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VOIP Codecs

 VOIP Codecs

When a SIP-to-SIP call is initiated, the SIP provider will negotiate a common protocol based on the available codecs on each phone, as well as which codecs the VOIP provider supports. Most codecs used in VOIP are defined by the ITU and have a G.xxx nomenclature. G.711 and G.722 are free to use, but G.729 is licensed by Sipro and requires a license to use.

G.711a/711u (48, 56, 64 kbps) (Free) is the default codec for VOIP (as well as fixed-line networks). Every VOIP phone should support the G.711 codec, and usually as well as G.729.

G.729/729a (8kbps) (Proprietary)  is a very common low-bit-rate codec, and can be used in cases where bandwidth is restricted, or high errors occur. G.729a is less processor intensive than G.729.

G.722 (48, 56, 64 kbps) (Free) is a wideband (or "High-Defenition") audio codec. It samples audio at a rate of 16 kHz, double the rate of standard voice codecs, and splits the signal into two subbands that are then encoded with ADPCM modulation to double the audio content, while still staying within the standard 64kbps channel. This results in a much clearer audio quality compared to the standard G.711. However to be effective, as with any codec, both ends need to have phones supporting the G.722 codec, and the VOIP provider needs to support it as well. The G.722 codec is gaining in popularity, but is not yet a standard codec in all VOIP phones.

Some other common codecs and their uses:

G.723.1 (Proprietary) (5.3, 6.3 Kbps) offers high quality and high compression (30kbps connection including overhead makes it a good choice for users with dial-up 56k modems)

G.726 - (Free) (16, 24, 32, 40 kbps) Low processing power.

GSM Full-Rate (13 kbps) (Free) Good quality, commonly used in GSM systems.

iLBC (13.3, 15 kbps) (Free) Handles errors well.

For a more detailed calculation of bandwidth required by codec including overhead, read Alcatel's VOIP Design Guide: Alcatel VOIP Design Guide

To hear a sample of a narrowband codec (13kbps) and a wideband (G.722.2) codec:

Narrowband        Wideband

Bandwidth and Latency Requirements

To calculate the actual bandwidth needed for a VOIP call including overhead can vary widely. However roughly speaking 100kbps bandwidth is needed for the G.711 (64kbps) codec,  and a total of 30kbps is needed for the G.729a (8kbps) codec.

The ITU-T G.114 recommendation limits the maximum acceptable round trip delay time (RTT) to 300 ms between the two VoIP gateways (or 150 ms one-way delay).

Download and read ITU-T G.114 "One-way transmission time": ITU-T G.114

To test the latency between two gateways, simply open a command line (in Windows: Start-> Run -> CMD) and type "ping" followed by the IP address of the far-end gateway:

The Round Trip Time (RTT) above is 80ms average, and is more than sufficient to handle VOIP.