Web SIP Phone

Keeps a WebRTC SIP extension registered over WS/WSS and shows a headset button with live connection status on the pages you allow.

Web SIP Phone keeps a WebRTC SIP connection alive in your browser and shows a small draggable headset button — with a status badge on its corner — on the business pages you allow. STATUS AT A GLANCE • Green badge — registered and ready • Amber badge, pulsing — connecting, registering, or reconnecting after a drop • Red badge — a fault, or nothing running (not configured, or no allowed page open) • The button itself tinted — a call is in progress; the badge keeps reporting connection health, so a call never hides a connection problem Status is never carried by colour alone: the same wording is in the button's tooltip and its accessible label, and the panel repeats every signal as a check, warning or cross shape. HOW IT WORKS • Registers a single SIP extension over secure WebSocket (WSS) • Auto-answers server-initiated calls (Call-Info answer-after) • Supports remote talk/hold control (BroadSoft Event: talk/hold) • Click the button for the Voice connection panel: which extension you are on which server, signalling health with the registration expiry counting down, your microphone with its device name and a live input level, and TURN readiness • A fault opens the panel on its own, names the server's own reason — "Registration failed (403 Forbidden) — check password in Settings" — counts down to the next retry, and offers the one action that fixes it. It stays open until the fault clears; a fault is dismissed by fixing it, not by hiding it • Copy diagnostics puts the whole picture on your clipboard for a support ticket. Your SIP password is never included — it never leaves Settings Web SIP Phone is a connection indicator, not a softphone: your business page keeps full control of dialing, answering and call information. Call state never reaches the page — no numbers, no timers, no call controls. It only runs on the HTTPS sites you explicitly allow in Settings, and only while one of those tabs is open. REQUIREMENTS • A SIP server reachable over WSS (e.g. FreeSWITCH with wss-binding) • Microphone access, granted once from Settings → Advanced → Test microphone • Chrome 116 or newer