Latency, jitter, and packet loss

Three numbers decide how a call actually sounds, and none of them is simply "internet speed." Here is what each one measures, and why they matter together.

Latency: how long the trip takes

Latency is the time it takes a single packet of your voice to travel from your mouth to the other person's ear. Anything under about 150 milliseconds one way is barely noticeable. Push past 300 and people start talking over each other, because the natural pause where you would expect a reply has stretched too long.

Jitter: how inconsistent the trip is

Jitter is not about speed, it is about consistency. If packet one takes 40 milliseconds, packet two takes 41, and packet three suddenly takes 120, that unevenness is jitter. Audio expects a steady drip of packets to sound smooth, and the shared, best-effort nature of the internet means that steady drip is never actually guaranteed.

Packet loss: the trip that never finished

Sometimes a packet simply never arrives at all, dropped somewhere along a congested or unreliable path. A small amount is usually invisible, filled in or smoothed over. Enough of it in a row is what produces the short robotic gaps and missing syllables people describe as a call breaking up.

Why they are usually discussed together

All three metrics are symptoms of the same underlying reality: your voice is riding on shared, best-effort packets instead of a dedicated line. A network can be fast on average and still sound bad if it is inconsistent, which is exactly why jitter, not raw speed, is often the real culprit behind a choppy call.