WebRTC Call Quality & Jitter Diagnostic
Test your microphone, network latency, jitter, and packet loss in real time.
The CHATR WebRTC Call Quality Diagnostic tests your local browser microphone, network latency, jitter buffer performance, and packet loss against global VoIP edge servers, verifying your network readiness for crystal-clear business voice and video calls.
Interactive Application Ready
Use this interactive utility right in your browser. All processing is private and performed client-side.
Frequently Asked Questions
What is an acceptable jitter for VoIP calls? ▼
Network jitter should be under 30ms with less than 1% packet loss for carrier-grade voice quality.