A free, comprehensive VoIP quality test. See how your network performs now!
Complete this short form to begin your VoIP network test. This test will open a socket-connection to your browser and pass simulated VoIP traffic to your computer to measure the quality and performance of your Internet connection between your network and our servers.
Note: This requires a web browser with Java 1.5 (Version 5.0) or greater. It does not work with the Chrome browser (version 42 and up). We recommend using Firefox to run the VoIP Test.
See explanations of the scores below.
The MOS score is a measure from 1 (being the worst) to 5 (being the best). MOS is quite subjective, as it originated from the phone companies and used human input from related quality tests. A VoIP simulation that drops below 3.5 is considered poor quality, a measure of 4.2-4.5 is considered good quality.
Packet loss occurs when one or more packets of data traveling across a network fail to reach their intended destination. Here it is measured as a percentage. Call quality deteriorates when this value exceeds 5%.
In VoIP, jitter is the variation in packet transit delays (from serialization, propagation, and buffering effects on the stream). Jitter is significant to real-time applictions because the receiver must dimension its jitter buffer based on maximum jitter, which adds delays for all packets and causes eventual loss when jitter values exceed buffer capacity.
Measured in bits transferred per second, bandwidth is the maximum throughput of a network. For VoIP, enough bandwidth must be supplied to allow for the transmission of voice data in real time. We recommend 100 kbps per simultaneous call. A good rule of thumb is to consider 1/10 of your organization is on the phone at any given time, or 100 kbps for every 10 employees.
Measured in milliseconds, latency measures the time it takes to move a VoIP packet from one point to another.