Jitter is the variation in time between packets sent and packets arriving caused by network difficulties such as route changes, congestion, packet loss, traffic regulators etc., and plays a major role in the quality of a VoIP call. VoIP works by sending voice data as a stream of packets from source to destination. These packets can take a varying amount of time to reach the destination and invariably do not arrive in the order in which they were sent.

For a VoIP telephone call to work well the packets sent from the source must arrive within a certain time window (or ‘buffer’) in order for the receiving end to reassemble the packets in the correct order and reproduce the spoken words. When there is excessive jitter the time delay is too long (high latency) and packets arrive outside the time window and get lost from the call (discarded). As a result, the recomposed sound no longer reflects exactly what was sent, and depending of the extent of the delay may not be understandable by the recipient.

Packet loss plays a key role in the quality of VoIP connections, as high packet loss causes some of the voice data not to arrive to the recipient. Packet loss occurs when voice packets are discarded by the jitter buffer (see previous FAQ), or dropped by network routers/switches due to high congestion. This speed test measures the percentage of packet loss and reports the associated level of sound quality.

The MOS, or Mean Opinion Score, is a numeric measure of the sound quality at the receiving end of a communication circuit. Although the score is subjective it provides a widely-used method to rate the quality of voice communication in a simple way that meaningful to end users. The score is normally between 1 and 5 with 5 being the best.

The MOS value is reported in the Speed Server summary tab once a connection test completes, a VoIP simulation that drops below 3.5 is considered poor quality, a measure of 4.2-4.5 is considered good quality.