この翻訳は不完全です。英語から この記事を翻訳 してください。
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。
WebRTCはこれらを実現するために、いくつかの相互に関係するAPI群とプロトコル群から構成されます。ここにあるドキュメント類は、あなたがWebRTCの基礎や、データコネクション／メディアコネクションの設定、利用方法などを理解するための手助けをしてくれます。
インターフェース
RTCPeerConnection- 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.
RTCSessionDescription- Represents the parameters of a session. Each
RTCSessionDescriptionconsists of a descriptiontypeindicating which part of the offer/answer negotiation process it describes and of the SDP descriptor of the session. RTCIceCandidate- Represents a candidate internet connectivity establishment (ICE) server for establishing an
RTCPeerConnection. RTCIceTransport- Represents information about an internet connectivity establishment (ICE) transport.
RTCPeerConnectionIceEvent- Represents events that occurs in relation to ICE candidates with the target, usually an
RTCPeerConnection. Only one event is of this type:icecandidate. RTCRtpSender- Manages the encoding and transmission of data through a
MediaStreamTrackfor anRTCPeerConnection. RTCRtpReceiver- Manages the reception and decoding of data through a
MediaStreamTrackfor anRTCPeerConnection. RTCTrackEvent- Indicates that a new incoming
MediaStreamTrackwas created and an associatedRTCRtpReceiverobject was added to theRTCPeerConnectionobject. RTCCertificate- Represents a certificate that an
RTCPeerConnectionuses to authenticate. RTCDataChannel- Represents a bi-directional data channel between two peers of a connection.
RTCDataChannelEvent- Represents events that occur while attaching a
RTCDataChannelto aRTCPeerConnection. The only event sent with this interface isdatachannel. RTCDTMFSender- Manages the encoding and transmission of dual-tone multi-frequency (DTMF) signaling for an
RTCPeerConnection. RTCDTMFToneChangeEvent- 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).
RTCStatsReport- Reports stats for a given
MediaStreamTrackasynchronously. RTCIdentityProviderRegistrar- Registers an identity provider (idP).
RTCIdentityProvider- Enables a user agent is able to request that an identity assertion be generated or validated.
RTCIdentityAssertion- 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 RTCIdentityEvent- Represents an identity assertion generated by an identity provider (idP). This is usually for an
RTCPeerConnection. The only event sent with this type isidentityresult. RTCIdentityErrorEvent- Represents an error associated with the identity provider (idP). This is usually for an
RTCPeerConnection. Two events are sent with this type:idpassertionerrorandidpvalidationerror.
ガイド
- 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
RTCDataChannelto 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.
チュートリアル
- 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
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. - 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.
リソース
Protocols
WebRTC-proper protocols
- 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
仕様書
| Specification | Status | Comment |
|---|---|---|
| WebRTC 1.0: Real-time Communication Between Browsers | 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.