A complete guide to understanding, monitoring and fixing network jitter.
The global adoption hybrid working has highlighted the importance of VoIP and video as primary communication solutions. With less in-office and in-person communication, it's vital that your UCC network is functioning optimally at all times, and across every device and location.
However, one of the most common challenges is poor network performance, where VoIP call quality is affected, and video quality is sub-par. One of the key culprits of poor communication quality is network jitter.
Jitter is the variation in time delay between when a signal is transmitted and when it's received over a network connection, measuring the variability in ping. This is often caused by network congestion, poor hardware performance and not implementing packet prioritization.
Performance of your VoIP and video services during video conferencing will be negatively impacted by jitter the longer the delay, and VoIP jitter can make calls choppy and incoherent - or drop out altogether.
This comprehensive guide will explain everything you need to know about how to minimize jitter in computer networks. We'll explore such areas as how to minimize unnecessary bandwidth usage, what is an acceptable level of jitter, Quality of Service (QOS), and how to achieve better quality of VoIP calls.
We’ll cover in-depth the definition of jitter in networking, how to monitor network jitter and how to reduce network jitter.
Download our complete guide to optimizing your network by understanding jitter, latency and packet loss.
Table of Contents
- What is Network Jitter?
- The technology behind network jitter
- Examples of jitter
- The effects of jitter
- What is acceptable network jitter?
- What can cause network jitter?
- Quality of Services (QoS) and jitter
- QoS tools to address jitter
- How is jitter measured
- How to fix network jitter issues
- Summary - network jitter definition and guide
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Ping and Jitter
Ping and jitter are most important during streaming media usage like video streaming, online gaming, or voice over internet (VoIP). Web browsing isn’t very affected by response times and the variation therein, but if you need real-time data, ping and jitter are important measures of your connection quality.
Ping is a measure of the time your connection takes to react, or how quickly you receive a response when a request is made. Ping is measured in milliseconds (ms) and the lower the ping number, the faster your connection. Ping is important during real-time application use like video streaming or online gaming.
Image source: Callhippo
What is network jitter?
As mentioned earlier, jitter is a variance in latency, or the time delay between when a signal is transmitted and when it is received. This variance is measured in milliseconds (ms) and is described as the disruption in the normal sequence of sending data packets. Good connections have a reliable and consistent response time, which is represented as a lower jitter score. The higher the jitter score, the more inconsistent response times are, which manifests in choppy calls or glitchy-sounding looking video quality.
Image source: GetVoip
The technology behind jitter
To define jitter in networking, it comes down to data packets, and packet loss.
All data, in fact everything you do on the internet involves packets. All text, images, audio or video is transmitted in packets over a given network path. When you send or receive an email, search for information on web pages, stream, game or shop online, digital information is dispatched, received, ‘unscrambled’ and ‘reassembled’ ready to view and listen to. Data packets in switched networks allow the exchange of all this information.
Most network data packets are split into three parts:
The header contains instructions about the data carried by packet such as:
Packet length - some networks have fixed-length packets, while others rely on the header to contain this information.
Synchronization - to help the packet align with the network.
Packet number – identifying which packet in the sequence.
Protocol - defines what type of packet is being transmitted whether it’s e-mail, Web page, streaming video.
Destination address - where the packet is going
Originating address - where the packet came from
IP packets are a little like postal letters: the header is the envelope with all the routing information that's needed by the post office, and the payload is the letter that's read only by the recipient.
Image source: Khan Academy
This is the actual data or body that is being delivering to the destination. If a packet is fixed-length, then the payload may be topped up with blank information to make it the right size.
The trailer, sometimes called the footer, tells the receiving device that it has reached the end of the packet. It may also have some type of error checking. The most common error checking used in packets is Cyclic Redundancy Check (CRC).
VoIP technology converts fragments of your voice into data packets which are transmitted digitally via the internet. One of the most common causes of jitter on VoIP services is the absence of packet prioritization. If voice packets aren’t prioritized, then the end user is very likely to experience jitter.
A massive amount of information is constantly being transmitted back and forth - millions of packets every second – and all this data takes a toll on network resources, often resulting in delay. The delay in sending these data packets may not be as apparent when downloading a file or an email, but when your voice arrives in disorganized packets, it will sound distorted and out of sequence.
