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Open-source software has always been a driving force in communications — from Asterisk powering PBXs to FreeSWITCH enabling scalable VoIP. Now, the same collaborative spirit is fuelling a new frontier: the open-source WebRTC softphone.
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In today’s workplace, communication tools are expected to be fast, secure, and flexible. That’s where the WebRTC SIP client comes in. This hybrid technology bridges the best of two worlds: the browser-based power of WebRTC and the industry-standard reliability of SIP (Session Initiation Protocol). Together, they’re transforming how businesses handle voice and video calls.
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For years, businesses relied on traditional SIP-based softphones — applications installed on desktops or mobiles — to connect with VoIP and PBX systems. But today, a new approach is rapidly overtaking them: the WebRTC softphone.
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If you’ve ever set up a VoIP system or explored real-time communications, you’ve likely come across two terms: WebRTC and SIP. While both enable voice and video communication over the internet, they operate in very different ways — and understanding these differences is crucial for businesses deciding on their next communication platform.
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WebRTC has transformed how we communicate online, offering low-latency audio, video, and data streaming directly between browsers. But what if you need to record a WebRTC call — for training, compliance, or quality assurance? The process isn’t always straightforward, and there are legal considerations to keep in mind.
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For years, traditional SIP softphones dominated VoIP deployments. But today, a new wave of technology is reshaping communication: WebRTC softphones. By combining the simplicity of browser-based communication with the reliability of SIP and PBX systems, WebRTC is bridging the gap between modern accessibility and enterprise-grade telephony.
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If you’ve been exploring WebRTC for real-time audio, video, and data, you may be wondering which transport protocol it uses: UDP or TCP. The truth is that WebRTC can use both — but prefers one over the other for good reasons.
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WebRTC has become the go-to technology for real-time audio, video, and data sharing across browsers and devices. But many developers wonder — does WebRTC actually use WebSockets under the hood? The short answer: not for media transport, but it can for signalling.
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Understanding what issues cause lag & freeze issues with Webrtc—and knowing how to address them—can make all the difference between a crystal-clear conversation and a pixelated mess.
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In SIP-based communication systems, few things trip up developers and VoIP engineers more than the confusing difference between a SIP dialog and a SIP transaction.
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