Sharpen WebRTC Connectivity Check
Overview
Confirms a browser can reach the STUN servers Sharpen's voice depends on and identifies the firewall and network conditions that stop calls from connecting. Where a speed test tells you how fast a connection is, this tells you whether voice can get through it at all, which is the more common cause of a failed call.
It contacts each of the five Google STUN servers listed in the Network Requirements article, several times each, and reports the following:
Check | What it tells you |
|---|---|
STUN reachability | Whether UDP 19302 is open to the servers voice relies on |
UDP allowed | Whether the network permits UDP at all, or is forcing everything to TCP |
NAT type | Whether the network re-maps ports per destination, which forces calls onto a relay |
Split tunnel | Whether voice and web traffic are leaving by different public addresses, which usually means a VPN |
This replaces the older manual methods, which required either a third-party website Sharpen does not control or pasting JavaScript into the browser's developer console. It takes about fifteen seconds, needs no microphone access, and places no call.
What it does not do. It does not place a call or measure audio quality, and it cannot measure RTP packet loss, which only exists once a real call is running. A pass here means the path is open, not that call quality will be good.
This is a snapshot, not a source of truth. The result reflects one moment on one machine. A VPN, virtual desktop, Wi-Fi, or a corporate proxy can all change it. Use it as a starting point for investigation rather than proof that the network was or was not the cause.
This article is part of a series on Admin Configuration topics.