A Practical Guide to VoIP CODECs
Making voice calls over the Internet through digital networks such as Voice over IP (VoIP) requires audio codecs to encode voice data into digital data and then decode it into analog again. Short for encoder-decoder, codecs are necessary for compressing data for faster transmission and improving the calling experience.
However, as there are several VoIP codecs for audio, video, fax and text, choosing the best audio VoIP codec for your network is a matter of preference. Do you want audio codecs that provide great sounding audio or are you okay with reducing the impact on your VoIP network?
Do you have to settle for limited codec features so they can work on your legacy devices or applications or are you looking for an audio codec that provides you with cutting edge technology.
In the end, everything just boils down to what you need. This is why in our practical guide to VoIP codecs, we will give you a short comparison of each VoIP codec and then explain where it is best to use them.
VoIP Audio Codec Comparison
|G.711||64||Known for delivering precise speech transmission. Requires minimal processor requirements. You will need at least 128 kbps for two-way communication. Although it one of the oldest codecs in the market, it requires high bandwidth for best performance. Such a high bandwidth requirement made this codec obsolete for the internet, but because it offers an excellent MOS of 4.2, many companies still want to use it in their LAN connections.|
|G.722||48/56/64||It has the ability to adapt to changing bandwidths and compressions whenever there is network congestion. Being a robust codec, it captures a range of frequencies, even twice as large as G.711. As a result, it offers better quality and clarity that isn’t far behind PSTN.|
|G.723.1||5.3/6.3||Balances high-quality audio with high compression. This allows users to implement it in both dial-ups, as well as, low bandwidth environments, owing to its low bit rate. The only downside is that it needs higher processor power.|
|G.726||16/24/32/40||Improvised version of G.723 and G.721 but different from G.723.1|
|G.729||8||Being error-tolerant, it offers excellent bandwidth utilization. Although it is an improved version of most similarly named audio codecs, it isn’t used by most providers. This is because the codec isn’t free and you have to buy the license for it. End users must indirectly pay for this license whenever they buy hardware such as gateways or phone sets for implementing it.|
|OPUS codec||6kbps to 510||OPUS is one of the most advanced versions of audio codecs available in the market. It maintains high audio quality using lossless compression of audio signals. However, since the codec is absolutely free, many providers and network administrators prefer it for their VoIP setup.
How to Choose the Best Audio Codec for Your VoIP Network?
Ensuring High Audio Quality
Many companies want their audio quality to be the same as what they used to have in their older digital telephones and legacy systems. That might not be that bad as audio quality in those telephones systems was good.
In such situations, it’s best to use G.711, as it gives users the exact same voice quality as digital telephones. Many VoIP network experts refer to this codec for its “toll quality audio.” G.711 can also be used for supporting touch-tones (DTMF) and even fax, if you set it up correctly.
Reducing Burden on Internet Bandwidth
For some users, ensuring the best voice quality becomes secondary when it comes to saving costs from IP telephony. This usually happens when companies have to process a high volume of audio calls and they must increase the number of calls sent to their network. In these situations, it’s best to choose codecs that reduce the amount of bandwidth consumed by the network.
For this, compression-enabled codecs such as G.729, G.726, G.728, iLBC, and G.723 are the best. Among them, G.729 is most commonly used. G.726 is also good for compressing audio calls. It requires less bandwidth compared to G.711, but has a significantly worse audio quality. Another problem with these codecs is that most carriers don’t currently support them.
This is why if your internet bandwidth is stretched thin, G.729 can be the best VoIP audio codec for your phone system. G.729 performs optimally even when there is low bandwidth.
Using an algorithm for extreme compression, it offers a bitrate of 8 Kbps. Over time, it will consistently achieve a bandwidth requirement of 24 Kbps, on average. However, G.729 is not free, and you must pay the license fee to use it. Moreover, compressing the bitrate too much can affect the call quality significantly.
Moreover, you should know that these codecs are not for fax signals or touch tones. To integrate your network system with these components, you must configure an out-of-band mechanism for transporting them similar to the one outlined in RFC 2833.
Offering the Best of Both Worlds
Although G.711 offers users toll quality audio, it only captures voice within a very narrow frequency range. Compared to that audio codecs with the latest wideband offer a more realistic sound. This happens despite these codecs having bandwidth consumption less than G.711.
G.722 is the most common codec used in this class. Although the codec had been relatively unknown a few years ago, many high-profile IP telephony vendors support it. What’s great about G.722 is that it sounds great over an IP handset and also plays well in hands-free mode.
It uses a bitrate ranging from 48 Kbps to 64 Kbps and ultimately requires an average bandwidth of 80 Kbps. While G.722 needs relatively high bandwidth consumption, the quality of audio is exceptional. This mostly happens because it uses lossless compression for converting audio signals.
Seeking for Advanced Features and Better Performance
Lastly, we will discuss codecs for users who are seeking the solutions that support the latest and most-advanced features. They need codecs that produce higher quality audio by using cutting-edge compression techniques.
Here, the Opus codec stands out as one of the best. It produces amazing sound quality and is ideal for both high fidelity audio and speech. While the codec algorithm was originally designed for WebRTC, many people use it outside of browser-based telephony, mostly in SIP telephones.
OPUS also utilizes lossless compression of audio signals for maintaining high call quality. Additionally, being completely free, it’s surprising that it offers so many advanced features such as echo cancel for enhancing call experience.
Moreover, the codec is known for its flexibility and can function in all kinds of conditions. It provides users with a robust and variable bitrate encoding range that starts from 6kbps to 510 kbps. This means, users can apply the Opus codec regardless of the bandwidth restrictions on their connection.
Although currently, the Opus codec isn’t as widespread as G.711 had been in its prime, the range of features it offers is too great of a deal to ignore. In the future, we can expect the Opus codec to become the preferred audio codec.
In this article, we analyzed different types of VoIP audio codecs, both in detail and at the high-level. If you want the best audio codec for your VoIP network, you should prioritize what you want in your solution.
By looking at the frequency ranges, bit rates, and various other parameters and matching them with your company needs, it will be easier for you to choose the VoIP audio codec best fit for your organization.