WebRTC is an open source project to enable realtime communication of audio, video and data in Web and native apps.
WebRTC是“網路即時通信”(Web Real Time Communication)的縮寫,它主要用來讓瀏覽器即時獲取和交換視頻、音訊和資料。
WebRTC共分三個API。
·
MediaStream(getUserMedia)
·
RTCPeerConnection
·
RTCDataChannel
getUserMedia主要用於獲取視頻和音訊資訊,後兩個API用於瀏覽器之間的資料交換。
https://webrtc.github.io/samples/src/content/getusermedia/gum/
Demo:
Display the video stream from getUserMedia() in a video element.
The MediaStream object stream passed to the getUserMedia() callback is in global scope, so you can inspect it from the console.
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