Configuring Security, Quality, and Network Features
Ensuring Voice Quality
Cisco SPA 500 Series and WIP310 IP Phone Administration Guide
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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
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