Jitter

When you experience choppy audio during a VoIP call or lag in a video meeting, the issue is often related to jitter. Here, we’ll explain what network jitter is, how it differs from latency, why it affects call quality, and what levels are considered acceptable for VoIP and real-time communication.

What Is Jitter?

Jitter is the variation in packet delay across a network. While latency measures the total time it takes for data to travel from sender to receiver, jitter measures how inconsistent that delay is between packets. It is typically measured in milliseconds (ms).

For VoIP and video calls, jitter should ideally remain under 30ms to maintain clear and stable communication.

How Does Jitter Work?

In real-time communication like VoIP, voice data is sent across the network in small packets. These packets are expected to arrive at consistent, evenly spaced intervals so the audio can be played smoothly on the receiving side.

When the network is stable, packets arrive in a predictable rhythm. However, factors like network congestion, bandwidth saturation, Wi-Fi interference, or route changes can cause some packets to arrive faster and others slower than expected. This uneven arrival time is jitter.

The receiving device tries to compensate using a jitter buffer, which briefly stores and reorders packets before playback. If the variation in arrival time is small, the user won’t notice. But if the jitter becomes too large, the buffer cannot smooth it out, and the result is choppy, robotic, or broken audio.

What Causes Network Jitter?

Network jitter is usually the result of instability somewhere along the data path. Here are the most common causes:

• Network congestion — When too many devices or applications use the same network at once, packets may be delayed inconsistently as routers prioritise traffic.

• Wi-Fi interference — Wireless networks are more prone to signal interference, which can cause unpredictable packet timing compared to wired Ethernet connections.

• Router or firewall performance issues — Overloaded or outdated networking equipment can struggle to process packets consistently, creating delay variation.

• Bandwidth saturation — When available bandwidth is nearly maxed out (for example, large file uploads running during calls), packet timing becomes irregular.

• Route changes across the internet — Data may take different network paths to reach its destination. If routes shift mid-session, packet delivery timing can fluctuate.

How Jitter Affects VoIP and Video Calls

Jitter directly impacts how smooth audio and video feel during real-time communication, and it’s one of the main VoIP issues. When packet timing becomes inconsistent, the receiving device struggles to reconstruct the media stream properly.

Common effects of high jitter include:

• Choppy or robotic audio — Voices may sound distorted, metallic, or broken.

• Words cutting out — Parts of sentences may disappear if packets arrive too late to be used.

• Delayed responses — Conversations may feel unnatural or slightly out of sync.

• Video and audio desynchronization — Lip movements may not match the spoken words.

For VoIP and video calls, typical jitter thresholds are:

• Under 20ms — Excellent quality

• Under 30ms — Generally acceptable for VoIP

• Around 50ms or higher — Noticeable quality issues

Jitter vs Latency: What’s the Difference?

Jitter and latency are related but measure different network performance issues. Latency refers to how long data takes to travel from sender to receiver, while jitter measures how inconsistent that delay is between packets.

For example, if a VoIP call has 150ms of latency, there may be a slight but consistent delay in conversation. However, if jitter is high, some packets may arrive in 100ms and others in 250ms, causing uneven playback. Even with acceptable latency, high jitter can result in robotic or fragmented audio because the packet timing is inconsistent.

How to Reduce or Fix Jitter

Reducing jitter focuses on stabilising packet delivery and minimising network variability. Here are practical steps that help improve VoIP and video call quality:

• Use wired Ethernet instead of Wi-Fi: Wired connections are more stable and less prone to interference than wireless networks.

• Implement QoS (Quality of Service): QoS settings prioritise voice and video traffic over less time-sensitive data, helping maintain consistent packet timing.

• Upgrade bandwidth if needed — Insufficient bandwidth increases congestion and packet delay variation, especially during peak usage.

• Reduce background network traffic — Limit large downloads, streaming, or cloud backups during important calls to avoid bandwidth saturation.

• Use a jitter buffer — A jitter buffer temporarily stores incoming packets and releases them at consistent intervals. This helps smooth out minor delay variations. However, if jitter is too high, even the buffer cannot fully compensate, and audio quality will still degrade.

Example of Jitter in a VoIP Call

Imagine you make a VoIP call using a business phone system.

First, SIP sets up the call, handling the signalling between both parties. Once the call is connected, RTP carries the voice packets across the network in real time.

Ideally, those RTP packets arrive at steady, consistent intervals. But if the network is congested or unstable, the packets may arrive unevenly — some too early, some too late. This timing variation is jitter.

A jitter buffer on the receiving side briefly stores and reorders the packets to smooth out small variations. If the jitter is minor, the user won’t notice anything unusual. However, if the delay variation becomes too large, the buffer can’t compensate, and the result is choppy, robotic, or broken audio.