ZyXEL Communications X6004 Manual Do Utilizador

Chapter 14 Authority Group

X6004 User’s Guide

A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital 
signals back into voice signals. The following table describes the codecs supported on the 
X6004 

Table 34   Voice Codecs Supported

G.711

This is a Pulse Code Modulation (PCM) waveform codec. PCM measures analog signal 

amplitudes at regular time intervals (sampling) and converts them into digital bits 

(quantization). Quantization "reads" the analog signal and then "writes" it to the nearest 

digital value. For this reason, a digital sample is usually slightly different from its analog 

original (this difference is known as "quantization noise").  
G.711 provides excellent sound quality but requires 64kbps of bandwidth.
There are two main algorithms defined in the G.711 standard, the µ-law algorithm (used 

in North America & Japan) and a-law algorithm (used in Europe and the rest of the 

world). 

G.722

G.722 is an ADPCM codec (see G.723) working at 48 ~ 64 Kbps, with an audio sample 

rate of 16 KHz. G.722 provides excellent sound quality.

Note: The X6004 supports G.722 pass-through, meaning that devices 

communicating via the X6004 must support this codec. 

G.722 AMR-

WB 

(also 

referred to 

G.722.2)

G.722.2 is similar to G.722, but with a lower compression rate that can vary according to 

the amount of available bandwidth. When there is plenty of bandwidth, the compression 

ratio decreases, and when there is network congestion the compression ratio increases. 

This is also known as Adaptive Multi Rate - WideBand (AMR-WB).

Note: The X6004 supports G.722.2 pass-through, meaning that devices 

communicating via the X6004 must support this codec. 

G.723.1

This is an ITU (International Telecommunication Union) standard for voice coding. The 

G.723.1 compresses voice audio in 30 ms frames.  The G.723.1 operates at two 

bitrates: 6.3 kbps when sampling at 24 bytes or 5.3 kbps when sampling at 20 bytes per 

30 ms frame. 

G.726

This is an Adaptive Differential Pulse Code Modulation (ADPCM) waveform codec that 

uses a lower bitrate than standard PCM conversion. G.726 operates at 16, 24, 32 or 40 

kbps. Differential (or Delta) PCM is similar to PCM, but encodes the audio signal based 

on the difference between one sample and a prediction based on previous samples, 

rather than encoding the sample’s actual quantized value. Many thousands of samples 

are taken each second, and the differences between consecutive samples are usually 

quite small, so this saves space and reduces the bandwidth necessary. 

G.729

This is an Analysis-by-Synthesis (AbS) hybrid waveform codec. It uses a filter based on 

information about how the human vocal tract produces sounds. The codec analyzes the 

incoming voice signal and attempts to synthesize it using its list of voice elements. It 

tests the synthesized signal against the original and, if it is acceptable, transmits details 

of the voice elements it used to make the synthesis. Because the codec at the receiving 

end has the same list, it can exactly recreate the synthesized audio signal.
G.729 provides good sound quality and reduces the required bandwidth to 8kbps.