Latency and Jitter

The top network problems that cause poor voice quality are latency and jitter, and it is important to understand the two parameters to improve performance.

Network latency, also known as transmission delay is the time between when a packet was transmitted to when it arrives at its intended destination. The two forms of latency are:

One-way latency—The difference in time between when the source sends the packet, and when the destination receives it.
Round-trip latency—The difference in time between the when the source sends the packet and receives a response to that packet sent by the destination.

Jitter is the variation of latency from packet to packet, which means that the packets must be received at the destination in the accurate order and accurate interval. If the latency is more than 108 milliseconds, it causes choppy audio due to lack of reordering. Hence, those packets are automatically discarded. Vocera employs a 108 milliseconds jitter buffer to help ensure uninterrupted audio on the badge. If a packet arrives out of sequence or is transmitted with a slight delay, the buffer allows for continuous audio if the delay does not exceed the buffer size.

Jitter occurs due to fluctuation in network delay, network overload, and interference. If preventive measures are not taken, there is packet loss, and the quality of the call is low.

Voice applications are sensitive to packet delays, latency, or jitter that affect data transmission. For example, depending upon the sensitivity of the listener, a delay of 150 milliseconds may cause interruption in a stream of spoken words.