WebRTC (Web Real-Time Communications) is a technology which enables Web applications and sites to capture and optionally stream audio and/or video media, as well as to exchange arbitrary data between browsers without requiring an intermediary. The set of standards that comprises WebRTC makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user install plug-ins or any other third-party software.
WebRTC consists of several interrelated APIs and protocols which work together to achieve this. The documentation you'll find here will help you understand the fundamentals of WebRTC, how to set up and use both data and media connections, and more.
- WebRTC architecture overview
- Beneath the APIs that developers use to create and use WebRTC connections lie a number of network protocols and connectivity standards. This brief overview covers these standards.
- WebRTC basics
- This article takes you through the creation of a cross-browser RTC App. By the end of it, you should have working peer-to-peer data channel and media channel.
- WebRTC protocols
- This article introduces the protocols on top of which the WebRTC API is built.
- WebRTC connectivity
- This article describes how the varous WebRTC-related protocols interact with one another in order to create a connection and transfer data and/or media among peers.
- WebRTC API overview
- WebRTC consists of several interrelated APIs and protocols which work together to support the exchange of data and media between two or more peers. This article provides a brief overview of each of these APIs and what purpose it serves.
- Lifetime of a WebRTC session
- WebRTC lets you build peer-to-peer communication of arbitrary data, audio, or video—or any combination thereof—into a browser application. In this article, we'll look at the lifetime of a WebRTC session, from establishing the connection all the way through closing the connection when it's no longer needed.
- Improving compatibility using WebRTC adapter.js
- Taking still photos with WebRTC
- This article shows how to use WebRTC to access the camera on a computer or mobile phone with WebRTC support and take a photo with it.
- A simple RTCDataChannel sample
RTCDataChannelinterface is a feature which lets you open a channel between two peers over which you may send and receive arbitrary data. The API is intentionally similar to the WebSocket API, so that the same programming model can be used for each.
- Application Layer Protocol Negotiation for Web Real-Time Communications
- WebRTC Audio Codec and Processing Requirements
- RTCWeb Data Channels
- RTCWeb Data Channel Protocol
- Web Real-Time Communication (WebRTC): Media Transport and Use of RTP
- WebRTC Security Architecture
- Transports for RTCWEB
Related supporting protocols
- Interactive Connectivity Establishment (ICE): A Protocol for Network Address Translator (NAT) Traversal for Offer/Answer Protocol
- Session Traversal Utilities for NAT (STUN)
- URI Scheme for the Session Traversal Utilities for NAT (STUN) Protocol
- Traversal Using Relays around NAT (TURN) Uniform Resource Identifiers
- An Offer/Answer Model with Session Description Protocol (SDP)
- Session Traversal Utilities for NAT (STUN) Extension for Third Party Authorization
|WebRTC 1.0: Real-time Communication Between Browser||Working Draft||The initial definition of the API of WebRTC.|
|Media Capture and Streams||Candidate Recommendation||The initial definition of the object conveying the stream of media content.|
|Media Capture from DOM Elements||Editor's Draft||The initial definition on how to obtain stream of content from DOM Elements|