Right now, whether it's for business or pleasure, almost everybody is using real-time communication to connect with coworkers, family members, and friends.
Web Real-Time Communication (WebRTC) is a free and open-source technology which enables web browsers and mobile applications to use real-time communication (RTC) in a simple and easy-to-implement way. Working inside of web pages, WebRTC facilitates peer-to-peer connections which can exchange both audio and video data between users.
WebRTC allows you to add real-time communication capabilities to your application and works on top of an open standard. It supports video, voice, and generic data transmission between users, allowing developers to build powerful voice- and video-communication solutions. The technology is available on all modern browsers as well as on native clients for all major platforms.
In addition to peer-to-peer video and audio connections, WebRTC can transmit video streams, text data, files, and instant messaging. This flexibility has allowed WebRTC to be integrated into countless websites and applications, adding advanced features which would otherwise require significant work for developers to implement.
The inclusion of WebRTC as a standard technology is one of the most important factors in ensuring this technology's versatility and ease of use. Since the technology is included by default in most web browsers, any worry about compatibility has been removed, making the implementation of real-time communication features much more accessible for developers and users alike. It is because of WebRTC that applications such as Facebook Messenger and Google Hangouts can be used to video chat from a web browser with the click of a mouse. Unlike other solutions for real-time communication, WebRTC does not require that users download any plugins or new applications, making it so easy that many people have used this technology without even realizing it.
The universal nature of WebRTC has also benefited the technology's security. As a default portion of modern web browsers, countless security reviews have been conducted and possible vulnerabilities have been identified and addressed since its introduction nearly a decade ago. This attention has ensured that WebRTC - which includes a number of security-preserving features in its default implementation - has remained secure not only theoretically but in real-world implementations as well.
StormFree is currently using a modified WebRTC deployment to implement our VoIP functionality because of its ease of use, and because its open-source nature permits it to be modified to suit particular user needs. By pairing WebRTC with a customized signaling server, the technology can be made highly secure while maintaining its simplicity and reliability. StormFree has further bolstered the built-in security of WebRTC by pairing it with a secure server that is used for data relay, improving the reliability of connections while keeping one-to-one calls secure with end-to-end encryption.
With WebRTC's inclusion as a default component in modern web browsers, we expect to see the protocol being used more and more widely in the future. We encourage developers who are seeking to implement VoIP or other communication functionality in their applications to consider using WebRTC, as there is endless potential in this simple yet effective piece of open-source tech.
