Network Jitter: How It Affects VoIP Phone Calls & How to Fix It

VoIP jitter is the variation in how long voice packets take to reach their destination. When packets arrive at uneven intervals, callers may hear choppy speech, gaps, or distorted audio. Reducing congestion, checking your connection, and configuring voice traffic correctly can reduce jitter and improve call quality.

You might have experienced a business phone call where the audio became choppy or cut out, then returned to normal a few moments later. Network jitter and packet loss are two common causes of these disruptions in Voice over Internet Protocol (VoIP) systems.

Troubleshooting these persistent VoIP problems is a headache, even for seasoned IT pros. This guide explains what exactly network jitter is, how to measure it, and which fixes to try based on the results.

What Is Network Jitter?

Network jitter measures how packet delay changes over time, usually in milliseconds (ms). For example, packets may take 20 ms, 25 ms, and 45 ms to reach their destination rather than taking roughly the same time throughout.

High delay does not necessarily mean high jitter. If every packet takes about 100 ms to arrive, latency is high, but the variation may be small.

Jitter affects online activities that depend on two-way, real-time communication. Examples include online gaming, audio and video conference calls, IP security cameras, and more.

Here’s a helpful video that provides a simple explanation of network jitter.

Why Does Network Jitter Matter?

Network jitter can make business conversations harder to follow. Customers may need to repeat account details, while employees may miss instructions or spend extra time confirming what was said. During support calls, these interruptions can delay resolving a customer’s issue.

When you start a movie on Netflix, the first few seconds are spent buffering the content. Downloading video ahead of playback can mask brief connection interruptions. A live conversation has less room for buffering because both people need to hear and respond without a long pause.

That makes stable packet delivery important for business communications. Whether your team is answering customer questions or discussing a project, interruptions can lead to misunderstandings and repeated work.

Here’s a visual representation that illustrates what happens behind the scenes.

How Jitter Affects VoIP Call Quality

During a VoIP call, the receiving device needs incoming voice packets in time for audio playback. A jitter buffer briefly holds packets to accommodate uneven arrivals. If a packet arrives after its playback deadline, the device may discard it, leaving a gap in the audio.

The effect depends on how much the packet timing varies and how the receiving device handles it. Small variations may pass unnoticed. Larger fluctuations can exceed the buffer’s capacity, while increasing the buffering delay can make the conversation feel less immediate.

This video explains some common VoIP problems and how you can fix them.

Think about it this way: if parts of your speech arrive in a different order, that impacts your conversation. VoIP isn’t any different.

Low jitter supports clear calls, but packet loss, latency, and your devices also affect audio quality. High-definition VoIP codecs such as G.722 support wideband audio, while G.729 is a narrowband codec designed to use less bandwidth.

What does VoIP jitter sound like?

Callers may hear clipped words, short gaps, or broken speech that returns to normal moments later. These symptoms can also occur with packet loss or device problems, so use call-quality measurements to confirm the cause.

This video by Cee Tee simulates network conditions that cause a high degree of jitter and some packet loss. The SIP media stream arrives out of order, which affects the audio clarity of the call.

What is acceptable jitter for VoIP?

Cisco recommends keeping jitter below 30 ms for high-quality voice calls. Your provider may set a stricter requirement; Nextiva’s published guidance calls for jitter below 20 ms and 0% packet loss.

Use your provider’s requirement when assessing whether the connection is suitable for its service. A general benchmark should not replace the requirements for your phone system.

An average below the recommended threshold can still hide short spikes. Where your diagnostic tool supports it, review peak values alongside the average, particularly when callers report broken audio. Note whether the problem is brief or recurring and whether it affects one user or several.

What Is the Difference Between Jitter, Latency, and Ping?

These terms describe different parts of network performance. A connection can have low average latency and still suffer from jitter, so a good ping result alone does not rule out call-quality problems.

Term	What it measures	What it means for VoIP
Latency	How long data takes to travel from one point to another	High one-way delay can make people talk over each other
Ping	A test that usually reports round-trip time using ICMP packets	Useful for basic connectivity checks, but it does not directly measure voice quality
Jitter	Variation in packet delay	Uneven arrivals can interrupt audio playback
Packet loss	Packets that fail to reach their destination	Missing voice data can leave gaps in speech
Bandwidth	The data capacity available on a connection	Limited capacity can contribute to congestion when calls compete with other traffic

The Technology Behind Jitter

It’s a good idea to brush up on network connectivity terms to better understand jitter.

