WebRTC
Introduction
Web Real-Time Communication (WebRTC) 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 Google, Microsoft, Mozilla, and Opera, WebRTC is being standardized by the World Wide Web Consortium (W3C) and the Internet Engineering Task Force (IETF).
History
The WebRTC project was first announced by Google in May 2011. The company had acquired the technologies from GIPS, a company they had purchased earlier. In the following years, the technology was standardized by the W3C and the IETF, two major Internet standardization bodies, and it was implemented by all major browsers.
Technical Details
WebRTC uses a set of protocols and APIs to achieve real-time communication. These include:
- Session Description Protocol (SDP): This protocol is used for negotiation of media capabilities, formats, and transports.
- Interactive Connectivity Establishment (ICE): ICE is used to traverse Network Address Translation (NAT) and firewalls.
- Secure Real-time Transport Protocol (SRTP): This protocol is used for encryption and secure transport of media.
- Real-time Transport Control Protocol (RTCP): RTCP is used for QoS monitoring and reporting.
Applications
WebRTC has a wide range of applications, including but not limited to:
Security
WebRTC is designed with strong security in mind. All WebRTC components, including signaling, data transport, and media, are required to be encrypted. The APIs are designed to respect user privacy and consent for sharing devices like microphones and cameras.
Future Development
WebRTC continues to be actively developed and standardized. Future improvements include better support for mobile devices, improved network resilience, and more efficient codecs.