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WebRTC (where RTC stands for Real-Time Communications) is a technology that enables audio/video streaming and data sharing between browser clients (peers). As a set of standards, WebRTC provides any browser with the ability to share application data and perform teleconferencing peer to peer, without the need to install plug-ins or third-party software. 

WebRTC components are accessed with JavaScript APIs: the Network Stream API, which represents an audio or video data stream, the PeerConnection API, which allows two or more users to communicate browser-to-browser, and the DataChannel API that enables communication of other types of data for real-time gaming, text chat, file transfer, and so forth.

Note: This documentation is in the process of moving to its new home.


Peer-to-peer communications with WebRTC
How to perform peer-to-peer communications using the WebRTC APIs.
Introduction to WebRTC architecture
(AKA "WebRTC and the Ocean of Acronyms") WebRTC has a lot of different parts to it, and it can be overwhelming and confusing to newcomers. This article aims to explain what all the parts are, and how they fit together.
WebRTC basics
Now you understand the WebRTC architecture, you can move on to this article, which takes you through the creation of a basic cross-browser RTC App.


The API to capture media (video/audio).
The interface handling the streaming of data between two peers.
The interface for sending arbitrary data across the peer connection.