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From Wikipedia, the free encyclopedia
WebRTC
Original author(s)Justin Uberti
Peter Thatcher
Initial release2011; 14 years ago (2011)
Stable release
1.0[1] / May 4, 2018; 7 years ago (2025-08-05)
Repositorywebrtc.googlesource.com
Written inC++,[2] JavaScript
Standard(s)w3.org/TR/webrtc/
LicenseBSD license[3]
Websitewebrtc.org
百度 她强调，一要进一步推动机关纪委规范运行和有效履职，主动承接起党风廉政建设日常工作职责，更好地发挥好机关纪委的作用，推动党风廉政建设主体责任向基层延伸；二要进一步加强工委机关党风廉政建设及相关工作，持之以恒抓好党风廉政建设基础性工作，让党员干部知敬畏、存戒惧、守底线；三要进一步重视和加强队伍建设，从健全组织架构、明确职责权限、完善工作机制、着力提升能力等方面入手，全面加强纪工委和机关纪委队伍建设，为推动机关党风廉政建设向纵深发展提供坚实保障。

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 and streaming to work inside web pages by allowing direct peer-to-peer communication, eliminating the need to install plugins or download native apps.[4]

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).[5][6]

ICE, STUN and TURN are the NAT traversal techniques used to connect to remote peers.[7]

History

[edit]

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.[8][9] In May 2011, Google released an open-source project for browser-based real-time communication known as WebRTC.[10] This has been followed by ongoing work to standardize the relevant protocols in the IETF[11] and browser APIs in the W3C.[12]

In January 2011, Ericsson Labs built the first implementation of WebRTC using a modified WebKit library.[13][14] In October 2011, the W3C published its first draft for the spec.[15] 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.[16]

The W3C draft API was based on preliminary work done in the WHATWG.[17] It was referred to as the ConnectionPeer API, and a pre-standards concept implementation was created at Ericsson Labs.[13] The WebRTC Working Group expects this specification to evolve significantly based on:

  • Outcomes of ongoing exchanges in the companion RTCWEB group at IETF[18] 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[19] 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[20]
  • 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.[21]

In January 2021, the WebRTC 1.0 specification transitioned from Candidate Recommendation to Recommendation.[5]

Design

[edit]

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).[22]
  • RTCPeerConnection enables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management.[23]
  • RTCDataChannel allows bidirectional communication of arbitrary data between peers. The data is transported using SCTP over DTLS.[24] It uses the same API as WebSockets and has very low latency.[25]

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.[26]

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 (XMPP), Message Queuing Telemetry Transport, Matrix, or another protocol. Signaling may depend on one or more servers.[27][28]

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.[29]

Applications

[edit]

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.[30] 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.[31][32][33] 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.[34][35]

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.[36]

Support

[edit]

WebRTC is supported by the following browsers (incomplete list; oldest supported version specified):

Codec support across browsers

[edit]

WebRTC establishes a standard set of codecs which all compliant browsers are required to implement. Some browsers may also support other codecs.[44]

Video codec compatibility
Codec name Profile Browser compatibility
H.264 Constrained Baseline (CB) Chrome (52+), Firefox[1], Safari
VP8 - Chrome, Firefox, Safari (12.1+)[45]
VP9 - Chrome (48+), Firefox
Audio codec compatibility
Codec name Browser compatibility
Opus Chrome, 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

Vulnerability

[edit]

In January 2015, TorrentFreak reported a serious security flaw in browsers supporting WebRTC, that compromised the security of VPN tunnels by exposing a user's true IP address.[46] The IP address read requests are not visible in the browser's developer console, and they are not blocked by most ad blocking, privacy and security add-ons, enabling online tracking despite precautions.[47]

It has been reported that the cause of the address leak is not a bug that can be patched, but is foundational to the way WebRTC operates; however, there are several solutions to mitigate the problem. WebRTC leakage can be tested for, and solutions are offered for most browsers.[48] WebRTC can be disabled, if not required, in most browsers. 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[49]).

