WebRTC#Concerns

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.{{cite web|title=Are the WebRTC components from Google's acquisition of Global IP Solutions?|url=https://webrtc.org/faq/#are-the-webrtc-components-from-googles-acquisition-of-global-ip-solutions|website=WebRTC|access-date=6 February 2018|archive-date=7 June 2011|archive-url=https://web.archive.org/web/20110607005550/https://webrtc.org/faq/#are-the-webrtc-components-from-googles-acquisition-of-global-ip-solutions|url-status=dead}}{{cite news|last1=Wauters|first1=Robin|title=Google makes $68.2 million cash offer for Global IP Solutions|url=https://techcrunch.com/2010/05/18/google-makes-68-2-million-cash-offer-for-global-ip-solutions|access-date=6 February 2018|work=TechCrunch|date=18 May 2010|archive-date=7 February 2018|archive-url=https://web.archive.org/web/20180207062936/https://techcrunch.com/2010/05/18/google-makes-68-2-million-cash-offer-for-global-ip-solutions/|url-status=live}} 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.{{cite news|author1=Stefan Håkansson|author2=Stefan Ålund|title=Beyond HTML5: Experiment with Real-Time Communication in a Browser|url=https://www.ericsson.com/research-blog/beyond-html5-experiment-real-time-communication-browser|access-date=6 February 2018|work=Ericsson Research blog|date=26 May 2011|archive-date=7 February 2018|archive-url=https://web.archive.org/web/20180207012115/https://www.ericsson.com/research-blog/beyond-html5-experiment-real-time-communication-browser/|url-status=live}} In October 2011, the W3C published its first draft for the spec.{{cite web|title=WebRTC 1.0: Real-time Communication Between Browsers (W3C Working Draft 27 October 2011)|url=https://www.w3.org/TR/2011/WD-webrtc-20111027/|website=World Wide Web Consortium|access-date=6 February 2018|date=27 October 2011|archive-date=29 October 2011|archive-url=https://web.archive.org/web/20111029173130/https://www.w3.org/TR/2011/WD-webrtc-20111027/|url-status=live}} 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.{{cite web|last1=Nowak|first1=Szymon|title=WebRTC: So Much More Than Videoconferencing|url=https://szimek.github.io/presentation-meetjs-summit-2014-webrtc/#16|website=GitHub|access-date=6 February 2018|archive-date=7 February 2018|archive-url=https://web.archive.org/web/20180207005022/https://szimek.github.io/presentation-meetjs-summit-2014-webrtc/#16|url-status=live}}

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 ({{IETF RFC|7118}}) is often used partially due to the applicability of SIP{{Cite web |title=SIP Trunking VoIP with WebRTC SDK |last=SIP Trunking |first=MTPL |website=Moon Technolabs |date=18 July 2023 |url=https://www.moontechnolabs.com/blog/sip-trunking/ |access-date=18 July 2023 |archive-date=5 August 2023 |archive-url=https://web.archive.org/web/20230805170916/https://www.moontechnolabs.com/blog/sip-trunking/ |url-status=live }} 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.{{cite web |url=https://www.w3.org/TR/2017/CR-webrtc-20171102/ |title=WebRTC 1.0: Real-time Communication Between Browsers (W3C Candidate Recommendation 02 November 2017) |date=2 November 2017 |access-date=25 March 2019 |archive-date=2 November 2017 |archive-url=https://web.archive.org/web/20171102204947/https://www.w3.org/TR/2017/CR-webrtc-20171102/ |url-status=live }}

