In yesterday’s article we looked at the importance of understanding your overall network performance and existing network demand through a site survey before implementing VoIP. We also stressed the importance of continual monitoring of VoIP on your network to develop an understanding of normal network performance.
In this article we’ll be looking at two of the key factors that adversely affect VoIP applications – delay and jitter.
Understanding Delay and Jitter
Real-time communications, like VoIP, are sensitive to delay and variation in packet arrival times, and require a steady, dependable stream of packets to provide reasonable playback quality. Delay is the time it takes a packet to reach its destination. Whenever packets travel across a network, some delay is inevitable. Because of this, the delay budget for reasonable two-way conversations is about 150 milliseconds each way. When a delay exceeds the budget, callers may be confused about who should speak or listen.
When packets arrive early, late, or out of sequence, this is referred to as jitter. Jitter is the result of variation having occurred in the time between packets transmitted and packets received. Excessive jitter causes the users to experience quality degradation during a call.
A way to compensate for excessive jitter is to increase the size of the jitter buffer. The jitter buffer on the phone is responsible for reassembling packet streams. When there is an issue with packets, such as arriving out of sequence or too late, the buffer will try and adjust to compensate or fill in with white comfort noise if necessary. Adjusting the buffer can minimize jitter problems, but it can also introduce other issues such as latency, which can cause conversations to be clipped. For example, adjusting the buffer to 300 milliseconds would make normal conversation difficult. Increasing buffer size can help, but only to a point. Just as critical as adjustments is the ability to measure and understand jitter. Tracking jitter measurements provide hard facts that an administrator can use to improve call quality.
[Stephen Brown and Charles Thompson of Network Instruments]
[tags]voip,delay,site survey,benchmark,network performance,voip metric,cdr,call detail record[/tags]