1. Data packets

Data packets are the basic units of information transmitted across a network.

Data sent over a network is broken down into smaller packets for efficient and reliable transmission. These packets are then routed individually through the network and reassembled at the receiving end to compile the original message.

Each packet consists of a header and a payload section. 

Headers contain addressing and other control information, with the exact fields depending on the protocols used. The payload contains the data being transmitted, such as text, images, audio, or video.

With many computers and active devices on a network, large chunks of data could slow everyone down. Packets allow everyone to send and receive information reliably.

Call signaling is separate from the voice stream. SIP commonly handles tasks such as starting and ending calls, while RTP carries the audio.

2. Transport protocols

Packets themselves are neither reliable nor unreliable. It’s data. Applications use transport protocols to reach their destinations quickly and reliably.

• User Datagram Protocol (UDP) does not wait for acknowledgments or automatically retransmit missing packets. VoIP audio commonly uses RTP over UDP because late audio may no longer be useful during a live conversation.
• Transmission Control Protocol (TCP) can be used for less urgent communications. Got it? Are you sure? Good. That’s an example of TCP in action. TCP provides ordered delivery, acknowledges received data, and retransmits missing data. Waiting for retransmissions can add delay. VoIP systems may use TCP for signaling and other functions, depending on their setup.

Glitches in transport protocols affect data transmission, causing latency and delays in the form of network jitter.

3. Jitter buffers

A jitter buffer briefly holds incoming voice packets before playback to smooth out uneven delivery. Some devices use adaptive buffers that adjust to changing network conditions. A larger buffer can accommodate more variation but also adds delay. It cannot recover packets that never arrive.

What Causes Network Jitter?

VoIP services are susceptible to network jitter and connectivity issues. Here are the top culprits to look for.

Cause #1: Network congestion

Routers support today’s bandwidth-heavy needs, such as video calls. By default, routers segment the Wide Area Network (WAN) from your local network. This segmentation efficiently routes, manages, and optimizes traffic between the two networks.

This hands-off approach is nice until everyone watches YouTube, streams music, and joins video conferences. Oh, and don’t forget all those laptops connected to the Wi-Fi, either.

What happens to a congested network? It experiences jitter.

How are VoIP, bandwidth, packet loss, and jitter related? 

Every packet matters since VoIP converts sound into data packets. A congested network sees frequent packet loss and delays, creating gaps in conversation or drops in connection quality. A packet’s path from your desk to a VoIP service provider isn’t always so direct.

As you exhaust available bandwidth, packets spend longer waiting in queues. When that waiting time varies, jitter increases. Depending on how much variation the receiving device can accommodate, the audio may become choppy or unintelligible.

While network congestion is the bedrock, many other network issues cause jitter.

Cause #2: Network interference and poor network systems

Electromagnetic interference (EMI), faulty wiring, or radio frequency interference can cause network interference and jitter.

Similarly, outdated or inadequate hardware, such as routers or switches, contributes to jitter. Poor network hardware struggles to handle high call volumes, causing inconsistencies in packet arrival times and degrading call quality.

Smartphone users should pay attention to their surroundings. You’re bound to have call quality issues if you get into an elevator, a supply room, or underground. Certain building materials (concrete, metal, and window films) affect and potentially block cellular and Wi-Fi signals.

Cause #3: Wireless signal quality degradation

Jitter problems can affect any network connection, but end-users often experience it on Wi-Fi. Wireless networks use wireless signals to transmit data packets between devices and the network access point. 

Unlike wired connections, wireless connections are more prone to interference from other wireless devices, Wi-Fi networks, and even environmental disturbances. Signal degradation or fluctuations from interference can lead to inconsistent packet delivery and trigger jitter. 

Wi-Fi 6 uses the IEEE 802.11ax standard; 802.11ac is Wi-Fi 5. Newer equipment can improve performance, but interference and signal strength still affect call quality.

