How to do a Network test

If you experience issues with the call quality in Dixa, we advise you to run a network test to measure the connectivity between your network and the telephony platform Dixa uses for phone communication. The network test is applicable in case you experience phone quality degradation or if you are not able to make calls within Dixa.

This guide explains how to run a network test and how to analyze the results. You can always send the test results to friends@dixa.com and we’ll analyze them and advise you on which actions to take in order to improve your network.

Running a network test

Do the network test on the computer that is connected to the network you use for Dixa and do it closely after or during the phone call having the quality issue. In other cases, test results may not be relevant as network connectivity may change.

Please note that a conventional speed test is not enough to estimate the quality of your network using VOIP.

Step 1. 
Do the network test here: https://networktest.twilio.com/ on Google Chrome browser (it may not work on a browser other than that) and allow the webpage to access your microphone (the system will perform an automatic test call to estimate the quality network can provide).

Step 2. 
Wait until the test is fully completed. Usually, it takes around 1 minute. The test is complete when "Log Output" on the right-hand side of the screen is complete and is not spinning anymore. Copy the entire log output on your right. You can copy the entire screen and paste it to your email for us or make several screenshots but please note that we need all the information from the test.

Analyzing the results 

Below you can see what we check in order to assess network connection quality (a wired connection is a huge plus, no VPN):

Jitter should be below 30 ms
Jitter is the measure of variability at which packets arrive at the endpoint. High jitter can result in audio quality problems on the call, such as crackling and choppy audio. Jitter should be below 30ms for the best sound.

Round-trip-time (RTT) below 400ms
Round-trip-time is the measure of latency in the network. Higher latency can result in perceptible delays in audio. RTT should be below 400ms.

Packet loss below 1%
Packet loss is measured as the percentage of packets that were sent but not received at the endpoint. High packet loss can result in choppy audio or a dropped call. Packet loss should be below 1%.

Mean Opinion Score (MOS) above 3.5
Mean Opinion Score is a measure of the overall network conditions that affect call quality. MOS should be above 3.5.

Lowest bandwidth
The lowest bandwidth should be more than 3000 kBits/sec for companies with 15+ agents (100-150 per agent min).

UDP connection established 
Make sure there’s an established UDP connection from your network. If not, please make sure to open UDP ports. Read more here in the section "Twilio Client WebRTC 1.x (Twilio.js) Port Requirements".

Please read this article if you want to troubleshoot the network or reach out to us if you have questions or need help troubleshooting poor call quality. 

Red errors in establishing connections
Suppose you get several red errors on establishing various connections like the below screenshot. In that case, there is a very high chance that it is due to the network firewall blocking the IPs for our subcontractor Twilio.

This external guide from Twilio explains how to resolve this.