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Real-time communication for the web

With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. It supports video, voice, and generic data to be sent between peers, 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. The technologies behind WebRTC are implemented as an open web standard and available as regular JavaScript APIs in all major browsers. For native clients, like Android and iOS applications, a library is available that provides the same functionality. The WebRTC project is open-source and supported by Apple, Google, Microsoft and Mozilla, amongst others. This page is maintained by the Google WebRTC team.

What can WebRTC do?

There are many different use-cases for WebRTC, from basic web apps that uses the camera or microphone, to more advanced video-calling applications and screen sharing. We have gathered a number of code samples to better illustrate how the technology works and what you can use it for.

Application flow

A WebRTC application will usually go through a common application flow. Accessing the media devices, opening peer connections, discovering peers, and start streaming. We recommend that new developers read through our introduction to WebRTC before they start developing.

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Next steps

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Start with our codelab to become familiar with the WebRTC APIs for the web (JavaScript).