Configuring Security, Quality, and Network Features
Ensuring Voice Quality
Cisco Small Business SPA 300 Series, SPA 500 Series, and WIP310 IP Phone Administration Guide 124
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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 may be
considered as lost and thus lowers the voice quality. Cisco 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 milliseconds or (10 milliseconds +
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. Select the appropriate setting: low,
medium, high, very high, or extremely high. Defaults to high.
Jitter Buffer Adjustment—Controls how the jitter buffer should be
adjusted. Select the appropriate setting: up and down, up only, down
only, or disable. Defaults to up and down.
• Echo—Impedance mismatch between the telephone and the IP Telephony
gateway phone port can lead to near-end echo. Cisco IP phones have a
near-end echo canceller with at least 8 ms tail length to compensate for
impedance match. Cisco IP phones implement an echo suppressor with
comfort noise generator (CNG) so that any residual echo is not noticeable.
• Hardware noise—Certain levels of noise can be coupled into the
conversational audio signals because of the hardware design. The source
can be ambient noise or 60 Hz noise from the power adaptor. The Cisco
hardware design minimizes noise coupling.
• End-to-end delay—End-to-end delay does not affect voice quality directly
but is an important factor in determining whether IP phone subscribers can
interact normally in a conversation. A reasonable delay should be about 50–
100 ms. End-to-end delay larger than 300 ms is unacceptable to most
callers. Cisco IP phones support end-to-end delays well within acceptable
thresholds.
• Adjustable Audio Frames Per Packet—Allows you to set the number of
audio frames contained in one RTP packet. Packets can be adjusted to
contain from 1–10 audio frames. Increasing the number of frames
decreases the bandwidth utilized, but it also increases delay and can affect
voice quality.
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