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Getting OpenSIPS Installed and Ready as a WebRTC Server: A Complete Guide

WebRTC (Web Real-Time Communication) is revolutionizing communication by enabling peer-to-peer audio, video, and data sharing in browsers without additional plugins. For developers and businesses, setting up a WebRTC server using OpenSIPS, an open-source SIP (Session Initiation Protocol) proxy/server, is a powerful way to enable efficient real-time communication. This article walks you through the process of installing OpenSIPS, configuring it for WebRTC, and preparing it to function as a WebRTC server.

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GeneralLatestOpenSIPSTech NewsTOP STORIESVOIP
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OpenSIPS vs. Kamailio: Key Differences to Know for Your Next VoIP Project

OpenSIPS and Kamailio are two popular open-source SIP (Session Initiation Protocol) server projects that originated from the same codebase (SER – SIP Express Router). They’re widely used for handling VoIP calls, SIP routing, instant messaging, and other real-time communication services. Though they are similar in many ways, they have diverged over the years, each developing distinct features and communities. Here’s a comparison of key aspects:

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