See also

[edit]

References

[edit]
  1. ^ "WebRTC 1.0: Real-time Communication Between Browsers". World Wide Web Consortium. 27 September 2018. Archived from the original on 7 April 2019. Retrieved 25 March 2019.
  2. ^ "Src/webrtc - Git at Google". Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  3. ^ "WebRTC License".
  4. ^ How WebRTC Is Revolutionizing Telephony Archived 2025-08-05 at the Wayback Machine. Blogs.trilogy-lte.com (2025-08-05). Retrieved on 2025-08-05.
  5. ^ 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. Archived from the original on 27 July 2022. Retrieved 27 Jan 2021.
  6. ^ "Rtcweb Status Pages". tools.ietf.org. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  7. ^ "Introduction". W3C WebRTC.
  8. ^ "Are the WebRTC components from Google's acquisition of Global IP Solutions?". WebRTC. Archived from the original on 7 June 2011. Retrieved 6 February 2018.
  9. ^ Wauters, Robin (18 May 2010). "Google makes $68.2 million cash offer for Global IP Solutions". TechCrunch. Archived from the original on 7 February 2018. Retrieved 6 February 2018.
  10. ^ Harald Alvestrand (2025-08-05). "Google release of WebRTC source code". public-webrtc@w3.org. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  11. ^ "Charter of the Real-Time Communication in WEB-browsers (rtcweb) working group". Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  12. ^ "WebRTC 1.0: Real-time Communication Between Browsers". W3.org. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  13. ^ 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.
  14. ^ Stefan H?kansson; Stefan ?lund (26 May 2011). "Beyond HTML5: Experiment with Real-Time Communication in a Browser". Ericsson Research blog. Archived from the original on 7 February 2018. Retrieved 6 February 2018.
  15. ^ "WebRTC 1.0: Real-time Communication Between Browsers (W3C Working Draft 27 October 2011)". World Wide Web Consortium. 27 October 2011. Archived from the original on 29 October 2011. Retrieved 6 February 2018.
  16. ^ Nowak, Szymon. "WebRTC: So Much More Than Videoconferencing". GitHub. Archived from the original on 7 February 2018. Retrieved 6 February 2018.
  17. ^ "Introduction — HTML Standard". Whatwg.org. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  18. ^ "Rtcweb Status Pages". Tools.ietf.org. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  19. ^ SIP Trunking, MTPL (18 July 2023). "SIP Trunking VoIP with WebRTC SDK". Moon Technolabs. Archived from the original on 5 August 2023. Retrieved 18 July 2023.
  20. ^ Jesup, Randell; Loreto, Salvatore; Tüxen, Michael (5 March 2012). "draft-jesup-rtcweb-data-protocol-00 - WebRTC Data Channel Protocol". Tools.ietf.org. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  21. ^ "WebRTC 1.0: Real-time Communication Between Browsers (W3C Candidate Recommendation 02 November 2017)". 2 November 2017. Archived from the original on 2 November 2017. Retrieved 25 March 2019.
  22. ^ "Media Capture and Streams: getUserMedia". W3C. 2025-08-05. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  23. ^ "WebRTC: RTCPeerConnection Interface". W3C. 2025-08-05. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  24. ^ Jesup, Randell; Loreto, Salvatore; Tüxen, Michael (January 2021). "RFC 8831 - WebRTC Data Channels". datatracker.ietf.org. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  25. ^ "WebRTC: RTCDataChannel". W3C. 2025-08-05. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  26. ^ "Identifiers for WebRTC's Statistics API". W3C. 2025-08-05. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  27. ^ Tsahi Levent-Levi (13 April 2020). "WebRTC Server: What is it exactly?". BlogGeek.me. Archived from the original on 11 May 2020. Retrieved 10 June 2020.
  28. ^ Tsahi Levent-Levi (13 November 2014). "Matrix.org and WebRTC: An Interview with Matthew Hodgson". BlogGeek.me. Archived from the original on 25 February 2021. Retrieved 10 June 2020.
  29. ^ "W3C ORTC (Object Real-time Communications) Community Group". 19 August 2014. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  30. ^ "WebTorrent FAQ". webtorrent.io. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  31. ^ "How to Transfer Files Between Linux, Android, and iOS Using Snapdrop". MUO. 2025-08-05. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  32. ^ Pinola, Melanie (2025-08-05). "The easiest and quickest way to transfer files between devices on the same network". Computerworld. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  33. ^ "FilePizza: share files without the middleman in your browser - gHacks Tech News". gHacks Technology News. 2025-08-05. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  34. ^ Foley, Mary Jo. "Microsoft acquires Peer5 to supplement Teams' live video streaming". ZDNet. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  35. ^ "Overview - Peer5 P2P Docs". docs.peer5.com. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  36. ^ "Catch the Babelfish: Irish telco devises a new kind of cloud phone". November 2017. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  37. ^ "ORTC API is now available in Microsoft Edge". Microsoft. 2025-08-05. Archived from the original on 2025-08-05. Retrieved 2025-08-05.
  38. ^ Firefox Notes - Desktop Archived 2025-08-05 at the Wayback Machine. Mozilla.org (2025-08-05). Retrieved on 2025-08-05.
  39. ^ "Safari 11.0". Apple Inc. Archived from the original on 14 November 2017. Retrieved 6 June 2017.
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Further reading

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