On the right is a live demo of the MyVoIP test available in MyConnection Server^®.

Go ahead and click the start button to see what results you get for your connection...

Upstream jitter - Jitter is a key measure of VoIP quality. Jitter refers to 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. VoIP works by sending multi-media data as a stream of packets from the 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 VoIP 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 picture and sound. 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 connection (discarded). As a result, the recomposed data no longer reflects exactly what was sent, and depending of the extent of the delay may result in unrecognizable display.

The upstream jitter refers to the variation in time between packets sent and packets arriving from the client to the server.

Upstream packet loss - Packet loss plays a key role in the quality of VoIP call quality, as high packet loss causes some of the voice data not to arrive to the destination.

The upstream packet loss metric refers to the percentage of voice data packets that are discarded by the jitter buffer, or dropped by network routers/switches due to high congestion, from the client to the server.

Downstream jitter - Downstream jitter refers to the variation in time between packets sent and packets arriving from the server to the client.

Downstream packet loss - Downstream packet loss refers to the percentage of voice data packets that are discarded by the jitter buffer, or dropped by network routers/switches due to high congestion, from the server to the client.

Mean Opinion Score (MOS) - The Mean Opinion Score, or MOS, is a numeric measure used for VoIP, indicating 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 MyVoIP 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.

Upstream packet order - Upstream packet order reports the percentage of video packets that arrived in the correct order from the client to the server.

Upstream packet discards - Upstream packet loss refers to the percentage of voice data packets that are discarded by the jitter buffer from the client to the server. A number greater than 0 indicates that packets arrived outside the time window (jitter buffer) and got lost from the connection (discarded). As a result, the recomposed data no longer reflects exactly what was sent, and depending of the extent of the delay may result in unrecognizable sound.