What Type of VoIP Codec Do I Need?

By Neil Zawacki
Updated: September 20, 2011

What Type of VoIP Codec Do I Need?

A VoIP codec is a complex algorithm that translates an analog voice signal into a digital version. They tend to have different bandwidth and computational requirements, and some of them require royalty payments. It’s therefore a good idea to take the time to look at different VoIP codecs and determine which one you need.

Some of the most popular VoIP codecs include:

Broadvoice Codec – This VoIP codec is 16Kbps narrowband and 32Kbps wideband. It has extremely low latency (the algorithmic buffering delay is just 5ms). It is also royalty free and available through an open source license.

ITU G.711 – G.711 is an old VoIP codec – it was standardized back in 1988. It uses a high bit rate (64 Kbps), but offers superior voice quality since it doesn’t compress the signal at all. It does requires a great deal of CPU power in order to properly function, however.

ITU G.726 – G.726 is mainly used by international trunks to reduce the amount of bandwidth required. The standard bitrate is 32 Kbps, but can be set to 16, 24, or 40 Kbps. It also uses the Adaptive Differential Pulse Code Modulation scheme.

ITU G.729 – This is an ITU standard codec that offer VoIP at the low rate of 8Kbps. It requires quite a bit of CPU power, however, so some VoIP phone (particularly those from Linksys and Cisco) will be only able to handle a single channel at a time. It also requires licensing in to use the codec.

GSM – The GSM codec has a great deal of popularity outside of the United States. It divides the voice signal into blocks of 20ms, and then passes them through a speech codec that can handle 13kbps. It should be noted that the newer GSM codecs are heavily patented.

iLBC – This VoIP codec was designed by Global IP sound is available under a free (but restricted) license. It has a payload bit rate of 13.33 Kbps and an encoding frame length of 30 ms. iLBC is also highly resistance to packet loss, which in turn helps to maintain the quality of phone calls.

Speex – The Speex codec supports 2.15 to 44.2 Kbps. It is highly flexible, but consumes a great deal of CPU power – quite a bit more than the ITU G.729 or GSM. Speex is completely free, however, and works well with most internet applications.

AMR Codec – This VoIP codec supports toll quality speech at 7.4 Kbps (or higher) and can encode narrowband signals at variable rates (4.75 to 12.2 Kbps). It is also a required codec for several 2.5G/3G wireless networks, including WDMA, EDGE, and GPRS.
 

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