How the packet loss test works
The test sends small real-time packets between your browser and a WebRTC test server. It records how many packets return, how long they take, and how much timing varies during the run.
Test packet loss, latency, and jitter in your browser. Useful for gaming, video calls, streaming, and real-time apps.
No download required. Works directly in modern browsers with WebRTC support.
Live packet path
Browser to test server and back
Connection status
Ready to test
Choose a preset or tune the packet profile for your connection.
Size of each test packet.
Packets sent per second.
How long the recorded test runs.
Packets above this round-trip time are marked late.
Connection settling time before recording packets.
Auto is enabled for v1.
Connection status, packet counts, and live timing.
0/150
0
0.0 sec
N/A
Current jitter: N/A
The test sends small real-time packets between your browser and a WebRTC test server. It records how many packets return, how long they take, and how much timing varies during the run.
Packet loss means some data packets never reach their destination. In real-time apps, missing packets can feel like stutter, rubber-banding, audio gaps, or frozen video.
Zero packet loss is ideal. Anything above 1% can be noticeable in competitive games, voice calls, and video meetings, even when ordinary browsing still feels fine.
Real-time apps care about stability more than raw download speed. Low loss, low latency, and low jitter usually matter more than a high bandwidth number.
Wi-Fi interference, overloaded routers, damaged cables, congested ISP routes, VPN issues, and busy servers can all drop packets before they return.
Test with Ethernet, restart network gear, reduce local traffic, update router firmware, remove VPNs from the path, and compare results across server regions.
FAQ
Packet loss happens when data packets traveling across a network fail to reach their destination. It can make games lag, calls break up, and real-time apps feel unstable.
For browsing, 1% packet loss may not be very noticeable. For competitive gaming, voice calls, and video meetings, even 1% packet loss can cause problems.
Common causes include Wi-Fi interference, overloaded routers, bad cables, congested ISP routes, VPN issues, and server-side network problems.
Try Ethernet instead of Wi-Fi, restart your router, reduce local network load, update firmware, test without VPN, and contact your ISP if packet loss continues across multiple services.
A speed test measures bandwidth. Real-time apps also need low latency, low jitter, and low packet loss. Download speed can be high while the connection is still unstable.
The test uses WebRTC DataChannel packets to approximate real-time UDP-like traffic in the browser. Results depend on browser support, the chosen server region, and current network conditions.