Network & QoS
Jitter
By Myroslav Orshak · operates licensed carrier infrastructure · Updated September 2026
Jitter is the variation in arrival timing of the data packets carrying your voice — when packets arrive unevenly instead of in a steady stream, calls sound warbly, garbled, or robotic even though nothing was technically lost.
How it works
Your voice leaves as a metronome-steady stream of packets, one every twenty milliseconds. The network doesn't promise to keep them steady — some packets take a slower route, wait behind a big download, or linger in a congested Wi-Fi link, so they arrive bunched and gapped. Play them back as they arrive and speech smears; that smearing is jitter.
Phones fight back with a jitter buffer: a tiny holding area that collects packets for a few hundredths of a second and releases them evenly, trading a little added delay for smoothness. The buffer is why modest jitter is inaudible — and why big jitter is doubly bad: packets that arrive later than the buffer can wait for get thrown away, which means severe jitter literally becomes packet loss.
Why it matters
Jitter is the early-warning light of call quality. It usually rises before loss does. The congestion that will drop packets at 2 p.m. often shows up as climbing jitter at 1:45. And it is almost always a local problem you can fix: busy Wi-Fi, an overloaded connection, or a router juggling traffic with no priority rules. If your calls sound robotic but a speed test looks fine, jitter is nearly always the cause. The fixes are the same cheap steps as for packet loss: wire the phones that matter, quiet the network, and turn on the router's voice-priority (QoS) setting.
Under 30 ms of jitter, calls sound smooth — jitter buffers (typically holding 20–60 ms) absorb it invisibly. From 30–50 ms, audio starts to warble. Past 50 ms, buffers overflow and jitter converts into audible packet loss.
Related terms
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Measure your jitter in 10 seconds →Frequently asked questions
What's the difference between jitter, latency, and packet loss?
Latency is how long audio takes to arrive (too much and you talk over each other). Jitter is how uneven that arrival is (calls warble). Packet loss is audio that never arrives at all (words vanish). They're related — the congestion that raises jitter eventually causes loss — but each has its own threshold: under 150 ms latency, under 30 ms jitter, under 1% loss.
What causes high jitter?
The same suspects as packet loss, one step earlier: Wi-Fi contention, a connection saturated by downloads or backups, and routers processing traffic in whatever order it arrives. Wireless links are the biggest offender — radio retransmissions make timing inherently uneven — which is why the first diagnostic is always comparing a wired test to a Wi-Fi one.
How do I fix jitter?
In order: run our test wired — if jitter collapses, fix the Wi-Fi, not the internet. Quiet the network during the re-test to spot saturation. Turn on QoS (voice priority) in your router so calls jump the queue ahead of downloads — the highest-value setting for this specific problem. Persistent high jitter on a wired, quiet line is your provider's to fix; bring the measurements.
Is some jitter normal?
Yes — no real network delivers metronome timing, and that's fine: buffers exist precisely to absorb everyday variation. A few milliseconds is healthy; 20 ms is unremarkable. The number to react to is sustained jitter above 30 ms, or jitter that spikes every day at the same busy hour — that's a congestion pattern, not background noise.