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 installs plug-ins or any other
third-party software.
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.
data and media connections, and more.
Interoperability
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
RTCPeerConnectionRTCPeerConnectionMediaStreamMedia streams can consist of any number of tracks of media information; tracks, which are represented by objects based on the
MediaStreamTrackMost 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
RTCDataChannelmore details and links to relevant guides and tutorials needed
WebRTC interfaces
Because WebRTC provides interfaces that work together to accomplish a variety of tasks, we have divided up the interfaces in the list below by category. Please see the sidebar for an alphabetical list.
Connection setup and management
These interfaces 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.
These interfaces 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.
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.
remote peer. It is used to handle efficient streaming of data between
the two peers.
Represents a bi-directional data channel between two peers of a connection.
Represents events that occur while attaching a
RTCDataChannelRTCPeerConnectiondatachannel Represents the parameters of a session. Each
RTCSessionDescriptiontypeThe RTCSessionDescriptionCallback is passed into the
RTCPeerConnectionProvides information detailing statistics for a connection or for an individual track on the connection; the report can be obtained by calling
RTCPeerConnection.getStats()Represents a candidate Internet Connectivity Establishment (ICE) server for establishing an
RTCPeerConnectionRepresents information about an ICE transport.
Represents events that occur in relation to ICE candidates with the target, usually an
RTCPeerConnectionicecandidateManages the encoding and transmission of data for a
MediaStreamTrackRTCPeerConnectionManages the reception and decoding of data for a
MediaStreamTrackRTCPeerConnectionContains information about a given contributing source (CSRC) including the most recent time a packet that the source contributed was played out.
The interface used to represent a
trackRTCRtpReceiverRTCPeerConnectionMediaStreamTrackRTCPeerConnectionProvides 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'Indicates the state of an
Identity and security
The WebRTC API includes a number of interfaces which are used to manage security and identity.
RTCSctpTransportIdentity and security
The WebRTC API includes a number of interfaces which are used to manage security and identity.
RTCIdentityProviderEnables a user agent is able to request that an identity assertion be generated or validated.
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.
Registers an identity provider (idP).
Represents an identity assertion generated by an identity provider (idP). This is usually for an
RTCPeerConnectionidentityresultRepresents an error associated with the identity provider (idP). This is usually for an
RTCPeerConnectionidpassertionerroridpvalidationerrorThese interfaces are related to interactivity with Public-Switched Telephone Networks (PTSNs).
Manages the encoding and transmission of Dual-Tone Multi-Frequency (DTMF) signaling for an
RTCPeerConnection Used by the
tonechangeGuides
This article introduces the protocols on top of which the WebRTC API is built.
A guide to how WebRTC connections work and how the various protocols and interfaces can be used together to build powerful communication apps.
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.
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.
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.
A tutorial and example which turns a WebSocket-based chat system
created for a previous example and adds support for opening video calls
among participants. The chat server's WebSocket connection is used for
WebRTC signaling.
created for a previous example and adds support for opening video calls
among participants. The chat server's WebSocket connection is used for
WebRTC signaling.
A guide to the codecs which WebRTC requires browsers to support as
well as the optional ones supported by various popular browsers.
Included is a guide to help you choose the best codecs for your needs.
well as the optional ones supported by various popular browsers.
Included is a guide to help you choose the best codecs for your needs.
This guide covers how you can use a peer connection and an associated
RTCDataChannelWebRTC's support for interacting with gateways that link to old-school telephone systems includes support for sending DTMF tones using the
RTCDTMFSenderTutorials
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.
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.
The
RTCDataChannelResources
Protocols
WebRTC-proper 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
WebRTC statistics
Specifications
In additions to these specifications defining the API needed to use WebRTC, there are several protocols, listed under resources.
See also
- MediaDevices
- MediaStreamEvent
- MediaStreamConstraints
- MediaStreamTrack
- MessageEvent
- MediaStream
- Media Capture and Streams API
- 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
- Web media technologies
Credits
- Source: https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API
- Published under Open CC Attribution ShareAlike 3.0 licence