Specifications

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Cisco 1751 Router Software Configuration Guide
OL-1070-01
Analog Compared with Digital
Analog transmission is not particularly robust or efficient at recovering from line noise. Because analog
signals degrade over distance, they need to be periodically amplified; this amplification boosts both the
voice signal and ambient line noise, resulting in degradation of the quality of the transmitted sound.
In response to the limitations of analog transmission, the telephony network migrated to digital
transmission using pulse code modulation (PCM) or adaptive differential PCM (ADPCM). In both
cases, analog sound is converted into digital form by sampling the analog sound 8000 times per second
and converting each sample into a numeric code.
CODECs
Pulse code modulation (PCM) and adaptive differential PCM (ADPCM) are examples of “waveform”
CODEC techniques. Waveform CODECs are compression techniques that exploit the redundant
characteristics of the waveform itself. In addition to waveform CODECs, there are source CODECs that
compress speech by sending only simplified parametric information about voice transmission; these
CODECs require less bandwidth. Source CODECs include linear predictive coding (LPC), code-excited
linear prediction (CELP) and multipulse-multilevel quantization (MP-MLQ).
Coding techniques for telephony and voice packet are standardized by the ITU-T in its G-series
recommendations. The Cisco 1751 router uses the following coding standards:
• G.711—Describes the 64-kbps PCM voice coding technique. In G.711, encoded voice is already in
the correct format for digital voice delivery in the PSTN or through PBXs.
• G.729—Describes CELP compression where voice is coded into 8-kbps streams. There are two
variations of this standard (G.729 and G.729 Annex A) that differ mainly in computational
complexity; both provide speech quality similar to 32-kbps ADPCM.
• G.723—Describes a compression technique that can be used for compressing speech or audio
signal components at very low bit rate as part of the H.324 family of standards. This CODEC has
two bit rates associated with it: 5.3 kbps and 6.3 kbps. The higher bit rate is based on ML-MLQ
technology and provides a somewhat higher quality of sound. The lower bit rate is based on CELP
and provides system designers with additional flexibility.
• G.726—Describes ADPCM coding at 40, 32, 24, and 16 kbps. ADPCM-encoded voice can be
interchanged between packet voice, PSTN, and PBX networks if the PBX networks are configured
to support ADPCM.
In Cisco’s voice implementations, compression schemes are configured using the codec command.
Mean Opinion Score
Each CODEC provides a certain quality of speech. The quality of transmitted speech is a subjective
response of the listener. A common benchmark used to determine the quality of sound produced by
specific CODECs is the mean opinion score (MOS). With MOS, a wide range of listeners judge the
quality of a voice sample (corresponding to a particular CODEC) on a scale of 1 (bad) to 5 (excellent).
The scores are averaged to provide the MOS for that sample. Table 1-1 shows the relationship between
CODECs and MOS scores.