In many modern web applications, WebSockets are used to implement realtime, live-updating user interfaces. When some data is updated on the server, a message is typically sent over a WebSocket connection to be handled by the client. This provides a more robust, efficient alternative to continually polling your application for changes.
WebRTC is an open source project to enable realtime communication of audio, video and data in Web and native apps.
getUserMedia(): capture audio and video.
MediaRecorder: record audio and video.
RTCPeerConnection: stream audio and video between users.
RTCDataChannel: stream data between users.
In Firefox, Opera and in Chrome on desktop and Android. WebRTC is also available for native apps on iOS and Android.
WebRTC uses RTCPeerConnection to communicate streaming data between browsers, but also needs a mechanism to coordinate communication and to send control messages, a process known as signaling. Signaling methods and protocols are not specified by WebRTC. In this codelab you will use Socket.IO for messaging, but there are many alternatives.
Build an app to get video and take snapshots with your webcam and share them peer-to-peer via WebRTC. Along the way you'll learn how to use the core WebRTC APIs and set up a messaging server using Laravel Echo.
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