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.

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.


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 (en-US) (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 (en-US) interface. Once a connection has been established and opened using RTCPeerConnection, media streams (MediaStream (en-US)s) and/or data channels (RTCDataChannel (en-US)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 (en-US) 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 (en-US) 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 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..

RTCPeerConnection (en-US)
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.
RTCDataChannel (en-US)
Represents a bi-directional data channel between two peers of a connection.
RTCDataChannelEvent (en-US)
Represents events that occur while attaching a RTCDataChannel (en-US) to a RTCPeerConnection (en-US). The only event sent with this interface is datachannel.
RTCSessionDescription (en-US)
Represents the parameters of a session. Each RTCSessionDescription consists of a description type (en-US) indicating which part of the offer/answer negotiation process it describes and of the SDP (en-US) descriptor of the session.
RTCSessionDescriptionCallback (en-US)
The RTCSessionDescriptionCallback is passed into the RTCPeerConnection (en-US) object when requesting it to create offers or answers.
RTCStatsReport (en-US)
Provides information detailing statistics for a connection or for an individual track on the connection; the report can be obtained by calling RTCPeerConnection.getStats() (en-US). Details about using WebRTC statistics can be found in WebRTC Statistics API.
RTCIceCandidate (en-US)
Represents a candidate Internet Connectivity Establishment (ICE (en-US)) server for establishing an RTCPeerConnection (en-US).
RTCIceTransport (en-US)
Represents information about an ICE transport.
RTCIceServer (en-US)
Defines how to connect to a single ICE server (such as a STUN (en-US) or TURN (en-US) server).
RTCPeerConnectionIceEvent (en-US)
Represents events that occur in relation to ICE candidates with the target, usually an RTCPeerConnection (en-US). Only one event is of this type: icecandidate.
RTCRtpSender (en-US)
Manages the encoding and transmission of data for a MediaStreamTrack (en-US) on an RTCPeerConnection (en-US).
RTCRtpReceiver (en-US)
Manages the reception and decoding of data for a MediaStreamTrack (en-US) on an RTCPeerConnection (en-US).
RTCRtpContributingSource (en-US)
Contains information about a given contributing source (CSRC) including the most recent time a packet that the source contributed was played out.
RTCTrackEvent (en-US)
The interface used to represent a track (en-US) event, which indicates that an RTCRtpReceiver (en-US) object was added to the RTCPeerConnection (en-US) object, indicating that a new incoming MediaStreamTrack (en-US) was created and added to the RTCPeerConnection.
RTCConfiguration (en-US)
Used to provide configuration options for an RTCPeerConnection.
RTCSctpTransport (en-US)
Provides information which describes a Stream Control Transmission Protocol (SCTP (en-US)) transport and also provides a way to access the underlying Datagram Transport Layer Security (DTLS (en-US)) transport over which SCTP packets for all of an RTCPeerConnection's data channels are sent and received.
RTCSctpTransportState (en-US)
Indicates the state of an RTCSctpTransport (en-US) instance.

Identity and security

The WebRTC API includes a number of interfaces which are used  to manage security and identity.

Enables a user agent is able to request that an identity assertion be generated or validated.
RTCIdentityAssertion (en-US)
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).
RTCIdentityEvent (en-US)
Represents an identity assertion generated by an identity provider (idP). This is usually for an RTCPeerConnection (en-US). The only event sent with this type is identityresult.
RTCIdentityErrorEvent (en-US)
Represents an error associated with the identity provider (idP). This is usually for an RTCPeerConnection (en-US). Two events are sent with this type: idpassertionerror and idpvalidationerror.
RTCCertificate (en-US)
Represents a certificate that an RTCPeerConnection (en-US) uses to authenticate.


These interfaces are related to interactivity with Public-Switched Telephone Networks (PTSNs).

RTCDTMFSender (en-US)
Manages the encoding and transmission of Dual-Tone Multi-Frequency (DTMF (en-US)) signaling for an RTCPeerConnection (en-US).
RTCDTMFToneChangeEvent (en-US)
Used by the tonechange (en-US) 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).


Introduction to WebRTC protocols
This article introduces the protocols on top of which the WebRTC API is built.
WebRTC connectivity
A guide to how WebRTC connections work and how the various protocols and interfaces can be used together to build powerful communication apps.
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.
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
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.
Codecs used by WebRTC
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.
Using WebRTC data channels
This guide covers how you can use a peer connection and an associated RTCDataChannel (en-US) 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 (en-US) interface. This guide shows how to do so.


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 (en-US) 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.



WebRTC-proper protocols

WebRTC statistics


Specification Status Comment
WebRTC 1.0: Real-time Communication Between Browsers Candidate Recommendation The initial definition of the API of WebRTC.
Media Capture and Streams Candidate Recommendation The initial definition of the object conveying the stream of media content.
Media Capture from DOM Elements Working 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