ZyXEL Communications IES-1248-51A, Руководство пользователя

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Chapter 45 VoIP
IES-1248-51V User’s Guide
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• G.711 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 very good sound quality but requires 64kbps of bandwidth.
• G.723.1 uses Low-Delay Code-Excited Linear Prediction (LD-CELP) to code
audio in 30-millisecond frames. The standard supports two bitrates, 6.3 kbps
and 5.3 kbps.
3
G.723.1 provides toll-quality sound and requires very little
bandwidth.
• G.726 is an Adaptive Differential Pulse Code Modulation (ADPCM) waveform
codec that uses a lower bitrate than standard PCM conversion.
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.
However, DPCM produces a high quality signal (high signal-to-noise ratio or
SNR) for high difference signals (where the actual signal is very different from
what was predicted) but a poor quality signal (low SNR) for low difference
signals (where the actual signal is very similar to what was predicted). This is
because the level of quantization noise is the same at all signal levels. Adaptive
DPCM solves this problem by adapting the difference signal’s level of
quantization according to the audio signal’s difference level. A low difference
signal is given a higher quantization level, increasing its signal-to-noise ratio.
This provides a similar sound quality at all signal levels.
G.726 operates at 16, 24, 32 or 40 kbps.
• G.729 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.
45.1.7 PSTN Call Setup Signaling
PSTNs (Public Switched Telephone Networks) use DTMF or pulse dialing to set up
telephone calls.
Dual-Tone Multi-Frequency (DTMF) signaling uses pairs of frequencies (one lower
frequency and one higher frequency) to set up calls. It is also known as Touch
3. At the time of writing, the IES-1248-51V supports the 5.3 kbps bitrate only.