Networks can suffer from several different ailments, each having an effect on data transfers. The main ones are delay or latency—two close cousins that are so similar that many consider them to be one and the same, packet loss and jitter.
While it is quite obvious what impact delay or latency can have on network communications and it is also easy to fathom what packet loss can cause, jitter is a more complicated concept. It is, however, what we’re going to be discussing today and while we have no intention of making jitter you experts, our hope is to give you enough information to better understand what it is, how it can affect your network, and what can be done to prevent or reduce it.
We’ll start our discussion with some basic information about jitter. We’ll do our best to explain what it is and how it happens. We’ll then discuss the adverse effects of excessive jitter and how much of it should be tolerated. Then, we’ll review the specific case of VoIP traffic as this is one of the common network services that is most affected by jitter. This will be followed by a discussion on measuring and monitoring jitter with a short review of some of the best tools available for that purpose. Our next order of business will be preventing or reducing jitter. We’ll see how jitter buffers and QoS packet prioritization are two of the best options.
So that we all start on the same page, let’s first define a couple of important terms. In fact. It’s not so much the terms themselves that are important but that we are aware of their differences and similarities. They are the four primary ailments of networks that we listed earlier.
Packet Loss is simply the failure of one or more packet to reach their destination. It is relatively frequent but its effects are mitigated by the built-in error correction of many network protocols.
Delay is the time it takes for data to go from one point to another. It is a factor of the distance between the source and destination and the speed at which data travels.
Latency is any extra delay added to the data transmission for various causes.
Note that delay and latency are often used interchangeably as they both pertain to the time it takes for data to go from its source to its destination. Furthermore, some people will tell you that they are two different names for the same concept
As for jitter, it is a fluctuation in a network’s latency. Let’s dig deeper…
Jitter is the term used to refer to the fluctuation in delay as packets are transferred across a network. It is the changing rate of delay across a network. Let’s clarify what jitter is with an example. Suppose we have two computers communicating with each other on the same network. There is a constant exchange of data packets between the two computers. In the absence of jitter, each packet will take the same time to traverse the network. Let’s assume it takes 10 ms.
On a network affected by jitter, the transit time of each packet would fluctuate. One packet would arrive in 10 ms while the next one would arrive in 50 ms and the third one in 15 ms. In this specific example, the third packet could arrive before the second. Jitter is measured in milliseconds (ms) or thousandths of a second and the figure represents the difference between the fastest and the slowest packets. In our example above, jitter is 40 ms. In many situations, that wouldn’t be a problem and the recipient would be able to reorder the packets and make sense of them.
Some protocols, however, don’t deal too well with jitter. It is, for instance, the case of any real-time protocol such as streaming video or Voice over IP, the technology behind many modern business telephone systems.
The Adverse Effects Of Jitter
The effects of jitter vary according to the specific service. For instance, file transfers have built-in reliability and will either reorder packets correctly or request a re-transmission if something is out of sequence or missing. Other services such as VoIP telephony and video streaming will be much more affected. The effects of jitter are the greatest—or at least the most noticeable—on real-time user services such as IP telephony.
The goal of Voice over IP is to achieve at least as good call quality as is obtained with standard telephony. You expect everything you say to be crystal clear to the other party and vice-versa. This means that the arriving audio packets must be maintained in sequence in order for the speech to remain intelligible. Anything less than real-time signal delivery will result in a conversation with chopped up audio. Actually, skips in audio quality and shaky audio signals are often this first tell-tale signs that excessive jitter is present on your network.
How Much Jitter Is Too Much Jitter?
Jitter is as much a fact of life of networks as taxes are for us. There’s simply no way around it. It should, however, remain within some acceptable boundaries in order not to cause any issues. A small degree of jitter won’t have a noticeable effect on even the most sensitive real-time data flows. For a good measure of what level of jitter is acceptable, let’s turn to Cisco. After all, it is the number one maker of networking equipment so the company should know what it’s talking about.
So, here’s what Cisco suggests as acceptable levels of jitter and, while we are there, what it suggests for packet loss and latency too.
- Jitter: <30 ms
- Packet Loss: <1%
- Overall network latency: <150 ms
For the best performance, you should try to keep jitter below 20ms. Anything above 30ms will have a noticeable impact on the quality of any real-time VoIP conversations you have. And when it goes too high above 30 ms, you will likely experience distortion that could make the other user more difficult to understand. By keeping these three key performance metrics below their respective thresholds, you can ensure that important services on your network don’t experience severe performance issues.
