W3C Consortium announced to give API , related to technology WebRTC , status recommended standard. At the same time, the Internet Engineering Task Force (IETF) committee for the development of Internet protocols and architecture published 11 RFC (8825-8835, 8854) with description architectures , protocol elements, modes of transport and error correction mechanisms used in WebRTC. These RFCs are now “Proposed Standard” status.
WebRTC Technologies was created by GIPS, a company specializing in the development of digital signal processing technologies and real-time multimedia streaming. In 2011, GIPS was acquired by Google, which opened all WebRTC-related developments under the BSD license and provided free access to patents. Since then, WebRTC support has been implemented in all modern browsers and has become widespread in communication applications that need a direct communication channel between browsers.
For example, WebRTC is actively used in applications for organizing video and audio conferencing, games, collaboration platforms, instant messengers, streaming systems and content distribution systems.
Using WebRTC, communication applications can process voice and video traffic in real time using only HTML and JavaScript, without the use of third-party proprietary technologies and external plugins.
WebRTC consists of four basic components: a user session management system, an audio processing engine, a video processing engine, and a transport layer. Audio and video processing engines allow using different codecs (VP8, H.264), as well as noise suppression methods. All data is transmitted only in encrypted form. For real-time data transmission, DTLS and SRTP (Secure Real-time Transport Protocol) protocols can be used in combination with technologies for organizing P2P communication channels and ensuring operation through firewalls and address translators (ICE, STUN, TURN, RTP-over-TCP , the ability to work through a proxy).
In addition to the standardized base parts, the W3C and IETF working groups are also developing so far on approved extensions that allow the use of the QUIC protocol as a transport and allow the use of the AV1 video codec. A working group has also been created to develop the WebTransport API to simplify streaming to multiple recipients and the Scalable Video Coding to adapt the video stream to the client’s bandwidth. For the next version of the WebRTC API, such capabilities are being developed as end-to-end encryption of video conferencing, live processing of audio and video streams (including using machine learning systems), the ability to establish a communication channel with sensors in IoT devices.