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WebRTC

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This is an old revision of this page, as edited by OMouse (talk | contribs) at 06:08, 15 January 2014 (→Design: updated names of components to match latest webrtc draft, added references to the relevant sections in the draft). The present address (URL) is a permanent link to this revision, which may differ significantly from the current revision.

WebRTC (Web Real-Time Communication) is an API definition being drafted by the World Wide Web Consortium (W3C) to enable browser-to-browser applications for voice calling, video chat, and P2P file sharing without plugins.

History

A project known as WebRTC, for browser-based real-time communication, was open sourced by Google in May 2011.[1] This has been followed by ongoing work to standardise the relevant protocols in the IETF[2] and browser APIs in the W3C.[3]

The W3C draft of WebRTC[4] is a work in progress with advanced implementations in the Chrome and Firefox browsers. The API is based on preliminary work done in the WHATWG.[5] It was referred as the ConnectionPeer API, and a pre-standards concept implementation was created at Ericsson Labs.[6] The Web Real-Time Communications Working Group expects this specification to evolve significantly based on:

  • The outcomes of ongoing exchanges in the companion RTCWEB group at IETF[7] to define the set of protocols that, together with this document, will enable real-time communications in Web browsers.
  • Privacy issues that arise when exposing local capabilities and local streams.
  • Technical discussions within the group, on implementing data channels in particular.[8]
  • Experience gained through early experimentation.
  • Feedback received from other groups and individuals.

Design

Major components of WebRTC include:

  • getUserMedia, which allows a web browser to access the camera and microphone and to capture media[9]
  • RTCPeerConnection, which sets up audio/video calls[10]
  • RTCDataChannels, which allow browsers to share data via peer-to-peer[11]

As of March 2012 the IETF WebRTC Codec and Media Processing Requirements draft[12] requires implementations to provide PCMA/PCMU (RFC 3551), Telephone Event as DTMF (RFC 4733), and Opus (RFC 6716), along with a number of video codec minimum capabilities. The Peerconnection, Data channels and a media capture browser APIs are detailed in the W3C.

Support

WebRTC is supported in the following browsers.

See also

References

  1. ^ "Google release of WebRTC source code from Harald Alvestrand on 2011-05-31". public-webrtc@w3.org. Retrieved 2012-09-12.
  2. ^ Charter of the Real-Time Communication in WEB-browsers (rtcweb) working group
  3. ^ "WebRTC 1.0: Real-time Communication Between Browsers". W3.org. Retrieved 2012-09-12.
  4. ^ "WebRTC 1.0: Real-time Communication Between Browsers". Dev.w3.org. Retrieved 2012-09-12.
  5. ^ "Introduction — HTML Standard". Whatwg.org. Retrieved 2012-09-12.
  6. ^ "Beyond HTML5: Peer-to-Peer Conversational Video | Ericsson Labs". Labs.ericsson.com. Retrieved 2012-09-12.
  7. ^ "Rtcweb Status Pages". Tools.ietf.org. Retrieved 2012-09-12.
  8. ^ "draft-jesup-rtcweb-data-protocol-00 - WebRTC Data Channel Protocol". Tools.ietf.org. Retrieved 2012-09-12.
  9. ^ "Media Capture and Streams: getUserMedia". W3C. 2013-09-03. Retrieved 2014-01-15.
  10. ^ "WebRTC: RTCPeerConnection Interface". W3C. 2013-09-10. Retrieved 2014-01-15.
  11. ^ "WebRTC: RTCDataChannel". W3C. 2013-09-10. Retrieved 2014-01-15.
  12. ^ "draft-cbran-rtcweb-codec-02 - WebRTC Codec and Media Processing Requirements". Tools.ietf.org. 2012-03-12. Retrieved 2012-09-12.
  13. ^ https://www.mozilla.org/en-US/firefox/22.0/releasenotes/
  14. ^ http://my.opera.com/ODIN/blog/opera-desktop-18-released
  15. ^ https://bugzilla.mozilla.org/show_bug.cgi?id=750010
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