WebRTC API | Web API

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互操作性
WebRTC concepts and usage
WebRTC reference
指南
教程
资源
规范
另请参阅
相关话题

WebRTC (Web Real-Time Communication) 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 comprise 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.

互操作性

Because implementations of WebRTC are still evolving, and because each browser has different levels of support for codecs and WebRTC features, you should strongly consider making use of the Adapter.js library provided by Google before you begin to write your code.

Adapter.js uses shims and polyfills to smooth over the differences among the WebRTC implementations across the environments supporting it. Adapter.js also handles prefixes and other naming differences to make the entire WebRTC development process easier, with more broadly compatible results. The library is also available as an NPM package .

To learn more about Adapter.js, see Improving compatibility using WebRTC adapter.js .

WebRTC concepts and usage

WebRTC serves multiple purposes; together with the Media Capture and Streams API , they provide powerful multimedia capabilities to the Web, including support for audio and video conferencing, file exchange, screen sharing, identity management, and interfacing with legacy telephone systems including support for sending DTMF (touch-tone dialing) signals. Connections between peers can be made without requiring any special drivers or plug-ins, and can often be made without any intermediary servers.

Connections between two peers are represented by the RTCPeerConnection interface. Once a connection has been established and opened using RTCPeerConnection , media streams ( MediaStream s) and/or data channels ( RTCDataChannel s) can be added to the connection.

Media streams can consist of any number of tracks of media information; tracks, which are represented by objects based on the MediaStreamTrack interface, may contain one of a number of types of media data, including audio, video, and text (such as subtitles or even chapter names). Most streams consist of at least one audio track and likely also a video track, and can be used to send and receive both live media or stored media information (such as a streamed movie).

You can also use the connection between two peers to exchange arbitrary binary data using the RTCDataChannel interface. This can be used for back-channel information, metadata exchange, game status packets, file transfers, or even as a primary channel for data transfer.

more details and links to relevant guides and tutorials needed

WebRTC reference

Because WebRTC provides interfaces that work together to accomplish a variety of tasks, we have divided up the reference by category. Please see the sidebar for an alphabetical list.

Connection setup and management

These interfaces, dictionaries, and types are used to set up, open, and manage WebRTC connections. Included are interfaces representing peer media connections, data channels, and interfaces used when exchanging information on the capabilities of each peer in order to select the best possible configuration for a two-way media connection.

接口

RTCPeerConnection
RTCDataChannel
RTCDataChannelEvent: Represents events that occur while attaching a RTCDataChannel 到 RTCPeerConnection . The only event sent with this interface is datachannel .
RTCSessionDescription: 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.
RTCStatsReport: Provides information detailing statistics for a connection or for an individual track on the connection; the report can be obtained by calling RTCPeerConnection.getStats() . Details about using WebRTC statistics can be found in WebRTC Statistics API .
RTCIceCandidate: Represents a candidate Interactive Connectivity Establishment ( ICE ) server for establishing an RTCPeerConnection .
RTCIceTransport: Represents information about an ICE transport.
RTCPeerConnectionIceEvent: Represents events that occur 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 for a MediaStreamTrack 在 RTCPeerConnection .
RTCRtpReceiver: Manages the reception and decoding of data for a MediaStreamTrack 在 RTCPeerConnection .
RTCTrackEvent: The interface used to represent a track event, which indicates that an RTCRtpReceiver object was added to the RTCPeerConnection object, indicating that a new incoming MediaStreamTrack was created and added to the RTCPeerConnection .

RTCSctpTransport: Provides information which describes a Stream Control Transmission Protocol ( SCTP ) transport and also provides a way to access the underlying Datagram Transport Layer Security ( DTLS ) transport over which SCTP packets for all of an RTCPeerConnection 's data channels are sent and received.

字典

RTCConfiguration: Used to provide configuration options for an RTCPeerConnection .
RTCIceServer: Defines how to connect to a single ICE server (such as a STUN or TURN server).
RTCRtpContributingSource

