VoIP Speed test
Start the VoIP network test to find out if your Internet connection and hardware meet the requirements needed to ensure high call quality with Aircall.
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.
The MOS (Mean Opinion Score) is an automated score between 1-5 of your experienced audio quality, represented as follows:
Bad
Poor
Fair
Good
Excellent
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.
Gathered candidate of Type:
Protocol:
Address:
Data successfully transmitted between peers.
Audio track created using device
Gathered candidate of Type:
Protocol:
How to interpret your VoIP test results
Your internet connection will impact the quality of your phone calls. It is important for you and your business to ensure that you have a strong download and upload rate. The best way to do this is to test your VoIP speed, to ensure that your network can support your team. Better network conditions means clearer, more reliable calls with your customers. The slower the speed, the higher the chances of having a call interrupted by network issues. Speed is not all though, and other factors in your connection will impact the quality of your phone calls as well. For more details on your results, please see below and visit How to interpret the Network Diagnostics results
Connection Throughput
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 64kbps.
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.
Available bitrate
The available bitrate is the measure of data transfer capacity on your internet connection within a specific timeframe.
It signifies the maximum speed at which voice data can be sent and received. A stable and adequate available bitrate is essential for optimal communication quality.
Ensure that your network maintains a reliable and recommended bitrate to prevent issues such as call interruptions or audio distortions.
Adjustments to your internet connection or network configurations may be necessary if the available bitrate falls below the recommended level for seamless call quality.
RTT
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.
Packet loss
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
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.
MOS Score
The MOS (Mean Opinion Score) is an automated score between 1-5 of your experienced audio quality, represented as follows:
Bad
Poor
Fair
Good
Excellent
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.
Elements that impact your network checker test
Are you experiencing issues with your internet connection? These issues may be your solution.
Internet Service Provider (ISP)
Your choice of internet service provider matters. Make sure you choose the right vendors based on your overall business needs. You need to make sure you have the right bandwidth to serve the amount of users who will be using the internet on a daily basis. Make sure your ISP can support a high speed environment depending on your needs.
Type of Internet Connection
Internet technology has developed quite a bit since the days of dial-up internet service (this type of service is not recommended). Today the most typical types of internet connections used by businesses are DSL (Digital Subscriber Line), ADSL (asymmetric digital subscriber line), VDSL (very high-speed digital subscriber line), wireless, fiber optics and cellular. Each type of connection comes with a different speed - be mindful of the download and upload speeds.
Strain on the Network
Like your computer, your internet connection can only handle so much activity on it. This is why the download and upload speeds are so important as they help dictate how quickly activities tied to the internet take. More importantly, how much strain they put on the network. Factors like number of users, number of applications being used and the overall hardware will all have an impact on the performance of your network. Be mindful of all of this when selecting the right provider.
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