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Cisco spa509g ip Manual

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Getting Started
Ensuring Voice Quality
Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide 33
1
• 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 SPA 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 by using a comfort
noise generator (CNG).
• Packet loss—Audio packets are transported by UDP. Packets might be lost
or contain errors that can lead to audio sample drop-outs and distortions
and lower the perceived voice quality. The Cisco SPA IP phones apply an
error concealment algorithm to alleviate the effect of packet loss.
• Network jitter—The IP network can induce varying delays of received
packets. The RTP receiver in Cisco SPA IP phones keep a reserve of
samples to absorb the network jitter, instead of playing out all the samples
as soon as they arrive. This reserve is known as a jitter buffer. The bigger the
jitter buffer, the more jitter it can absorb, but this also introduces bigger
delay.
- Jitter buffer size should be kept to a relatively small size whenever
possible. If jitter buffer size is too small, many late packets might be
considered lost and thus lower the voice quality. Cisco SPA IP phones
dynamically adjust the size of the jitter buffer according to the network
conditions that exist during a call.
- The minimum jitter buffer size is 30 ms or 10 ms plus the current RTP
frame size, whichever is larger, for all jitter level settings. However, the
starting jitter buffer size value is larger for higher jitter levels. This setting
controls the rate at which the jitter buffer size is adjusted to reach the
minimum.
- Jitter Buffer Adjustment—Controls how the jitter buffer should be
adjusted.
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