Examples of jitter
This is a roughly constant level of packet to packet delay variation.
Characterized by a substantial gradual delay that may be incurred by a single packet.
Short term delay variation
An increase in delay that persists for some number of packets and may be accompanied by an increase in packet to packet delay variation. This type of jitter is usually due to congestion and route changes.
The effects of jitter
Packet jitter can cause flickering display monitors, delayed data transmission and poor processor performance.
IP jitter in VoIP communication can severely impact the call quality of telephony and video conferencing, even causing conversations to ‘drop out’, and become jumbled and difficult to understand.
As a result, high jitter is a big problem for real-time applications like digital voice and video communication, as well as streaming and online gaming.
What is acceptable network jitter?
Some applications and services have a higher level of tolerance for jitter than others. So what is acceptable network jitter?
For example, jitter doesn’t affect sending emails as much as it would a VoIP calls, so, it depends on what we’re willing to accept from your internet service provider as irregularities and fluctuations in data transfers. But poor audio quality in VoIP calls and video quality leads to a poor user experience and can impact an organization’s bottom line.
All networks experience some amount of latency, especially wide area networks. Ideally, over a normally functioning network, packets travel in equal intervals, with a 10ms delay between packets. With high jitter, this could increase to 50ms, severely disrupting the intervals and making it difficult for the receiving computer to process the data.
Ideally, jitter should be below 30ms. Packet loss should be no more than 1%, and network latency shouldn’t exceed 150 ms one-way (300 ms return).
If you only have control of one endpoint, you can execute a ping jitter test by working out the mean round-trip time and the minimum round-trip time for a series of packets.
If you have control of both endpoints, you can check jitter with what’s called an instantaneous jitter measurement. This refers to the variation between transmitting and receiving intervals for a single packet. In this case, jitter is calculated as the average difference between instantaneous jitter measures, and the average instantaneous jitter across the transmission of multiple packets.
What can cause network jitter?
Managing network jitter comes down to understanding what causes jitter in computer networks. Doing a regular network jitter test can reduce the prevalence of jitter within your network.
Network congestion – Network congestion caused by insufficient bandwidth is a common problem. Networks become overcrowded with traffic congestion when too many active devices are consuming bandwidth. There are steps you can take to reduce network congestion which we'll discuss later.
Poor Hardware Performance – Older networks with outdated equipment including routers, cables or switches could be the causes of jitter.
Wireless jitter – One of the downsides of using a wireless network is a lower-quality network connection. Wired connections will help to ensure that voice and video call systems deliver a higher quality user experience.
Not implementing packet prioritization – For VoIP systems in particular, jitter occurs when audio data is not prioritized to be delivered before other types of traffic.
Quality of Services (QoS) and jitter
QoS is the technology that manages data traffic in order to reduce jitter on your network and prevent or reduce the degradation of quality. QoS controls and manages network resources by setting priorities by which data is sent on the network.
There are tools and techniques which are often included in an organization’s network Service Level Agreement (SLA) to guarantee an acceptable level of performance.
QoS tools to address jitter
Queuing - Enables you to prioritize or order packets so that delay-sensitive packets leave their queues more quickly than delay-insensitive packets.
Link fragmentation and interleaving (LFI) - Routers do not pre-empt a packet that is currently being transmitted, so LFI reduces the sizes of larger packets into smaller fragments before sending them.
Compression - Payload or headers can be compressed, and this reduces the overall number of bits required to transmit the data. This requires less bandwidth, meaning queues shrink, which in turn reduces delay.
Traffic shaping - Artificially increases delay to reduce drops inside a Frame Relay or ATM network.
How is network jitter measured?
Where your network has control over just one of the endpoints (aka single-ended), jitter is determined by measuring the mean round-trip time (RTT), and the minimum RTT of a series of voice packets.
In a double-ended path, the measurement used is the instantaneous jitter, or the variation between the intervals for transmitting and receiving a single packet. Jitter is the average difference between instantaneously measured jitter and the average instantaneous jitter throughout the transmission of a series of data packets.
Performing a bandwidth test can also determine the level of jitter. A bandwidth test assesses your internet connection’s upload and download speeds, jitter times and your network’s overall capacity.
