WebRTC allows for real-time communications directly in the browser via JavaScript APIs. It aims to enable rich, high-quality real-time communications applications. The first implementation was built by Ericsson in 2011 and it has since progressed, with Firefox and Chrome achieving interoperability in 2013. The key APIs included are MediaStream for camera/mic access, PeerConnection for audio/video calls, and DataChannels for other data transfer like in gaming. Signaling is needed to exchange session information before peer-to-peer streaming can begin. WebRTC continues to be supported by browsers and has potential uses like video chat, file sharing, and real-time collaboration.