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WebRTC

Summary

WebRTC (Web Real-Time Communication) is a free and open-source project providing web browsers and mobile applications with real-time communication (RTC) via 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.[3] Supported by Apple, Google, Microsoft, Mozilla, and Opera, WebRTC specifications have been published by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF).[4][5]

WebRTC
WebRTC Logo.svg
Original author(s)Justin Uberti
Peter Thatcher
Initial release2011; 11 years ago (2011)
Stable release
1.0[1] / May 4, 2018; 4 years ago (2018-05-04)
Repositorywebrtc.googlesource.com
Written inC++,[2] JavaScript
Standard(s)w3.org/TR/webrtc/
LicenseBSD license
Websitewebrtc.org

According to the webrtc.org website, the purpose of the project is to "enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols".[6]

HistoryEdit

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.[7][8] In May 2011, Google released an open-source project for browser-based real-time communication known as WebRTC.[9] This has been followed by ongoing work to standardize the relevant protocols in the IETF[10] and browser APIs in the W3C.[11]

In January 2011, Ericsson Labs built the first implementation of WebRTC using a modified WebKit library.[12][13] In October 2011, the W3C published its first draft for the spec.[14] 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.[15]

The W3C draft API was based on preliminary work done in the WHATWG.[16] It was referred to as the ConnectionPeer API, and a pre-standards concept implementation was created at Ericsson Labs.[12] The WebRTC Working Group expects this specification to evolve significantly based on:

  • Outcomes of ongoing exchanges in the companion RTCWEB group at IETF[17] 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[18]
  • Experience gained through early experimentation
  • Feedback from other groups and individuals

In November 2017, the WebRTC 1.0 specification transitioned from Working Draft to Candidate Recommendation.[19]

In January 2021, the WebRTC 1.0 specification transitioned from Candidate Recommendation to Recommendation.[4]

DesignEdit

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).[20]
  • RTCPeerConnection enables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management.[21]
  • RTCDataChannel allows bidirectional communication of arbitrary data between peers. The data is transported using SCTP over DTLS.[22] It uses the same API as WebSockets and has very low latency.[23]

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.[24]

The WebRTC API includes no provisions for signaling, that is discovering peers to connect to and determine how to establish connections among them. Applications use Interactive Connectivity Establishment for connections and are responsible for managing sessions, possibly relying on any of Session Initiation Protocol, Extensible Messaging and Presence Protocol, Message Queuing Telemetry Transport, Matrix, or another protocol. Signaling may depend on one or more servers.[25][26]

RFC 7478 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 specification.

W3C is developing ORTC (Object Real-Time Communications) for WebRTC.[27]

ApplicationsEdit

WebRTC allows browsers to stream files directly to one another, reducing or entirely removing the need for server-side file hosting. WebTorrent uses a WebRTC transport to enable peer-to-peer file sharing using the BitTorrent protocol in the browser.[28] Some file-sharing websites use it to allow users to send files directly to one another in their browsers, although this requires the uploader to keep the tab open until the file has been downloaded.[29][30][31] A few CDNs, such as the Microsoft-owned Peer5, use the client's bandwidth to upload media to other connected peers, enabling each peer to act as an edge server.[32][33]

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.[34]

SupportEdit

WebRTC is supported by the following browsers:

GStreamer directly provides a free WebRTC implementation.[40]

Codec support across browsersEdit

Support for individual codecs is not uniform. WebRTC establishes a standard set of codecs which all compliant browsers are required to implement. Some browsers may choose to allow other codecs as well. [41]

Video codec compatibility
Codec name Profile(s) Browser compatibility
H.264 Constrained Baseline (CB) Chrome (52+), Edge, Firefox[1], Safari
VP8 - Chrome, Edge, Firefox, Safari (12.1+)[42]
VP9 - Chrome (48+), Firefox
Audio codec compatibility
Codec name Browser compatibility
Opus Chrome, Edge, Firefox, Safari
G.711 PCM (A-law) Chrome, Firefox, Safari
G.711 PCM (µ-law) Chrome, Firefox, Safari
G.722 Chrome, Firefox, Safari
iLBC Chrome, Safari
iSAC Chrome, Safari

ConcernsEdit

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.[43] 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,[44] although the uBlock Origin add-on can fix this problem (as some browsers now fix this problem by themselves, from uBlock Origin v1.38 onwards this option has been disabled on these browsers[45]). As of September 2019, this WebRTC flaw still surfaces on Firefox 69.x and still by default exposes the user's internal IP address to the web.[46] In Firefox WebRTC can be disabled via media.peerconnection.enabled config-option.

