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Each time you make a call or send a fax over a VoIP system, it uses something called a codec to transmit the sound from your phone to the person you’re calling. Many of these codecs produce high definition sound for top quality audio, video and faxes. These are just some of the many audio codecs that best VoIP systems use.

G.711

Originally developed in 1972, this codec is famous for delivering precise speech while not asking much of your computer’s processor. For two-way communication, the codec needs a connection of at least 128 kbps. While it may not be suited to long-distance communication online, it is still useful for LAN communications.

G.723.1

This codec is high-compression and high-quality, a rare combination in an audio codec. It can also work with limited internet connections, even dial-up. In exchange, the codec needs more processor power than average to function. Even so, having the ability to potentially have HD audio at dial-up speeds could be worth it.

G.729

The most advanced of the G family of codecs, it makes efficient use of the bandwidth available to it, making it good for both large and small businesses. It is a licensed codec, so not all VoIP providers will have this codec available. It is also tolerant of user errors and won’t immediately malfunction if the user makes a mistake.

GSM

If you suspect this is the same GSM that cell phones use, you are absolutely correct. The GSM codec is freely available, so many telecommunications providers offer it to their customers. With its high compression ratio, it delivers sound quickly, but may not be as clear as a few of the other codecs on this list.

iLBC

This open source codec is free to use, so many VoIP providers also have this codec as an option. It handles data packet loss quite well, meaning it won’t drop calls easily if the connections aren’t quite even in speed. Some VoIP apps, like SessionTalk, use this particular codec.

Speex

Billing itself as “a free codec for free speech,” Speex uses a variable bit rate to minimize how much bandwidth it uses. While it can be a bit of a CPU hog, it is customizable, a unique feature among audio codecs. Many VoIP apps make use of Speex, including Samsung Voice Recorder and Polaris Office for LG.

Vorbis

Also known as Ogg Vorbis because of its close association with the audio format, Vorbis originally began development in 1993. It is another freely available audio codec and is the audio format of choice for many video games due to its high quality. One potential issue is the fact that Windows computers do not natively recognize Vorbis audio, so you may need additional software to use it. This will likely be provided by your VoIP service of choice, if it uses Vorbis.

voip codecs

SILK

Developed by the makes of Skype, SILK is freely available and widespread in its usage. It has a low delay between sending and receiving audio signals, making it a high-performance codec. SILK is commonly used on the Steam computer game client and with the in-game voice chat for the popular computer game “Team Fortress 2.”

Opus

An offshoot of the SILK codec, Opus is a low-latency codec that doesn’t demand much from your computer’s processor. Where most codecs have about 100 milliseconds of delay, Opus has only 5 milliseconds of delay on average. The low latency diminishes quality slightly as a result, but to most people it’s barely noticeable, if at all.

AMR

AMR is the standard codec in WDMA, GPRS and EDGE cell phone networks, so the codec already sees a lot of widespread usage. In fact, the codec is required if you use a 2.5G or 3G network. It was initially developed as part of the GSM codec, but is also available on its own.

The Caveat

One important thing to note with all of these audio codecs floating around is one crucial fact: for there to be a successful HD VoIP call, most of the people involved must use the same audio codec. Otherwise, the signal will have to be “translated” into the recipient’s codec, which can cause latency issues or – even worse – cause everyone to drop down to the lowest common denominator, which is often G.711. This in turn takes more processor resources and a potential loss in quality. When arranging a VoIP conference call, be sure to find out which codec everyone uses before beginning.