What Is VoIP Jitter and Latency: Causes and How to Troubleshoot

VoIP (Voice over Internet Protocol) has completely changed the way we communicate, offering a cost-effective and flexible alternative to traditional telephony systems.

However, one of the key challenges in ensuring high-quality VoIP communication is managing jitter, a phenomenon that can significantly degrade the overall user experience. 

Jitter can lead to choppy audio, distorted speech, and even dropped calls. Understanding the root causes of jitter and implementing effective troubleshooting strategies is crucial for maintaining seamless and reliable VoIP communication.

In this context, we will delve into the various factors that contribute to jitter in VoIP networks, explore its impact on call quality, and provide practical solutions to identify and mitigate jitter-related issues.

What Is VoIP Jitter and Latency: Causes and How to Troubleshoot

What is Jitter in Voip?

Jitter in Voice over IP (VoIP) refers to the variation in the time between packets arriving at the destination. It measures the consistency of the delay in packet delivery times.

In simpler terms, it’s the uneven spacing or irregular gaps between the arrival of voice packets.

In VoIP systems, voice data is transmitted over the internet or IP networks in the form of packets.

These packets are expected to arrive at a consistent rate for the voice to sound natural and continuous. However, due to network congestion, route changes, or other factors, the packets may experience varying delays, resulting in jitter.

High jitter can lead to several issues in VoIP communication, including:

  1. Degraded voice quality: Jitter can cause the received audio to sound choppy, distorted, or robotic, affecting the overall voice quality and intelligibility.
  2. Dropped packets: Excessive jitter can cause some packets to arrive too late, forcing the receiver to discard them, leading to lost voice data and potential audio dropouts.
  3. Increased latency: Jitter can contribute to increased end-to-end latency, as the receiver needs to buffer incoming packets to compensate for the varying delays, introducing additional delay in the audio playback.

To mitigate the effects of jitter, VoIP systems often employ jitter buffers, which temporarily store incoming packets and release them at a consistent rate, smoothing out the variation in packet delivery times.

However, jitter buffers can also introduce additional latency, so a balance must be struck between acceptable jitter levels and latency.

Minimising jitter is crucial for ensuring high-quality VoIP communications, and it is often addressed through network optimization, quality of service (QoS) mechanisms, and efficient codec designs that can better handle packet loss and jitter.

What is an acceptable jitter for the internet?

There is no universally defined “acceptable” level of jitter for the internet, as the acceptable jitter level can vary depending on the specific application and its requirements.

However, in the context of Voice over IP (VoIP) communications, some general guidelines and recommendations exist:

  1. For VoIP calls:
    • Typical recommendations suggest keeping jitter below 20-30 milliseconds (ms) for good voice quality.
    • Jitter below 10 ms is considered excellent.
    • Jitter above 50 ms may cause noticeable voice quality degradation.
  1. For video conferencing and multimedia applications:
    • Jitter should typically be kept below 30-50 ms for high-quality video and audio.
    • Jitter above 100 ms can significantly impact the user experience.

It’s important to note that these values are general guidelines, and the actual acceptable jitter level may depend on factors such as:

  • Codec used: Some codecs are more resilient to jitter than others.
  • Jitter buffer size: Larger jitter buffers can compensate for higher jitter levels but introduce additional latency.
  • Application requirements: Real-time applications like VoIP and videoconferencing have stricter jitter requirements than non-real-time applications like file transfers.
  • Network conditions: Higher network congestion and packet loss can exacerbate the effects of jitter.

In general, lower jitter levels are desirable for real-time communication applications to ensure high-quality audio and video transmission.

Network optimization techniques, quality of service (QoS) mechanisms, and efficient buffer management can help minimise jitter and improve the overall performance of internet-based communication services.

How do I fix the Internet Jitter?

1. Upgrade your internet connection

Higher bandwidth connections generally have lower jitter and latency due to reduced network congestion.

Fiber-optic internet connections tend to have the lowest jitter levels compared to cable or DSL.

If upgrading is not an option, consider switching to a different service provider that offers better performance in your area.

2. Use wired Ethernet connections

Wi-Fi connections are more susceptible to interference, signal strength fluctuations, and packet loss, which can contribute to jitter.

Connect your devices directly to your router or modem using Ethernet cables for a more stable and consistent connection.

If using Wi-Fi is unavoidable, ensure you have a strong signal strength and minimize potential interference sources (e.g., microwaves, cordless phones, Bluetooth devices).

3. Optimize your network

Use a wired connection for devices running VoIP or video conferencing applications to minimize wireless interference.

Limit other bandwidth-intensive activities during important calls or conferences, such as large file downloads, video streaming, or online gaming.

Configure Quality of Service (QoS) settings on your router to prioritize real-time traffic like VoIP and video conferencing over other types of traffic.

4. Use a dedicated VoIP router or gateway

These devices are specifically designed to prioritize and optimize VoIP traffic, reducing jitter and latency.

