MySpeed PC VoIP Advanced Edition simulates VoIP traffic over your Internet connection, measures key diagnostics including Jitter and Packet Loss, and provides a route analysis of your connection.

Test your connection for jitter and packet loss, see the corresponding level of VoIP sound quality

See the network route with packet loss and latency detail for each hop

Get the MOS Score VoIP performance rating for your connection

Automatically test your connection speeds at regular intervals, view your results over time

See the number of VoIP lines supported by your Internet connection

View graphical reports of the data transfers for detailed analysis to help identify the cause of poor performance

Measure the consistency of your download speeds, a critical measure for time-sensitive applications such as VoIP

   

For information on hosting MySpeed VoIP on your website see MySpeed Server VoIP.

Network Route Testing
Route analysis is helpful in diagnosing where connectivity problems occur by identifying each network device, or hop, that data travels through from the source to the destination.

MySpeed PC VoIP identifies each network device of your connection and shows you exactly where packet loss and latency occur, which can adversely affect VoIP sound quality. By identifying the exact router or switch where a connectivity problem occurs significantly reduces the time for technical staff to isolate and resolve the problem.

Reports the Mean Opinion Score (MOS)
MOS is a numeric measure of the sound quality of a communication connection. Although the score is subjective it provides a widely-used method to rate the quality of voice communication.

The score is normally between 1 and 5 with 5 being the best. In MySpeed, a VoIP simulation that drops below 3.5 is considered poor quality, a measure of 4.2-4.5 is considered good quality.

Measures Jitter and Packet Loss
The primary measures of VoIP connection quality are Jitter and Packet Loss. MyVoIPSpeed simulates voice traffic on your Internet connection, determines the levels of Jitter and Packet Loss, and rates the corresponding voice sound quality.

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, or ‘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, or dropped by network routers/switches due to high congestion. MyVoIPSpeed measures the percentage of packet loss and reports the associated level of sound quality. For more accurate results Packet Loss is measured in one direction only, which is normally not possible using traditional methods such as ping, where routing can affect the reply.

NEW! VoIP Test Graph
The VoIP graph shows the variance of UDP jitter during the simulation test. This variance must be kept to a minimum otherwise call quality will be degraded. The packet loss distribution is shown in red, high packet loss will result in broken sound during calls.