Web Real-Time Communications Working Group Charter

The mission of the Web Real-Time Communications Working Group, part of the Ubiquitous Web Applications Activity, is to define client-side APIs to enable Real-Time Communications in Web browsers.

These APIs should enable building applications that can be run inside a browser, requiring no extra downloads or plugins, that allow communication between parties using audio, video and supplementary real-time communication, without having to use intervening servers (unless needed for firewall traversal, or for providing intermediary services). APIs enabling supplementary functions, such as recording, image capture and screen sharing are also in scope.

Join the Web Real-Time Communications Working Group.

End dateMarch 2018
ConfidentialityProceedings are public
  • Harald Alvestrand (Google)
  • Stefan Håkansson (Ericsson)
Team Contacts
(FTE %: 30)
  • Dominique Hazaël-Massieux
  • Vivien Lacourba
Usual Meeting ScheduleTeleconferences: approximately 1 per month
Face-to-face: up to 3-4 per year


Enabling real-time communications between Web browsers require the following client-side technologies to be available:

Success Criteria

To advance to Proposed Recommendation, each specification is expected to have two independent implementations of each feature defined in the specification.

To advance to Proposed Recommendation, interoperability between the independent implementations (that is, bidirectional audio and video communication between the implementations) should be demonstrated.

Out of Scope

The definition of the network protocols used to establish the connections between peers is out of scope for this group; in general, it is expected that protocols considerations will be handled in the IETF.

The definition of any new codecs for audio and video is out of scope.


Recommendation-Track Deliverables

The Working Group will deliver specifications that cover at least the following functions, unless they are found to be fully specified within other Working Groups' finished results:

Media Stream Functions
API functions to manipulate media streams for interactive real-time communications, connecting various processing functions to each other, and to media devices and network connections, including media manipulation functions for e.g. allowing to synchronize streams. Supplementary functions such as recording of media streams are also in scope.
Audio Stream Functions
An extension of the Media Stream Functions to process audio streams, to enable features such as automatic gain control, mute functions and echo cancellation.
Video Stream Functions
An extension of the Media Stream Functions to process video streams, to enable features such as bandwidth limiting, image manipulation or "video mute".
Functional Component Functions
API functions that allow to query for the components present in an implementation, instantiate them, and connect them to media streams.
P2P Connection Functions
API functions to provide interfaces that enable the conveyance of parameters necessary to establish peer to peer connections, based on the protocols selected by the IETF RTCWeb Working Group. Included in this category are also API functions to allow identification of the peer.

The Working Group may decide to group the specified functions in one or more specifications. The Working Group has already started and will continue work on the following specifications:

WebRTC 1.0: Real-time Communication Between Browsers
JavaScript APIs to allow media and data to be sent to and received from another browser or device implementing the appropriate set of real-time protocols
Identifiers for WebRTC's Statistics API
JavaScript APIs that allow access to statistical information about a peer-to-peer connection established via the WebRTC API

as well as on the following specifications jointly developed with the Device APIs Working Group:

Media Capture and Streams
JavaScript APIs that allow local media, including audio and video, to be requested from a platform
MediaStream Recording
a JavaScript API to record MediaStreams
MediaStream Image Capture
a JavaScript API to capture still images from a video MediaStream
Media Capture Depth Stream Extensions
An extension to the Media Capture and Streams API to capture depth streams (e.g. from 3D cameras)
Media Capture from DOM Elements
An extension to DOM elements to allow to capture a media stream from their content
Audio Output Devices API
JavaScript APIs that let a Web application manage how audio is rendered on the user audio output devices
Screen Capture
An extension to the Media Capture and Streams API to use a user's display, or parts thereof, as the source of a MediaStream.

This work will be done in collaboration with the IETF. The W3C will define APIs to ensure that application developers can control the components or the architecture for selection and profiling of the wire protocols that will be produced by the IETF Real-Time Communication in WEB-browsers (RTCWeb) Working Group. While the specified API Functions will not constrain implementations into supporting a specific profile, they will be compatible with the Profile that will be specified by the RTCWeb Working Group.

