Feeling jittery? We’re not talking about that feeling you get when you have too much coffee before lunch. We’re talking about network jitter, which can make all the difference in the success or failure of your VoIP calls.
Poor call quality can literally be a deal-breaker; if a phone call fails at a critical moment, your sales agent might never get another chance to close the deal. Even when the stakes aren’t quite that high, network jitter can cause delays and inconveniences that negatively impact your customers’ experience and decrease your employees’ productivity. So eliminating network jitter is often a top priority for VoIP users.
Jitter refers to the differences in latency between packet flows from one client to another. Measured in milliseconds, it has the most impact on streaming video and audio services. Signs of jitter usually include the following:
The underlying cause of jitter is the way that data gets divided up into packets for transmission. Each of these packets can use a different path to get to its endpoint, so the data packets will all arrive at the same place–just not in the same order.
Because jitter causes garbled calls and inconsistent call quality, it’s sometimes also called “stuttering” or “ping spikes.” Meanwhile, the terms “jitter” and “latency” are often used interchangeably, but they’re not exactly the same. Latency simply refers to the time that it takes for data packets to travel from one point to another. Like jitter, it’s most noticeable with streaming audio or video.
Jitter, however, is a measure of inconsistency in latency across your network. Inconsistent latency–that is, jitter–can be just as frustrating as high latency, when it takes data a longer time to travel from point to point. While high latency causes delays, high jitter causes inconsistent call quality.
The first, most obvious sign of jitter will be changes in call or connection quality. If you’ve noticed this, it might be time to get to know your network better. This will require a network monitoring tool, and the right VoIP provider will actually do this monitoring for you. The monitor will help you understand how severe the jitter is, along with providing other network performance insights.
When it comes to your VoIP network, your testing approach will depend on whether both endpoints are part of your network.
Sound complicated? It can be intimidating for a beginner, but the best VoIP providers will handle all of this testing for you, along with any necessary bandwidth testing.
Even the most robust network might sometimes have short-term jitter that doesn’t impact call quality. This little jitter isn’t anything to worry about. The industry has some accepted standards regarding jitter:
If your network exceeds these standards and you notice lapses in call quality, you can eliminate jitter with a few simple steps.
Placed between data endpoints, a jitter buffer works by holding on to data packets and releasing them for a set amount of time. Most jitter buffers are configured to hold the data packets from 30 to 200 milliseconds. Essentially the buffer ensures that packets arrive at their final destination in the correct order. One drawback of jitter buffers, however, is that because they hold data packets, they can also increase latency.
Jitter buffers are a popular solution to improve the quality of VoIP and video streaming services, where consistency is often more important than a small decrease in overall quality. They can be deployed directly at the source of the jitter, which can be pinpointed with a network monitoring tool. However, once you’ve identified the underlying cause, it might be preferable to fix that issue.
If network congestion is the cause of your jitter, packet prioritization is an excellent way to boost VoIP call quality. Using this method, certain types of data always get transmitted first, reducing network congestion.
Which traffic should you prioritize? That depends. Usually, organizations implement packet prioritization when they need reliable services that require constant high performance. If you prioritize VoIP calls, for instance, other, non-prioritized data won’t get delivered until all your VoIP data has been transmitted.
Because packet prioritization requires real-time transport protocols, every router might require a different procedure and configuration. Work with your VoIP provider to ensure that everything is set up as simply as possible and that nothing is misconfigured.
Most of us have an old flip phone lying around the house somewhere, or perhaps even a fax machine. And you probably wouldn’t consider using either these days, when there are much faster options available.
On the other hand, do you remember the last time you upgraded your network hardware? Outdated routers and switches or even old cables can cause data transmission issues. Your VoIP provider can help you identify hardware that needs to be upgraded to reduce or eliminate jitter.
If you’re experiencing jitter, latency, or other call quality issues, we’re happy to help! Contact us to learn more about how you can maximize your VoIP communication system.