The Aircall Network Diagnostics Panel will help you assess your internet connection. A good and reliable internet connection is essential for a good call quality experience when using Aircall.
⚠️ Please note that this feature is not available on iOS or Android apps.
How to open the Network Diagnostics section
Open the Desktop app, Web App, or CTI.
Click on your User Icon on the bottom right-hand corner to open Settings.
Select Preferences and click on Quality.
Under the Quality section, click on Network Diagnostics.
Select Run Diagnostics.
Clicking the Run Diagnostics button will execute a number of processes that simulate conditions during a voice call and analyze the data throughput.
The tests performed may take a little time, but once complete, they will output a number of values pertaining to each identifiable characteristic of your network quality. Here, values that are presented in green are considered as being within an ideal range for voice calls, while values in yellow or red may identify a potential concern.
How to interpret the Network Diagnostics results
The MOS (Mean Opinion Score) is an automated score between 1-5 of your experienced audio quality, represented as follows:
This is calculated based on the values of latency, packet-loss, and jitter for the duration of the call.
A value between 4.0 and 4.3 is desirable, over 4.3 is ideal. If the MOS score is 4 or less, you might face some issues understanding the other person during a voice conversation.
Bandwidth represents the available throughput of your internet connection, consisting of download and upload speeds.
The measurement shown here is your available bandwidth for calls over Aircall. To ensure that your network supports good call quality, it needs to provide at least 2Mbps.
If your bandwidth is not sufficient, this may result in latency, jitter and/or packet loss during a call. Reducing the number of running applications that use the internet may help.
TURN is a technology that acts as an intermediate between a user’s device and another device or server.
If the result of the test is ‘NO’, there are no issues expected. A ‘YES’ may also be acceptable, but it could suggest that the user’s firewall or network configuration is affecting the connection.
If your network is using TURN, be sure that you are not restricting UDP packet transmission on your network. Failure to do so will likely result in increased latency during your calls.
Aircall uses essentially two protocols for voice conversation: UDP and TCP.
UDP is the preferred protocol for voice calls, it is faster and more efficient.
However, if the result is TCP, consider doing the following to reduce overall latency during a call:
Update your network or firewall configuration to allow UDP.
Disconnect from your VPN if you are using one.
Try to turn off any antivirus or content blocker software.
This refers to the network connection type - Wifi, Ethernet, or Cellular - used by your internet connection.
We strongly recommend a connection via Ethernet cable as Wifi is less reliable.
When you are using Wifi, you could be exposed to several issues as increased packet-loss or jitter can be caused by:
Your distance to the router
Any physical obstacle between your device and the router
The amount of devices connected to the router
The selected Wifi channel (you should select a channel with low usage)
The selected Wifi band (you should select 5GHz for short distances and 2.4GHz for long distances)
Audio data is transmitted across the internet in small data blocks called packets. Some of these packets may get lost and never reach the recipient impacting call quality considerably.
The ideal rating for packet loss is 0%. A value of 1% or more should be investigated.
Please ensure you have a reliable connection to the internet, using a good quality Ethernet cable, and avoid using a VPN.
If you’re already using an ethernet cable, please try a new cable or a new port. If the issue persists, try changing your router, or directly contact your ISP.
Jitter is the variance of latency in time. When jitter is high it means your latency is not stable.
To ensure good call quality, the jitter level should stay below 30 ms. Values above this level may incur call drops, audio gaps, audio distortion, audio delay, echo, or other call quality issues.
Jitter can be caused by a number of means, including;
Using a wireless internet connection.
Congestion within your router due to heavy traffic on your network.
A generally unstable internet connection through your Internet Service Provider (ISP).
Packets traversing multiple pathways across the internet, which can occur when using a VPN.
If you are experiencing issues with Jitter, please ensure you have a reliable connection to the Internet using a good quality Ethernet cable and avoid using a VPN. If needed, contact your internet provider.
Round trip time
RTT (Round Trip Time) is the representation of the network latency for outgoing and incoming packets. This is the time it takes between a request being sent by a user and its response being received (round-trip).
To ensure good call quality, RTT should be 300 milliseconds or less. Higher values will result in noticeable delays in the audio (for example, the time between you speaking and the other user receiving the audio on their end).
Latency could be caused by using a TCP network type, a VPN, or by network congestion.
If you are experiencing latency issues, please ensure you have a reliable connection to the internet using a good quality Ethernet cable, a good quality router and sufficient broadband.
When carrying out intensive processes, such as bidirectional voice calls, it is important to ensure you have enough processing power.
If you have a “Poor” result on this component, you might be using a device with a low performance processor. To use Aircall, an average machine running minimal applications may require at least 2.5 Gigahertz of speed. Check our minimum requirements in this article: Device & Headset Recommendations.
Running the Aircall application with too little processing power could cause the application to periodically freeze, perform sluggishly, or cause audio quality problems during calls.
Performing bidirectional voice calls on a device can use a lot of memory. Besides executing the application itself, the device also needs to encode and decode the audio data, which means storing the audio in memory so that it can be processed.
If you have a “Poor” result on this component, you might be using a device with low RAM. To use Aircall, an average machine running minimal applications may require at least 4 Gigabytes of RAM. Check our minimum requirements in this article: Device & Headset Recommendations.
Be aware that if the result for any component is NaN (Not a Number), that means Aircall was not able to test and retrieve values for that parameter. If this result is consistent, you might be blocked by a firewall, VPN, anti-virus, content-blocker, or similar components.
Try to disconnect from your VPN. If the issue only happens when using the VPN, your company needs to follow Aircall’s Network Requirements and Recommendations;
If you’re working from the office, your company may have a firewall setup. A firewall may be blocking some of the Aircall traffic. Your team should follow Aircall’s Network Requirements and Recommendations;
Try to turn off any antivirus or content blocker software. If the issue is one of those pieces of software, please reach out to your IT team.
Still in need of assistance? Please feel free to reach out to the Support Team and we will be happy to assist!