As the name indicates, WebRTC 1.0: Real-time Communication Between Browsers is a first version of APIs for real-time communication, sometimes referred to as the PeerConnection API. The activities in the ORTC (Object Real-time Communications) Community Group indicate that there is interest in additional APIs to provide more direct control over WebRTC than what the PeerConnection API offers.

In recognition of this interest, the Working Group will, once WebRTC 1.0: Real-time Communication Between Browsers reaches Candidate Recommendation, start work on a new set of object-oriented APIs for real-time communication.

In developing these new APIs, the Working Group will adhere to the following principles:

The Working Group will take the work done by the ORTC Community Group as a source of input, and when contemplating similar APIs in the Working Group, make efforts to align with the ORTC CG on API methodologies and nomenclature. This may include scheduled design meetings with relevant WG and CG stakeholders to foster convergence of the APIs.

The specified API Functions and the requirements on their implementation must offer functionality that ensures that users' expectations of privacy and control over their devices are met - this includes, but is not limited to, ensuring that users can control what local devices an application can access for capturing media, and are able to at any time revoke that access.

Similarly, all the deliverables must address issues of security. The security and privacy goals and requirements will be developed in coordination with the IETF RTCWeb Working Group.

Similarly, all deliverables must address issues of accessibility including relevant requirements listed in the Media User Accessibility Requirements document (MAUR), such as multiple well-synchronized instances of the same media type. The accessibility goals and requirements will be developed in coordination with the Accessible Platform Architectures Working Group.

Other Deliverables

A comprehensive test suite for all features of a specification is necessary to ensure the specification's robustness, consistency, and implementability, and to promote interoperability between User Agents. Therefore, each specification must have a companion test suite, which should be completed by the end of the Last Call phase, and must be completed, with an implementation report, before transition from Candidate Recommendation to Proposed Recommendation. Additional tests may be added to the test suite at any stage of the Recommendation track, and the maintenance of a implementation report is encouraged.

In particular, since WebRTC deals with communications, specific attention will be brought to the interoperability testing of the WebRTC API implementations (both browser-to-browser and browser-to-native).

Other non-normative documents may be created such as:

Given sufficient resources, this Working Group should review other Working Groups' deliverables that are identified as being relevant to the Working Group's mission.


Note: The group will document significant changes from this initial schedule on the group home page.
Specification FPWD CR PR Rec
Media Capture and Streams 2011-10-27 Q3 2015 Q1 2016 Q2 2016
WebRTC 1.0: Real-time Communication Between Browsers 2011-10-27 Q4 2015 Q4 2016 Q1 2017
MediaStream Recording 2013-02-25 Q4 2015 Q3 2016 Q1 2017
MediaStream Image Capture 2013-07-09 Q2 2016 Q2 2017 Q3 2017
Media Capture Depth Stream Extensions 2014-10-07 Q2 2016 Q2 2017 Q3 2017
Identifiers for WebRTC's Statistics API 2014-10-21 Q4 2015 Q4 2016 Q1 2017
Media Capture from DOM Elements 2015-02-19 Q4 2015 Q3 2016 Q4 2016
Audio Output Devices API 2015-02-10 Q3 2015 Q2 2016 Q3 2016
Screen Capture 2015-02-10 Q4 2015 Q3 2016 Q4 2016
WebRTC next version Q1 2016 Q2 2017 Q4 2017 Q1 2018

