VoIP quality test

Sends traffic shaped like a voice call and tells you whether your connection will carry one cleanly.

Jitter –ms
Packet loss –%
Ping –ms

Each tick is a packet. Taller ticks took longer. Red marks are packets that never came back.

Test length

Sends 1500 small UDP packets to your nearest Cloudflare edge and back.

Updated

A test shaped like a call

Almost every internet phone system, from office VoIP handsets to WhatsApp, Discord and Teams voice, sends audio the same way: a small UDP packet every 20 milliseconds, each carrying 20 ms of sound. Nothing is resent if it goes missing, because a replacement would arrive after the listener needed it. This VoIP quality test sends exactly that pattern, 50 packets a second of 160 bytes each, which is the size of a standard G.711 voice frame, and measures what happens to them.

Three numbers decide call quality. Packet loss removes slices of sound, which the listener hears as clipped words or robotic gaps. Jitter makes packets arrive unevenly, forcing the app to hold audio back in a buffer before playing it. Latency delays everything, and once the one-way delay passes about 150 ms people start talking over each other. The verdict list under the result rates your connection for voice specifically, and names whichever of the three is holding it back.

VoIP quality thresholds

Modern codecs such as Opus hide a little loss well, so a call at 1% loss may sound fine while an older desk phone on G.711 sounds noticeably choppy. The thresholds above are the conservative ones, which is what matters if you are checking a line before rolling out phones to an office.

Targets used by voice equipment makers and the ITU-T G.114 guidance
MeasureClear callsDegradedPoor
Packet lossUnder 1%1 to 3%Over 3%
JitterUnder 30 ms30 to 50 msOver 50 ms
Ping (round trip)Under 150 ms150 to 300 msOver 300 ms

Before you blame the phone system

  • Test from the device and location with the bad calls, at the time they happen. A clean test from the server room does not help the person on wifi in the back office.
  • Test on wifi and on a cable. Voice is very sensitive to wifi jitter, and the fix for that is not in the phone system at all.
  • Test while the office is busy. A line that is fine at 8am and congested at 11am needs quality-of-service rules on the router, or more upload capacity.
  • Keep the copied result. Voice providers and ISPs both ask for loss and jitter figures before they will investigate.

For video meetings rather than voice, see the Zoom and Teams connection test.

Frequently asked questions

What is a good jitter for VoIP?

Under 30 ms, and ideally under 10 ms. Above 30 ms, most voice apps have to add enough buffering that the delay is noticeable, and above 50 ms they start dropping late packets.

How much packet loss is acceptable for VoIP?

Under 1%. Between 1% and 3% calls are usable but choppy on older codecs. Above 3% calls break up badly.

Why do my VoIP calls sound robotic?

Robotic or metallic audio is the sound of a codec filling in missing or late packets. It points to packet loss or high jitter, most often from wifi or a congested upload.