WebRTC ("Web Real-Time Communication") is a collection of communications protocols and application programming interfaces that enable real-time communication over peer-to-peer connections. This allows web browsers to not only request resources from backend servers, but also real-time information from browsers of other users.
WebRTC is being standardized by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF). The reference implementation is released as free software under the terms of a BSD license. OpenWebRTC provides another free implementation based on the multimedia framework GStreamer.
WebRTC uses Real-time Transport Protocol to transfer audio and video.
WebRTC is supported in the following browsers.
As of September 2015[update], Internet Explorer still lacks the native support of WebRTC but ORTC was already added to the new Microsoft browser, Edge. Several plugins are available to add the support of WebRTC to these browsers. At WWDC 2017, Apple announced Safari would get WebRTC support in Safari 11 , and it became available in release 32 of the Safari Technology Preview. 
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 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 Labs. The Web Real-Time Communications Working Group expects this specification to evolve significantly based on:
Major components of WebRTC include:
getUserMedia, which allows a web browser to access the camera and microphone and to capture media
RTCPeerConnection, which sets up audio/video calls
RTCDataChannel, which allow browsers to share data via peer-to-peer
The WebRTC API also includes a statistics function:
getStats, 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.
RFC 7874 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.
W3C is developing ORTC (Object Real-Time Communications) for WebRTC. This is commonly referred to as WebRTC 1.1.
WebRTC technology is facilitating a move towards browser-based VoIP telephony, also called cloud phones or web phones. These applications allow calls to be made and received from within a internet search browser, replacing the requirement to download and install a softphone.
In January 2015, TorrentFreak reported that browsers supporting WebRTC suffer from a serious security flaw that compromises the security of 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 most ad blocking/privacy/security add-ons, enabling online tracking by advertisers and other entities despite precautions (however the uBlock Origin add-on can fix this problem).
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