Integrating RTC technology with existing content, data, and services has been difficult and time-consuming, particularly on the web. Historically, RTC has been corporate and complex, requiring expensive audio and video technologies to be licensed or developed in house. Without it, you're limited in your ability to innovate and develop new ways for people to interact. RTC should be as natural in a web app as entering text in a text input. One of the last major challenges for the web is to enable human communication through voice and video: real-time communication or RTC for short.
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Having trouble with your machine and WebRTC? Visit the WebRTC Troubleshooter.Īlternatively, jump straight into the WebRTC codelab, a step-by-step guide that explains how to build a complete video chat app, including a simple signaling server.Can’t wait and just want to try WebRTC right now? Try some of the more-than 20 demos that exercise the WebRTC JavaScript APIs.
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To learn how WebRTC uses servers for signaling, and firewall and NAT traversal, see the code and console logs from appr.tc.To learn about the RTCPeerConnection API, see the following example and RTCPeerConnection.If you haven't used the getUserMedia API, see Capture audio and video in HTML5.To get an overview of WebRTC, watch the following Google I/O video or view these slides: Haven't got time to read this article or only want code? Open the URL displayed at the end of the page in a new tab or, better still, on a different computer.
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