WebRTC

WebRTC
Original author(s) Justin Uberti
Peter Thatcher
Initial release 2011 (2011)
Stable release
1.1 / May 4, 2017 (2017-05-04)
Repository webrtc.googlesource.com
Written in C++[1], JavaScript
License BSD license
Website webrtc.org

WebRTC (Web Real-Time Communication) is a free, open-source project that provides web browsers and mobile applications with real-time communication (RTC) via simple application programming interfaces (APIs). It allows audio and video communication to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps.[2] Supported by Google, Microsoft, Mozilla, and Opera, WebRTC is being standardized through the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF).[3]

Its mission is to "enable rich, high-quality RTP applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols".[3] The reference implementation is released as free software under the terms of a BSD license. OpenWebRTC provides another free implementation based on the multimedia framework GStreamer. JavaScript inventor Brendan Eich called it a "new front in the long war for an open and unencumbered web".[4]

History

In May 2010, Google bought Global IP Solutions or GIPS, a VoIP and videoconferencing software company that had developed many components required for RTC, such as codecs and echo cancellation techniques. Google open-sourced the GIPS technology and engaged with relevant standards bodies at the IETF and W3C to ensure industry consensus.[5][6] In May 2011, Google released an open-source project for browser-based real-time communication known as WebRTC.[7] This has been followed by ongoing work to standardize the relevant protocols in the IETF[8] and browser APIs in the W3C.[9]

In May 2011, Ericsson Labs built the first implementation of WebRTC using a modified WebKit library.[10] In October 2011, the W3C published its first draft for the spec.[11] WebRTC milestones include the first cross-browser video call (February 2013), first cross-browser data transfers (February 2014), and as of July 2014 Google Hangouts was "kind of" using WebRTC.[12]

The W3C draft of WebRTC[13] 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.[14] It was referred to as the ConnectionPeer API, and a pre-standards concept implementation was created at Ericsson Labs.[15] The WebRTC Working Group expects this specification to evolve significantly based on:

  • Outcomes of ongoing exchanges in the companion RTCWEB group at IETF[16] to define the set of protocols that, together with this document, define real-time communications in web browsers. While no one signaling protocol is mandated, SIP over Websockets ( RFC 7118) is often used partially due to the applicability of SIP to most of the envisaged communication scenarios as well as the availability of open-source software such as JsSIP.
  • Privacy issues that arise when exposing local capabilities and local streams
  • Technical discussions within the group, on implementing data channels in particular[17]
  • Experience gained through early experimentation
  • Feedback from other groups and individuals

Overview

Design

Major components of WebRTC include several JavaScript APIs:

  • getUserMedia acquires the audio and video media (e.g., by accessing a device's camera and microphone).[18]
  • RTCPeerConnection enables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management.[19]
  • RTCDataChannel allows bidirectional communication of arbitrary data between peers. It uses the same API as WebSockets and has very low latency.[20]

The WebRTC API also includes a statistics function:

  • getStats allows the web application to retrieve a set of statistics about WebRTC sessions. These statistics data are being described in a separate W3C document.[21]

RFC 7874 requires implementations to provide PCMA/PCMU ( RFC 3551), Telephone Event as DTMF ( RFC 4733), and Opus ( RFC 6716) audio codecs as minimum capabilities. The PeerConnection, data channel and media capture browser APIs are detailed in the W3C.

W3C is developing ORTC (Object Real-Time Communications) for WebRTC.[22] This is commonly referred to as WebRTC 1.1.

Examples

Although initially developed for web browsers, WebRTC has applications for non-browser devices, including mobile platforms and IoT devices. Examples include browser-based VoIP telephony, also called cloud phones or web phones, which allow calls to be made and received from within a web browser, replacing the requirement to download and install a softphone.[23]

Support

WebRTC is supported by the following browsers:

Support was not included in Internet Explorer prior to its final feature release in October 2013[29], but 3rd party plugins are available to add the support of WebRTC to IE and Safari for MacOS.[30][31] At WWDC 2017, Apple announced Safari would get WebRTC support in Safari 11,[26] and it was made available in release 32 of the Safari Technology Preview.[32]

Concerns

In January 2015, TorrentFreak reported a serious security flaw in browsers that support WebRTC, saying that it compromised the security of VPN tunnels, by exposing the true IP address of a user.[33] The IP address read requests are not visible in the browser's developer console, and they are not blocked by most ad blocking/privacy/security add-ons, enabling online tracking by advertisers and other entities despite precautions[34] (however the uBlock Origin add-on can fix this problem).[35]

