Would you use this? - VOIP
I’m considering building a 100% browser-based WebRTC network diagnostic tool to help remote users troubleshoot choppy VoIP/video calls without installing native CLI tools or needing admin rights.
Standard speed tests use HTTP/TCP, which hides the UDP packet loss and jitter that ruins WebRTC traffic. This tool runs directly in the user's browser, establishes a real-time RTCPeerConnection against your own self-hosted STUN/TURN or media servers, and simulates an actual audio call stream. Using the browser's native getStats() API, it pulls second-by-second telemetry on packet loss, jitter, round-trip latency, and ICE candidate paths (ensuring they aren't falling back to TCP). At the end of a test, it generates a simple Pass/Fail gauge and a one-click "Download JSON Log" button for users to paste into a support ticket.
Is this a tool you would actually use, or does something like this already exist in your workflow? I want to make sure I’m building something enterprise teams actually need. If you'd find this helpful for your helpdesk, let me know.