|Original author(s)||Justin Uberti|
1.0 / May 4, 2018
WebRTC (Web Real-Time Communication) is a free, open-source project that provides web browsers and mobile applications with real-time communication (RTC) via simple application programming interfaces (APIs). It allows audio and video communication 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 is being standardized through the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF).
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 May 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.
getUserMediaacquires the audio and video media (e.g., by accessing a device's camera and microphone).
RTCPeerConnectionenables audio and video communication between peers. It performs signal processing, codec handling, peer-to-peer communication, security, and bandwidth management.
RTCDataChannelallows bidirectional communication of arbitrary data between peers. It uses the same API as WebSockets and has very low latency.
The WebRTC API also includes a statistics function:
getStatsallows 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 somehow manage sessions, possibly relying on any of Session Initiation Protocol, Extensible Messaging and Presence Protocol, Message Queuing Telemetry Transport, Matrix (protocol), or another protocol. Signaling may depend on one or more servers.
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.
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.
WebRTC is supported by the following browsers:
- Desktop PC
- Google Chrome 28+ (enabled by default since 29)
- Mozilla Firefox 24+
- Opera Mobile 12+
- Chrome OS
- Firefox OS
- BlackBerry 10
- MobileSafari/WebKit (iOS 11+)
- Tizen 3.0
In January 2015, TorrentFreak reported a serious security flaw in browsers that support WebRTC, saying that it compromised the security of VPN tunnels by exposing the true IP address of a user. 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). As of September 2019, this WebRTC flaw still surfaces on Firefox 69.x and still by default exposes the user's internal IP address to the web.
- Global IP Solutions (GIPS)
- Real-time Transport Protocol (RTP)
- Session Description Protocol (SDP)
- WebRTC Gateway
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- "W3C ORTC (Object Real-time Communications) Community Group".
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- "GStreamer 1.14 release notes". gstreamer.freedesktop.org. Retrieved 2019-12-19. since version 1.14
- Huge Security Flaw Leaks VPN Users’ Real IP-addresses TorrentFreak.com (2015-01-30). Retrieved on 2015-02-21.
- STUN IP Address requests for WebRTC Retrieved on 2015-02-21.
- Raymond Hill (26 March 2016). "Prevent WebRTC from leaking local IP address". uBlock Origin documentation. Retrieved 1 Sep 2016.
- 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.