Karl Jones: /* External links */

2016-04-25T01:12:18Z

‎External links

Karl Jones at 13:33, 6 February 2016

2016-02-06T13:33:11Z

Karl Jones: /* See also */

2015-09-14T15:31:47Z

‎See also

Karl Jones: Created page with "'''WebRTC''' ('''Web Real-Time Communication''') is an API definition drafted by the World Wide Web Consortium (W3C) that supports ..."

2015-09-14T15:31:14Z

Created page with "'''WebRTC''' ('''Web Real-Time Communication''') is an API definition drafted by the World Wide Web Consortium (W3C) that supports ..."

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'''WebRTC''' ('''Web Real-Time Communication''') is an [[Application programming interface|API]] definition drafted by the [[World Wide Web Consortium]] (W3C) that supports [[Web browser|browser]]-to-browser applications for [[voice calling]], [[video chat]], and [[P2P file sharing]] without the need of either internal or external [[Plug-in (computing)|plugins]].

(TO DO: expand, organize, cross-reference, illustrate.)

== History ==

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 standardise the relevant protocols in the [[Internet Engineering Task Force|IETF]] and browser APIs in the W3C.

The W3C draft of WebRTC 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]].

It was referred to as the ConnectionPeer API, and a pre-standards concept implementation was created at [[Ericsson|Ericsson Labs]].

The Web Real-Time Communications Working Group expects this specification to evolve significantly based on:

* Outcomes of ongoing exchanges in the companion RTCWEB group at [[IETF]]<ref>{{cite web|url=http://tools.ietf.org/wg/rtcweb/ |title=Rtcweb Status Pages |publisher=Tools.ietf.org |date= |accessdate=2012-09-12}}</ref> to define the set of [[Communications protocol|protocols]] that, together with this document, define [[real-time communication]]s in Web browsers. While no one signalling protocol is mandated, [[Session Initiation Protocol|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<ref>{{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 |publisher=Tools.ietf.org |date= |accessdate=2012-09-12}}</ref>
* Experience gained through early experimentation
* Feedback from other groups and individuals

== Design ==

Major components of WebRTC include:

* <code>getUserMedia</code>, which allows a web browser to access the camera and microphone and to capture media<ref>{{cite web|url=http://www.w3.org/TR/mediacapture-streams/#dom-navigator-getusermedia |title=Media Capture and Streams: getUserMedia |publisher=W3C |date=2013-09-03 |accessdate=2014-01-15}}</ref>
* <code>RTCPeerConnection</code>, which sets up audio/video calls<ref>{{cite web|url=http://www.w3.org/TR/webrtc/#rtcpeerconnection-interface |title=WebRTC: RTCPeerConnection Interface |publisher=W3C |date=2013-09-10 |accessdate=2014-01-15}}</ref>
* <code>RTCDataChannel</code>, which allow browsers to share data via peer-to-peer<ref>{{cite web|url=http://www.w3.org/TR/webrtc/#rtcdatachannel |title=WebRTC: RTCDataChannel |publisher=W3C |date=2013-09-10 |accessdate=2014-01-15}}</ref>

The WebRTC API also includes a statistics function:
* <code>getStats</code>, which allows the web application to retrieve a set of statistics about WebRTC sessions. These statistics data are being described in a separate W3C document.

As of March 2015, the IETF WebRTC Codec and Media Processing Requirements draft requires implementations to provide [[PCMA]]/[[PCMU]] (RFC 3551), Telephone Event as [[DTMF]] (RFC 4733), and [[Opus (audio format)|Opus]] (RFC 6716) [[Audio codec| audio codecs]], along with the [[H.264]] and [[VP8]] [[video codec| video codecs]] as minimum capabilities.

The Peerconnection, Data channels and a media capture browser APIs are detailed in the W3C.

W3C is developing ORTC (Object Real-time Communications) for WebRTC. This is commonly referred to as WebRTC 1.1.

== Benefits ==

WebRTC enables all kinds of real time communication such as audio, video and text between users by utilising the browsers. Using WebRTC bears different benefits for different market segments. For end users it has two major advantages:

* Ease of use: Real-time communication is supported without the need for additional applications or plug-ins.
* Security: WebRTC enforces the usage of encryption for both the media and the signalling. Thereby, WebRTC provides a higher security level than most currently available commercial telephony systems.

For enterprises WebRTC can provide even more benefits:

* Cost savings:
** Save on the costs of [[toll-free telephone number]] for call centres

* Rich communication: Enhance the communication to users and between employers with video and messaging without the need for special applications and servers.
* Un-interrupted communication: Keep the customers on the web page and at the same time start a voice and video call with customer.
* Security: Secure the communication with the customers as well as employees in the home office and remote branches using state of the art encryption standards.

