WebRTC API
WebRTC (Web Real-Time Communications) 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 comprises 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.
Interfaces
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
RTCSessionDescription
consists of a descriptiontype
indicating 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
MediaStreamTrack
for anRTCPeerConnection
. RTCRtpReceiver
- Manages the reception and decoding of data through a
MediaStreamTrack
for anRTCPeerConnection
. RTCTrackEvent
- Indicates that a new incoming
MediaStreamTrack
was created and an associatedRTCRtpReceiver
object was added to theRTCPeerConnection
object. RTCCertificate
- Represents a certificate that an
RTCPeerConnection
uses to authenticate. RTCDataChannel
- Represents a bi-directional data channel between two peers of a connection.
RTCDataChannelEvent
- Represents events that occur while attaching a
RTCDataChannel
to 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
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 isidentityresult
. RTCIdentityErrorEvent
- Represents an error associated with the identity provider (idP). This is usually for an
RTCPeerConnection
. Two events are sent with this type:idpassertionerror
andidpvalidationerror
.
Guides
- 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.
Tutorials
- 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.
Resources
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
Specifications
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 also
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
License
© 2016 Mozilla Contributors
Licensed under the Creative Commons Attribution-ShareAlike License v2.5 or later.
https://developer.mozilla.org/en-us/docs/web/api/webrtc_api