webRTC

WebRTC（Web实时通讯）是一种技术，使Web应用程序和网站来捕捉和可选的音频和/或视频流媒体，以及交换任意数据之间的浏览器不须要中介。该系列标准包括WebRTC可以共享数据和执行会议的对等，而不须要用户安装插件或任何其余第三方软件。

WebRTC由几个相互关联的应用程序接口和协议共同来实现。你会发现这里将帮助你理解WebRTC的基础，如何设置和使用数据和媒体的链接、更多。

接口

1.RTCPeerConnection

    表示本地计算机和远程节点之间的链接的WebRTC。它是用来处理两个对等体之间的高效的数据流

2.RTCSessionDescription

    表示会话的参数。每一个 rtcsessiondescription 由描述 型 指示提供/应答谈判过程的描述和对 SDP 描述符的会议的一部分。

3.RTCIceCandidate

    表明候选人的网络链接创建（ice）创建一个 rtcpeerconnection服务器。

4.RTCIceTransport

    表明一个 互联网链接创建信息（ice）运输。

5.RTCPeerConnectionIceEvent

    表示与目标对象有关的事件发生的事件，一般一个rtcpeerconnection。只有一个事件是这种类型：icecandidate。

6.RTCRtpSender

    Manages the encoding and transmission of data through a MediaStreamTrack 对于 RtcPeerConnection

7.RTCRtpReceiver

    负责接收和解码的数据经过一个 rtcpeerconnection 对于 mediastreamtrack。

8.RTCTrackEvent

Indicates that a new incoming MediaStreamTrack was created and an associated RTCRtpReceiver object was added to the RTCPeerConnection object.

9.RTCCertificate

Represents a certificate that an RTCPeerConnection uses to authenticate.

10.RTCDataChannel

Represents a bi-directional data channel between two peers of a connection.

RTCDataChannelEvent

Represents events that occur while attaching a RTCDataChannel to a RTCPeerConnection. The only event sent with this interface is datachannel.

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 MediaStreamTrack asynchronously.

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 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.

GuidesEDIT

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.

TutorialsEDIT

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.

ResourcesEDIT

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

SpecificationsEDIT

Specification	Status	Comment
WebRTC 1.0: Real-time Communication Between Browser	Working Draft	The initial definition of the API of WebRTC.
Media Capture and Streams	Editor's Draft	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

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

See alsoEDIT

• MediaDevices
• MediaStreamEvent
• MediaStreamConstraints
• MediaStreamTrack
• MessageEvent
• MediaStream
• Firefox multistream and renegotiation for Jitsi Videobridge
• Peering Through the WebRTC Fog with SocketPeer
• Inside the Party Bus: Building a Web App with Multiple Live Video Streams + Interactive Graphics

SDPEDIT

Session Description Protocol (SDP) is a standard for describing the multimedia content of the connection such as resolution, formats, codecs, encryption, etc. so that both peers can understand each other once the data is transferring. This is, in essence, the metadata describing the content and not the media content itself.

 

 

The offer/answer process is performed both when a call is first established, but also any time the call's format or other configuration needs to change. Regardless of whether it's a new call, or reconfiguring an existing one, these are the basic steps which must occur to exchange the offer and answer, leaving out the ICE layer for the moment:

1. The caller calls RTCPeerConnection.createOffer() to create an offer.
2. The caller calls RTCPeerConnection.setLocalDescription() to set that offer as the local description (that is, the description of the local end of the connection).
3. The caller uses the signaling server to transmit the offer to the intended receiver of the call.
4. The recipient receives the offer and calls RTCPeerConnection.setRemoteDescription()to record it as the remote description (the description of the other end of the connection).
5. The recipient does any setup it needs to do for its end of the call, including adding outgoing streams to the connection.
6. The recipient then creates an answer by calling RTCPeerConnection.createAnswer().
7. The recipient calls RTCPeerConnection.setLocalDescription() to set the answer as its local description. The recipient now knows the configuration of both ends of the connection.
8. The recipient uses the signaling server to send the answer to the caller.
9. The caller receives the answer.
10. The caller calls RTCPeerConnection.setRemoteDescription() to set the answer as the remote description for its end of the call. It now knows the configuration of both peers. Media begins to flow as configured.

 

The entire exchange in a complicated diagramEDIT（整个交换中的一个复杂的图表）