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Real-Time Latency Stream
Stable High Latency Loss
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About

This Packet Loss & Jitter Tester is a precision diagnostic tool designed to evaluate the stability of your internet connection, not just its speed. While standard speed tests measure bandwidth (capacity), this tool measures quality (reliability). High bandwidth means nothing if data packets are dropped or arrive irregularly.

Why this matters: In real-time applications like Online Gaming (CS:GO, Valorant) or VoIP (Zoom, Discord), consistency is king. Packet Loss occurs when data fails to reach its destination, causing rubber-banding or robotic audio. Jitter is the variance in latency; a ping that spikes from 20ms to 150ms creates a stuttery experience even if the average is low.

This tool simulates high-frequency traffic to detect micro-outages and strictly analyzes UDP/TCP reliability against industry standards for different use cases.

packet loss jitter ping test latency network diagnostic

Formulas

We calculate Jitter (J) using the RFC 3550 smoothing algorithm, which approximates the statistical variance of packet inter-arrival times. This is more accurate for real-time streams than simple standard deviation.

{
Ji = Ji-1 + |Di-1,i| Ji-116Where Di-1,i is the difference in delay between packets.

Packet Loss Percentage (L) is strictly defined as:

Lost PacketsTotal Sent × 100

Where a packet is considered "Lost" if latency > Thresholdtimeout (default 1000ms).

Reference Data

MetricExcellent (Esports/Comp)Good (Casual Gaming)Fair (Browsing/Streaming)Poor (Unusable)
Packet Loss0.0%< 1.0%1.0 - 2.5%> 2.5%
Jitter< 1ms< 5ms< 30ms> 50ms
Latency (Ping)< 20ms< 50ms< 100ms> 150ms
MOS Score4.5 - 5.04.0 - 4.43.0 - 3.9< 3.0

Frequently Asked Questions

Speed tests use large TCP streams that automatically retry lost packets to maximize throughput numbers. They hide instability. This tool uses high-frequency, non-retrying requests (simulating UDP behavior) to expose the micro-stutters and drops that actually affect gaming and VoIP.
For competitive gaming, Jitter should be under 1ms. For VoIP calls (Zoom/Teams), anything under 30ms is usually imperceptible. Jitter above 50ms will cause "robotic" voice and rubber-banding in games.
Sometimes. First, switch from Wi-Fi to Ethernet (cable); this fixes 90% of local jitter. If on Wi-Fi, move closer to the router or use 5GHz. If the issue persists on Ethernet, the problem lies with your ISP (Internet Service Provider) or the physical lines to your home.
This tool primarily measures the Round Trip Time (RTT) reliability. Since HTTP/WebRTC handshakes require bidirectional confirmation, a loss in either direction registers as a failure here. It effectively tests the weakest link in the chain.