Cause #4: Packet prioritization, QoS, and network configuration

An easy-to-overlook but critical factor that underpins high-quality VoIP calls is packet prioritization. Routers can prioritize traffic from specific devices and classes of traffic. This strategy is known as Quality of Service (QoS) and can be simple to put in place. However, many overlook or fail to configure it correctly on their network infrastructure. 

Missing or misconfigured QoS can leave voice packets waiting behind less time-sensitive traffic on a congested link.

How to Measure Network Jitter

Different tools measure jitter in different ways, but you can add plenty of useful tools to measure jitter to your toolbox. Keep in mind that jitter measures the variability of your network latency. 

Most diagnostic tools report jitter in milliseconds. Measurement methods and test destinations differ, so compare results from the same tool over time.

1. Online speed tests

Bandwidth tests don’t always tell the full story. They do clue you into problems with your internet speed and connection. You can confirm issues like bandwidth, packet loss, and latency in seconds.

• Nextiva’s VoIP Network Quality Test – This VoIP speed test determines your network’s quality and how well it can handle VoIP calls. It’s highly accurate and tailor-made for VoIP connections. It simulates realistic network loads for simultaneous VoIP calls. Choose the starting point of the concurrent calls (location), enter the number of VoIP lines to simulate, and get the results in a few minutes. 
• Cloudflare’s Internet Speed Test – This tech-focused speed test shows your vitals beautifully. It displays bandwidth, latency, network jitter, and performance by file size.
• Ookla’s Speed Test – This simple speed test also measures your connection speed and ping. Its endpoint testing can be useful to confirm high bandwidth conditions. It’s also handy to use their free mobile app.

2. Terminal-based ping jitter test

Browser-based testing can leave you with a misleading view of your network congestion. Business VoIP calls depend on low-latency connectivity to a specific VoIP server.

Open up a terminal (“Command Prompt” for Windows users) and conduct manual ping tests. 

This command shows you the speed for each packet to reach that network.

• macOS and Linux: ping -c 20 8.8.8.8
• Windows: ping -n 20 8.8.8.8

This command will ping a server (Google public DNS) with 20 packets. Observe the values displayed at the end.

--- 8.8.8.8 ping statistics ---
20 packets transmitted, 20 packets received, 0.0% packet loss
round-trip min/avg/max/stddev = 21.897/28.623/38.247/3.979 ms

The stddev value shows about 4 ms of variation in the sampled round-trip times, not a direct measurement of RTP jitter. Windows’ standard ping summary does not include this value.

This short test reports no packet loss, but it can miss intermittent problems. It checks the path to Google’s public DNS, not necessarily your VoIP provider, and cannot separate upstream from downstream delay.

3) Advanced network monitoring tools

For enterprise phone systems, you might already have access to robust network diagnostics tools. These tools monitor all inbound and outbound traffic at the router level and analyze traffic from different endpoints, including SIP.

• LogicMonitor
• Cisco Catalyst Center
• Dynatrace
• ManageEngine OpManager

These alert you to high jitter, packet loss, and real-time metrics to troubleshoot. Plus, you can hold vendors accountable with Service Level Agreement (SLA) monitoring.

How to Fix Jitter (Troubleshooting Tips)

Use your test results to choose a fix. If restarting your modem or router is appropriate, schedule it when it will not interrupt active calls, then test again.

Here are eight solutions to fix high jitter that you can try:

1. Test your connection for jitter issues

Your approach should be to determine the root cause of high latency. Adjusting VoIP Quality of Service settings can only get you so far. 

Use network diagnostic tools to monitor jitter issues as they occur. Run tests at different times of the day to see if the jitter intensifies during specific periods.

Digging underneath jitter helps you determine the pain point, whether it’s with your network, internet service provider (ISP), or other external factors. Compare the times of poor calls with backups, uploads, and other network activity. When you know the exact issue, you get the right head start to solve it. 

2. Use a wired internet connection.

Wi-Fi calling remains vulnerable to interference from microwaves and electrical motors. If users experience jitter during lunch when the network is less busy, it’s a cue to check for environmental causes. If jitter occurs during peak business hours, check your device frequency. 

There may be interference from other wireless devices operating in the same frequency band.