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

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.{{Cite web |title=RFC 8831 - WebRTC Data Channels |url=https://datatracker.ietf.org/doc/html/rfc8831 |access-date=2022-03-10 |website=datatracker.ietf.org |date=January 2021 |language=en |archive-date=2022-03-10 |archive-url=https://web.archive.org/web/20220310133453/https://datatracker.ietf.org/doc/html/rfc8831 |url-status=live |last1=Jesup |first1=Randell |last2=Loreto |first2=Salvatore |last3=Tüxen |first3=Michael }} 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.{{cite web |title=WebRTC Server: What is it exactly? |url=https://bloggeek.me/webrtc-server/ |date=13 April 2020 |author=Tsahi Levent-Levi |work=BlogGeek.me |access-date=10 June 2020 |archive-date=11 May 2020 |archive-url=https://web.archive.org/web/20200511114630/https://bloggeek.me/webrtc-server/ |url-status=live }}{{cite web |title=Matrix.org and WebRTC: An Interview with Matthew Hodgson |url=https://bloggeek.me/matrix-webrtc-interview/ |date=13 November 2014 |author=Tsahi Levent-Levi |work=BlogGeek.me |access-date=10 June 2020 |archive-date=25 February 2021 |archive-url=https://web.archive.org/web/20210225150612/https://bloggeek.me/matrix-webrtc-interview/ |url-status=live }}

{{IETF RFC|7478}} requires implementations to provide PCMA/PCMU ({{IETF RFC|3551}}), Telephone Event as DTMF ({{IETF RFC|4733}}), and Opus ({{IETF 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.

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.{{Cite web |title=WebTorrent FAQ |url=https://webtorrent.io/faq |access-date=2022-03-10 |website=webtorrent.io |language=en |archive-date=2022-03-11 |archive-url=https://web.archive.org/web/20220311144830/https://webtorrent.io/faq |url-status=live }} 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.{{Cite web |date=2021-08-04 |title=How to Transfer Files Between Linux, Android, and iOS Using Snapdrop |url=https://www.makeuseof.com/transfer-files-between-linux-android-ios-snapdrop/ |access-date=2022-03-10 |website=MUO |language=en-US |archive-date=2022-01-29 |archive-url=https://web.archive.org/web/20220129164205/https://www.makeuseof.com/transfer-files-between-linux-android-ios-snapdrop/ |url-status=live }}{{Cite web |last=Pinola |first=Melanie |date=2014-04-07 |title=The easiest and quickest way to transfer files between devices on the same network |url=https://www.computerworld.com/article/2697955/the-easiest-and-quickest-way-to-transfer-files-between-devices-on-the-same-network.html |access-date=2022-03-10 |website=Computerworld |language=en |archive-date=2022-06-28 |archive-url=https://web.archive.org/web/20220628235829/https://www.computerworld.com/article/2697955/the-easiest-and-quickest-way-to-transfer-files-between-devices-on-the-same-network.html |url-status=live }}{{Cite web |date=2015-05-12 |title=FilePizza: share files without the middleman in your browser - gHacks Tech News |url=https://www.ghacks.net/2015/05/12/filepizza-share-files-without-the-middleman-in-your-browser/ |access-date=2022-03-10 |website=gHacks Technology News |language=en-US |archive-date=2022-01-23 |archive-url=https://web.archive.org/web/20220123091008/https://www.ghacks.net/2015/05/12/filepizza-share-files-without-the-middleman-in-your-browser/ |url-status=live }} 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.{{Cite web |last=Foley |first=Mary Jo |title=Microsoft acquires Peer5 to supplement Teams' live video streaming |url=https://www.zdnet.com/article/microsoft-acquires-peer5-to-supplement-teams-live-video-streaming/ |access-date=2022-03-10 |website=ZDNet |language=en |archive-date=2022-03-10 |archive-url=https://web.archive.org/web/20220310140207/https://www.zdnet.com/article/microsoft-acquires-peer5-to-supplement-teams-live-video-streaming/ |url-status=live }}{{Cite web |title=Overview - Peer5 P2P Docs |url=https://docs.peer5.com/overview/ |access-date=2022-03-10 |website=docs.peer5.com |archive-date=2022-03-16 |archive-url=https://web.archive.org/web/20220316194439/https://docs.peer5.com/overview/ |url-status=live }}

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.{{Cite web| url=https://www.siliconrepublic.com/comms/babelfish-softphone-cloud-goldfish| title=Catch the Babelfish: Irish telco devises a new kind of cloud phone| date=November 2017| access-date=2017-11-20| archive-date=2017-11-01| archive-url=https://web.archive.org/web/20171101172651/https://www.siliconrepublic.com/comms/babelfish-softphone-cloud-goldfish| url-status=live}}

