Web Real-Time Communication (WebRTC) is an open-source project used to provide real-time communication to mobile applications and web browsers with the help of application programming interfaces (APIs). With support for video, voice, and generic data formats, WebRTC helps developers build communication solutions.
It allows direct peer-to-peer communication and eliminates the need to install plugins.
With WebRTC, service providers and organizations can enable real-time communications to any device via any network. Browsers can directly exchange real-time media with other browsers in a peer-to-peer manner using a TURN server.
It works with all common browsers and supports native Android and iOS apps.
WebRTC is highly secure, as it doesn't use any third-party software or plugins.
It is highly customizable and therefore has a vast scope of implementation.
WebRTC functions as three components using APIs.
To acquire the media stream, i.e. audio or video.
To make WebRTC calls to stream video and audio, and exchange data. Video streaming, screen sharing, and video calling among peers are established using this.
To enable bilateral data communication between peers.
WebRTC is commonly used in enabling video, voice, or chat support on customer-facing websites and mobile apps. It is also used for online meetings and conferences.