ZyXEL ISG50-PSTN User Guide - Page 475
Authority Group Technical Reference
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Chapter 29 Extension Management 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 Table 163 Voice Codecs Supported CODEC G.711 DESCRIPTION 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. G.722 G.723.1 G.726 G.729 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 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. 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. 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. 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. 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 ISG50 User's Guide 475