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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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Building browser-based calling is powerful, but a handful of recurring issues create most of the user pain. Here’s a concise, field-tested playbook to diagnose and fix them fast.
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Great microphones and codecs won’t save a call if your network treats real-time packets like background syncs. On Wi-Fi 6/6E, quality of service (QoS) is the difference between crisp audio and “can you hear me now?”. This guide explains how DSCP and WMM map voice traffic across wired and wireless hops—and the practical tweaks that actually move the needle.
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Real-time apps now have three strong browser options for moving data: WebRTC DataChannel, WebTransport, and WebSockets. They overlap, but they’re not interchangeable. Choosing the right one affects latency, reliability, back-pressure handling and deployment complexity.
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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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Most teams discover WebRTC through a simple peer-to-peer demo—and then hit a wall the moment real users, real networks, and real scale arrive. The hidden lever is your media topology: P2P, SFU, or MCU. Pick the wrong one and you’ll fight jitter, ballooning CPU bills, or fragile group calls. Pick the right one and you’ll deliver crisp audio/video with predictable costs.
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When it comes to real-time communication, two acronyms dominate the conversation: WebRTC and SIP. Both play vital roles in enabling modern voice and video calling, yet they serve different purposes. For businesses and IT professionals, understanding the relationship between the two is essential for making the right technology choices.
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The way we make voice and video calls is undergoing a quiet revolution. Businesses and developers alike are moving away from heavy, app‑dependent VoIP systems and leaning into WebRTC SIP clients — lightweight, browser‑native solutions that blend the power of SIP protocols with the flexibility of modern web technology.
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