Dependencies and Liaisons

W3C Groups

HTML Working Group
The HTML Working Group defines a number of markup elements and APIs that serves as the basis on which the RTC APIs have been developed; in particular, several specifications of this group extends the <audio> and <video> elements.
Web Applications Working Group
Some of the Web Applications Working Group APIs (such as the Web Sockets API) have served as inspiration or starting points for the APIs developed by the WebRTC Working Group. The work on Push Notifications provides an important feature for many WebRTC use cases. In addition, all the APIs developed by this group are based on WebIDL which the Web Applications Working Group is specifying.
Device APIs Working Group
The Device APIs Working Group jointly develops the media capture-related APIs with this group.
Audio Working Group
The API developed by the Audio Working Group builds upon the MediaStream object built by this group; further collaboration on the management of audio output device is expected.
Web Application Security Working Group
The Web Application Security Working Group is developing guidance on APIs that expose sensitive information, and an API to manage permissions, both of which matter to several of this group specifications.
Web Cryptography Working Group
WebRTC connections are encrypted end-to-end; collaboration is expected with the Web Cryptography Working Group on exposing and manipulating some of the cryptography functions used.
Second Screen Presentation Working Group
The Second Screen Presentation Working Group is developing APIs to allow rendering of media on secondary devices; potential overlap with features enabled by the Audio Output Devices API will need to be looked at.
WAI Protocols and Formats Working Group
Reviews from the WAI PF Working Group will be required to ensure the APIs allow to create an accessible user experience.
Web and TV Interest Group
Work on gathering use cases and requirements for Home Networking scenarios within the Web and TV Interest Group may uncover aspects that affect the design of real-time communications functions. The WebRTC Working Group will coordinate with the Web and TV Interest Group on these use cases and requirements as appropriate.
Privacy Interest Group
Several of the specifications developed by this group have potential impact on the privacy of users; reviews from the privacy interest group will be sought on these specifications.
Web Security Interest Group
Several of the specifications developed by this group have a complex impact on the security model of the Web; reviews from the Web Security Interest Group will be sought.

External Groups

IETF Real-Time Communication in WEB-browsers group (RTCWeb)
The RTC APIs developed by this group will build upon the protocols and formats developed in the IETF RTCWeb Working Group, which will in general handle dependencies to other IETF Working Groups.


To be successful, the Web Real-Time Communications Working Group is expected to have 10 or more active participants for its duration. Effective participation to Web Real-Time Communications Working Group is expected to consume one work day per week for each participant; two days per week for editors. The Web Real-Time Communications Working Group will allocate also the necessary resources for building Test Suites for each specification.

Participants are reminded of the Good Standing requirements of the W3C Process.


This group primarily conducts its work on the public mailing list public-webrtc@w3.org (archives).

The group uses a Member-confidential mailing list for administrative purposes and, at the discretion of the Chairs and participants of the group, for Member-only discussions in special cases when a particular participant requests such a discussion.

Information about the group (deliverables, participants, face-to-face meetings, teleconferences, etc.) is available from the Web Real-Time Communications Working Group home page.

Decision Policy

As explained in the Process Document (section 3.3), this group will seek to make decisions when there is consensus. When the Chair puts a question and observes dissent, after due consideration of different opinions, the Chair should record a decision (possibly after a formal vote) and any objections, and move on.

After mailing list and other informal discussion, substantive change proposals should be submitted as GitHub pull requests. These can come from the editors or from WG members.

Chairs are responsible for determining whether or not there is WG consensus for the changes contained in a pull request.

Editors are responsible for “curating” the pull requests to reject frivolous ones and substantive ones that the Chairs have determined do not comply with the IPR policies.

In cases where the editors make substantive changes without WG consensus, those changes must be labelled as provisional. The chairs are responsible for resolving the status of such changes.

Patent Policy

This Working Group operates under the W3C Patent Policy (5 February 2004 Version). To promote the widest adoption of Web standards, W3C seeks to issue Recommendations that can be implemented, according to this policy, on a Royalty-Free basis.

For more information about disclosure obligations for this group, please see the W3C Patent Policy Implementation.

About this Charter

This charter for the Web Real-Time Communications Working Group has been created according to section 6.2 of the Process Document. In the event of a conflict between this document or the provisions of any charter and the W3C Process, the W3C Process shall take precedence.

This charter updates and replaces the first WebRTC Working Group charter approved in 2011.

This charter was udpated on September 13 2016 to reflect the change of Chairs of the group, as Erik Lagerway stepped down, having chaired since June 2015.

Initial charter written by François Daoust and Dominique Hazaël-Massieux, based on initial input from Harald Alvestrand; the updated charter reflects much broader input from the WebRTC Working Group.