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How WebRTC Works with SIP: A Simple Technical Breakdown

For many VoIP and PBX professionals, WebRTC can seem like a mystery wrapped inside a browser tab. It’s the technology that allows real-time voice and video communication directly from a website — no plugins, no downloads, and no desktop apps. But how exactly does it interact with SIP, the long-standing signalling protocol that powers most of today’s VoIP infrastructure?

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WebRTC Softphones: The Future of Browser VoIP

For decades, telephony revolved around physical desk phones and, later, downloadable SIP softphones. These tools brought voice over IP (VoIP) into the workplace, allowing employees to make calls using software instead of hardware. But in today’s digital-first economy, expectations have shifted. Businesses need faster deployment, more mobility, and airtight security. This is where WebRTC softphones have emerged as the clear frontrunner.

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SIP over WebSocket: Browser SIP Client for WebRTC PBXs

Most “click-to-call” demos end the moment a real PBX enters the picture. The missing piece is SIP over WebSocket (WSS): a standard that lets browsers speak SIP for signalling while WebRTC handles the encrypted media. The result is a true browser SIP client that can register, place calls, and integrate with your existing telephony—without installing anything.

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