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Jitter in VoIP

Understand variation in packet arrival and the delay trade-off made by a playout buffer.

Reviewed 2026-10-15Foundational

Uneven arrival is different from slow arrival

Jitter describes variation in packet timing at a measurement point. A stream can have steady delay with little variation, or relatively low average delay with occasional late bursts. The receiver's playout process must decide how long to wait before audio is needed.

Delay, jitter and loss are different
Packets leave regularly; variable arrival spacing illustrates jitter, while a missing numbered packet illustrates loss.

Illustration, not measured data. A jitter buffer trades playback delay against late-packet loss; it cannot recreate every missing packet.

What the buffer trades

A jitter buffer can absorb some timing variation by delaying playback. If it waits too little, late packets may miss their playout deadline. If it waits too much, conversation feels delayed. Adaptive behavior depends on the endpoint and implementation.

Find the source of variation

Compare affected and unaffected directions and check whether the symptom appears under uploads, wireless contention or a particular route. Preserve the measurement location and interval. An RTCP statistic from one receiver is not a universal summary of the entire call.

Validate perceptually

After improving queueing or adjusting a supported buffer setting, test both sound continuity and conversational delay. Do not maximize the buffer by default. Include busy-hour calls and hold/transfer paths that may negotiate different media handling.

Keep Asterisk jitter values scoped to one stream

Use the Asterisk RTCP statistics guide to preserve channel, media type, direction, SSRC and report source before comparing jitter values.

Separate measured arrival variation from buffer policy

The Asterisk jitter-buffer guide treats network jitter, read-side playout buffering and endpoint DSP as separate observations and controls.

Watch the explanation

What are the reasons for WebRTC to discard audio packets?Tsahi Levent-Levi · 1:57 · Conceptual

Explains why an audio packet can arrive yet be discarded before playback. This distinction helps separate network loss from jitter-buffer and timing problems.

Read applicability and editorial notes →
Watch this video here

Conceptual browser-media discussion from 2024. Specific receiver algorithms and statistics vary. Review used the available transcript and primary references; audio and screen readability remain unverified.

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Sources & applicability

Primary references for the technical details above. Operational examples and planning checklists are VoIP.info editorial guidance.

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