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Chapter 6 SIP
6.1.7.4 Outbound Proxy
Your VoIP service provider may host a SIP outbound proxy server to handle all of the P-2024's
VoIP traffic. This allows the P-2024 to work with any type of NAT router and eliminates the
need for STUN or a SIP ALG. Turn off a SIP ALG on a NAT router in front of the P-2024 to
keep it from retranslating the IP address (since this is already handled by the outbound proxy
server).
6.1.8 Voice Coding
A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital
signals back into voice signals. The P-2024 supports the following codecs.
• 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 excellent sound quality but requires 64kbps of bandwidth.
• G.726 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.
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.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.
6.1.9 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.
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P-2024 User's Guide
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Troubleshooting
