Running Network Tests and Interpreting Results
If you are experiencing intermittent connectivity issues or dropped calls, you may be asked to run a network test to help determine the root cause. A fast internet connection does not always guarantee high-quality audio. Environmental factors like distance from your Wi-Fi router or congestion from other devices can affect your connection and lead to lower bandwidth, packet loss, and jitter.
Tip: Run the diagnostics while you are actively experiencing call issues. Running them when everything seems fine will not capture the actual network state when problems occur.
Running Network Tests
If you are running network tests to understand the quality and throughput of your connection, follow the instructions below.
If you are encountering intermittent connectivity, it’s important to run these tests while the issue is occurring so the results accurately reflect the network behavior.
Navigate to the WebRTC Diagnostics Test Page.
Complete the reCAPTCHA to verify you are not a bot.
Allow microphone access if prompted.
During the test, you will hear the message “Record a message in 3, 2, 1.” Say a few words or recite a short phrase to capture your audio. The recording will play back to confirm your microphone is working properly.
You may be prompted to allow camera access. Revenue.io does not utilize video, so you can safely block camera access if you prefer to skip that portion.
As the tests complete, you will see a Log Output appear along the right-hand side of your screen. Take a screenshot or save the log output to a text file so that you can share it with Support or your IT administrator.
Understanding the Log Output
| Metric | What It Means |
|---|---|
| UDP, TCP, or TLS Connection Status | Both UDP and TCP must be allowed for calls to connect properly. If either is blocked, you may experience dropped calls or missing audio. |
| Bandwidth (Uplink/Downlink) | Recommended minimums: Opus: 40 kbps / 40 kbps; PCMU: 100 kbps / 100 kbps. |
| RTT (Roundtrip Time) | Recommended < 200 ms for a stable call experience. Higher values can introduce delays or lag. |
| Jitter | Recommended < 30 ms. Higher jitter can cause distorted or robotic audio. |
| Packet Loss | Recommended < 3%. Higher levels will cause choppy audio or dropped calls. |
Getting Help
If you suspect your calls are being impacted by network quality:
- Take screenshots or save the test logs to a text file.
- Share them with your Network or IT Administrator to check firewall or router settings.
If your test results look healthy but you are still experiencing call quality problems:
- Contact our Support Team
- Paste your network test logs and include details about when the issues occur.
Technical Insight
Even if your internet speed appears strong, VoIP quality depends more on packet stability and low latency than raw bandwidth. If you see evidence of packet loss or jitter:
- Try from a different network (for example, a home Wi-Fi or mobile hotspot) to rule out local restrictions.
- Disable VPNs or proxies, as they can block or reroute RTP (audio) packets.
- Avoid running large downloads, video streaming, or cloud syncs during calls.
Running diagnostics during active issues and saving the log results provides the most accurate view of what is affecting your connection quality.