Diagnose your network connection. Test download, upload, ping, jitter, packet loss, and bufferbloat to see if your connection is ready for streaming and gaming.
Professional Grade internet analyzer evaluating bandwidth metrics, loaded latency, stability, and real-world application performance.
TURN connectivity was unavailable, so packet loss was not measured.
Calculated using a rigorous multi-weighted quality score formula.
Calculated using standard deviation variance across sample latency arrays and packet loss dropouts.
Your latency increased by only 0 ms under heavy traffic. Online gaming and video calls will remain perfectly responsive.
Authentic details fetched directly from edge headers and client browser APIs. Testing your network securely shares your IP address with Cloudflare edge network nodes.
Comprehensive statistical distributions, consistency ratings, and phase performance profiling.
Converts dry measured statistics into practical capability ratings for popular web services and workloads.
Your internet speed represents how much data your connection can transfer per second. Download speed affects how quickly you can load web pages, download files, and stream video. Upload speed affects how quickly you can send data, such as uploading files, sending emails, or broadcasting video during a call.
Latency (Ping) is the baseline round-trip time for a packet of data to travel from your device to a server and back. High latency makes connections feel sluggish, even if bandwidth is high.
Jitter measures the variation in your latency. A highly unstable connection with high jitter will cause stuttering in voice calls and lag in competitive gaming.
Bufferbloat occurs when your router buffers too much data during active heavy traffic. If someone else on your network starts a large download, your latency might spike from 20ms to 500ms, causing extreme lag. Our test actively measures your loaded latency to detect bufferbloat.
Packet loss occurs when data fails to reach its destination. Even a 1% packet loss can significantly degrade real-time applications like VoIP and video conferencing. We measure this directly through WebRTC UDP sessions to detect exact dropped packet metrics.
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