Jitter (VoIP)
Jitter is the variation in arrival time between consecutive RTP packets. Too much jitter causes choppy or robotic audio; a jitter buffer trades delay for smoothness.
Published 8 November 2026 · 1 min read · By SpaGreen Creative
Short answer
Jitter is the variation in the time between arriving RTP packets, measured as described in RFC 3550, which a receiver smooths with a jitter buffer at the cost of added delay.
A sender transmits RTP packets at a steady rhythm, typically one every 20 milliseconds. The network does not deliver them at the same rhythm. Some packets take a longer route, queue behind a large download, or wait for a Wi-Fi retransmission. The resulting irregularity in arrival times is jitter. RFC 3550 section 6.4.1 defines how it is estimated and reported in RTCP receiver reports.
Audio playback needs a steady stream. If a packet arrives late, there is a gap in the sound; if the receiver plays it anyway, the voice stutters or sounds robotic. The remedy is a jitter buffer: the receiver holds incoming packets for a short time before playing them, so that late arrivals still make their slot. A larger buffer absorbs more jitter but adds delay to the conversation. Modern softphones size the buffer adaptively, growing it when jitter rises and shrinking it when the network settles.
Jitter values below about 20 to 30 milliseconds are usually unnoticeable. Values above that start to degrade audio, and above 100 milliseconds calls become difficult. Wi-Fi, congested links and VPNs are common sources. Prioritising voice traffic on the router, using a wired connection, and choosing a codec with good loss concealment such as Opus all help.
Sources and further reading
This article is published by SpaGreen Creative, the maker of SIPdia. Product names mentioned belong to their owners. Where it describes other software, treat the steps as general guidance and check the official documentation linked above.
