Cisco SPA525G Administration Guide - Page 124

6 122 Configuring Security, Quality, and Network Features Ensuring Voice Quality 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 packets decreases the bandwidth utilized, but it also increases delay and may affect voice quality. Cisco SPA and Wireless IP Phone Administration Guide