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WebRTC (ウェブリアルタイムコミュニケーション)はWebアプリケーションやWebサイトにて、仲介を必要とせずにブラウザ間で直接、任意のデータの交換や、キャプチャしたオーディオ/ビデオストリームの送受信を可能にする技術である。WebRTCに関する一連の標準規格は、ユーザがプラグインやサードパーティ製ソフトウェアをインストールすることなく、ピア・ツー・ピアにて、データ共有や遠隔会議を実現することを可能にします。



Represents a WebRTC connection between the local computer and a remote peer. It is used to handle efficient streaming of data between the two peers.
Represents the parameters of a session. Each RTCSessionDescription consists of a description type indicating which part of the offer/answer negotiation process it describes and of the SDP descriptor of the session.
Represents a candidate internet connectivity establishment (ICE) server for establishing an RTCPeerConnection.
Represents information about an internet connectivity establishment (ICE) transport.
Represents events that occurs in relation to ICE candidates with the target, usually an RTCPeerConnection. Only one event is of this type: icecandidate.
Manages the encoding and transmission of data through a MediaStreamTrack for an RTCPeerConnection.
Manages the reception and decoding of data through a MediaStreamTrack for an RTCPeerConnection.
Indicates that a new incoming MediaStreamTrack was created and an associated RTCRtpReceiver object was added to the RTCPeerConnection object.
Represents a certificate that an RTCPeerConnection uses to authenticate.
Represents a bi-directional data channel between two peers of a connection.
Represents events that occur while attaching a RTCDataChannel to a RTCPeerConnection. The only event sent with this interface is datachannel.
Manages the encoding and transmission of dual-tone multi-frequency (DTMF) signaling for an RTCPeerConnection.
Indicates an occurrence of a of dual-tone multi-frequency (DTMF). This event does not bubble (except where otherwise stated) and is not cancelable (except where otherwise stated).
Reports stats for a given MediaStreamTrack asynchronously.
Registers an  identity provider (idP).
Enables a user agent is able to request that an identity assertion be generated or validated.
Represents the identity of the a remote peer of the current connection. If no peer has yet been set and verified this interface returns null. Once set it can't be changed
Represents an identity assertion generated by an identity provider (idP). This is usually for an RTCPeerConnection. The only event sent with this type is identityresult.
Represents an error associated with the identity provider (idP). This is usually for an RTCPeerConnection. Two events are sent with this type: idpassertionerror and idpvalidationerror.


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.
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.
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.
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.
Using WebRTC data channels
This guide covers how you can use a peer connection and an associated RTCDataChannel to exchange arbitrary data between two peers.
WebRTC connectivity
This article describes how the various WebRTC-related protocols interact with one another in order to create a connection and transfer data and/or media among peers.


Improving compatibility using WebRTC adapter.js
The WebRTC organization provides on GitHub the WebRTC adapter to work around compatibility issues in different browsers' WebRTC implementations. The adapter is a JavaScript shim which lets your code to be written to the specification so that it will "just work" in all browsers with WebRTC support.
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
The RTCDataChannel interface 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.
Signaling and two-way video calling
Sample, we take the web socket chat system we've created in another example and add the ability to make video calls. The chat server is augmented to handle the WebRTC signaling.



WebRTC-proper protocols


Specification Status Comment
WebRTC 1.0: Real-time Communication Between Browser 草案 The initial definition of the API of WebRTC.
Media Capture and Streams 勧告改訂案 The initial definition of the object conveying the stream of media content.
Media Capture from DOM Elements 勧告改訂案 The initial definition on how to obtain stream of content from DOM Elements

In additions to these specifications defining the API needed to use WebRTC, there are several protocols, listed under resources.


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