Free packet loss test

Your speed test says you're fine.
Your game says otherwise.

Speed tests measure bandwidth over TCP, which silently retransmits dropped packets and hides the problem. PacketProbe measures what actually breaks games and calls: packet loss, jitter, and latency, over real UDP.

Disconnected

Numbers only mean something next to a target. See what Valorant, CS2, Warzone, Teams, Zoom or Meet actually need, and whether yours clear it.

Check your game or call app

What we measure

  • Upload packet loss: packets that never reached the server. Even 1% upload loss can cause voice calls to cut out and game actions to fail to register.
  • Download packet loss: packets the server sent back that never arrived. Download loss causes missing game state updates, frozen video frames, and audio dropouts.
  • Round-trip time (RTT) / latency: how long a packet takes to travel to the server and back, measured in milliseconds. This is what gamers call "ping." Under 50ms is excellent, 50-100ms is playable, and over 150ms creates noticeable input delay.
  • Jitter: the variation in latency between consecutive packets. Even with low ping, high jitter causes stuttering, rubber-banding in games, and choppy VoIP audio. Under 30ms is good, over 50ms is problematic.
  • Late packets: packets that arrived but exceeded the acceptable delay threshold. In real-time applications, a packet that arrives too late is as useless as one that was lost entirely.

How it works

Your browser establishes a WebRTC data channel to one of our test servers. It then sends numbered packets at a configurable rate and size. The server records which packets arrive (measuring upload loss) and echoes them back (measuring download loss and round-trip latency). Jitter is calculated from the variation in successive round-trip times. The entire test runs in your browser with no software to install and no app to download.

Packet Loss FAQ

What is packet loss?

Packet loss occurs when data packets traveling across a network fail to reach their destination. On the internet, data is broken into small packets that travel independently. If any packet is dropped along the way (by a congested router, a faulty cable, or a weak Wi-Fi signal) it's considered lost.

What's a good packet loss percentage?

0% is ideal. Under 1% is generally acceptable for most applications. Between 1-2.5% you'll start noticing issues in gaming and voice calls. Above 5% is considered severe, and most real-time applications will struggle significantly.

How do I fix packet loss?

  • Use a wired connection: Ethernet eliminates Wi-Fi as a variable
  • Restart your router/modem: clears congested buffers
  • Check cables: replace any damaged Ethernet cables
  • Reduce network load: pause large downloads, limit connected devices
  • Change Wi-Fi channel: use a less congested channel or 5GHz band
  • Contact your ISP: if loss persists on a wired connection, the problem may be upstream

Why is this different from a speed test?

Speed tests measure throughput: how much data your connection can move per second over TCP. TCP automatically retransmits lost packets, so speed tests mask packet loss entirely. PacketProbe uses an unreliable (UDP-like) channel that reveals the true loss and jitter your connection produces under real-time conditions.

What do the presets mean?

Each preset configures the test to mimic a specific application's network behavior. For example, the "Gaming (FPS)" preset sends 64 small packets per second with a tight 80ms delay threshold, matching what a competitive shooter actually demands from your connection. This tells you whether your connection can handle that specific use case.

Is packet loss the same as lag?

Not exactly. "Lag" is a general term that can mean high latency (slow response), high jitter (inconsistent response), or packet loss (missing data). They feel similar to the user (things stutter, freeze, or feel unresponsive) but the causes and fixes are different. PacketProbe helps you identify which specific problem is affecting your connection.