Packet Loss Test — Measure Latency & Jitter

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About PacketProbe

PacketProbe measures real packet loss, latency, and jitter on your internet connection — not just speed or bandwidth. Most speed tests only measure throughput over TCP, which hides packet loss by automatically retransmitting failed packets. That tells you nothing about your connection quality for gaming, video calls, or VoIP.

We use WebRTC unreliable data channels — the same technology browsers use for real-time communication — configured to behave like raw UDP. When a packet is lost, it stays lost, and we count it. This gives you an accurate picture of what your connection actually does under real-time conditions.

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 — no software to install, no app to download.

Why testing packet loss matters

A speed test might say you have 500 Mbps, but if you're dropping 3% of packets, your Zoom calls will still freeze, your Valorant matches will rubber-band, and your VoIP phone will cut out. Packet loss, latency, and jitter are the metrics that actually determine real-time connection quality — and they're invisible to standard speed tests. PacketProbe reveals what your connection is really doing.

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.

Why does packet loss matter?

For real-time applications like online gaming, video calls, and VoIP, even 1-2% packet loss can cause noticeable problems: rubber-banding in games, choppy audio, frozen video frames, and dropped calls. Unlike file downloads (where lost packets are simply re-sent), real-time data has a deadline — if a packet arrives late, it's useless.

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 — most real-time applications will struggle significantly.

What causes packet loss?

  • Network congestion — routers drop packets when traffic exceeds their capacity
  • Wi-Fi interference — other networks, walls, microwaves, and distance from your router
  • Faulty hardware — damaged cables, failing network cards, or overheating routers
  • ISP issues — oversubscribed nodes, routing problems, or throttling
  • Server distance — more hops between you and the server means more chances for loss

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

What is jitter?

Jitter is the variation in packet arrival times. Even if no packets are lost, inconsistent timing causes problems. A voice call with high jitter sounds choppy. A game with high jitter feels laggy and unpredictable. Under 30ms jitter is ideal; over 50ms will be noticeable.

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.

What is a good latency (ping) for gaming?

For competitive online gaming, under 30ms is excellent, 30-60ms is good, 60-100ms is playable but you'll be at a disadvantage in fast-paced shooters, and over 150ms creates obvious input lag. Latency depends on your physical distance from the game server, your ISP's routing, and network congestion. Use the Gaming preset to test whether your connection meets these thresholds.

How do I reduce jitter on my connection?

  • Switch to Ethernet — Wi-Fi jitter is typically 5-10x higher than wired connections
  • Enable QoS — Quality of Service settings on your router prioritize real-time traffic over downloads
  • Close background apps — cloud sync, updates, and streaming consume bandwidth unpredictably
  • Upgrade your router — older routers have smaller buffers and weaker packet scheduling
  • Try a different DNS — slow DNS resolution can add variable latency to new connections

What's the difference between latency, jitter, and packet loss?

Latency (ping) is how long a round trip takes — it determines your reaction delay. Jitter is how much that latency varies from packet to packet — it causes stuttering and instability. Packet loss is when data never arrives at all — it causes rubber-banding, audio gaps, and missed inputs. A good connection has low values in all three. PacketProbe measures all of them simultaneously.

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.