They often include advanced features like jitter buffers, packet prioritization, and traffic shaping to ensure smooth VoIP performance.

5. Adjust jitter buffer settings

Most VoIP applications and devices have built-in jitter buffers that can be adjusted. Increasing the jitter buffer size can help compensate for jitter by temporarily storing incoming packets and releasing them at a consistent rate.

However, larger jitter buffers can also introduce additional latency, so finding the right balance is important.

6. Consider a VPN connection

In some cases, using a Virtual Private Network (VPN) can provide a more stable and optimized connection for VoIP and video conferencing by routing traffic through a private, managed network.

VPNs can help reduce jitter caused by internet congestion or routing issues, but may introduce additional latency.

7. Update network drivers and firmware

Ensure that your network devices (routers, modems, network adapters) have the latest drivers and firmware installed, as these updates often include performance improvements and bug fixes.

Check with your device manufacturers for the latest updates and follow their instructions for properly updating the firmware.

8. Contact your internet service provider

If the issue persists after trying the above steps, contact your ISP and report the jitter problems you’re experiencing.

They may be able to identify and resolve any issues on their end, such as network congestion, routing problems, or faulty equipment. In some cases, they may be able to provide you with a higher-quality service or suggest additional solutions.

By implementing a combination of these measures, you should be able to reduce jitter and improve the quality of your internet-based voice and video communications.

How is network jitter measured?

Network jitter is typically measured by analysing the variation in packet delivery times between consecutive packets sent over the network.

There are several methods used to measure jitter, including:

  1. Packet Delay Variation (PDV): This method measures the difference in one-way delay between consecutive packets. The one-way delay is calculated as the time it takes for a packet to travel from the sender to the receiver. PDV is calculated by taking the difference between the one-way delays of consecutive packets.
  2. Inter-Packet Delay Variation (IPDV): IPDV measures the variation in the delay experienced by packets traveling in the same direction. It calculates the difference between the delays of successive packets, providing a measure of the jitter seen by the receiver.
  3. RFC 3550 Jitter Estimation: This method, defined in RFC 3550, is commonly used in real-time multimedia applications like VoIP and video conferencing. It calculates the smoothed estimate of the statistical variance of the packet delay, taking into account the arrival times of past packets.
  4. Mean Absolute Packet Delay Variation: This method calculates the absolute difference between the one-way delay of each packet and the overall mean one-way delay, then averages these differences to obtain the mean absolute packet delay variation.

To measure jitter using these methods, network monitoring tools or specialized software are typically employed.

These tools capture and analyze packet timestamps at both the sender and receiver ends, allowing them to calculate the various jitter metrics.

Some common tools and utilities used for measuring network jitter include:

  • Ping and TraceRoute: While primarily used for measuring latency and network path, these tools can also provide basic jitter information by analyzing the variation in round-trip times.
  • Wireshark and tcpdump: These packet capture and analysis tools can be used to analyse packet timestamps and calculate jitter metrics like PDV and IPDV.
  • Specialised jitter measurement tools: Several commercial and open-source tools are available specifically for measuring jitter, such as IxChariot, OPNET IT Guru, and D-ITG.

It’s important to note that network jitter can be affected by various factors, such as network congestion, routing changes, hardware performance, and link quality.

As a result, jitter measurements can vary depending on the network conditions and the specific path taken by the packets.


Jitter can significantly impact the quality of voice and video communications over the internet. While some level of jitter is unavoidable, understanding its causes and implementing effective mitigation strategies is crucial.

By optimising networks, prioritising real-time traffic, and leveraging advanced technologies like VPNs and dedicated VoIP routers, users can minimise jitter and enjoy seamless communication experiences.

Ultimately, proactively addressing jitter is essential for ensuring reliable and high-quality internet-based voice and video services in today’s digital landscape.


Ques 1. What is a good Jitter speed?

Ans: A good jitter speed is generally considered to be below 20ms. Anything above 30 ms is considered high and may cause quality issues.

Ques 2. What is Jitter on a network?

Ans: Jitter is the variation in latency or delay for data packets travelling over a network, measured in milliseconds.

Ques 3. What is considered a bad Jitter?

Ans: Jitter above 30 ms is generally considered bad, as it can cause quality issues in real-time applications like VoIP, video conferencing, and online gaming.

Ques 4. Does WIFI cause Jitter?

Ans: Yes, WiFi connections are more prone to jitter compared to wired connections due to potential interference and signal fluctuations.

Ques 5. How do I get rid of Internet Jitter?

Ans: To reduce jitter, you can use a wired connection, enable Quality of Service (QoS), optimise your network, and use a higher-quality internet plan.

Evelyn Brown
Evelyn Brown

Evelyn Brown is a knowledgeable and dedicated reviewer of business communication softwares. When she's not testing the latest platforms or providing in-depth analyses for his readers, you can find her playing guitar and hiking local trails.