This in-depth guide will walk you through everything you need to know if you’re facing jitter issues or want to learn more about jitter in general. To find exactly what you’re looking for, click on a line in the table of contents to navigate to the section you need. Alternatively, you can also scroll through to read the bullet-pointed summary at the bottom of each section (highlighted in red).
Table of Contents
- The technology behind jitter
- What is jitter
- Why jitter matters
- How much jitter is acceptable
- How is jitter measured
- How to fix jitter issues
‘Jitter’ is a problem that can impact any online activity, and generally it can mean that your Internet connection suffers from poor sound quality. This issue is of utmost importance when considering Voice Over Internet Protocol (or VoIP) because superior sound quality is a primary selling point to such a service.
In this article we will discuss how jitter can occur with anytime you’re dealing with an Internet connection or in a situation where you’re sending large data streams across a computer network (this can include online gaming, network communications, etc.). The principles we’ll talk about here apply to any technology where high jitter is an issue. Specifically, we will talk about how it affects VoIP, a technology that allows telephone calls to be transmitted over the Internet.
Lets start with discussing how jitter works.
Data packets are the backbone of online communication and is defined by Techopedia as a unit of data made into a single package that travels along a given network path.
For example, every time you log onto a website information is sent to request access to a specific page. In response to that request, data is sent backwards and forwards across thousands of different virtual networks.
The same applies for every email you write and every telephone call you make using VoIP. The networks that exchange all this information and allow you to communicate online are known as packet switched networks.
An email, for example, can be broken into any number of packets depending on its size and the type of media being sent. To give you an idea of the vast number of data packets that comprise a single email, each one might include:
- Part of the body of your message
- Details of your IP address – that is, a specific numerical code which identifies the computer the message is sent from
- The IP address of the person the message is being sent to
- An indication telling the network how many packets the message has been split into
Similarly, the content of VoIP telephone calls is broken down into tiny packets for transmission over a network. These data packets travel at high speeds. Whenever there is a delay in this process, however small, it can result in a deterioration in audio quality.
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The International Standards Organization (ISO) has a standard format for the layers of information contained in each data packet. This standard identifies the layers that make up each packet and the standards in which they must adhere.
Since data is used and shared in a wide variety of ways across countless devices, the ISO’s aim is simply to make sure that data packets can be put together, then transmitted effectively across network systems no matter the device. Once that data is split into packets, it needs to be capable of being used and accessed across a wide variety of networks.
In order to make the process as efficient as possible, each data packet will be sent through the best route available, as determined by your internet service provider. Think of this as a massive, virtual transport system.
For example, if there is a problem that causes a ‘roadblock’ and prevents part of a message from reaching its destination, in a millisecond your network will figure out the best alternative route around the problem and will make sure that the whole message is delivered as efficiently as possible.
- Data transferred over the internet is converted into data packets
- A single message can be broken up into numerous packets
- Packets then travel to their intended destination where they are re-ordered into the proper sequence
- Any delay in transmission could result in mis-ordered packets or gaps in the data
VoIP converts your voice into data so that it can be transmitted via the Internet instead of the old-fashioned way of transmitting electrical signals along massive networks of copper wires (how telephone calls were originally transmitted).
The most obvious benefit of VoIP is that it does away with the need for expensive hardware, most notably the old-fashioned telephone exchanges. Instead, your voice – like all other information transmitted across the Internet – is broken down into those tiny data packets as described earlier.
Each of those data packets then contains fragments of your voice and has to compete with the huge amounts of other information being sent back and forth over the web, like traffic during rush hour. Each packet is fighting to get to their intended destination on time. And just like every other piece of information being transmitted to enable us to use the Internet efficiently, sometimes these bits of information experience what is called ‘packet delay’ (aka a lag in packet transmission).
These tiny delays which can result from network congestion (as millions of data packets are transmitted every second) are more apparent in a ‘live’ VoIP telephone call. It means that the voice can break up, and as your speech is sent instantaneously as huge numbers of packets of data, there is no time for it to be re-assembled at the receiver’s end before they hear it.
However, your internet service provider (ISP) will try to ensure that network jitter is avoided as much as possible by sending the individual packets in a steady stream at fixed intervals. In instances such as an email, all these tiny pieces of information can be re-assembled before the recipient receives them, so as not to be aware that the messages have been broken into tiny fragments, transmitted, and then re-assembled before they read it.
That’s why your voice can sometimes sound garbled to the person on the other end of the line.
- VoIP (Voice over Internet Protocol) converts audio into data, making internet-based telephones possible
- Jitter can cause data packet delays or packet loss
- This can result in reduced audio quality
Jitter is the congestion that results when many millions of Internet connections are trying to compete with each other at the same time. This means many tiny packets of information are trying to use the same IP network.
A technical jitter definition is the variability over time of the latency across a network. Latency is the time taken for a single packet of data to pass along its route. And of course, given that each packet can travel a different route and still reach the same destination, you can begin to see why high jitter can become an issue.
The practical result of this is that some words in a VoIP call may be jumbled due to some packets taking longer to reach their destination than others. High jitter impacts sound quality of VoIP telephone calls. And in extreme cases, that can mean that a call is difficult to understand. Parts of the conversation may ‘drop out’ and make what’s being said unclear.
We at Nextiva want to reassure you that there are ways to combat, and even prevent, jitter issues. In the next sections we’ll explain exactly how much jitter is acceptable and how you can measure it.
- Jitter is the variability in the time it takes for data packets to reach their destination
- High amounts of jitter result in packet delays and lead to poor data transmissions
- VoIP jitter can easily be reduced
High jitter is a data traffic jam. It’s a nuisance.
