Datasheet

Configuring Voice Ports
Configuring Digital Voice Ports
VC-92
Cisco IOS Voice, Video, and Fax Configuration Guide
Voice Quality Tuning Commands
The commands in this section configure parameters to improve voice quality. Common voice quality
issues include the following:
• Delay in Voice Networks
• Jitter Adjustment
• Echo Adjustment
• Voice Level Adjustment
Delay in Voice Networks
Delay is the time it takes for voice packets to travel between two endpoints. Excessive delay can cause
quality problems with real-time traffic such as voice. However, because of the speed of network links
and the processing power of intermediate devices, some delay is expected.
When listening to speech, the human ear normally accepts up to about 150 ms of delay without noticing
delays. The ITU G.114 standard recommends no more than 150 ms of one-way delay for a normal voice
conversation. Once the delay exceeds 150 ms, a conversation is more like a “walkie-talkie” conversation
in which one person must wait for the other to stop speaking before beginning to talk.
You can measure delay fairly easily by using ping tests at various times of the day with different network
traffic loads. If network delay is excessive, it must be reduced for adequate voice quality.
Several different types of delay combine to make up the total end-to-end delay associated with voice
calls:
• Propagation delay—Amount of time it takes the data to physically travel over the media.
• Handling delay—Amount of time it takes to process data by adding headers, taking samples,
forming packets, etc.
• Queuing delay—Amount of time lost due to congestion.
• Variable delay or jitter—Amount of time that causes the conversation to break and become
unintelligible. Jitter is described in detail below.
Propagation, handling, and queuing delay are not addressed by voice-port commands and fall outside the
scope of this chapter.
Jitter Adjustment
Delay can cause unnatural starting and stopping of conversations, but variable-length delays (also known
as jitter) can cause a conversation to break and become unintelligible. Jitter is not usually a problem with
PSTN calls because the bandwidth of calls is fixed and each call has a dedicated circuit for the duration
of the call. However, in VoIP networks, data traffic might be bursty, and jitter from the packet network
can become an issue. Especially during times of network congestion, packets from the same conversation
can arrive at different interpacket intervals, disrupting the steady, even delivery needed for voice calls.
Cisco voice gateways have built-in jitter buffering to compensate for a certain amount of jitter; the
playout-delay command can be used to adjust the jitter buffer.
Normally, the defaults in effect are sufficient for most networks. However, a small playout delay from
the jitter buffer can cause lost packets and choppy audio, and a large playout delay can cause
unacceptably high overall end-to-end delay.