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.{{Cite web|url=https://gstreamer.freedesktop.org/releases/1.14/|title=GStreamer 1.14 release notes|website=gstreamer.freedesktop.org|access-date=2019-12-19|archive-date=2018-03-20|archive-url=https://web.archive.org/web/20180320084417/https://gstreamer.freedesktop.org/releases/1.14/|url-status=live}} since version 1.15
• OvenMediaEngine
• Ant Media Server{{Cite web |date=2017-06-07 |title=Scalable Streaming Solutions with Ant Media Server |url=https://antmedia.io |access-date=2024-12-11 |website=antmedia.io |language=en-US}}

WebRTC establishes a standard set of codecs which all compliant browsers are required to implement. Some browsers may also support other codecs.{{Cite web|title=Codecs used by WebRTC - Web media technologies {{!}} MDN|url=https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs|access-date=2021-07-29|website=developer.mozilla.org|language=en-US|archive-date=2021-07-27|archive-url=https://web.archive.org/web/20210727121111/https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs|url-status=live}}

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.{{Cite web |title=WebRTC leaks real IP addresses (even with VPN) |last=Timmerman |first=Crystal |website=IPVanish |date=28 February 2022 |url=https://www.ipvanish.com/blog/webrtc/ |access-date=12 August 2022 |archive-date=13 August 2022 |archive-url=https://web.archive.org/web/20220813032544/https://www.ipvanish.com/blog/webrtc/ |url-status=live }} 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{{cite web |title=Prevent WebRTC from leaking local IP address |author=Raymond Hill |work=uBlock Origin documentation. |url=https://github.com/gorhill/uBlock/wiki/Prevent-WebRTC-from-leaking-local-IP-address |date=17 Sep 2021 |access-date=18 Dec 2021 |archive-date=21 February 2016 |archive-url=https://web.archive.org/web/20160221222622/https://github.com/gorhill/uBlock/wiki/Prevent-WebRTC-from-leaking-local-IP-address |url-status=live }}).

• CU-RTC-WEB
• Real-time Transport Protocol (also known as RTP) is used internally by WebRTC (specifically, it uses SRTP) {{cite web |title=Introduction to the Real-time Transport Protocol (RTP) - Web APIs {{!}} MDN |url=https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Intro_to_RTP |website=developer.mozilla.org |date=26 July 2024}}

{{Reflist|30em|refs=

[http://blogs.trilogy-lte.com/post/77427158750/how-webrtc-is-revolutionizing-telephony How WebRTC Is Revolutionizing Telephony] {{Webarchive|url=https://web.archive.org/web/20140407071345/http://blogs.trilogy-lte.com/post/77427158750/how-webrtc-is-revolutionizing-telephony |date=2014-04-07 }}. Blogs.trilogy-lte.com (2014-02-21). Retrieved on 2014-04-11.

{{cite web |url=http://lists.w3.org/Archives/Public/public-webrtc/2011May/0022.html |title=Google release of WebRTC source code |author=Harald Alvestrand |date=2011-05-31 |publisher=public-webrtc@w3.org |access-date=2012-09-12 |archive-date=2013-02-27 |archive-url=https://web.archive.org/web/20130227012310/http://lists.w3.org/Archives/Public/public-webrtc/2011May/0022.html |url-status=live }}

{{Cite web |url=http://datatracker.ietf.org/wg/rtcweb/charter/ |title=Charter of the Real-Time Communication in WEB-browsers (rtcweb) working group |access-date=2013-11-11 |archive-date=2013-11-11 |archive-url=https://web.archive.org/web/20131111040415/http://datatracker.ietf.org/wg/rtcweb/charter/ |url-status=live }}

{{cite web |url=http://www.w3.org/TR/webrtc/ |title=WebRTC 1.0: Real-time Communication Between Browsers |publisher=W3.org |access-date=2012-09-12 |archive-date=2012-09-05 |archive-url=https://web.archive.org/web/20120905093824/http://www.w3.org/TR/webrtc/ |url-status=live }}

