WebRTC stands for Web Real-Time Communications, which is an open-source project that enables real-time voice, text, and video communications capabilities between web browsers and devices. It provides web browsers and mobile applications with real-time communication via application programming interfaces (APIs) . WebRTC allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps.
WebRTC is designed to make audio, video, and data communication between browsers simple and efficient. It enables real-time audio and video communication simply by opening a webpage. WebRTC serves multiple purposes, including support for audio and video conferencing, file exchange, screen sharing, identity management, and interfacing with legacy telephone systems. Connections between peers can be made without requiring any special drivers or plug-ins, and can often be made without any intermediary servers.
WebRTC uses JavaScript, APIs, and Hypertext Markup Language to embed communications technologies within web browsers. It is supported by Apple, Google, Microsoft, Mozilla, and Opera, and its specifications have been published by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF) .
In summary, WebRTC is an open-source project that enables real-time voice, text, and video communications capabilities between web browsers and devices. It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps. WebRTC is designed to make audio, video, and data communication between browsers simple and efficient.