事件

bufferedamountlow: The amount of data currently buffered by the data channel—as indicated by its bufferedAmount property—has decreased to be at or below the channel's minimum buffered data size, as specified by bufferedAmountLowThreshold .
close: The data channel has completed the closing process and is now in the closed state. Its underlying data transport is completely closed at this point. You can be notified before closing completes by watching for the closing event instead.
closing: RTCDataChannel has transitioned to the closing state, indicating that it will be closed soon. You can detect the completion of the closing process by watching for the close 事件。
connectionstatechange: The connection's state, which can be accessed in connectionState , has changed.
datachannel: 新的 RTCDataChannel is available following the remote peer opening a new data channel. This event's type is RTCDataChannelEvent .
error: RTCErrorEvent indicating that an error occurred on the data channel.
error: RTCErrorEvent indicating that an error occurred on the RTCDtlsTransport . This error will be either dtls-failure or fingerprint-failure .
gatheringstatechange: RTCIceTransport 's gathering state has changed.
icecandidate: RTCPeerConnectionIceEvent which is sent whenever the local device has identified a new ICE candidate which needs to be added to the local peer by calling setLocalDescription() .
icecandidateerror: RTCPeerConnectionIceErrorEvent indicating that an error has occurred while gathering ICE candidates.
iceconnectionstatechange: Sent to an RTCPeerConnection when its ICE connection's state—found in the iceconnectionstate property—changes.
icegatheringstatechange: Sent to an RTCPeerConnection when its ICE gathering state—found in the icegatheringstate property—changes.
message: A message has been received on the data channel. The event is of type MessageEvent .
negotiationneeded: Informs the RTCPeerConnection that it needs to perform session negotiation by calling createOffer() followed by setLocalDescription() .
open: The underlying data transport for the RTCDataChannel has been successfully opened or re-opened.
selectedcandidatepairchange: The currently-selected pair of ICE candidates has changed for the RTCIceTransport on which the event is fired.
track: track event, of type RTCTrackevent is sent to an RTCPeerConnection when a new track is added to the connection following the successful negotiation of the media's streaming.
signalingstatechange: Sent to the peer connection when its signalingstate has changed. This happens as a result of a call to either setLocalDescription() or setRemoteDescription() .
statechange: The state of the RTCDtlsTransport 已改变。
statechange: The state of the RTCIceTransport 已改变。
statechange: The state of the RTCSctpTransport 已改变。

类型

RTCSctpTransportState: Indicates the state of an RTCSctpTransport 实例。
RTCSessionDescriptionCallback: The RTCSessionDescriptionCallback is passed into the RTCPeerConnection object when requesting it to create offers or answers.

Identity and security

These APIs are used to manage user identity and security, in order to authenticate the user for a connection.

RTCIdentityProvider
RTCIdentityAssertion: Represents the identity of the 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.
RTCIdentityProviderRegistrar
RTCIdentityEvent: 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 .
RTCIdentityErrorEvent: 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 .
RTCCertificate: Represents a certificate that an RTCPeerConnection uses to authenticate.

Telephony

These interfaces and events are related to interactivity with Public-Switched Telephone Networks (PTSNs). They're primarily used to send tone dialing sounds—or packets representing those tones—across the network to the remote peer.

接口

RTCDTMFSender: Manages the encoding and transmission of Dual-Tone Multi-Frequency ( DTMF ) signaling for an RTCPeerConnection .
RTCDTMFToneChangeEvent: Used by the tonechange event to indicate that a DTMF tone has either begun or ended. This event does not bubble (except where otherwise stated) and is not cancelable (except where otherwise stated).

事件

tonechange: Either a new DTMF tone has begun to play over the connection, or the last tone in the RTCDTMFSender 's toneBuffer has been sent and the buffer is now empty. The event's type is RTCDTMFToneChangeEvent .

指南

Introduction to WebRTC protocols
WebRTC connectivity
Lifetime of a WebRTC session
Establishing a connection: The perfect negotiation pattern: Perfect negotiation is a design pattern which is recommended for your signaling process to follow, which provides transparency in negotiation while allowing both sides to be either the offerer or the answerer, without significant coding needed to differentiate the two.
Signaling and two-way video calling
Codecs used by WebRTC
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.
Using DTMF with WebRTC: WebRTC's support for interacting with gateways that link to old-school telephone systems includes support for sending DTMF tones using the RTCDTMFSender interface. This guide shows how to do so.

教程

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
A simple RTCDataChannel sample: 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.

资源

协议

WebRTC-proper protocols

Related supporting protocols

WebRTC statistics

WebRTC Statistics API

规范

规范	状态	注释
WebRTC 1.0: Real-time Communication Between Browsers	候选推荐	The initial definition of the API of WebRTC.
媒体捕获和流	候选推荐	The initial definition of the object conveying the stream of media content.
Media Capture from DOM Elements	工作草案	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 .

另请参阅

元数据

最后修改： Sep 1, 2020

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