Packet Loss Stream Test

Use this page to test how a browser, network, VPN, firewall, or Wi-Fi connection behaves when playing progressively larger video streams. Start with a low-resolution stream and work up toward 4K.

Packet loss often appears as buffering, dropped frames, resolution changes, long startup time, audio/video glitches, or complete playback failure. This page is intended as a simple front-end test and should be used together with Chrome DevTools, HAR exports, chrome://media-internals, and chrome://webrtc-internals when testing WebRTC.

How to use this test

Start with 360p or 480p. Confirm the video starts quickly and plays without buffering.
Move up to 720p and 1080p. Watch for buffering, dropped frames, or long startup time.
Try 1440p and 4K only after lower levels are stable.
Repeat the same test with VPN on and off to compare routing and throughput.
Open DevTools Network or export a HAR file to compare download timing and failed requests.

Important WebRTC note

This page tests browser video playback and network delivery behavior. It does not directly measure RTP packet loss from a WebRTC peer connection. For WebRTC media packet loss, open chrome://webrtc-internals and check packetsLost, fractionLost, selected candidate pairs, bitrate, frames decoded, and round-trip time.