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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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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, 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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Nothing tanks adoption like a user’s first call failing. A lightweight pre-call test catches device, permission, and network issues before a session starts—slashing “it doesn’t work” tickets and improving CSAT.
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Your agents make a legitimate call, but the customer’s phone says “Spam Likely” or blocks it outright. That’s lost revenue and trust. The fix isn’t just CNAM—it’s deploying STIR/SHAKEN so downstream carriers can verify your caller identity.
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Security is now a buying criterion, not a checkbox. Teams want end-to-end encryption (E2EE) for browser calls, but run into the same blockers: “Will it break the SFU?”, “How do we manage keys?”, “What about recordings and moderation?”. Here’s a practical, product-centred guide to E2EE that answers those questions and shows how to ship it.
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