See also

References

  1. ↑ https://webrtc.googlesource.com/src/webrtc/
  2. ↑ How WebRTC Is Revolutionizing Telephony. Blogs.trilogy-lte.com (2014-02-21). Retrieved on 2014-04-11.
  3. 1 2 "WebRTC". WebRTC. Archived from the original on 8 January 2018. Retrieved 6 February 2018.
  4. ↑ Justin Uberti; Sam Dutton. "WebRTC: Plugin-free realtime communication". Retrieved 6 February 2018.
  5. ↑ "Are the WebRTC components from Google's acquisition of Global IP Solutions?". WebRTC. Retrieved 6 February 2018.
  6. ↑ Wauters, Robin (18 May 2010). "Google makes $68.2 million cash offer for Global IP Solutions". TechCrunch. Retrieved 6 February 2018.
  7. ↑ Harald Alvestrand (2011-05-31). "Google release of WebRTC source code". public-webrtc@w3.org. Retrieved 2012-09-12.
  8. ↑ Charter of the Real-Time Communication in WEB-browsers (rtcweb) working group
  9. ↑ "WebRTC 1.0: Real-time Communication Between Browsers". W3.org. Retrieved 2012-09-12.
  10. ↑ Stefan Håkansson; Stefan Ålund (26 May 2011). "Beyond HTML5: Experiment with Real-Time Communication in a Browser". Ericsson Research blog. Retrieved 6 February 2018.
  11. ↑ "WebRTC 1.0: Real-time Communication Between Browsers". World Wide Web Consortium. 27 October 2011. Retrieved 6 February 2018.
  12. ↑ Nowak, Szymon. "WebRTC: So Much More Than Videoconferencing". GitHub. Retrieved 6 February 2018.
  13. ↑ "WebRTC 1.0: Real-time Communication Between Browsers". Dev.w3.org. Retrieved 2012-09-12.
  14. ↑ "Introduction — HTML Standard". Whatwg.org. Retrieved 2012-09-12.
  15. ↑ "Beyond HTML5: Peer-to-Peer Conversational Video". Labs.ericsson.com. Retrieved 2012-09-12.
  16. ↑ "Rtcweb Status Pages". Tools.ietf.org. Retrieved 2012-09-12.
  17. ↑ "draft-jesup-rtcweb-data-protocol-00 - WebRTC Data Channel Protocol". Tools.ietf.org. Retrieved 2012-09-12.
  18. ↑ "Media Capture and Streams: getUserMedia". W3C. 2013-09-03. Retrieved 2014-01-15.
  19. ↑ "WebRTC: RTCPeerConnection Interface". W3C. 2013-09-10. Retrieved 2014-01-15.
  20. ↑ "WebRTC: RTCDataChannel". W3C. 2013-09-10. Retrieved 2014-01-15.
  21. ↑ "Identifiers for WebRTC's Statistics API". W3C. 2014-09-29.
  22. ↑ "W3C ORTC (Object Real-time Communications) Community Group".
  23. ↑ https://www.siliconrepublic.com/comms/babelfish-softphone-cloud-goldfish
  24. ↑ "ORTC API is now available in Microsoft Edge". Microsoft.
  25. ↑ Firefox Notes - Desktop. Mozilla.org (2013-06-25). Retrieved on 2014-04-11.
  26. 1 2 "Safari 11.0". Apple Inc. Retrieved 6 June 2017.
  27. ↑ Opera News. blogs.opera.com (2013-11-19). Retrieved on 2015-09-17.
  28. ↑ Firefox Notes - Desktop. Mozilla.org (2013-09-17). Retrieved on 2014-08-04.
  29. ↑ "Internet Explorer Web Platform Status and Roadmap". Microsoft. Archived from the original on 2015-05-04. Retrieved 7 September 2014.
  30. ↑ "Priologic Releases First Open Source WebRTC Plugin for Internet Explorer". 2014-06-10. Archived from the original on 2014-06-26.
  31. ↑ Wilcox, Charles (2014-05-12). "Temasys Plugin Supports webRTC in Internet Explorer and Apple Safari, on Desktops" (Press release). Temasys Communications Pte Ltd. PRWeb.
  32. ↑ "Safari Technology Preview Release Notes". Retrieved 12 June 2017.
  33. ↑ Huge Security Flaw Leaks VPN Users’ Real IP-addresses TorrentFreak.com (2015-01-30). Retrieved on 2015-02-21.
  34. ↑ STUN IP Address requests for WebRTC Retrieved on 2015-02-21.
  35. ↑ Raymond Hill (26 March 2016). "Prevent WebRTC from leaking local IP address". uBlock Origin documentation. Retrieved 1 Sep 2016.

Further reading

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