Network Jitter

Measure network jitter to keep operations running smoothly

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Check Jitter Levels

Leverage VoIP Metrics

Drill Down on Jitter

Reduce VoIP Jitter

Decrease Jitter

Easily check jitter levels and uncover what could be affecting VoIP call quality

Minimizing network jitter helps ensure VoIP call quality. With SolarWinds^® VoIP & Network Quality Manager (VNQM), you can closely monitor VoIP calls and call detail records (CDRs) generated by Cisco Unified Communications Manager and Avaya Aura Communication Manager. VNQM is specially designed to check VoIP call quality for current jitter and maximum jitter, which can enable you to gauge performance at a more granular level, monitor the quality of the VoIP traffic, and troubleshoot quickly.

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Leverage VoIP call metrics to measure VoIP jitter across your enterprise

VNQM allows you to drill down on network jitter by using detailed VoIP call metrics and call records. With this VoIP jitter tool, monitoring jitter can consist of searching and filtering VoIP calls based on network jitter metrics found in call detail records. You can check jitter from multiple angles by filtering VoIP calls by most common error codes or call quality metrics, so you can more easily see where network jitter exceeds predefined threshold limits.

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Drill down on network jitter issues and troubleshoot as efficiently as possible

SolarWinds VNQM helps you pinpoint and measure network jitter within your network, so you can start fixing jitter immediately. VNQM is designed to provide you with an overview of your call detail records and IP SLA operations, which can help you identify VoIP call quality and performance issues. These powerful VoIP monitoring tools for Cisco devices can also help you see how a call moves through your network, so you get added insight to enhance troubleshooting with detailed CDR analysis to help you more easily pinpoint VoIP performance issues.

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Reduce VoIP jitter with an enterprise-grade tool specially designed to tackle VoIP performance issues

Network jitter can be a major problem for VoIP calls, but it’s not the only performance issue with a negative impact on call quality. The PerfStack^™ dashboard available for VNQM is designed to help you correlate VoIP jitter with other performance metrics, such as SIP and CUBE trunk availability, CPU, memory utilization, and more, all on a single timeline. This visibility can help you reduce VoIP jitter now and in the future by seeing how VoIP jitter may be impacted by the larger picture of your network performance.

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Proactively decrease network jitter with tools to improve capacity planning

VoIP Call Quality Test Monitor VoIP Metrics

One of the best ways to prevent network jitter is to make sure your network has enough bandwidth to handle both its business-critical operations and VoIP calls. VNQM uses VoIP gateway and PRI trunk monitoring to display exactly how your VoIP capacity is being used. It can also create fake VoIP traffic, so you can see how VoIP call quality would be affected if you made certain changes to your resource allocation. These unique features are designed to help you make smarter capacity planning decisions.

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Get More on Network Jitter

Do you find yourself asking…

What is network jitter in VoIP?
What causes network jitter?
What is the difference between jitter and latency?
Why is jitter a problem for VoIP phone calls?
What is acceptable jitter?
How can network jitter be calculated?
How does network jitter monitoring work in SolarWinds VNQM?

Network jitter, also known as packet delay variance, refers to small delays as packets travel across a network. When a network has high jitter, packets are delivered at irregular intervals as opposed to a steady stream. A few packets might make it to their destinations on time, while other packets might be sent all at once, out of order, or not at all.

Jitter can eventually lead to packet loss, which causes a noticeable decline in the quality of real-time services. VoIP jitter is the same as network jitter but in relation to Voice over Internet Protocols.

