VoIP Network Test | Network Readiness

Why is VoIP Network Readiness So Important?

Your phone system is the lifeline between your company, your customers, and your employees. That’s why reliability and quality of service are essential parts of your communications network. There are certain aspects of your current network that need to be considered to ensure that you are capable of delivering high-quality voice, video, and Unified Communications (UC) features to your office.

Below, you will find tips and tools, including a comprehensive VoIP network test, to help you understand your network and determine whether or not it’s fully ready to take advantage of a hosted phone system.

VoIP Network Test

How Do I Read the VoIP Test Results?

To the right you’ll find the top 3 factors in determining VoIP network readiness.

A good result on all of these tests does not guarantee your network will be reliable, but it can quickly detect an inadequate network. If you receive a result in these ranges, keep reading for more tips on how to address the issue.

  1. Upload Speed or Download Speed < 3Mbps 
  2.  Latency > 250ms
  3.  Jitter > 50 ms

Our VoIP specialists are here to help you get your network ready for Switchvox Cloud!

Definitions to Know

Here are some explanations of each network assessment test point and what you can do to improve your results:

Bandwidth speeds can vary widely based on your Local Area Network (LAN), your Internet connection (cable modem, DSL, T1, etc.), or your Internet provider’s network. Verify you are receiving the Internet service purchased from your provider.

In order to carry voice traffic over your Internet connection, it may require you to upgrade your connection or change providers in some circumstances.  

Delay occurs when there is a higher than normal latency on your network. Latency is the amount of time it takes between one person saying something and the other person hearing what was said. Latency doesn’t typically cause audio quality issues, but when the latency is over about 150ms, the delay is noticeable to your users.

If you have a latency issue, simple network tuning can solve the issue.

Packet loss is the failure of one or more transmitted packets to arrive at their destination. This event can cause noticeable effects in all types of digital communications. With audio calls, packet loss can provide jitter and silent gaps in communication.

Jitter occurs when voice packets arrive with varying delays. This is typically caused with changes in network traffic and can usually be fixed by using Quality of Service (QoS), or reducing the amount of traffic that your network equipment is handling. Users will often report jitter as poor audio quality.

If jitter is too high, you’ll have to fix the network underneath to improve the call quality.

Quality of Service (QoS) is a feature available in quality managed network switches that allows you to prioritize your voice traffic. This allows you to ensure that your phone calls are going to get the bandwidth needed regardless of what else is happening on the network.

QoS is a very important feature to ensure the quality and reliability of IP phone systems.

Now that you’ve tested your network, explore your options with Switchvox Cloud.

Resources

10 Reasons Why Hosted UC is Risk-Free with Switchvox Cloud eBook

Making the switch to hosted Unified Communications? Find out how Sangoma’s award-winning Switchvox Cloud can make your deployment easy and risk-free.

Network Readiness Guide

Is your network ready for an IP phone system? Find out why your communications quality is dependent upon your network and what you can do to get your network ready to support high-quality UC.

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