ZyXEL Communications IES-1248-51 Betriebsanweisung PDF herunterladen (Seite 322)

Chapter 45 VoIP

IES-1248-51V User’s Guide

322

• 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.

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.

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