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The communication between peers can be video, audio or arbitrary binary data (for clients supporting the rtcdatachannel api). For most webrtc applications to function a server is required for relaying the traffic between peers, since a direct socket is often not possible between the clients (unless they reside on the same local network). 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 with the various apis in the webrtc standard, by explaining a number of common use cases and code snippets for solving those. The application is called firebasertc and works as a simple example that will teach you the basics of building webrtc enabled applications. When developing for the web, the webrtc standard provides apis for accessing cameras and microphones connected to the computer or smartphone
These devices are commonly referred to as media devices and can be accessed with javascript through the navigator.mediadevices object, which implements the mediadevices interface.
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轨道，当我们想要将媒体传输到远程对等方时，会将这些轨道单独添加到 RTCPeerConnection。 const localStream = await getUserMedia({video: true. 在进行 Web 开发时，WebRTC 标准提供了一些 API，用于访问 摄像头和麦克风已连接到计算机或智能手机。 这些设备 通常称为媒体设备，可通过 JavaScript 进行访问 通过 navigator.mediaDevices 对象实现，该对象会实现 MediaDevices 界面。 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 getusermedia() to the rtcpeerconnection
A media stream consists of at least one media track, and these are individually added to the rtcpeerconnection when we want to transmit the media to the remote peer. In this codelab, you'll learn how to build a simple video chat application using the webrtc api in your browser and cloud firestore for signaling
