results in a lower voice quality. However, the voice quality is still on par with a good mobile phone call.
The G.729 is a licensed codec, so it isn’t free. You won’t really notice the cost of paying for this codec, because the cost is usually bundled in with the cost of the hardware that you purchase.
SILK is another popular codec often used by applications that have voice communication capabilities. SILK was actually developed by Skype, but it is now available as open-source freeware. Since it doesn’t cost anything for other businesses to use it as part of their apps, many other apps and services use SILK.
You’ve probably used apps and services that have made use of the SILK codec at some point in time. For example, SILK was used by the Steam game platform and videoconferencing application Zoom. The SILK codec is actually the basis for Opus, which is used as a codec in applications such as WhatsApp and Sony’s PlayStation 4.
These are just some of the popular codecs that are available. This list is by no means conclusive. However, it gives you a quick overview of some of the codecs that are available for your business to use. In my personal experience, the G.711 and G.729 are the most widely used codecs in VoIP telephone systems for business. So really, these are probably the best codecs for you to choose between.