Results
Waiting
Ping test
Measure the latency, jitter and packet loss on the connection you are using now. The test sends a stream of small packets to our server at the rate a voice call does, and tells you whether any loss is on the way up or the way down.
What the numbers mean
- Ping (latency) is the time for a packet to reach our server and come back. Under 30 ms is excellent for a UK connection, under 60 ms is fine for calls and games, and satellite links are usually far higher.
- Jitter is how much that time varies from one packet to the next. Calls and video need it low, ideally under 20 ms, because the receiving end has to buffer to smooth it out.
- Packet loss is the share of packets that never arrive. Anything above 1% is noticeable on a call. Because our server counts what reaches it, the test can tell loss on your upload from loss on your download, which points to where to look.
- Call quality is an estimate on the MOS scale from 1 to 5, using a simplified version of the ITU E-model: 4.0 or more is good, below 3.6 callers will notice.
The packets travel over WebRTC, the same technology as browser-based calls, without retransmission, so a lost packet stays lost just as it would on a call. If your network blocks it, the test falls back to timing ordinary web requests, which measures latency and jitter but cannot see packet loss. For packet-size problems that make large transfers hang, run the MTU and CGNAT test.
Connectivity built for calls
We run our own network, broadband and hosted voice, engineered for low latency and no loss.