VoIP

Jitter

Jitter is the variation in the time between successive RTP voice packets arriving at their destination. In a perfect network, voice packets would arrive at perfectly regular intervals — for example, every 20 milliseconds for a standard G.711 stream. Jitter occurs when some packets arrive early and others arrive late, causing irregular spacing that the receiving system must compensate for. Excessive jitter is one of the primary causes of poor VoIP call quality in VICIdial environments.

How It Works in VICIdial

Voice calls in VICIdial are carried as RTP streams — a continuous sequence of small UDP packets, each containing 20-30 milliseconds of encoded audio. These packets travel from the VICIdial Asterisk server through the network to the carrier or agent endpoint. Network conditions — congestion, routing changes, QoS policy enforcement, and competing traffic — cause variable delays, resulting in jitter.

To compensate for jitter, VoIP endpoints (including Asterisk) use a jitter buffer — a small memory queue that collects incoming packets, reorders any out-of-sequence packets, and plays them out at regular intervals. The jitter buffer introduces a small fixed delay (typically 20-60ms) but smooths out timing variations. If jitter exceeds the buffer size, packets are discarded, causing momentary audio dropouts or gaps.

Asterisk in VICIdial uses both fixed and adaptive jitter buffers depending on the channel driver. PJSIP endpoints support adaptive jitter buffering that automatically adjusts buffer size based on observed network conditions. For IAX2 connections between clustered servers, jitter buffering can be configured per trunk.

Common causes of jitter in VICIdial deployments include: shared internet connections without QoS prioritization, overloaded network switches, VPN tunnels for remote agents, WiFi connections on agent softphones, and undersized server network interfaces during peak dialing.

Why It Matters

Jitter above 30ms begins to noticeably degrade call quality — manifesting as robotic or choppy audio, words cutting in and out, and garbled speech. At jitter levels above 50ms, conversations become difficult and agents may struggle to communicate effectively. Jitter impacts are reflected in the MOS score for the call, with high-jitter calls scoring well below the acceptable 3.5 threshold.

For VICIdial deployments, jitter control is critical because poor call quality directly reduces conversion rates, increases average handle time (agents asking callers to repeat themselves), and damages the professional image of the operation. Prioritize voice traffic with QoS rules on your network, use wired connections for agent phones, and monitor jitter metrics through Asterisk’s RTP statistics to maintain call quality.

Related Terms

Mean Opinion Score (MOS) View definition → Voice over Internet Protocol (VoIP) View definition → Real-time Transport Protocol (RTP) View definition → Codec View definition →

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