ReferencesEdit

  1. ^ "WebRTC 1.0: Real-time Communication Between Browsers". World Wide Web Consortium. 27 September 2018. Retrieved 25 March 2019.
  2. ^ "Src/webrtc - Git at Google".
  3. ^ How WebRTC Is Revolutionizing Telephony. Blogs.trilogy-lte.com (2014-02-21). Retrieved on 2014-04-11.
  4. ^ a b "Web Real-Time Communications (WebRTC) transforms the communications landscape as it becomes a World Wide Web Consortium (W3C) Recommendation and Internet Engineering Task Force (IETF) standards". World Wide Web Consortium. 26 Jan 2021. Retrieved 27 Jan 2021.{{cite web}}: CS1 maint: url-status (link)
  5. ^ "Rtcweb Status Pages". tools.ietf.org. Retrieved 2021-02-18.
  6. ^ "WebRTC". WebRTC. Archived from the original on 9 January 2018. Retrieved 6 February 2018.
  7. ^ "Are the WebRTC components from Google's acquisition of Global IP Solutions?". WebRTC. Retrieved 6 February 2018.
  8. ^ Wauters, Robin (18 May 2010). "Google makes $68.2 million cash offer for Global IP Solutions". TechCrunch. Retrieved 6 February 2018.
  9. ^ Harald Alvestrand (2011-05-31). "Google release of WebRTC source code". public-webrtc@w3.org. Retrieved 2012-09-12.
  10. ^ Charter of the Real-Time Communication in WEB-browsers (rtcweb) working group
  11. ^ "WebRTC 1.0: Real-time Communication Between Browsers". W3.org. Retrieved 2012-09-12.
  12. ^ a b "Beyond HTML5: Peer-to-Peer Conversational Video". Ericsson Research blog. Labs.ericsson.com. 25 January 2011. Archived from the original on 25 February 2017. Retrieved 21 March 2021.
  13. ^ Stefan Håkansson; Stefan Ålund (26 May 2011). "Beyond HTML5: Experiment with Real-Time Communication in a Browser". Ericsson Research blog. Retrieved 6 February 2018.
  14. ^ "WebRTC 1.0: Real-time Communication Between Browsers (W3C Working Draft 27 October 2011)". World Wide Web Consortium. 27 October 2011. Retrieved 6 February 2018.
  15. ^ Nowak, Szymon. "WebRTC: So Much More Than Videoconferencing". GitHub. Retrieved 6 February 2018.
  16. ^ "Introduction — HTML Standard". Whatwg.org. Retrieved 2012-09-12.
  17. ^ "Rtcweb Status Pages". Tools.ietf.org. Retrieved 2012-09-12.
  18. ^ "draft-jesup-rtcweb-data-protocol-00 - WebRTC Data Channel Protocol". Tools.ietf.org. Retrieved 2012-09-12.
  19. ^ "WebRTC 1.0: Real-time Communication Between Browsers (W3C Candidate Recommendation 02 November 2017)". 2 November 2017.
  20. ^ "Media Capture and Streams: getUserMedia". W3C. 2013-09-03. Retrieved 2014-01-15.
  21. ^ "WebRTC: RTCPeerConnection Interface". W3C. 2013-09-10. Retrieved 2014-01-15.
  22. ^ "RFC 8831 - WebRTC Data Channels". datatracker.ietf.org. Retrieved 2022-03-10.
  23. ^ "WebRTC: RTCDataChannel". W3C. 2013-09-10. Retrieved 2014-01-15.
  24. ^ "Identifiers for WebRTC's Statistics API". W3C. 2014-09-29.
  25. ^ Tsahi Levent-Levi (13 April 2020). "WebRTC Server: What is it exactly?". BlogGeek.me.
  26. ^ Tsahi Levent-Levi (13 November 2014). "Matrix.org and WebRTC: An Interview with Matthew Hodgson". BlogGeek.me.
  27. ^ "W3C ORTC (Object Real-time Communications) Community Group".
  28. ^ "WebTorrent FAQ". webtorrent.io. Retrieved 2022-03-10.
  29. ^ "How to Transfer Files Between Linux, Android, and iOS Using Snapdrop". MUO. 2021-08-04. Retrieved 2022-03-10.
  30. ^ Pinola, Melanie (2014-04-07). "The easiest and quickest way to transfer files between devices on the same network". Computerworld. Retrieved 2022-03-10.
  31. ^ "FilePizza: share files without the middleman in your browser - gHacks Tech News". gHacks Technology News. 2015-05-12. Retrieved 2022-03-10.
  32. ^ Foley, Mary Jo. "Microsoft acquires Peer5 to supplement Teams' live video streaming". ZDNet. Retrieved 2022-03-10.
  33. ^ "Overview - Peer5 P2P Docs". docs.peer5.com. Retrieved 2022-03-10.
  34. ^ "Catch the Babelfish: Irish telco devises a new kind of cloud phone". November 2017.
  35. ^ "ORTC API is now available in Microsoft Edge". Microsoft. 2015-09-18.
  36. ^ Firefox Notes - Desktop. Mozilla.org (2013-06-25). Retrieved on 2014-04-11.
  37. ^ "Safari 11.0". Apple Inc. Retrieved 6 June 2017.
  38. ^ Opera News. blogs.opera.com (2013-11-19). Retrieved on 2015-09-17.
  39. ^ Firefox Notes - Desktop. Mozilla.org (2013-09-17). Retrieved on 2014-08-04.
  40. ^ "GStreamer 1.14 release notes". gstreamer.freedesktop.org. Retrieved 2019-12-19. since version 1.14
  41. ^ "Codecs used by WebRTC - Web media technologies | MDN". developer.mozilla.org. Retrieved 2021-07-29.
  42. ^ Fablet, Youenn (2019-03-12). "On the Road to WebRTC 1.0, Including VP8". WebKit. Retrieved 2021-07-29.
  43. ^ Huge Security Flaw Leaks VPN Users’ Real IP-addresses TorrentFreak.com (2015-01-30). Retrieved on 2015-02-21.
  44. ^ STUN IP Address requests for WebRTC Retrieved on 2015-02-21.
  45. ^ Raymond Hill (17 Sep 2021). "Prevent WebRTC from leaking local IP address". uBlock Origin documentation. Retrieved 18 Dec 2021.
  46. ^ Raymond Hill (26 March 2016). "Prevent WebRTC from leaking local IP address". uBlock Origin documentation. Retrieved 1 Sep 2016.

