Jitter
What is Jitter?
Jitter is the variation in delay between consecutive voice packets arriving over an IP network — even when packets are sent at a steady, even interval, network congestion or route changes can make them arrive unevenly, which distorts real-time audio if not corrected. A voice platform smooths jitter with a jitter buffer, which briefly holds and re-times incoming packets before playback, trading a small amount of added latency for smoother, more natural-sounding audio.
More detail
Jitter is typically measured in milliseconds; call quality tools flag it alongside latency and packet loss as one of the three core network metrics behind poor VoIP audio.
A jitter buffer that's too small lets jitter through as choppy audio; one that's too large adds noticeable delay to the conversation — sizing it is a real tradeoff, not a fixed setting.
Frequently asked
- What does a jitter buffer actually do?
- It temporarily holds incoming voice packets and plays them back at a steady, evenly-paced rate, absorbing the uneven arrival timing (jitter) so the listener hears smooth audio instead of the underlying network's timing variability.
- Is jitter the same problem as packet loss?
- No. Jitter is variation in how long packets take to arrive, causing choppy timing; packet loss is when packets never arrive at all, causing gaps or dropped audio — both degrade VoIP quality but are measured and mitigated differently.
See also
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