web端实时通话功能开发源代码
时间: 2023-08-02 14:10:36 浏览: 47
实现Web端实时通话功能需要前后端协作,以下是实现实时通话功能的Web前端代码示例:
```html
<!-- HTML代码 -->
<html>
<head>
<meta charset="utf-8">
<title>WebRTC 实时通话</title>
<script src="https://cdn.bootcdn.net/ajax/libs/jquery/3.6.0/jquery.min.js"></script>
</head>
<body>
<video id="localVideo" autoplay></video>
<video id="remoteVideo" autoplay></video>
<button id="startButton">开始通话</button>
<button id="hangupButton">结束通话</button>
<script>
var localStream;
var remoteStream;
var pc1;
var pc2;
var offerOptions = {
offerToReceiveAudio: 1,
offerToReceiveVideo: 1
};
var constraints = {
audio: true,
video: true
};
var startButton = document.getElementById('startButton');
var hangupButton = document.getElementById('hangupButton');
startButton.addEventListener('click', startCall);
hangupButton.addEventListener('click', hangupCall);
function startCall() {
navigator.mediaDevices.getUserMedia(constraints)
.then(function(stream) {
localStream = stream;
document.getElementById('localVideo').srcObject = localStream;
pc1 = new RTCPeerConnection();
pc1.onicecandidate = function(event) {
if (event.candidate) {
pc2.addIceCandidate(new RTCIceCandidate(event.candidate));
}
};
pc2 = new RTCPeerConnection();
pc2.onicecandidate = function(event) {
if (event.candidate) {
pc1.addIceCandidate(new RTCIceCandidate(event.candidate));
}
};
pc2.ontrack = function(event) {
remoteStream = event.streams[0];
document.getElementById('remoteVideo').srcObject = remoteStream;
};
localStream.getTracks().forEach(track => {
pc1.addTrack(track, localStream);
});
pc1.createOffer(offerOptions)
.then(function(offer) {
pc1.setLocalDescription(offer);
pc2.setRemoteDescription(offer);
pc2.createAnswer()
.then(function(answer) {
pc2.setLocalDescription(answer);
pc1.setRemoteDescription(answer);
});
});
})
.catch(function(error) {
console.error('获取媒体设备权限失败:', error);
});
}
function hangupCall() {
pc1.close();
pc2.close();
localStream.getTracks().forEach(track => {
track.stop();
});
document.getElementById('localVideo').srcObject = null;
document.getElementById('remoteVideo').srcObject = null;
}
</script>
</body>
</html>
```
这段代码实现了在Web端发起实时通话,并在本地和远程展示视频流。当点击“开始通话”按钮时,会调用`startCall`函数,获取本地媒体流、创建PeerConnection对象、进行信令交换等操作,最终建立点对点连接,完成实时通话。当点击“结束通话”按钮时,会调用`hangupCall`函数,关闭点对点连接,释放媒体设备资源。
需要注意的是,由于WebRTC协议涉及到NAT穿透和转发等技术,因此需要使用信令服务器完成客户端之间的信令交换。在实际开发过程中,需要配合后端开发完成信令服务器的搭建和调用。