Factors Affecting Voice Quality - Cisco SPA942 - - IP Phone Administration Manual

Configuring Security, Quality, and Network Features
Ensuring Voice Quality
Cisco SPA and Wireless IP Phone Administration Guide
For more information about bandwidth calculation, refer to the following websites:
http://www.erlang.com/calculator/lipb/
http://www.packetizer.com/voip/diagnostics/bandcalc.html

Factors Affecting Voice Quality

The following factors contribute to voice quality:
• Audio compression algorithm—Speech signals are sampled, quantized,
and compressed before they are packetized and transmitted to the other
end. For IP Telephony, speech signals are usually sampled at 8000 samples
per second with 12–16 bits per sample. The compression algorithm plays a
large role in determining the voice quality of the reconstructed speech
signal at the other end. Cisco IP phones support the most popular audio
compression algorithms for IP Telephony: G.711 a-law and u-law, G.726,
G.729a, and G.723. 1 . (G.723. 1 is not supported on the 525G or WIP310.)
• The encoder and decoder pair in a compression algorithm is known as a
codec. The compression ratio of a codec is expressed in terms of the bit
rate of the compressed speech. The lower the bit rate, the smaller the
bandwidth required to transmit the audio packets. Although voice quality is
usually lower with a lower bit rate, it is usually higher as the complexity of
the codec gets higher at the same bit rate.
• Silence suppression—Cisco IP phones apply silence suppression so that
silence packets are not sent to the other end to conserve more transmission
bandwidth. IP bandwidth is used only when someone is speaking. Voice
activity detection (VAD) with silence suppression is a means of increasing
the number of calls supported by the network by reducing the required
bidirectional bandwidth for a single call. A noise level measurement is sent
periodically during silence suppressed intervals so that the other end can
generate artificial comfort noise (comfort noise generator, or CNG).
• Packet loss—Audio packets are transported by UDP, which does not
guarantee the delivery of the packets. Packets may be lost or contain errors
that can lead to audio sample drop-outs and distortions and lower the
perceived voice quality. SPA9X2 phones apply an error concealment
algorithm to alleviate the effect of packet loss.
• Network jitter—The IP network can induce varying delay of received
packets. The RTP receiver in Cisco IP phones keeps a reserve of samples
to absorb the network jitter, instead of playing out all the samples as soon
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