WebRTC

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WebRTC (Web Real-Time Communication) is an API definition drafted by the World Wide Web Consortium (W3C) that supports browser-to-browser applications for voice calling, video chat, and P2P file sharing without plugins.[1]

History[edit]

In May 2011, Google released an open source project for browser-based real-time communication known as WebRTC.[2] This has been followed by ongoing work to standardise the relevant protocols in the IETF[3] and browser APIs in the W3C.[4]

The W3C draft of WebRTC[5] 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.[6] It was referred to as the ConnectionPeer API, and a pre-standards concept implementation was created at Ericsson Labs.[7] The Web Real-Time Communications Working Group expects this specification to evolve significantly based on:

  • Outcomes of ongoing exchanges in the companion RTCWEB group at IETF[8] to define the set of protocols that, together with this document, define 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[9]
  • Experience gained through early experimentation
  • Feedback from other groups and individuals

Design[edit]

Major components of WebRTC include:

  • getUserMedia, which allows a web browser to access the camera and microphone and to capture media[10]
  • RTCPeerConnection, which sets up audio/video calls[11]
  • RTCDataChannels, which allow browsers to share data via peer-to-peer[12]

As of March 2012, the IETF WebRTC Codec and Media Processing Requirements draft[13] 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[edit]

WebRTC is supported in the following browsers.

As of August 2014, Internet Explorer and Safari still lack the native support of WebRTC. Several plugins are available to add the support of WebRTC to these browsers.[17][18]

See also[edit]

References[edit]

  1. ^ How WebRTC Is Revolutionizing Telephony. Blogs.trilogy-lte.com (2014-02-21). Retrieved on 2014-04-11.
  2. ^ "Google release of WebRTC source code from Harald Alvestrand on 2011-05-31". public-webrtc@w3.org. Retrieved 2012-09-12. 
  3. ^ Charter of the Real-Time Communication in WEB-browsers (rtcweb) working group
  4. ^ "WebRTC 1.0: Real-time Communication Between Browsers". W3.org. Retrieved 2012-09-12. 
  5. ^ "WebRTC 1.0: Real-time Communication Between Browsers". Dev.w3.org. Retrieved 2012-09-12. 
  6. ^ "Introduction — HTML Standard". Whatwg.org. Retrieved 2012-09-12. 
  7. ^ "Beyond HTML5: Peer-to-Peer Conversational Video | Ericsson Labs". Labs.ericsson.com. Retrieved 2012-09-12. 
  8. ^ "Rtcweb Status Pages". Tools.ietf.org. Retrieved 2012-09-12. 
  9. ^ "draft-jesup-rtcweb-data-protocol-00 - WebRTC Data Channel Protocol". Tools.ietf.org. Retrieved 2012-09-12. 
  10. ^ "Media Capture and Streams: getUserMedia". W3C. 2013-09-03. Retrieved 2014-01-15. 
  11. ^ "WebRTC: RTCPeerConnection Interface". W3C. 2013-09-10. Retrieved 2014-01-15. 
  12. ^ "WebRTC: RTCDataChannel". W3C. 2013-09-10. Retrieved 2014-01-15. 
  13. ^ "draft-cbran-rtcweb-codec-02 - WebRTC Codec and Media Processing Requirements". Tools.ietf.org. 2012-03-12. Retrieved 2012-09-12. 
  14. ^ Firefox Notes - Desktop. Mozilla.org (2013-06-25). Retrieved on 2014-04-11.
  15. ^ Dev.Opera. My.opera.com. Retrieved on 2014-04-11.
  16. ^ Firefox Notes - Desktop. Mozilla.org (2013-09-17). Retrieved on 2014-08-04.
  17. ^ Priologic Releases First Open Source WebRTC Plugin for Internet Explorer
  18. ^ http://www.prweb.com/releases/2014/05/prweb11844308.htm

External links[edit]