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The RTC in WebRTC stands for Real-Time Communications, technology that enables audio/video streaming and data sharing between browser clients (peers). As a set of standards, WebRTC provides any browser with the ability to share application data and perform teleconferencing peer to peer, without the need to install plug-ins or third-party software. 

WebRTC components are accessed with JavaScript APIs. Currently in development are the Network Stream API, which represents an audio or video data stream, and the PeerConnection API, which allows two or more users to communicate browser-to-browser. Also under development is a DataChannel API that enables communication of other types of data for real-time gaming, text chat, file transfer, and so forth.

Note: Much of this content is out of date. We will be updating it soon.

Want to discover WebRTC? Watch this introductive web-enhanced video!


Documentation about WebRTC

Introduction to WebRTC
An introductory guide to what WebRTC is and how it works.
Using the Network Stream API
A guide to using the Network Stream API to stream audio and video.
Peer-to-peer communications with WebRTC
How to perform peer-to-peer communications using the WebRTC APIs.
Capturing images from a webcam
An introductory guide to what WebRTC is and how it works.
MediaStream API
The API supporting the generation and manipulation of media stream objects.
The navigator function that provides access to the system's media devices.

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Getting help from the community

When developing web sites and web applications that will take advantage of WebRTC technologies, it may be useful to engage in conversation with others doing likewise.

  • Ask your question on Mozilla's Media IRC channel: #media

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Etiquetas do documento e colaboradores

 Colaboradores desta página: Sheppy
 Última atualização por: Sheppy,