What Is WebRTC?
Read this article to find out more about web real-time communications, an exciting, powerful, and highly disruptive cutting-edge technology and standard.
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WebRTC stands for web real-time communications. It is a very exciting, powerful, and highly disruptive cutting-edge technology and standard. WebRTC leverages a set of plugin-free APIs that can be used in both desktop and mobile browsers and is progressively becoming supported by all major modern browser vendors. Previously, external plugins were required in order to achieve similar functionality as is offered by WebRTC.
WebRTC leverages multiple standards and protocols, most of which will be discussed in this article. These include data streams, STUN/TURN servers, signaling, JSEP, ICE, SIP, SDP, NAT, UDP/TCP, network sockets, and more.
History of WebRTC
One of the most important challenges for the web is to enable human communication through voice and video: real-time communication or RTC for short. Especially after Covid-19, we all need to communicate with each other regardless of distances and location. So, RTC is our new way to communicate.
Once upon a time, RTC was complex, requiring expensive audio and video technologies to be licensed or developed in-house. Integrating RTC technology with existing content, data, and services has been difficult and time-consuming, particularly on the web.
Google joined the video chat market with Gmail video chat in 2008. In 2011, Google introduced Hangouts and it was originally a feature of Google+, Hangouts became a stand-alone product in 2013. Google loved RTC technologies. It acquired GIPS, a company that develops many components required for RTC, such as codecs and echo cancellation techniques. Google open-sourced the technologies developed by GIPS and engaged with relevant standards bodies at the Internet Engineering Task Force (IETF) and World Wide Web Consortium (W3C) to ensure industry consensus. In May 2011, Ericsson built the first implementation of WebRTC. Ericsson created the first implementation of WebRTC in May 2011. And, this started the future of the communication industry.
The purpose of WebRTC was so clear, building a communication method for real-time, plugin-free video, audio, and data communication.
Many web services used RTC such as Skype, Facebook, and Hangouts but to use them, downloads, native apps, or plugins were needed. Downloading, installing, or updating plugins is complex. Usually, some error occurs and the plugins don't work properly. Using plugins may require some expensive licenses or technologies. It is also really hard to make people download a plugin or anything, really.
WebRTC emerged based on these principles that its APIs should be open source, free, standardized, and built into web browsers.
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