{{cite web |url=http://www.whatwg.org/specs/web-apps/current-work/multipage/introduction.html#history-1 |title=Introduction — HTML Standard |publisher=Whatwg.org |access-date=2012-09-12 |archive-date=2014-08-16 |archive-url=https://web.archive.org/web/20140816032701/http://www.whatwg.org/specs/web-apps/current-work/multipage/introduction.html#history-1 |url-status=live }}

{{cite web|url=http://www.ericsson.com/research-blog/context-aware-communication/beyond-html5-peer-peer-conversational-video/ |title=Beyond HTML5: Peer-to-Peer Conversational Video |publisher=Labs.ericsson.com|date=25 January 2011|access-date=21 March 2021|website = Ericsson Research blog| archive-url=https://web.archive.org/web/20170225045214/http://www.ericsson.com/research-blog/context-aware-communication/beyond-html5-peer-peer-conversational-video/ |archive-date=25 February 2017}}

{{cite web |url=http://tools.ietf.org/wg/rtcweb/ |title=Rtcweb Status Pages |publisher=Tools.ietf.org |access-date=2012-09-12 |archive-date=2012-09-05 |archive-url=https://web.archive.org/web/20120905214006/http://tools.ietf.org/wg/rtcweb/ |url-status=live }}

{{cite web |url=http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-00 |title=draft-jesup-rtcweb-data-protocol-00 - WebRTC Data Channel Protocol |date=5 March 2012 |publisher=Tools.ietf.org |access-date=2012-09-12 |archive-date=2012-10-31 |archive-url=https://web.archive.org/web/20121031070900/http://tools.ietf.org/html/draft-jesup-rtcweb-data-protocol-00 |url-status=live |last1=Jesup |first1=Randell |last2=Loreto |first2=Salvatore |last3=Tüxen |first3=Michael }}

{{cite web |url=https://www.w3.org/TR/mediacapture-streams/#dom-mediadevices-getusermedia |title=Media Capture and Streams: getUserMedia |publisher=W3C |date=2013-09-03 |access-date=2014-01-15 |archive-date=2014-01-02 |archive-url=https://web.archive.org/web/20140102135721/http://www.w3.org/TR/mediacapture-streams/#dom-mediadevices-getusermedia |url-status=live }}

{{cite web |url=http://www.w3.org/TR/webrtc/#rtcpeerconnection-interface |title=WebRTC: RTCPeerConnection Interface |publisher=W3C |date=2013-09-10 |access-date=2014-01-15 |archive-date=2012-09-05 |archive-url=https://web.archive.org/web/20120905093824/http://www.w3.org/TR/webrtc/#rtcpeerconnection-interface |url-status=live }}

{{cite web |url=http://www.w3.org/TR/webrtc/#rtcdatachannel |title=WebRTC: RTCDataChannel |publisher=W3C |date=2013-09-10 |access-date=2014-01-15 |archive-date=2012-09-05 |archive-url=https://web.archive.org/web/20120905093824/http://www.w3.org/TR/webrtc/#rtcdatachannel |url-status=live }}

{{cite web |url=https://w3c.github.io/webrtc-stats/webrtc-stats.html |title=Identifiers for WebRTC's Statistics API |publisher=W3C |date=2014-09-29 |access-date=2017-12-02 |archive-date=2017-07-15 |archive-url=https://web.archive.org/web/20170715052823/https://w3c.github.io/webrtc-stats/webrtc-stats.html |url-status=live }}

{{cite web|url=http://www.w3.org/community/ortc/|title=W3C ORTC (Object Real-time Communications) Community Group|date=19 August 2014 |access-date=2014-10-29|archive-date=2014-10-29|archive-url=https://web.archive.org/web/20141029141846/http://www.w3.org/community/ortc/|url-status=live}}

{{cite web | url=https://blogs.windows.com/msedgedev/2015/09/18/ortc-api-is-now-available-in-microsoft-edge/ | title=ORTC API is now available in Microsoft Edge | publisher=Microsoft | date=2015-09-18 | access-date=2015-09-20 | archive-date=2015-10-09 | archive-url=https://web.archive.org/web/20151009090318/http://blogs.windows.com/msedgedev/2015/09/18/ortc-api-is-now-available-in-microsoft-edge/ | url-status=live }}

[https://www.mozilla.org/en-US/firefox/22.0/releasenotes/ Firefox Notes - Desktop] {{Webarchive|url=https://web.archive.org/web/20140821215925/https://www.mozilla.org/en-US/firefox/22.0/releasenotes/ |date=2014-08-21 }}. Mozilla.org (2013-06-25). Retrieved on 2014-04-11.