Network jitter, also known as packet delay variance, refers to small delays as packets travel across a network. When a network has high jitter, packets are delivered at irregular intervals as opposed to a steady stream. A few packets might make it to their destinations on time, while other packets might be sent all at once, out of order, or not at all.
Jitter can eventually lead to packet loss, which causes a noticeable decline in the quality of real-time services. VoIP jitter is the same as network jitter but in relation to Voice over Internet Protocols.
The exact cause of network jitter can be hard to pinpoint in a large enterprise with many applications and endpoints competing for bandwidth.
However, four of the most common causes of jitter are:
Network congestion: If you’re having a problem with jitter in your network, it’s most likely due to network congestion. If there are too many users, devices, applications, or servers using bandwidth at the same time, your internet will slow down to keep up. Packets will get dropped as your connection slows down, causing jitter.
Wi-Fi connections: Wi-Fi networks are notorious for dropping packets and causing network jitter. When packets travel through the air on a Wi-Fi connection, some packets can get lost along the way, making it harder to maintain high-quality VoIP calls.
Old or faulty hardware: An old modem, faulty Ethernet cable, or misconfigured router can weaken your internet connection and cause jitter.
Poor packet prioritization: If issues with your network’s internet connection aren’t the culprit, jitter can also be caused by your Quality of Service (QoS) settings. QoS settings allow you to prioritize certain kinds of traffic to receive the most bandwidth, and if these settings aren’t on or being used effectively, you could be unknowingly starving your VoIP calls of the bandwidth they need.
To reduce jitter, you can:
Use a jitter buffer
Use a fiber optic or Ethernet connection
Upgrade your internet connection
Re-evaluate your bandwidth and QoS settings
Update your hardware
Before you jump into troubleshooting jitter, however, you’ll need a way to measure network jitter. You can monitor and measure jitter using a standard network monitoring tool, but you can achieve more detailed results with a tool specially designed to monitor VoIP jitter.
Latency measures how long it takes a packet to make it to its destination, while jitter measures any delays in a packet’s journey.
Jitter is defined as the delay in packet transmission measured in milliseconds, while latency is the time it takes for data to successfully travel from source to destination measured in milliseconds. Jitter can be considered a subsection of latency.
Latency and jitter also share a few common causes, like poor Wi-Fi connectivity, old hardware, overloaded networks, and insufficient capacity. However, unlike jitter, latency has more causes unrelated to the internet connection. For instance, a misconfigured network can cause latency issues.
Network latency ultimately comes down to how long packets stay in transit, which is determined by the strength of your internet connection and how well organized and optimized your network is. When troubleshooting latency, you have more variables to consider.
The most common way to measure latency is by calculating “round-trip time” or RTT. As the name suggests, this is the amount of time it takes, in milliseconds, for a packet to successfully complete a journey from source to destination. You can also measure latency according to “time to first byte” or TTFB. TTFB measures the time difference between the moment the first byte of a packet leaves the source and the moment the first byte of that same packet arrives at its destination.
Another key difference between jitter and latency is latency can be controlled and eliminated, while jitter cannot.
Common ways to help resolve latency include:
Isolating problematic endpoints
Finding and eliminating bottlenecks
Using a Content Delivery Network (CDN)
When it comes to jitter, the best thing you can do is take measures to reduce it when it arises and try to create an environment where your network generates the least amount of jitter possible.
All networks will have jitter. If network jitter is within the acceptable range, you might not experience any disruptions in service at all.
The following levels of jitter are typically considered acceptable:
Jitter below 30ms, preferably below 20ms
Less than 1% packet loss
Overall network latency less than 150ms
If any of these thresholds are surpassed, you may notice a sharp decline in call quality. Your voice might sound distorted or warbled and the call itself might go in and out.
SolarWinds VoIP & Network Quality Manager is designed to be a highly intelligent, highly specialized network jitter monitoring tool with the tools you need to monitor, manage, and mitigate the effects of jitter. With this tool, you can capture and analyze VoIP traffic directly from the packet stream and use those findings to calculate jitter and latency. With routine network jitter monitoring, VNQM can help you maintain call quality in VoIP communications.
Other notable features include:
Cisco SIP and CUBE trunk monitoring
VoIP gateway and PRI trunk monitoring
Automatic IP SLA setup
Real-time WAN monitoring and alerting of site-to-site WAN performance
Proactive VoIP QoS management
With SolarWinds VoIP & Network Quality Manager, you have everything you need to make sure VoIP calls come through loud and clear.

Easily measure, manage, and troubleshoot network jitter in VoIP calls

VoIP & Network Quality Manager

Drill down on the cause of call failures by correlating network jitter with other metrics.
Create fake VoIP traffic to understand how call quality would be affected by certain changes.
Measure network jitter, gauge performance at a more granular level, and troubleshoot quickly.

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Fully functional for 30 days

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