The Specific Case Of Voice Over IP
VoIP telephone is arguably the service that is most affected by jitter. Actually, video or audio streaming is affected just as much but, the relative importance of these types of services is not as high. This has to do with the way VoIP data transfers occur. When you speak into an IP phone, your voice is converted into data which is transmitted via the network. The voice data is broken down into many different packets then transmitted to the caller on the other end.
That voice data, while it is in transit on the network, is competing against all other traffic. Especially in situations where there is over-utilization of network circuits, packets could be delayed. This delay may not be obvious when transferring a file which could end up taking only a few extra seconds. But when voice traffic suffers delays and jitter, packets can get out of sequence, resulting in some major distortion of the voice signal.
About Monitoring Tools
Network monitoring tools are by far one of the best ways one can keep your eye out for the occurrence of jitter. A decent network monitoring tool will be able to tell you when the network is suffering—or is about to suffer—from jitter. It can also help you to see when you are about to exceed your current bandwidth limits.
It’s easy to see the importance of monitoring for jitter. It is the best way to ensure you react as soon as a problem arise. In fact, you’ll often be able to intervene before jitter gets so high as to cause a degradation of the service. Your monitoring tool can also give you some context about any performance issue so that you can perform informed troubleshooting. By giving you the ability to see network jitter emerging, the monitoring tool will make your efforts to solve and prevent jitter more effective.
The Best Tools For Measuring Jitter
Now that we’ve got you convinced of the usefulness of jitter monitoring tool, let’s have a look at a few of the best ones. None of our tools is solely dedicated to measuring jitter but all of them will do it. All of these tools have vastly different purposes and the best one for your specific needs is the one which has features that can be helpful to you. We’ll let you be the judge of that. But rest assured that all the tools reviewed here will do a fine job of measuring jitter.
SolarWinds has been making some of the best network administration tools for the past 20 years or so. Its flagship product, the Network Performance Monitor, consistently scores among the best SNMP network monitoring tools. The company is also famous for its free tools. Made to address specific needs of network administrators, they include products such as the TFTP Server or the Advanced Subnet Calculator.
The SolarWinds VoIP and Network Quality Manager is a dedicated VoIP monitoring tool that is packed with great features. This tool can be used to monitor VoIP call quality metrics including jitter, of course, but also latency, packet loss, and MOS. It can help troubleshoot VoIP call performance by correlating call issues and network performance. Real-time WAN monitoring is using Cisco IP SLA technology is also included. A VoIP call path trace feature lets you see and pinpoint call problems along the entire network path.
- FREE Trial: SolarWinds VoIP and Network Quality Manager
- Download Link: https://www.solarwinds.com/voip-network-quality-manager/registration
This tool provides real-time monitoring of site-to-site WAN performance and it also has alerting features to notify you of any abnormal situation. It can help ensure that WAN circuits are performing as expected by utilizing Cisco IP SLA metrics, synthetic traffic testing, and custom performance thresholds and alerts.
But the SolarWinds VoIP and Network Quality Manager won’t only monitor your WAN circuits, it can also display the utilization and performance metrics of your VoIP gateways and PRI trunks. It can help with capacity planning by allowing you to evaluate voice quality when planning new VoIP deployments.
Prices for the SolarWinds VoIP and Network Quality Manager start $1 615 for up to 5 IP SLA source devices and 300 IP phones. Other licensing levels–including a device-unlimited license–are also available. A free 30-day trial is available should you want to take the product for a test run.
2- PRTG Network Monitor
The PRTG Network Monitor from Paessler is a multi-purpose network monitoring system. Through the use of sensors, which can be compared to add-ons although they are included with the product, PRTG can be used to monitor many different parameters of networks and systems. The tool can monitor virtually any system, device, traffic, and application in your IT infrastructure.
Of particular interest in the context of this discussion, the tool includes a Ping Jitter Sensor designed to measure how much jitter is impacting your network. Other relevant sensors include a QoS Round Trip Sensor and a QOS One Way Sensor. For IP-SLA enabled Cisco devices, an IP-SLA sensor reads relevant metrics. Both methods show you the quality of your VoIP connection and enable you to define what level of latency, jitter, etc. are acceptable. You can choose to be notified via email, SMS or push notifications on a mobile device whenever the threshold is exceeded so you can take appropriate measures.
