Instant latency, packet loss & jitter — no download, no login, no BS.
A ping test measures the round-trip time (RTT) for a data packet to travel from your device to a remote server and back — expressed in milliseconds (ms). It's the single most fundamental diagnostic for checking network latency and connectivity health. Originally based on the ICMP echo-request protocol, modern browser-based ping tools (like this one) use HTTP HEAD requests to measure how fast a server responds, which reflects real-world web performance more accurately than raw ICMP.
Packet loss occurs when one or more packets never reach their destination. Even 1–2% loss can significantly degrade real-time services like VoIP calls, online gaming, and live video streams. Jitter measures the variability between successive ping times — a stable connection has low jitter; a congested or poorly-routed connection produces high jitter even if average latency looks acceptable.
Raw numbers only matter in context. Here's what your results actually mean across common use cases — and when to act on them.
Jitter rule of thumb: Jitter under 10ms is imperceptible. 10–30ms is acceptable for most uses. Above 30ms you may experience choppy audio in calls. Above 50ms, real-time gaming becomes unreliable regardless of your average latency.
High latency is rarely random. Every extra millisecond has a cause. Here are the most common culprits, in order of likelihood.
Not all of these will apply to your situation, but each one is worth testing in order. Work through them from easiest to most involved.
These three terms are often used interchangeably but they measure different things. Understanding the distinction helps you diagnose connection problems more precisely.
| Metric | What It Measures | Gaming | VoIP | Streaming |
|---|---|---|---|---|
| Ping (RTT) | Round-trip time for a single packet — your "raw" latency number. Lower is always better. | ✓ Critical |
✓ Important |
~ Minor |
| Latency | Broader term for any delay in a network path — includes processing time at the server, not just transit time. Ping is a specific latency measurement. | ✓ Critical |
✓ Critical |
~ Buffered |
| Jitter | The variance between successive ping measurements. A connection with 40ms average but 0ms jitter is smoother than one averaging 20ms with 30ms jitter. | ✓ Important |
! Critical |
~ Buffered |
| Packet Loss | Percentage of packets that never arrive. Even 0.5% loss can degrade VoIP calls and cause TCP retransmission storms that compound latency. | ! Severe |
! Severe |
! Rebuffering |
The practical takeaway: For gaming, optimise ping first, then jitter. For VoIP and video calls, prioritise jitter and packet loss — a slightly higher but stable latency is far better than a low but erratic one. Streaming video is the most tolerant because players buffer several seconds of content, absorbing both jitter and brief packet loss invisibly.
| 🟢 <20ms | Exceptional |
| 🟢 20–50ms | Excellent |
| 🔵 50–100ms | Good |
| 🟡 100–200ms | Fair |
| 🔴 200ms+ | Poor |
Run a lightweight HTTP ping to trusted targets or your own endpoint.