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Chapter 29 Extension Management
ISG50 User’s Guide
475
29.6 Authority Group Technical Reference
This section contains technical background information about the Authority Group screens.
Voice Codecs
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 ISG50
Video Codecs
Video codecs are used by video phones to compress the amount of information sent between two
devices. Video codecs encode video signals into digital signals and decode the digital signals back
Table 163 Voice Codecs Supported
CODEC DESCRIPTION
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.726) working at 48 ~ 64 Kbps, with an audio sample rate of
16 KHz. G.722 provides excellent sound quality.
G.723.1 This is an ITU (International Telecommunication Union) standard for voice coding. The G.723.1
codec 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.