Web实时音视频(WebRTC)是一种支持网页浏览器进行实时音视频通信的技术。它允许用户通过简单的API调用,在不需要任何插件的情况下,在浏览器之间直接传输音频和视频数据。
WebRTC的核心组件包括:
以下是一个简单的WebRTC一对一通信的示例代码:
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<title>WebRTC Demo</title>
</head>
<body>
<video id="localVideo" autoplay playsinline></video>
<video id="remoteVideo" autoplay playsinline></video>
<button id="startButton">Start</button>
<button id="callButton">Call</button>
<button id="hangupButton">Hang Up</button>
<script src="app.js"></script>
</body>
</html>
const localVideo = document.getElementById('localVideo');
const remoteVideo = document.getElementById('remoteVideo');
const startButton = document.getElementById('startButton');
const callButton = document.getElementById('callButton');
const hangupButton = document.getElementById('hangupButton');
let localStream;
let remoteStream;
let peerConnection;
const servers = {
iceServers: [
{ urls: 'stun:stun.l.google.com:19302' }
]
};
startButton.onclick = async () => {
localStream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
localVideo.srcObject = localStream;
};
callButton.onclick = async () => {
peerConnection = new RTCPeerConnection(servers);
peerConnection.onicecandidate = event => {
if (event.candidate) {
// Send the candidate to the remote peer
}
};
peerConnection.ontrack = event => {
remoteVideo.srcObject = event.streams[0];
};
localStream.getTracks().forEach(track => {
peerConnection.addTrack(track, localStream);
});
const offer = await peerConnection.createOffer();
await peerConnection.setLocalDescription(offer);
// Send the offer to the remote peer
};
hangupButton.onclick = () => {
peerConnection.close();
peerConnection = null;
};
通过以上基础概念、优势、类型、应用场景以及常见问题及解决方法的介绍,希望能帮助你更好地理解和实现WebRTC实时音视频功能。