Configuring Security, Quality, and Network Features
Ensuring Voice Quality
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide 134
5
The following table approximates the bandwidth budget for each side of the
conversation (in each direction) using different codecs and number of calls. This
table is based on the following assumptions:
• Bandwidth calculated with no silence suppression
• 20 millisecond of payload per RTP packet
NOTE The use of silence suppression can reduce the average bandwidth budget by 30%
or more.
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,
Codec Est.
Bandwidth
Budget
2 Calls 4 Calls 6 Calls 8 Calls
G.711 110 kbps 220 kbps 440 kbps 660 kbps 880 kbps
G.722 110 kbps 220 kbps 440 kbps 660 kbps 880 kbps
G.726-40 87 kbps 174 kbps 348 kbps 522 kbps 696 kbps
G.726-32 79 kbps 158 kbps 316 kbps 474 kbps 632 kbps
G.726-24 71 kbps 142 kbps 284 kbps 426 kbps 568 kbps
G.726-16 63 kbps 126 kbps 252 kbps 378 kbps 504 kbps
G.729 55 kbps 110 kbps 220 kbps 330 kbps 440 kbps
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