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VoIP & Gaming Quality Grader

Grade a connection for calls, gaming and streaming against real thresholds.

What you paste

Parsed in your browser — never uploaded

How this reads your output

Latency, jitter and loss are checked against the requirements of each use in turn. The thresholds are grounded in ITU-T G.114, which places one-way mouth-to-ear delay below 150 ms as acceptable for conversation, and in the loss levels at which concealment stops hiding gaps. Paste ping output directly and all three figures are extracted; write them out and they are read from the text.

What the results mean

Jitter
Variation between consecutive packets. It matters more than latency for calls, because a receiver must buffer against it and the buffer is itself delay.
Packet loss
Above roughly 1% is audible on a voice call regardless of latency. Concealment fills small gaps and cannot fill many.
Suitability grid
Each use graded separately. A connection can be perfect for streaming and poor for competitive gaming, since they need very different things.

Common problems and fixes

Speed tests are fast but calls still break up
Bandwidth is rarely the problem with calls. Check jitter and loss, which a speed test usually does not report. The most common cause on a home connection is bufferbloat: an upload saturating the link fills a buffer in the router, and latency climbs into the hundreds of milliseconds while the speed test still reads full. Queue management in the router fixes it.
Everything looks fine but gaming feels laggy
Look at the peaks rather than the average. A connection averaging 20 ms with occasional spikes to 200 ms feels far worse than a steady 60 ms, because prediction in the game cannot compensate for irregular arrival. Run a longer test and look at the distribution rather than the summary.

Frequently asked questions

What latency do I need for gaming?

Under 50 ms to the server feels responsive for competitive play, and under 100 ms is fine for most games. Consistency matters more than the number: a stable 70 ms beats a 30 ms connection that spikes, because prediction handles constant delay and cannot handle variation.

How much packet loss is acceptable?

Under 0.5% for anything real-time, and ideally zero. Above 1%, voice calls develop audible gaps and video calls freeze. TCP hides loss by retransmitting, which is why a browser feels fine on a connection where calls are unusable.

Does more bandwidth improve any of this?

Only by relieving congestion. If a link is saturated, adding capacity reduces queueing and therefore latency and jitter. If it is not saturated, more bandwidth changes nothing about any of the three figures here, which is why upgrading a connection so often fails to fix call quality.

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