Diagnostics
VoIP Quality Test
Real-time voice quality analysis over a real media path. Test up to four simultaneous calls for up to three minutes. No microphone permissions required.
Free · no account required · up to 4 calls, 3 minutes
Free account required
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How it works
Real Media Path
Sends traffic that mirrors actual voice calls -- same codecs, same packet sizes, same timing. Not a synthetic speed test or ICMP ping.
No Microphone Required
The test runs entirely in your browser. No mic permissions needed, no audio is captured, and nothing is installed.
ITU-T MOS Scoring
Calculates Mean Opinion Score using the ITU-T G.107 E-model. Industry-standard scoring that providers and ISPs both recognize.
Jitter & Packet Loss
Measures inter-arrival jitter, packet loss rate, and round-trip latency over the actual media path -- not just ICMP pings.
Up to 4 Calls at Once
Run up to four simultaneous calls, each over its own connection. Measures what actually happens when several calls share your uplink -- not a projection from a single path.
Runs Up to 3 Minutes
Choose 60 seconds for a quick check, or up to three minutes to catch intermittent drops and congestion that a short test walks straight past.
Exportable Reports
Results include raw data, methodology notes, and standardized metrics so the evidence speaks for itself in any dispute.
Who it's for
MSPs and IT Teams
Validate network readiness before a VoIP deployment. Run tests from the same desktops your users will make calls from and get hard numbers to share with the provider.
The MSP workflow →VoIP Providers
Give customers an independent, auditable test they can run themselves. When a quality complaint comes in, share a test link instead of trading blame with the ISP.
Network Engineers
Diagnose intermittent call quality problems with per-packet timing data. Identify whether the issue is jitter, loss, or latency -- and whether it correlates with time of day or traffic patterns.
Frequently asked questions
Do I need to install anything to run the test?+
No. The test runs entirely in your browser. There is nothing to download or install, and no browser extensions are required.
Why not just use a speed test?+
Speed tests measure bulk throughput -- how fast you can download a large file. VoIP traffic is small, constant-rate UDP packets that are sensitive to timing, not bandwidth. A connection can pass a speed test with flying colors and still produce terrible call quality. This test sends traffic that matches real voice calls and measures what actually matters: jitter, packet loss, and latency.
Does the test require microphone access?+
No. The test generates a synthetic audio signal in the browser. No microphone permissions are needed, and no live audio is captured or transmitted.
What metrics does the test measure?+
Jitter (variation in packet arrival timing), packet loss (percentage of packets that never arrive), round-trip latency, and MOS (Mean Opinion Score calculated using the ITU-T G.107 E-model). When you run more than one call at once, you also get a per-call breakdown alongside the combined result, which is what shows whether the calls degraded each other.
How does the concurrent call test work?+
Each simulated call opens its own peer connection with its own media stream, so four calls means four independent RTP streams competing for your uplink -- the same thing four real phones would do. That is a genuine measurement rather than an estimate extrapolated from one path. The report shows every call individually plus the combined result, and flags the worst performer, because the weakest call is the one people complain about.
Is the test free?+
Yes — the test is free and no account is needed to run it, including multi-call runs and the full three-minute duration. Creating a free account adds saved test history, and will unlock premium features like WebTransport per-packet accuracy and scheduled monitoring as they launch.
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