Have you ever been on a VoIP call and could hardly make sense of what the other person was saying? Do long unnatural silences, jumbled up words, or missing phrases ring a bell at all?
Then you’ve experienced VoIP jitter.
Ironically, VoIPs vulnerability lies in its key strength – the reliance on the internet instead of traditional Publicly Switched Telephone Network.
When the internet is not up to speed, interruptions can severely damage the quality of your conversation. These can vary in their severity, however, whether it will be just a momentary frustration or a failed attempt at delivering a sales pitch – why take a chance anyway?
How Do VoIP Calls Work
A few basic components make calling over the internet possible. Before understanding the causes of jitter, it’s a good idea to look into what makes VoIP calls successful in the first place.
Data Packets
When on a call, your voice is converted into smaller chunks called data packets that travel over the internet to get to the receiver.
Typically, a single packet consists of 10, 20 or 40 milliseconds of audio and the speed at which these packets are transferred is determined at large by the transport protocols.
Transport Protocols
There are two protocols involved in transporting data packets from sender to receiver – User Datagram Protocol (UDP) and Transmission Control Protocol (TCP).
- UDP – reduces transit time. In other words, the focus here is on the interval at which the data packets are being sent out and received.
- TCP – ensures that the receiver reassembles data packets in the correct order. This is necessary so that the receiver can make sense out of the information sent over in the data packets.
As these protocols depend largely on the available bandwidth, they are frequently involved in call congestion issues.
Bandwidth
Bandwidth describes the number of data packets that can travel given the parameters of your internet connection. This means that if you exhaust the available bandwidth, the queuing delays are expected and most likely will result in jumbled conversations.
Bandwidth problems are typically provider issues. Keep in mind that on average you need around 90 Kbps for the best quality and if you’re getting anything below 30 Kbps, you’re likely to experience interruptions.
Causes of VoIP Jitter
The traffic congestion and network interference cause jitter and incoherent conversations. In some cases the root of the problem is difficult to identify, but as a rule of thumb – anything that affects the quality of your connection directly affects the quality of your call.
Usually, jitter is caused by:
Configuration Errors
If the VoIP system isn’t configured to reduce transmission and faulty queuing issues, there’ll be call quality issues. Router misconfiguration where voice and data are routed over the same network can also cause transmission delays.
Faulty queuing problems are usually due to a combination of the transport protocols. The two commonly used options are the IP RTP Priority and Low Latency protocols. Keep in mind, only one should be used across the network at a time.
Network Congestion
When there are too many devices trying to use the same network, it causes overcrowding, and data packets begin to get delayed or dropped because the available bandwidth can’t accommodate all the devices, resulting in high jitter.
Wireless Connection
Aside from network congestion, wireless networks have an additional layer of vulnerability. VoIP calls need a stable connection to proceed without issues, and you won’t always get that with a WiFi service.
Using a wireless network may be risky, as signal fluctuations can interrupt your calls or make them completely inaudible.
Bad Hardware
If you need ethernet cables, modems, routers, or any other physical devices to establish and maintain an internet connection, the hardware can also become outdated or broken. Faulty equipment can result in poor call quality.
How to Measure VoIP Jitter
To assess the extent of the experienced interruptions, you can always measure it. There are are two ways in which you can go about it:
1. Online Tools
There are various online tools that provide information on bandwidth, latency, and jitter within seconds.
These online tools can help you to confirm high bandwidth conditions, call quality reports and even help with troubleshooting.
Different tools come with unique features, so the best way is to dig a little deeper and find which one fits your needs best.
2. Advanced Monitoring Tools
These tools monitor inbound and outbound traffic, provide more accurate metrics than the online tools and are usually used by VoIP providers and companies with heavy internet call usage. Some of these also make use of IP SLA UDP operation for measurement.
LogicMonitor, Cisco DNA, Dynatrace are just a few examples of these advanced tools.
What Is Acceptable Jitter
Every single network is prone to jitter, and while it might not be possible to completely eliminate it, it’s generally recommended that it should be kept at levels below 30ms to prevent any interruptions during transmission.
So if the max value doesn’t exceed 30ms, call quality won’t be affected as it’s within acceptable norm. However, if you are picking up on the signs that it could go beyond that, our advice to you – look into improving network performance as soon as possible.
How to Fix Jitter Issues
Often, troubleshooting these problems isn’t entirely in your hands, as some of them have to do with the internet provider. But, there are still a few things you should look into, before contacting your provider to resolve the issue:
1. Use a Jitter Buffer
A buffer can help you reduce the max value to acceptable levels.
When data packets queue, the buffer collects them and ensures they’re delivered at the receiver destination.
This would usually work for minor problems, but if these exceed the tolerance levels, increasing the buffer size will improve performance.
2. Replace Outdated Hardware
Bad hardware causes congestion issues, so improving these devices would improve the quality of your calls.
You can start by upgrading the ethernet cables, routers, and other devices that make up your internet connection to handle calls.
3. Deploy Quality of Service (QoS) Prioritization
This involves limiting the number of non-essential devices connected to the network saturated with voice traffic, and it ensures your bandwidth doesn’t get exhausted with non-voice traffic.
You can do this by running a test to see the different traffic segments of your connection and assigning all VoIP traffic the highest priority.
Final Words
Jitter is probably the most frustrating of all VoIP system issues that can hamper call quality, but the fixes in this post can help you with that. It is important to remember that some of the causes have to do with the provider alone, and securing a stable connection should be a priority if you are relying on VoIP technology.
Feel free to check out the other resources on VoIP-info.org to help you with getting set up on a provider that actively reduces these problems.