Further readingEdit

  • Proust, S., ed. (May 2016). Additional WebRTC Audio Codecs for Interoperability. IETF. doi:10.17487/RFC7875. RFC 7875. Retrieved 2016-10-12.
  • Valin, J. M.; Bran, C. (May 2016). WebRTC Audio Codec and Processing Requirements. IETF. doi:10.17487/RFC7874. RFC 7874. Retrieved 2016-10-12.
  • Roach, A. B. (March 2016). WebRTC Video Processing and Codec Requirements. IETF. doi:10.17487/RFC7742. RFC 7742. Retrieved 2016-10-12.
  • Perumal, M.; Wing, D.; Ravindranath, R.; Reddy, T.; Thomson, M. (October 2015). Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness. IETF. doi:10.17487/RFC7675. RFC 7675. Retrieved 2016-10-12.
  • Holmberg, C.; Hakansson, S.; Eriksson, G. (March 2015). Web Real-Time Communication Use Cases and Requirements. IETF. doi:10.17487/RFC7478. RFC 7478. Retrieved 2016-10-12.

External linksEdit

  • Official website  
  • W3C Web Real-Time Communications Working Group
  • IETF Real-Time Communication in WEB-browsers (rtcweb) Working Group
  • Video chat demo app based on WebRTC
  • libdatachannel, open-source WebRTC network library

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