Compare Wi-Fi results with a wired connection using a known-good Ethernet cable. Replace damaged cables or those that do not support the required connection speed. Upgrading to Category 6 alone will not fix congestion or problems outside your local network.

3. Review packet inspection and firewall settings

Firewalls that analyze incoming voice packets can contribute to network latency. Inspection can add processing delay, particularly if the device is overloaded.

Check device utilization and logs, then review settings against your VoIP provider’s instructions. Adjust problematic inspection features with your IT team rather than disabling firewall protection across the network.

4. Set up a jitter buffer.

In situations with constant jitter, you can set up a buffer to accommodate jitter. Jitter buffers work by delaying VoIP audio enough to reorder voice packets correctly.

Too much buffering can add noticeable delay. Too little buffering can cause more late packets to be discarded. Check the settings on your phone, app, or media gateway and follow the device or provider’s guidance.

5. Enable Quality of Service (QoS).

Many networks become saturated with non-voice traffic, which can limit the availability of VoIP phones and access to VoIP networks. To avoid packet prioritization concerns, schedule large data transfers outside of business hours.

Configure voice prioritization according to your provider’s guidance. RTP voice traffic commonly uses DSCP 46, the Expedited Forwarding marking, but your equipment must apply the corresponding queue policy.

QoS manages how traffic shares a congested link. It does not create more bandwidth, and prioritizing voice can reduce capacity available to other traffic during congestion.

Base bandwidth reservations and traffic shaping on measured connection capacity rather than a fixed percentage. Local QoS settings cannot control every network along the internet path.

6. Increase your WAN network bandwidth.

If all else fails, you’ll likely want to switch WAN providers. Consider fiber if tests show your current connection is inadequate, but compare actual performance before switching. This step is more of a last resort but a worthwhile upgrade. Afterward, set up Quality of Service for Voice over Internet Protocol traffic.

We recommend consulting with your VoIP provider to analyze your network configuration. Sometimes, the fix could be as simple as a firmware upgrade on select VoIP systems. Disabling SIP ALG has little to do with VoIP jitter but is highly recommended.

Quick story here…

I was troubleshooting why I could only achieve about half of my ISP’s gigabit network performance. After testing everything, it turned out the router’s WAN port simply couldn’t reach 1000 Mbps despite the advertised hardware specs. Sometimes, upgrading your router is the fix.

Once you have lowered jitter, you’ve solved one obstacle to business communication.

7. Opt for a high-quality VoIP provider.

You can fix thousands of problems with your network, but you still have issues without quality and reliable VoIP service.

To avoid and mitigate network jitter, you need robust network infrastructure and optimized routing mechanisms without worrying about managing the underlying infrastructure. Your VoIP service should reduce the jitter and simplify your life, not add bottlenecks.

When choosing a VoIP phone system:

• Prioritize stability and advanced tech to minimize packet loss and delay.
• Look for excellent diagnostics and faster issue resolution. 
• Consider its network infrastructure, customer reviews, service level agreements (SLAs), and ability to handle and manage real-time communication traffic effectively.

8. Regularly maintain and monitor your network.

Finally, effective VoIP calling boils down to closely monitoring your network, detecting jitter problems early, and mitigating them before they become bigger. 

Maintain: Keep your hardware and software up to date. Regularly update firmware and drivers for routers, switches, and other network devices. Outdated or buggy software contributes to network instability and increases jitter. Additionally, configure your network equipment to optimize it to handle large VoIP traffic volumes.

Monitor: Track key performance indicators (KPIs), including latency, packet loss, and jitter levels. Your diagnostic tools should provide real-time insight into your network’s health and alert you to any anomalies or excessive jitter to effectively monitor network jitter.

Easy and Reliable VoIP Calls Ahead

Virtual phone service is already optimized for peak performance. You provide a fast internet connection, and the provider handles the rest.

Nextiva has built one of the most reliable VoIP networks for businesses. We leverage several data centers across North America to provide ultra-low latency. You get superior call quality backed by Amazing Service anytime you need it.

Skip the guesswork when it comes to your VoIP phone service. Our cloud telephony experts will walk you through every step. We’ve helped over 100,000 companies — we know what works and what doesn’t.

Frequently Asked Questions About Jitters