Not only is jitter frustrating for call quality, but it can also lead to customer dissatisfaction and revenue loss. How can it be measured and reduced? Find out in this VoIP jitter guide. Subspace, because it’s designed differently, is far better suited to these more demanding and emerging applications. Decide for yourself by trying Subspace for free now
Estimated read time: 7 minutes
“– is – the most – things that can happen – voice call.”
Let’s try that sentence again.
Jitter is one of the most frustrating things that can happen on a voice call.
Also sometimes called stuttering or ping spikes, jitter occurs when packets of data that flow between parties on a VoIP call don’t reach their destination as expected.
There’s lots of information that gets exchanged between devices over the course of a voice call. This data is divided into small pieces of information, known as packets, which are transmitted over the internet in a continuous stream, each taking their own path. When these packets reach their destination—the other device involved in the call—they are reassembled in order and converted into an audio signal the receiving party can hear via a digital signal processor (DSP).
While this may sound pretty straightforward, at any given time, there are millions of packets of data all flowing across the internet, trying to reach their destinations. This means that the journey doesn’t always go as planned. Some packets may be delayed and arrive out of order, and some may never reach their destination at all.
To fix jitter, you’ll need to improve how packets move between the VoIP call’s endpoints.
What Are the Overall Impacts of Jitter?
If only a small amount of data is lost or delayed during a VoIP session, it’s generally not a problem. The DSP can interpolate what was lost and fill in the missing data, with no noticeable impact on the receiving end. But if packet loss or delays exceed the DSP’s capabilities, it will affect call quality.
The uneven transmission of data packets is technically known as packet delay variation (PDV), but a user experiences it as jitter, stuttering, or ping spikes.
What does jitter sound like? Unfortunately, just about anyone who has ever made a VoIP call probably knows all too well what it sounds like: dropped words, jumbled sentences, and other annoying audio problems.
Anyone who has ever tried to have a conversation while experiencing a high level of jitter knows how frustrating this problem can be.
With more people working from home, including remote call center reps, jitter is more than just an annoyance. For example, if businesses use VoIP to support phone interactions with customers, calls plagued with jitter can erode customer satisfaction and phone agents’ ability to effectively close sales and resolve customer queries. (Don’t think that’s a big deal? Today, 96% of consumers will consider switching to another business after a bad customer experience
.) Adding to the business risk, jitter can also make it difficult for employees to work together and collaborate remotely.
While the occasional lost or delayed packet won’t likely cause disruption, there’s a threshold at which jitter becomes noticeable. As a rule of thumb, less than 1% packet loss won’t cause dropped words or other call quality problems. Jitter is also less likely to be an issue as long as individual packet delivery delays are less than 30 milliseconds (according to Cisco), and overall network latency is less than 150 milliseconds.
How to Measure and Test Jitter
Because of the potential impacts of jitter, it’s a good idea to understand what’s going on with your voice apps.
Jitter can be measured using the average packet-to-packet delay time. This is done using proprietary or open-source software that analyzes various quality of service and VoIP call metrics that identify potential contributors to jitter, such as network congestion and packet loss. Some examples include SolarWinds Network Jitter Monitoring
and the open-source StarTrinity Jitter and Packet Loss Test Tool
The analytics provided by some networking products, such as Cisco IOS voice gateways and routers, can also be used to diagnose excessive jitter.
But regardless of what tool you use to test jitter, your ability to fully illuminate what’s happening on your VoIP calls depends on whether you have control over one or both endpoints that are involved.
One endpoint. If you only have access to performance data from one endpoint, you’ll have to conduct a ping jitter test to measure and work out both the mean and minimum round-trip times for a series of packets (again, keeping in mind the thresholds outlined above at which jitter becomes problematic). Round-trip time (RTT) is the amount of time it takes a packet to travel from the source to its destination and back again.
Two endpoints. With two endpoints, you’ll have greater visibility into how packets are exchanged. In this case, you’ll use instantaneous jitter measurement, which is the variation between transmission and receipt of each packet. This can then be used to determine average instantaneous jitter on a call.
How to Reduce Jitter for VoIP
If you’ve ever experienced jitter, then we probably don’t need to tell you it makes for a bad calling experience. And you’re probably wondering how to improve latency and consistently reduce jitter on all your calls.
There are a number of tactics that are helpful for reducing jitter on VoIP calls.
Optimize your bandwidth. Think of how much faster you can reach your destination on a road trip when the highways are clear. Similarly, you can reduce latency—and jitter—by deploying practices that reduce traffic by limiting network usage and reducing bandwidth hogs. These may include:
- Blocking or limiting streaming sites
- Throttling cloud backup applications
- Using hosted filtering
- Centralizing application/OS updates
Use an ISP that provides enterprise-grade VoIP services. Many consumer-grade ISPs aim to deliver a great web surfing experience. But while they may all generally support the protocols involved in making VoIP calls, they don’t necessarily prioritize this type of traffic. A business-class high-speed solution may provide calling that’s more consistently jitter-free.
Use a router that supports packet prioritization. It may also be that your router isn’t up to the task of providing low-latency, jitter-free calling. This can be improved with a router that offers packet prioritization, so that the data associated with voice calls is given priority—and thus reaches its destination faster—than other types of traffic.
Deploy a jitter buffering device. Another way to keep calls flowing is by installing a jitter buffer on your VoIP system. While a jitter buffer doesn’t help to improve latency in transit (in fact, it slows VoIP traffic down), it improves call quality by holding packets until they arrive, putting them back in order, and retransmitting them in the correct sequence. It’s important to note that a buffer won’t fix the cause of jitter (as the above tactics can). But it can help to provide clearer calls with fewer interruptions.
Adjust your hardware. There are a couple of quick hardware adjustments that can help to improve the quality of your calls. For example, running IP phone devices above the 5 GHz range may cause network interference. Ensuring phones are running in the 2.4 GHz range can help. In addition, newer ethernet cables (Cat 6 and above) are capable of transmitting data at much higher speeds, improving VoIP performance.
At the end of the day, reducing jitter is all about speed—and for the best-quality call experience, you’ll need a network that’s designed to meet the needs of VoIP.
Subspace’s network runs parallel to the Internet, giving companies a high-speed alternative for their VoIP services. This means low latency, reduced jitter, and less dropped packets.