ZyXEL Communications MSC1000G Series, Руководство пользователя

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Chapter 16 VoIP
Management Switch Card 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.
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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.
16.1.7 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 Tone®. Each of the keys on a DTMF
telephone corresponds to a different pair of frequencies.
Pulse dialing sends a series of clicks to the local phone office in order to dial numbers.
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16.1.7.1 VoIP VLAN
Virtual Local Area Network (VLAN) allows a physical network to be partitioned into multiple logical
networks. Only stations within the same VLAN can communicate with each other.
4. At the time of writing, the VOP supports the 5.3 kbps bitrate only.
5. The VOP supports DTMF at the time of writing.