[http://blogs.opera.com/news/2013/11/opera-18/ Opera News] {{Webarchive|url=https://web.archive.org/web/20150907003321/http://blogs.opera.com/news/2013/11/opera-18/ |date=2015-09-07 }}. blogs.opera.com (2013-11-19). Retrieved on 2015-09-17.

[https://www.mozilla.org/en-US/mobile/24.0/releasenotes/ Firefox Notes - Desktop] {{Webarchive|url=https://web.archive.org/web/20210401044319/https://www.mozilla.org/en-US/mobile/24.0/releasenotes/ |date=2021-04-01 }}. Mozilla.org (2013-09-17). Retrieved on 2014-08-04.

{{cite web | url=https://developer.apple.com/library/content/releasenotes/General/WhatsNewInSafari/Safari_11_0/Safari_11_0.html | title=Safari 11.0 | publisher=Apple Inc. | access-date=6 June 2017 | archive-date=14 November 2017 | archive-url=https://web.archive.org/web/20171114193322/https://developer.apple.com/library/content/releasenotes/General/WhatsNewInSafari/Safari_11_0/Safari_11_0.html | url-status=live }}

[https://torrentfreak.com/huge-security-flaw-leaks-vpn-users-real-ip-addresses-150130/ Huge Security Flaw Leaks VPN Users’ Real IP-addresses] {{Webarchive|url=https://web.archive.org/web/20210108120117/https://torrentfreak.com/huge-security-flaw-leaks-vpn-users-real-ip-addresses-150130/ |date=2021-01-08 }} TorrentFreak.com (2015-01-30). Retrieved on 2015-02-21.

[https://github.com/diafygi/webrtc-ips STUN IP Address requests for WebRTC] {{Webarchive|url=https://web.archive.org/web/20150218134636/https://github.com/diafygi/webrtc-ips |date=2015-02-18 }} Retrieved on 2015-02-21.

}}

• {{cite IETF |title=Additional WebRTC Audio Codecs for Interoperability |rfc=7875 |editor1-last=Proust |editor1-first=S. |date=May 2016 |publisher=IETF |access-date=2016-10-12}}
• {{cite IETF |title=WebRTC Audio Codec and Processing Requirements |rfc=7874 |last1=Valin |first1=J. M. |last2=Bran |first2=C. |date=May 2016 |publisher=IETF |access-date=2016-10-12}}
• {{cite IETF |title=WebRTC Video Processing and Codec Requirements |rfc=7742 |last1=Roach |first1=A. B. |date=March 2016 |publisher=IETF |access-date=2016-10-12}}
• {{cite IETF |title=Session Traversal Utilities for NAT (STUN) Usage for Consent Freshness |rfc=7675 |last1=Perumal |first1=M. |last2=Wing |first2=D. |last3=Ravindranath |first3=R. |last4=Reddy |first4=T. |last5=Thomson |first5=M. |date=October 2015 |publisher=IETF |access-date=2016-10-12}}
• {{cite IETF |title=Web Real-Time Communication Use Cases and Requirements |rfc=7478 |last1=Holmberg |first1=C. |last2=Hakansson |first2=S. |last3=Eriksson |first3=G. |date=March 2015 |publisher=IETF |access-date=2016-10-12}}

• {{official website}}
• [https://www.w3.org/2011/04/webrtc/ W3C Web Real-Time Communications Working Group]
• [https://tools.ietf.org/wg/rtcweb/ IETF Real-Time Communication in WEB-browsers (rtcweb) Working Group]
• [https://github.com/webrtc/apprtc Video chat demo app based on WebRTC]
• [https://libdatachannel.org libdatachannel, open-source WebRTC network library]

{{Web browsers|fsp}}

{{Web interfaces}}

Category:Peer-to-peer

Category:Software using the BSD license

Category:Web development

Category:Web standards