For operators, WebRTC can additionally provide new opportunities:

* Mobile Telephony: By relying on WebRTC technology, service providers can enable users to access their [[VoIP]] service while on the go without specialised applications.
* Hosted Services: By deploying a WebRTC gateway end users would be able to access the [[Session Initiation Protocol|SIP]] based hosted [[Business telephone system#Private branch exchange|PBX]] and call centers without the need to change these services.
* WebRTC as a Service: Similar to hosted PBX services, service providers can host [[WebRTC Gateway]]s on behalf of enterprises. WebRTC calls destined to the enterprise are handled by a [[WebRTC Gateway]] of the service provider. Incoming WebRTC calls would be translated into SIP calls and routed to the enterprise. The enterprise would not have to change anything in its infrastructure, as it will still be only handling SIP calls

== Support ==
WebRTC is supported in the following browsers.
* Desktop PC
** [[Google Chrome]] 23
** [[Mozilla Firefox]] 22<ref>[https://www.mozilla.org/en-US/firefox/22.0/releasenotes/ Firefox Notes - Desktop]. Mozilla.org (2013-06-25). Retrieved on 2014-04-11.</ref>
** [[Opera (web browser)|Opera]] 18<ref>[http://web.archive.org/web/20131119173936/http://my.opera.com/ODIN/blog/opera-desktop-18-released Dev.Opera]. My.opera.com. Retrieved on 2014-04-11.</ref>
* Android
** Google Chrome 28 (Enabled by default since 29)
** Mozilla Firefox 24<ref>[https://www.mozilla.org/en-US/mobile/24.0/releasenotes/ Firefox Notes - Desktop]. Mozilla.org (2013-09-17). Retrieved on 2014-08-04.</ref>
** Opera Mobile 12
* [[Chrome OS]]
* [[Firefox OS]]
* iOS
** Bowser<ref>http://www.openwebrtc.io/bowser/</ref>

As of August 2014, [[Internet Explorer]] and [[Safari (web browser)|Safari]] still lack the native support of WebRTC.

Several plugins are available to add the support of WebRTC to these browsers.

== Concerns ==

In January 2015, [[TorrentFreak]] reported that browsers supporting WebRTC suffer from a serious security flaw that compromises the security of [[Virtual private network|VPN-tunnels]], by allowing the true [[IP address]] of the user to be read.

The IP address read requests are not visible in the browser's developer console, and they are not blocked by common [[Ad filtering|ad blocking]]/privacy plugins (enabling online tracking by advertisers and other entities despite precautions).

WebRTC can be enabled and disabled in Firefox by toggling the value of "media.peerconnection.enabled" in <code>[[about:config]]</code>, and WebRTC settings can be changed in <code>about:webrtc</code>. WebRTC cannot be disabled in the desktop version of Google Chrome, although there is a plug-in available for blocking it.<ref>[https://www.browserleaks.com/webrtc#webrtc-disable WebRTC — Local IP Address Discovery — Media Device ID Fingerprints — BrowserLeaks.com] BrowserLeaks.com. Retrieved on 2015-03-23.</ref>

== See also ==

* [[CU-RTC-WEB]]
* [[Jingle (protocol)]]
* [[Datagram Transport Layer Security]] (DTLS), [[Secure Real-time Transport Protocol]] (SRTP) - DTLS-SRTP is an essential protocol for WebRTC key management
* [[Web application]]

== External links ==

* [http://www.webrtc.org/ Official website]
* [https://en.wikipedia.org/wiki/WebRTC WebRTC] @ Wikipedia
* [http://www.w3.org/2011/04/webrtc/ W3C Web Real-Time Communications Working Group]
* [http://tools.ietf.org/wg/rtcweb/ IETF Real-Time Communication in WEB-browsers (rtcweb) Working Group]
* [http://www.html5rocks.com/en/tutorials/webrtc/basics/ Getting Started With WebRTC]
* [http://knowledge.santanu.net/what-is-webrtc-current-scenario-and-why-we-should-follow/ WebRTC Interest and History]
* [http://www.frafos.com/wp-content/uploads/2014/11/FRAFOS_WebRTC_Deployment.pdf WebRTC:How an Why?]
* [http://diafygi.github.io/webrtc-ips/ WebRTC: IP Address Exposure Demo]

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	* [http://www.frafos.com/wp-content/uploads/2014/11/FRAFOS_WebRTC_Deployment.pdf WebRTC:How an Why?]		* [http://www.frafos.com/wp-content/uploads/2014/11/FRAFOS_WebRTC_Deployment.pdf WebRTC:How an Why?]
	* [http://diafygi.github.io/webrtc-ips/ WebRTC: IP Address Exposure Demo]		* [http://diafygi.github.io/webrtc-ips/ WebRTC: IP Address Exposure Demo]
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 '''WebRTC''' ('''Web Real-Time Communication''') is an [[Application programming interface|API]] definition drafted by the [[World Wide Web Consortium]] (W3C) that supports [[Web browser|browser]]-to-browser applications for [[voice calling]], [[video chat]], and [[P2P file sharing]] without the need of either internal or external [[Plug-in (computing)|plugins]].
 == History ==
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 * [[Jingle (protocol)]]
 * [[Datagram Transport Layer Security]] (DTLS), [[Secure Real-time Transport Protocol]] (SRTP) - DTLS-SRTP is an essential protocol for WebRTC key management
 * [[Web application]]

WebRTC - Revision history

Karl Jones: /* External links */

Karl Jones at 13:33, 6 February 2016

Karl Jones: /* See also */

Karl Jones: Created page with "'''WebRTC''' ('''Web Real-Time Communication''') is an API definition drafted by the World Wide Web Consortium (W3C) that supports ..."