WebRTC 技术正在以非常惊人的速度发展，目前相应的技术文档和草案也逐步发布。因为WebRTC更新的速度非常快，文档编写的内容也是经常更新。目前发布的是浏览器之间的实时通信草案 1.0 版本。我们相信具有化时代的意义。
参与的作者都是行业的权威人士：
Adam Bergkvist, Ericsson
Daniel C. Burnett, Invited Expert
Cullen Jennings, Cisco
Anant Narayanan, Mozilla (until November 2012)
Bernard Aboba, Microsoft Corporatio
具体内容包括：
文档介绍
2.确认
3.专有名词
4.点对点通信
4.11.1Dictionary
RTCCertificateExpiration4.11.2RTCCertificate Interface
4.10.1RTCPriorityType Enum
4.9.1RTCIceCandidate Interface
4.9.2RTCPeerConnectionIceEvent
4.9.3RTCPeerConnectionIceErrorEvent
4.9.1.1RTCIceProtocol Enum
4.9.1.2RTCIceTcpCandidateType Enum
4.9.1.3RTCIceCandidateType Enum
4.8.1Setting Negotiation-Needed
4.8.2Clearing Negotiation-Needed
4.8.3Updating the Negotiation-Needed flag
4.7.1RTCSdpType
4.7.2RTCSessionDescription Class
4.6.1General Principles
4.5.1RTCPeerConnectionErrorCallback
4.5.2RTCSessionDescriptionCallback
4.4.1RTCSignalingState Enum
4.4.2RTCIceGatheringState Enum
4.4.3RTCPeerConnectionState Enum
4.4.4RTCIceConnectionState Enum
4.3.1Operation
4.3.2Interface Definition
4.3.3Legacy Interface Extensions
4.3.4Garbage collection
4.2.1RTCConfiguration Dictionary
4.2.2RTCIceCredentialType Enum
4.2.3RTCIceServer Dictionary
4.2.4RTCIceTransportPolicy Enum
4.2.5RTCBundlePolicy Enum
4.2.6RTCRtcpMuxPolicy Enum
4.2.7Offer/Answer Options
4.1Introduction
4.2Configuration
4.3RTCPeerConnection Interface
4.4State Definitions
4.5Callback Definitions
4.6Error Handling
4.7Session Description Model
4.8Session Negotiation Model
4.9Interfaces for Connectivity Establishment
4.10Priority and QoS Model
4.11Certificate Management
5.RTP Media API
5.5.1The RTCDtlsFingerprint Object
5.4.1"Hold" functionality
5.1.1Processing Remote MediaStreamTracks
5.1RTCPeerConnection Interface Extensions
5.2RTCRtpSender Interface
5.3RTCRtpReceiver Interface
5.4RTCRtpTransceiver Interface
5.5RTCDtlsTransport Interface
5.6RTCIceTransport Interface
5.7RTCTrackEvent
6.Peer-to-peer Data API
6.1.1RTCSctpTransport Interface
6.1RTCPeerConnection Interface Extensions
6.2RTCDataChannel
6.3RTCDataChannelEvent
6.4Garbage Collection
7.Peer-to-peer DTMF
7.1RTCRtpSender Interface Extensions
7.2RTCDTMFSender
7.3RTCDTMFToneChangeEvent
8.Statistics Model
8.1Introduction
8.2RTCPeerConnection Interface Extensions
8.3RTCStatsCallback
8.4RTCStatsReport Object
8.5RTCStats Dictionary
8.6Derived Stats Dictionaries
8.7Example
9.Identity
9.3.1User Login Procedure
9.2.1Interface Exposed by Identity Providers
9.2.2Identity Assertion and Validation Results
9.1.1Identity Provider Selection
9.1.2Instantiating an IdP Proxy
9.1.2.1Implementing an IdP Securely
9.1Identity Provider Interaction
9.2Registering an IdP Proxy
9.3Requesting Identity Assertions
9.4Verifying Identity Assertions
9.5IdP Error Handling
9.6RTCPeerConnection Interface Extensions
9.7Examples
10.Media Stream API Extensions for Network Use
10.4.1Extended MediaStreamTrack Properties
10.4.2Isolated Streams and RTCPeerConnection
10.4.3Protection Afforded by Media Isolation
10.3.1MediaTrackSupportedConstraints, MediaTrackCapabilities, MediaTrackConstraints and MediaTrackSettings
10.2.1id
10.1Introduction
10.2MediaStream
10.3MediaStreamTrack
10.4Isolated Media Streams
11.Examples and Call Flows
11.1Simple Peer-to-peer Example
11.2Simple Peer-to-peer Example with Warm-up
11.3Simple Peer-to-peer Example with media before signaling
11.4Simple Simulcast Example
11.5Advanced Peer-to-peer Example
11.6Peer-to-peer Data Example
11.7Call Flow Browser to Browser
11.8DTMF Example
12.Event summary
13.Privacy and Security Considerations
13.1Impact on same origin policy
13.2Revealing IP addresses
13.3Impact on local network
13.4Confidentiality of Communications
13.5Persistent information exposed by WebRTC
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