Web SIP Phone

WebRTC SIP phone connection indicator. Registers your SIP extension over WSS and shows live status: ready, busy, or offline.

Web SIP Phone keeps a WebRTC SIP connection alive in your browser and shows a small draggable phone icon with a live status dot on the business pages you allow. STATUS AT A GLANCE • Green — registered and ready • Amber — a call is in progress • Red — not connected (blinking while connecting or reconnecting) 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 icon to inspect SIP registration, WebSocket, microphone and media health • Errors pop up with a one-click recovery action Web SIP Phone is a connection indicator, not a softphone: your business page keeps full control of dialing, answering and call information. It only runs on the HTTPS sites you explicitly allow in Settings. REQUIREMENTS • A SIP server reachable over WSS (e.g. FreeSWITCH with wss-binding) • Chrome 116 or newer