And when you’re on the phone, it’s far more noticeable than when you’re simply navigating around the Internet. Just think of the impact that an inconsistent connection might have on the people you call. The person on the other end of the line might just start to get frustrated at not hearing the whole message. So no matter how well crafted your phone scripts are and how professionally they’re delivered, you might lose any chance you had of that call being productive.
So while network jitter most often only results in minor, such tiny glitches can mean the difference between success and failure in a call.
- Effective communication is a key to business growth and long-term success
- High jitter can result in poor business communication
When it comes to VoIP, we’ve found that a delay of 30 milliseconds or greater can result in distortion or calls momentarily dropping-out
Shorter delays than this won’t result in any appreciable loss of sound quality, but once a delay reaches 30 milliseconds (or 0.03 of a second), it will start to affect the quality of the audio heard at the receiving end.
At Nextiva, we carry out our equipment set-ups with the aim of restricting the maximum delay to 15 to 20 milliseconds. We also maintain a target for total packet loss being under 1 percent. This means that Nextiva customers can expect consistent, reliable service no matter where they’re at.
- Delays greater than 30 milliseconds can cause audio distortion
- Nextiva’s VoIP network and devices are designed to maintain a maximum delay of up to 20 milliseconds, meaning no distortion of data
Measuring jitter requires calculating the average packet-to-packet delay time, or the variation between absolute packet delays in any sequence of online communications. This is done in a number of ways depending on the type of traffic.
For voice traffic, e.g. VoiP, jitter can be measured based on whether you have control over just one endpoint, or both.
Where your network has control over just one of the endpoints (aka single-ended), it 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.
Rather than doing the math yourself, you can also determine the level of jitter you’re dealing with by performing a bandwidth test. Nextiva has a VoIP speed test tool that can calculate important factors in your internet connection’s performance such as:
- Download and upload speeds
- Jitter times
- Your network’s overall capacity
- Jitter is measured differently depending on whether you have control over a single endpoint or double endpoint
- The easiest way to test jitter is through a bandwidth test
The most annoying aspect of jitter is how it can vary in degree, even during the course of a single VoIP conversation. We hinted earlier that one of the surest ways to reduce jitter to a minimum is proper initial set-up. A correctly set-up VoIP network, like Nextiva, will include what’s called a ‘jitter buffer’.
A jitter buffer is a device installed in a VoIP system that intentionally delays each incoming data packet. That way the person on the receiving end of the call will hear the sound as clearly as possible with a minimum amount of sound distortion or delay.
A properly set-up jitter buffer will re-group data packets impacted by transmission delays. It will then play them back in a steady stream through processors that re-convert the information back to audio.
Nextiva always includes jitter buffering with its devices. This buffering will store your data packets in the right sequence and then transmit them across your network at evenly-spaced intervals. You will quickly notice that this delivers clearer voice data–closely resembling the way in which it was spoken.
With your computer plugged directly into your modem, if high jitter is found, your problem is likely to be an issue with your Internet Service Provider (ISP). If you can’t test while plugged directly into your modem, your ISP should be able to do this for you and resolve any jitter issues found at its end.
If you get good results with your computer and modem connected directly, contact your VoIP phone system provider. They’ll be able to help you troubleshoot the issue.
Once you know where you need to focus your attention to solve your jitter issues, you can let your VoIP provider recommend a solution. Common solutions include:
- Upgrade your Ethernet. The easiest fix of all – your Ethernet cable may need upgrading. The newest-generation Category 6 cables can transmit data at about double the speed of their predecessors (250MHz as against 125 MHz), so a new cable usually solves most high jitter issues with your network.
- Check your device frequency. If you have an internet phone which operates at a higher frequency than the standard 2.4GHz, this may also cause interference on your network. With connected phones that run at frequencies as high as 5.8GHz, these may be hampering your potential to minimize the jitter being experienced across your network.
- Reduce unnecessary bandwidth usage during work hours. If your staff are in the habit of using large amounts of bandwidth for non-essential and non-work related activities – such as network gaming, or quietly streaming content from Netflix or Pandora – tell them to restrict these, or ban them altogether during work hours. This will reduce jitter experienced in your more important business-related tasks. Also, setting updates for applications and operating systems to be carried out outside work times will free up capacity for more essential communications.
- Switch to Nextiva. Our network is pre-conditioned to combat jitter, making your calls clear and your business communication effective.
- To fix jitter issues you should:
- Activate jitter buffering for your network
- Consult your VoIP provider before implementing. Some providers like Nextiva pre-emptively include this with their devices
- Conduct a bandwidth test
- Upgrade your Ethernet cables from a Cat-5 to a Cat-6
- Restrict unnecessary bandwidth consumption in the office
- Consult your VoIP provider
- Switch to Nextiva!
- Activate jitter buffering for your network
There’s plenty you can do to ensure that your business is set up to get the most from VoIP telephone systems. And once you have effectively reduced jitter, you’ll have removed the only potential obstacle to clear communication.
Still having issues with VoIP? Contact Nextiva! Our phone systems are held to the highest possible standards and are designed to help your business succeed in all its business communication needs. Speak to a Nextiva sales advisor today by calling (800) 799-0600, or by emailing firstname.lastname@example.org.
Cameron Johnson is a market segment leader at Nextiva. Along with his articles on Nextiva’s blog, Cameron has written for a variety of publications including Inc. and Business.com. Cameron was recently recognized as Utah’s Marketer of the Year.