How to fix network jitter issues
Troubleshooting network jitter can be tricky because of its unpredictability. Keeping jitter to a minimum begins by ensuring that your network is initially properly set up. Ensuring a quality network connection, enough bandwidth, and predictable latency can help reduce network jitter.
Jitter buffering - VoIP endpoints such as desk phones and ATAs usually include jitter buffers to intentionally delay incoming data packets. A jitter buffer ensures that the receiving device can store a set number of packets and then realign them into the proper order, so that the receiver experiences minimum sound distortion.
Jitter buffers are one way to address network jitter and latency but will not always work. If a jitter buffer is too small then too many packets may be discarded, meaning bad call quality. If a jitter buffer is too large, then the additional delay can lead to conversational difficulty.
A typical jitter buffer configuration is 30ms to 50ms in size. You can increase your jitter buffer size to a point, but usually they are only effective for delay variations of less than 100 ms.
Perform a bandwidth test – Bandwidth testing sends files over a network to a specific computer, then measures the time required for the files to download at the destination. This determines a theoretical data speed between the two points, measured in kilobits per second (Kbps) or megabits per second (Mbps).
Bandwidth tests can vary greatly. Factors that affect testing can be internet traffic, noise on data lines, file sizes, and load demand on the server at the time of testing. Bandwidth testing should ideally be carried out several times to determine an average throughput.
Improvements from within - Solving your VoIP network jitter problems may not be as challenging as you think.
Use an ethernet cable - If you are using a desktop computer at a fixed working point, it may be worthwhile to substitute an ethernet cable for a WiFi internet connection. It can provide a more powerful connection with less jitter, and you’ll often experience higher internet internet speeds. maximum jitter
Upgrade your ethernet cable - If you are using an ethernet cable, you may find that outdated cables and switches can often cause high jitter issues. The latest cables are capable of transmitting data at 250 MHz, as opposed to 125MHz, potentially solving ethernet jitter.
Check your device frequency - A VoIP phone that operates at a higher frequency than a standard 2.4 GHz could cause interference on your network. Some phones run at frequencies as high as 5.8GHz, which could potentially exacerbate jitter across your network.
Reduce unnecessary bandwidth usage during work hours - Using large amounts of bandwidth for activities not related to work, like network gaming, or streaming video content can make jitter worse.
Schedule updates outside of business hours – Updating applications and operating systems should be carried out outside work times to free up capacity for more essential communications.
Reducing network congestion
Network congestion is an overload of data that slows down traffic across your entire network. Slow internet connection speeds, a faulty internet connection, or buffering videos, can impact productivity, user experience and ultimately cost you time and money. There are a number of ways to help reduce network congestion
Monitor and analyze network traffic
The right network monitoring tool can help provide answers. For example, your network discovery program can scan your company’s virtual networks, cloud servers, and all other wireless networks and devices to help identify devices, servers, and even users, accounting for significant portions of the available bandwidth.
Prioritize network traffic
For important online processes to run smoothly, you can prioritize VoIP traffic and video traffic in ways that reserve bandwidth for certain users, devices, or platforms. Prioritizing voice traffic or video network traffic means you might need to slow down network connections for non-essential or lower-priority functions or devices.
You can often reduce congestion within your network simply by increasing the available bandwidth. When you increase your network’s bandwidth, the network itself will be able to handle more data and more devices at the same time. Increasing your network’s bandwidth, means you'll reduce jitter, and users will typically enjoy faster connection speeds and fewer interruptions.
This guide has been created to define what is meant by network jitter, and to help identify, understand and troubleshoot the most common problems related to jitter in computer networks.
The key takeaways are that network jitter, packet loss and network latency, are major obstacles standing in the way of clear communication and can universally affect your user experience. This highlights the need for the right monitoring and troubleshooting tools.
IR's Collaborate suite of hybrid-cloud performance management tools brings together reliability, agility, and innovation to solve the complexities of managing critical technologies that keep you in business.
With the shift to hybrid working happening at an astounding pace and scale, organizations all over the globe are tasked with managing increasingly complex unified communications environments to ensure the lines of communication are always open. In a complex, multi-vendor unified communications ecosystem, we help you avoid, and quickly find and resolve performance issues in real-time – across your on-premises, cloud or hybrid environments.
For further insightful information on network performance complications, download our additional guides on Packet Loss and Latency:
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