Chapter 11 SIP Account Setup
V500 Series User’s Guide
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• G. 7 11 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. 7 2 2 is an Adaptive Differential Pulse Code Modulation (ADPCM) waveform codec.
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.722 samples audio at 16 kHz; twice the traditional rate of 8 kHz. G.722 provides
excellent quality audio and requires 48 to 64 kbps.
• G.722.2 is similar to G.722, but with a lower compression rate that can vary according to
the amount of available bandwidth. When there is plenty of bandwidth, the compression
ratio decreases, and when there is network congestion the compression ratio increases.
G.722.2 is also known as Adaptive Multi Rate - WideBand (AMR-WB).
• G.723.1 is a Code Excited Linear Prediction (CELP) codec that compresses voice audio in
30 ms frames. 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. 7 2 6 is an ADPCM waveform codec that uses a lower bitrate than standard PCM
conversion. 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.
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.
