What is WebRTC?
Web Real-Time Communication (WebRTC) is an open-source project and technology that enables real-time communication capabilities directly in web browsers without requiring additional plugins or applications.
SIP Over WebRTC
SIP over WebRTC integrates the robustness of Session Initiation Protocol (SIP) with the versatility of Web Real-Time Communication (WebRTC), allowing seamless voice and video communication.
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Understanding SIP Transactions, Dialogs, and Sessions
A SIP transaction is a core element of the SIP protocol, encompassing a single request made by a client and the corresponding responses from a server.
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How secure is my password
In SIP communications, passwords play a key role in the authentication process, typically using a method known as Digest Authentication.
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What is DTLS (Datagram Transport Layer Security)?
DTLS, or Datagram Transport Layer Security, is a protocol that provides privacy and data integrity for communications over datagram protocols, which are typically used for applications that require real-time communication and low latency, such as streaming media, voice over IP (VoIP), and online gaming.
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What is SIP (Session Initiation Protocol)?
Session Initiation Protocol (SIP) is a signaling protocol used for initiating, maintaining, and terminating real-time sessions that involve voice, video, messaging, and other communications applications and services between two or more endpoints on IP networks.
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Siperb WebRTC Client: Web, Tablet, and Mobile
SIPERB provides robust WebRTC client formats to meet diverse communication needs across various devices. Our solutions include clients tailored for web, tablet, and mobile interfaces, each designed to optimize your communication experience regardless of your device type.
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Call Media Transcoding
Transcoding is a crucial function within the SIPERB platform, especially when dealing with legacy PBX systems that do not natively support WebRTC’s modern codecs and protocols.
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Understanding WebRTC’s End-to-End Encryption
WebRTC (Web Real-Time Communication) is an innovative technology that enables direct communication between browsers and devices.
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OpenSIPS WebRTC
This guide explores how to integrate WebRTC with OpenSIPS, enabling browser-based voice and video calls. It covers essential OpenSIPS modules, TLS setup, and using SIP.js for WebRTC clients, complete with code examples for making and receiving calls. Perfect for building SIP-compatible, real-time communication in web applications.
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How Safe is WebRTC?
WebRTC supports video, voice, and generic data to be sent between peers, building a powerful basis for building real-time communication applications, but how safe is it?