Packet Loss Test

Every digital interaction across the internet—from loading a webpage to playing competitive multiplayer games—relies on breaking data into discrete units called network packets. A packet contains routing headers (source and destination IP addresses, ports, and sequence numbers) and a data payload. Packet loss occurs when one or more of these data packets fail to reach their intended destination across intermediate routing hops or return to your client within the expected time window.

The impact of packet loss depends critically on the transport protocol in use:

• TCP (Transmission Control Protocol): Connection-oriented protocol used for web browsing, secure banking, and file downloads. TCP guarantees delivery through ACK (acknowledgement) packets. If a packet is lost, TCP initiates retransmission, which prevents data corruption but causes severe latency spikes and throughput degradation (TCP window throttling).
• UDP (User Datagram Protocol): Connectionless protocol used for real-time multiplayer games (Valorant, CS2, Fortnite), VoIP calls (Discord, Zoom), and live video streams. UDP prioritizes real-time delivery over reliability by transmitting without waiting for acknowledgements. When UDP packets are dropped, players experience rubber-banding, desync, hit registration failure, and robotic audio.

This diagnostic tool measures packet delivery reliability by streaming sequenced micro-probes over WebRTC and low-overhead HTTP data streams, calculating instantaneous delivery success, late arrival jitter, and total percentage loss.