VoIP Speed & Quality Test

Check your network's latency, jitter, and bandwidth to ensure crystal-clear VoIP calls.

This tool runs a basic latency and throughput test against our servers.

Why VoIP Speed Matters

When you make a phone call over the internet using Voice over IP (VoIP), your voice is converted into digital data packets. These packets are sent across your local network, out through your ISP, and over the public internet to your VoIP provider's data center.

Unlike downloading a file or buffering a video—where a slight delay is unnoticeable—voice communication happens in real-time. If those data packets arrive late, out of order, or don't arrive at all, the conversation breaks down. You'll experience echoes, robotic voices, delays, and dropped calls.

That is why a standard internet speed test (which only measures raw download and upload bandwidth) is not enough to determine if your network is ready for VoIP. You need to measure the consistency and stability of your connection.

Understanding Your Test Results

Our VoIP speed test above measures the three most critical metrics for voice call quality. Here is what they mean and what scores you should be aiming for:

1. Latency (Ping)

What it is: Latency is the time it takes for a data packet to travel from your computer to the server and back. It is measured in milliseconds (ms).

The Standard: For a natural, conversational flow without people talking over each other, your latency should be under 150ms (ideally under 50ms). Anything over 200ms will result in a noticeable delay.

2. Jitter

What it is: Jitter measures the variation in your latency. If your ping fluctuates wildly (e.g., 20ms, then 100ms, then 30ms), you have high jitter. This causes packets to arrive out of order.

The Standard: Jitter must be kept under 30ms. High jitter is the primary cause of "robotic" sounding voices or words being scrambled during a call.

3. Bandwidth (Download/Upload)

What it is: The raw capacity of your connection, typically measured in Megabits per second (Mbps).

The Standard: VoIP actually uses very little bandwidth—typically about 100 kbps (0.1 Mbps) per concurrent call. However, you need enough total bandwidth to ensure that your voice traffic isn't choked out by other users streaming video or downloading large files.

Why VoIP Speed Matters

When you make a phone call over the internet using Voice over IP (VoIP), your voice is converted into digital data packets. These packets are sent across your local network, out through your ISP, and over the public internet to your VoIP provider's data center.

Unlike downloading a file or buffering a video—where a slight delay is unnoticeable—voice communication happens in real-time. If those data packets arrive late, out of order, or don't arrive at all, the conversation breaks down. You'll experience echoes, robotic voices, delays, and dropped calls.

That is why a standard internet speed test (which only measures raw download and upload bandwidth) is not enough to determine if your network is ready for VoIP. You need to measure the consistency and stability of your connection.

Understanding Your Test Results

Our VoIP speed test above measures the three most critical metrics for voice call quality. Here is what they mean and what scores you should be aiming for:

1. Latency (Ping)

What it is: Latency is the time it takes for a data packet to travel from your computer to the server and back. It is measured in milliseconds (ms).

The Standard: For a natural, conversational flow without people talking over each other, your latency should be under 150ms (ideally under 50ms). Anything over 200ms will result in a noticeable delay.

2. Jitter

What it is: Jitter measures the variation in your latency. If your ping fluctuates wildly (e.g., 20ms, then 100ms, then 30ms), you have high jitter. This causes packets to arrive out of order.

The Standard: Jitter must be kept under 30ms. High jitter is the primary cause of "robotic" sounding voices or words being scrambled during a call.

3. Bandwidth (Download/Upload)

What it is: The raw capacity of your connection, typically measured in Megabits per second (Mbps).

The Standard: VoIP actually uses very little bandwidth—typically about 100 kbps (0.1 Mbps) per concurrent call. However, you need enough total bandwidth to ensure that your voice traffic isn't choked out by other users streaming video or downloading large files.

How to Improve Your VoIP Connection

If you failed the test or are experiencing poor call quality, try these actionable steps to stabilize your network:

  • Ditch the Wi-Fi: Wi-Fi is incredibly susceptible to interference and jitter. Hardwiring your phones or computers directly to your router with an Ethernet cable is the single most effective way to improve call quality.
  • Enable QoS (Quality of Service): Log into your router's administration panel and look for QoS settings. You can configure your router to prioritize VoIP traffic (usually UDP ports 5060 for SIP and 10000-20000 for RTP) over standard web browsing or video streaming.
  • Upgrade Your Router: If you are using a cheap, ISP-provided modem/router combo for a business with multiple employees, the router's CPU may be struggling to process the packets fast enough, leading to bufferbloat and latency. Upgrade to a business-class firewall or router.
  • Check for Background Downloads: Ensure no one on your network is syncing large files to cloud storage (like Dropbox or Google Drive) during business hours, as this can easily saturate your upload bandwidth.

If your network is healthy but you are still experiencing issues, the problem may lie with your current VoIP provider's routing or infrastructure. In that case, it might be time to use our comparison tools to find a provider with a better track record for reliability and high-definition audio.