User`s guide

Chapter 11 The Service Configuration Screens
User’s Guide
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Figure 63 STUN
11.2.1.2 Outbound Proxy
Your VoIP service provider may host a SIP outbound proxy server to handle all of the WiMAX
Modem’s VoIP traffic. This allows the WiMAX Modem 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 WiMAX Modem to keep it from re-translating the IP address (since this is already
handled by the outbound proxy server).
11.2.1.3 Voice Coding
A codec (coder/decoder) codes analog voice signals into digital signals and decodes the digital
signals back into voice signals. The WiMAX Modem supports the following codecs.
• 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 excellent sound
quality but requires 64kbps of bandwidth.
• G. 7 2 3 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 strength. 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.723 provides high quality sound and
requires 20 or 40 kbps.
• G. 7 2 9 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.
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