WebRTC (Web real-time communication) is an HTML5 specification that lets developers set up real-time media communications directly between web browsers and hardware devices. WebRTC enables voice and video communication on web pages without the need for browser plugins. Before WebRTC, voice and video web traffic often originated from a web-connected server. Different software platforms provided proprietary plugins and protocols for accessing their content from their servers (remember RealAudio?). This kind of fragmentation went against the notion of a globally accessible Web. WebRTC was proposed as a World Wide Web Consortium (W3C) standard in 2011. The WebRTC spec comprises a JavaScript API that developers can use inside the browser and a suite of communications protocols. Over the next few years web browser developers began incorporating support for the technology into their software; since 2017 it has been supported in all major browsers — Google Chrome, Mozilla Firefox, Apple Safari, and Microsoft Edge. The W3C finalized the standard in 2021. WebRTC allows access to device hardware — typically microphones and cameras — in real time on the machines that run it. It can also capture the screen displays and share or record them. But WebRTC isn’t limited to voice and video — it enables sending any type of data. It allows any connected device, on any network, to be a communication endpoint on the Web. Use Cases WebRTC is an open source project and has been embedded in browsers, but anyone can adopt it for their own needs. Developers can incorporate voice and video into complex use case scenarios, such as: Contact center operations — client/agent conversations, remote assistance. Watch parties for sporting events and awards shows. Trivia contests or other events or meetings that rely on audience participation. Promotional events, such as fashion shows. Online education, including tutoring. Online fitness classes. Public meetings and conferences. Telehealth and emergency response For one example, cloud communications platform Plivo has built WebRTC support into its . Plivo is a communications platform as a service (CPaaS) — it provides a voice API and that let businesses add voice calling and messaging to their existing applications. Browser SDK SMS API The Browser SDK lets developers make and receive calls and access Plivo’s cloud communications platform directly from a web browser. How to Get Started With WebRTC The best way to get familiar with WebRTC is to code a prototype application. The project’s site has sample code, but if you want to put WebRTC to work quickly, we suggest getting familiar with the Plivo Browser SDK, which wraps easy-to-call functions around WebRTC APIs to make developers more productive. Plivo makes freely available via GitHub so developers can see how to code voice applications that use WebRTC. Browser SDK sample applications

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What is WebRTC (Web Real-Time Communication)?

What is WebRTC (Web Real-Time Communication)?

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