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What is a jitter buffer?

A jitter buffer is the small trick that turns an unevenly arriving stream of packets into smooth, listenable audio, on purpose, at the cost of a tiny bit of delay.

Smoothing out an uneven drip

Because packets arrive with uneven jitter, playing each one the instant it lands would sound stuttery and uneven. A jitter buffer solves this by deliberately holding incoming packets for a brief moment, then releasing them to be played at a smooth, steady pace, even if they did not arrive at a steady pace.

A small, deliberate delay you are trading for smoothness

The buffer only works by adding a small amount of extra delay on purpose. A larger buffer absorbs more jitter and produces smoother audio, but adds more lag to the conversation. A smaller buffer feels more immediate but is more likely to run dry when a burst of jitter hits, causing an audible gap.

Fixed versus adaptive buffers

Older systems used a fixed buffer size, set once and never changed, which meant it was tuned wrong the moment network conditions shifted. Most modern VoIP and WebRTC systems use an adaptive jitter buffer instead, one that continuously grows or shrinks itself based on the jitter it is actually observing, trying to hold the smallest delay that still keeps audio smooth.

Try it yourself

See jitter measured on a real call

The Connection Quality Analyzer graphs live jitter alongside latency and packet loss, so you can see exactly what a jitter buffer is compensating for.

Open the Quality Analyzer →