The PRTG Network Monitor is super easy and quick to install. The tool’s auto-discovery system will scan network segments and automatically recognize a wide range of devices and systems. It will then create sensors from predefined device templates. Specific VoIP sensors sometimes need to be manually set up afterwards, making the installation a bit longer but this is still one of the fastest tools to set up.
The PRTG Network Monitor is available in a free, full-featured version limited to 100 sensors. Note that any single monitored parameter counts as one sensor. To monitor more than 100 sensors, you’ll need a license. Prices vary with the number of sensors and start at $ 600 for 500 sensors up to $14 500 for unlimited sensors. A free device-unlimited 30-day trial version is available.
3- ManageEngine OpManager With VoIP Monitor
The ManageEngine OpManager is another excellent network monitoring tool. It will monitor the vital signs of your equipment and alert you as soon as something is out of specs. The tool features an intuitive user interface that will let you easily find the information you need. It also features an excellent reporting engine along with some pre-built and custom reports. To complete the package, the product’s alerting features are also very comprehensive.
When it comes to monitoring for jitter, the ManageEngine OpManager‘s VoIP monitor option can proactively monitor and report on your infrastructure’s capacity to handle VoIP calls. The tool uses Cisco IP SLA to continuously monitor critical Quality of Service parameters of VoIP networks. The monitored VoIP parameters include packet loss, delay, jitter, the Mean Opinion Score (MOS) and Round Trip Time (RTT).
The ManageEngine OpManager is priced based on the number of monitored devices. Prices range from $715 for 25 devices to $14 995 for 1 000 devices. The VoIP monitor option adds $125 per device that requires it. A free 30-day trial is available so you can try the product and see how it fits your specific needs.
VoIPmonitor is an open source network packet sniffer with a commercial front end for monitoring most VoIP protocols. It runs on Linux and is designed to analyze the quality of ongoing VoIP calls based on network parameters such as jitter and packet loss according to the ITU-T G.107 E-model. Call information, along with their metrics, is saved to a database. Each call can be saved to a pcap file for further analysis with external tools such as Wireshark.
VoIPmonitor can also decode speech and play it over its web-based GUI as well as save it to disk as a .WAV file. Out of the box, the product supports the G.711 alaw and ulaw codecs and commercial plugins add support for G.722, G.729a, G.723, iLBC, Speex, GSM, Silk, iSAC, and OPUS. VoIPmonitor is also able to convert T.38 FAX to PDF.
The VoIPmonitor GUI front end is available either as a locally hosted server at prices ranging from $42/month for 10 channels to $917/month for 6 000 channels or as a cloud-based service with prices varying from $20/month for 3 channels to $200/month for 200 channels. Both versions are available in a free and unlimited 30-day trial.
Preventing Or Reducing Jitter
It’s one thing to measure jitter but, once you find it, you need to do something about it. Let’s have a look at two of the most popular ways jitter can be prevented or reduced. Both techniques have the extra advantage that will also address latency issues.
A jitter buffer is a device that is used to counter delay or latency variances by storing arriving packets for a short time period before passing them on to their destination. They are typically set to buffer traffic for 30 – 200 ms, depending on how much jitter has been measured. This buffering results in less jitter and a conversation that stays intelligible to both parties. A jitter buffer holds data packets before sending them and can ensure they arrive in proper sequence. The end result is minimized jitter and less adverse effect on VoIP call quality.
By their very nature, jitter buffers will increase the overall delay present across your network. By holding back packets a jitter buffer is adding latency to the service. You need to be careful about setting up jitter buffers when implementing full-duplex communication. Another problem with jitter buffers is that they are a band-aid solution. They don’t address the cause of the jitter, they only address the symptoms. For that, what you need is to set up Quality of Service (QoS) in your routers.
Packet prioritization is a type of Quality of Service setting where you prioritize certain types of traffic in order to reduce network congestion. Your prioritized traffic will have precedence over other types of traffic and be sent first no matter how much other traffic there is. Packet prioritization is generally applied to those mission-critical applications that demand real-time performance, such as VoIP.
To protect VoIP traffic you would typically prioritize Real-Time Transport Protocol (RTP) packets. How this is done will depend on the design on your router. Traffic can also be prioritized based on its DSCP marking. Differentiating Services Code Point, or DSCP, uses a six-bit code in the header of each packed to mark it according to several classes of increasing priority. Typical DSCP values range from 0, the least important traffic to 48, the most important one. Most VoIP devices such as IP phones and gateways mark all voice traffic as DSCP 46. It is then easy to prioritize that traffic in your router’s QoS configuration.