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WebRTC

Original author(s)	Justin Uberti Peter Thatcher
Initial release	2011; 13 years ago (2011)
Stable release	1.0 / May 4, 2018; 6 years ago (2018-05-04)
Repository	webrtc.googlesource.com
Written in	C++, JavaScript
Standard(s)	w3.org/TR/webrtc/
License	BSD license
Website	webrtc.org

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.

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).

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

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

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

• Outcomes of ongoing exchanges in the companion RTCWEB group at IETF 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
• 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.

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

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

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.

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.

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.

Applications

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. 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. 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.

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.

Support

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

• Desktop PC
  • Microsoft Edge 12+
  • Google Chrome 28+
  • Mozilla Firefox 22+
  • Safari 11+
  • Opera 18+
  • Vivaldi 1.9+
  • Brave
• Android
  • Google Chrome 28+ (enabled by default since 29)
  • Mozilla Firefox 24+
  • Opera Mobile 12+
• ChromeOS
• Firefox OS
• BlackBerry 10
• iOS
  • MobileSafari/WebKit (iOS 11+)
• Tizen 3.0
• GStreamer directly provides a free WebRTC implementation.
• OvenMediaEngine
• Ant Media Server

Codec support across browsers

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

Vulnerability

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. 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.

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. 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).

See also

• CU-RTC-WEB

References

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 2018-04-23. Retrieved 2018-04-22.
3. How WebRTC Is Revolutionizing Telephony Archived 2014-04-07 at the Wayback Machine. Blogs.trilogy-lte.com (2014-02-21). Retrieved on 2014-04-11.
4. ^ "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.
5. "Rtcweb Status Pages". tools.ietf.org. Archived from the original on 2020-04-20. Retrieved 2021-02-18.
6. "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.
7. 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.
8. Harald Alvestrand (2011-05-31). "Google release of WebRTC source code". public-webrtc@w3.org. Archived from the original on 2013-02-27. Retrieved 2012-09-12.
9. "Charter of the Real-Time Communication in WEB-browsers (rtcweb) working group". Archived from the original on 2013-11-11. Retrieved 2013-11-11.
10. "WebRTC 1.0: Real-time Communication Between Browsers". W3.org. Archived from the original on 2012-09-05. Retrieved 2012-09-12.
11. ^ "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.
12. Stefan Håkansson; Stefan Ålund (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.
13. "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.
14. Nowak, Szymon. "WebRTC: So Much More Than Videoconferencing". GitHub. Archived from the original on 7 February 2018. Retrieved 6 February 2018.
15. "Introduction — HTML Standard". Whatwg.org. Archived from the original on 2014-08-16. Retrieved 2012-09-12.
16. "Rtcweb Status Pages". Tools.ietf.org. Archived from the original on 2012-09-05. Retrieved 2012-09-12.
17. 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.
18. "draft-jesup-rtcweb-data-protocol-00 - WebRTC Data Channel Protocol". Tools.ietf.org. Archived from the original on 2012-10-31. Retrieved 2012-09-12.
19. "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.
20. "Media Capture and Streams: getUserMedia". W3C. 2013-09-03. Archived from the original on 2014-01-02. Retrieved 2014-01-15.
21. "WebRTC: RTCPeerConnection Interface". W3C. 2013-09-10. Archived from the original on 2012-09-05. Retrieved 2014-01-15.
22. "RFC 8831 - WebRTC Data Channels". datatracker.ietf.org. Archived from the original on 2022-03-10. Retrieved 2022-03-10.
23. "WebRTC: RTCDataChannel". W3C. 2013-09-10. Archived from the original on 2012-09-05. Retrieved 2014-01-15.
24. "Identifiers for WebRTC's Statistics API". W3C. 2014-09-29. Archived from the original on 2017-07-15. Retrieved 2017-12-02.
25. 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.
26. 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.
27. "W3C ORTC (Object Real-time Communications) Community Group". Archived from the original on 2014-10-29. Retrieved 2014-10-29.
28. "WebTorrent FAQ". webtorrent.io. Archived from the original on 2022-03-11. Retrieved 2022-03-10.
29. "How to Transfer Files Between Linux, Android, and iOS Using Snapdrop". MUO. 2021-08-04. Archived from the original on 2022-01-29. 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. Archived from the original on 2022-06-28. Retrieved 2022-03-10.
31. "FilePizza: share files without the middleman in your browser - gHacks Tech News". gHacks Technology News. 2015-05-12. Archived from the original on 2022-01-23. Retrieved 2022-03-10.
32. Foley, Mary Jo. "Microsoft acquires Peer5 to supplement Teams' live video streaming". ZDNet. Archived from the original on 2022-03-10. Retrieved 2022-03-10.
33. "Overview - Peer5 P2P Docs". docs.peer5.com. Archived from the original on 2022-03-16. Retrieved 2022-03-10.
34. "Catch the Babelfish: Irish telco devises a new kind of cloud phone". November 2017. Archived from the original on 2017-11-01. Retrieved 2017-11-20.
35. "ORTC API is now available in Microsoft Edge". Microsoft. 2015-09-18. Archived from the original on 2015-10-09. Retrieved 2015-09-20.
36. Firefox Notes - Desktop Archived 2014-08-21 at the Wayback Machine. Mozilla.org (2013-06-25). Retrieved on 2014-04-11.
37. "Safari 11.0". Apple Inc. Archived from the original on 14 November 2017. Retrieved 6 June 2017.
38. Opera News Archived 2015-09-07 at the Wayback Machine. blogs.opera.com (2013-11-19). Retrieved on 2015-09-17.
39. Firefox Notes - Desktop Archived 2021-04-01 at the Wayback Machine. Mozilla.org (2013-09-17). Retrieved on 2014-08-04.
40. "GStreamer 1.14 release notes". gstreamer.freedesktop.org. Archived from the original on 2018-03-20. Retrieved 2019-12-19. since version 1.15
41. "Scalable Streaming Solutions with Ant Media Server". antmedia.io. 2017-06-07. Retrieved 2024-12-11.
42. "Codecs used by WebRTC - Web media technologies | MDN". developer.mozilla.org. Archived from the original on 2021-07-27. Retrieved 2021-07-29.
43. Fablet, Youenn (2019-03-12). "On the Road to WebRTC 1.0, Including VP8". WebKit. Archived from the original on 2021-07-29. Retrieved 2021-07-29.
44. Huge Security Flaw Leaks VPN Users’ Real IP-addresses Archived 2021-01-08 at the Wayback Machine TorrentFreak.com (2015-01-30). Retrieved on 2015-02-21.
45. STUN IP Address requests for WebRTC Archived 2015-02-18 at the Wayback Machine Retrieved on 2015-02-21.
46. Timmerman, Crystal (28 February 2022). "WebRTC leaks real IP addresses (even with VPN)". IPVanish. Archived from the original on 13 August 2022. Retrieved 12 August 2022.
47. Raymond Hill (17 Sep 2021). "Prevent WebRTC from leaking local IP address". uBlock Origin documentation. Archived from the original on 21 February 2016. Retrieved 18 Dec 2021.

Further reading

• 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 links

• 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

• WebRTC
• Peer-to-peer
• Software using the BSD license
• Web development
• Web standards