Webrtc Leak Test Full Media Package #825
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It supports video, voice, and generic data to be sent between peers,. When writing automated tests for your webrtc applications, there are useful configurations that can be enabled for browsers that make development and testing easier. Creating a new application based on the webrtc technologies can be overwhelming if you're unfamiliar with the apis
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In this section we will show how to get started. Neste codelab, você aprenderá a criar um aplicativo simples de chat por vídeo usando a api webrtc no navegador e o cloud firestore para sinalização When developing for the web, the webrtc standard provides apis for accessing cameras and microphones connected to the computer or smartphone
RTCPeerConnection 连接到远程对等方后，便可以在它们之间流式传输音频和视频。此时，我们将从 getUserMedia() 收到的数据流连接到 RTCPeerConnection。媒体串流至少包含一个媒体.
WebRTC 标准还涵盖通过 RTCPeerConnection。 可通过对 createDataChannel() RTCPeerConnection 对象，该对象会返回 RTCDataChannel 对象。 Once a rtcpeerconnection is connected to a remote peer, it is possible to stream audio and video between them This is the point where we connect the stream we receive from.
