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If you run a PBX, you’ve probably heard people talk about “WebRTC softphones” and “browser calling” as if they’re the same thing. Sometimes they are. Often, they’re not. And in the middle of it all you still have real users, a real Asterisk or FreePBX box, and the same uptime targets as always.
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If you have ever joined a video meeting that refused to connect, suffered from one-way audio, or dropped unexpectedly, there is a good chance the problem occurred before the call even started — at the network layer.
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Most WebRTC and VoIP conversations fail for one simple reason: devices can’t reach each other through firewalls and NATs. And the most important — and least understood — component solving this problem is the TURN server.
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When a modern business tries to connect browser users with SIP phones, PBX systems, mobile VoIP apps, or external carriers, something important has to sit in the middle — a WebRTC gateway. It’s the bridge that translates WebRTC’s encrypted, browser-native communication into traditional SIP/RTP, ensuring everything works smoothly.
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For years, business telephony revolved around desk phones, proprietary VoIP handsets, and on-premise PBX hardware. That era is disappearing quickly. Today, browser-based SIP clients are becoming the preferred way for teams to make and receive calls — without installing apps, deploying hardware, or maintaining ageing desk phones.
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When you make a browser-based call or join an online meeting, you probably don’t give much thought to how your audio and video stay private. Yet behind every WebRTC session is a powerful encryption framework that ensures your data can’t be intercepted.
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When users connect over WebRTC, the experience feels seamless — instant video, clear audio, and smooth data transfer between browsers. But beneath that simplicity lies a sophisticated negotiation process designed to handle one of the internet’s biggest headaches: firewalls and NATs (Network Address Translators).
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Thanks to WebRTC data channels, developers can now build secure, ultra-fast, peer-to-peer applications for everything from file sharing to live collaboration — no central server required.
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Most people never think about what happens when they make a VoIP call. You click “dial,” your softphone connects, and the conversation just works. But beneath that simple experience is one of the most essential — and misunderstood — parts of the VoIP infrastructure: the SIP proxy.
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For decades, business phone calls relied on centralised infrastructure — copper lines, hardware PBX systems, and trusted networks. But as remote work and browser-based apps became standard, a question emerged: how can real-time voice remain both reliable and private when it runs entirely online?
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