What is jitter on a phone call?

Voice is chopped into small packets, one every twenty milliseconds or so. They should arrive at that same pace.

Networks rarely cooperate. Packets take different routes, queue behind other traffic, and land in bursts followed by gaps.

Equipment compensates with a buffer that holds arriving audio briefly and plays it out evenly. That buffer costs delay, which is why a heavily buffered call feels slow even when it sounds clear.

Past roughly 30 milliseconds of variation, the buffer cannot keep up and you get the chopped, robotic sound everyone recognizes.

Your phone sends a packet of audio every 20 milliseconds, as regular as a metronome. On a quiet network they arrive the same way: 20 milliseconds apart.

Imagine someone in the next office starts uploading 300 photos to a shared drive. The router's outbound queue fills with their data, and your voice packets wait in line behind it. Arrival gaps that were 20, 21, 19 become 20, 58, 3, 2, 37, 20. Nothing was lost, but the beat is gone.

The receiving end holds a small buffer, often around 40 milliseconds, and plays audio out of it at an even pace. A 58-millisecond gap drains that buffer, and for a moment there's nothing to play. The listener hears a stutter, then a burst as three packets land together. Stretch that across a two-minute photo upload and the person on your call sounds like they're talking through a fan.

People mix up jitter and latency because both are measured in milliseconds. Latency is how long audio takes to cross the network. Jitter is how much that time varies from one packet to the next.

A call with a steady 120-millisecond delay sounds normal. Telecom guidance has long held that one-way delay up to about 150 milliseconds goes unnoticed in conversation. Now take a call whose delay swings between 30 and 90 milliseconds. It has lower latency on average and sounds worse, because the buffer can't predict what's coming.

Most IP desk phones will show you both numbers. During a live call, open the phone's status or call statistics screen and you'll usually find jitter, packet loss and the codec listed together.

Check a few calls at different times of day. Jitter of 5 to 10 milliseconds all day with spikes to 60 at 4:30 every afternoon points at something scheduled on your network. If it's high on every call at every hour, suspect the connection itself. That's one to raise with your internet provider.

Audio reaching GreetKeeper crosses your carrier's network before it reaches us, so the same conditions that make a call sound bad to a person make it transcribe worse.

Where a transcript is full of gaps, the line is worth checking before the assistant is. Run a call from a wired connection and compare.

The transcript makes the problem visible instead of leaving you guessing, because you can see exactly which stretch of the call turned to noise.