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One-way audio is the most common (and most frustrating) issue when you connect browsers to a SIP PBX. The call rings, you answer, and only one party can hear the other. The good news: it’s almost always solvable with a methodical checklist that targets signalling, NAT traversal, and media paths.
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The era of the desk phone is fading fast. As businesses migrate to cloud communications, browser-based softphones have become the bridge between traditional PBX systems and the new world of WebRTC-powered VoIP.
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Modern teams live in the browser. Installing and maintaining desktop softphones slows everyone down. A WebRTC softphone (browser SIP client) puts secure, crystal-clear calling into any modern browser—no downloads, no VPN, no IT tickets.
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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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Business communication has changed dramatically over the past two decades. Where companies once relied on bulky on-site Private Branch Exchange (PBX) systems, many are now migrating to flexible Voice over IP (VoIP) solutions. Yet for many decision-makers, the distinction between PBX and VoIP is still blurred.
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For decades, Asterisk has been the open-source backbone of countless PBXs and VoIP systems. Meanwhile, WebRTC has emerged as the browser-native standard for secure, real-time communication. Put them together, and you unlock a powerful solution: WebRTC Asterisk integration, enabling direct browser-to-PBX calling without plugins or dedicated softphone apps.
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VoIP (Voice over Internet Protocol) solutions have become the backbone of day-to-day communication. But with increased connectivity comes heightened risks, making end-to-end encrypted VoIP not just desirable, but essential.
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The telecommunications landscape has undergone a seismic shift in recent years, with open source solutions emerging as powerful alternatives to proprietary communication systems.
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While traditional telephony systems like Asterisk PBX have served organisations reliably for decades, the emergence of WebRTC (Web Real-Time Communication) has revolutionised how we think about browser-based communications.
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The debate between OpenSIPS and Kamailio has been echoing through VoIP and telecom communities for over a decade. As we enter 2025, the question remains just as relevant — which SIP server is better for high-performance, carrier-grade, and WebRTC-ready applications?
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