WebRTC
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.
Here are 6,332 public repositories matching this topic...
An implementation for a multiplayer Bomberman Game via WebRTC Peer to Peer Communication using peer.js.
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Aug 12, 2013 - JavaScript
testing webrtc performance with holla abstraction
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Aug 17, 2013 - JavaScript
In browser interview app with video call and code sharing functionality
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Dec 7, 2013 - JavaScript
This is the Wrapper Library for WebRTC Voice Engine. Including Acoustic Echo Cancellation (AEC), Noise Suppression (NS), VAD (Voice Active Detection) and so on.
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Jul 3, 2014 - C++
WebRTC based video chat with a possibility to modify your voice with special effects (Web Audio Api)
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Jul 28, 2014 - JavaScript
Created by Google
Released May 4, 2018
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