nodejs搭建信令服务器
由于最近在研究webRTC视频推流，查了很多webRTC的资料，准备搭建rtsp的流媒体推送服务器，属于个人研究，非团队。进度慢之又慢！资料查了又查，始终找不到webrtc推送rtsp流的开源项目，自己后续慢慢研究吧！
虽然没开发出来什么东西，但是学习不少的东西！例如：创建nodejs搭建的聊天室，创建视频通话。生命就是在不断学习的过程！谢谢大佬的博客https://huangxiaoguo.blog.csdn.net/article/details/104229233
虽然需要修改一部分变量，但最终还是跑通了！如果对于流程及api感到比较陌生的话可以看看这篇文章webrtc开发入门教程
一下代码是修改过后的代码可以直接运行！在gitee上开源源码地址gitee码云，单独复制代码，会运行报错！
'use strict'
//引入express框架
var express = require('express');
//引入http
var http = require('http');
//引入文件管理的fs
var fs = require('fs');
//引入路径
var path = require('path');
//引入静态文件内容插件
var serveIndex = require('serve-index');
//初始化 express框架;
let app = express();
//express挂载
app.use(express.static('./public'));
//引入scoket
var { ioinit, logger, USERCOUNT } = require('./scoket');
app.use('/index', function (req, res, next) {
logger.debug('主页面请求')
console.log('主页面请求', req.url, req.url.indexOf('/favicon.ico') != -1);
if (req.url.indexOf('/index') != -1) { }
if (req.url.indexOf('/tall') != -1) {
console.log('进到消息通话页面');
res.sendFile(path.resolve(__dirname, './static/tall.html'));
} else if (req.url.indexOf('/webrtc') != -1) {
console.log('进到视频通话页面123');
res.sendFile(path.resolve(__dirname, './view/index.html'))
} else if (req.url.indexOf('/js') != -1) {
res.sendFile(path.resolve(__dirname, './static/client.js'));
} else if (req.url.indexOf('/tjs') != -1) {
res.sendFile(path.resolve(__dirname, './view/tjs/client.js'));
} else {
res.send({ code: 400, message: '请求的页面不见了', success: false })
}
});
//挂载静态文件夹下所有文件名称路径
app.use('/filepath', serveIndex('./public'));
app.use('/', (req, res, next) => {
console.log('荣阔所有路径', req.url);
res.sendFile(path.resolve(__dirname, './static/favicon.ico'));
})
let serve = http.createServer(app)
let io = ioinit(serve)
serve.listen(3100)
io.sockets.on('connection', (socket) => {
console.log('创建socket', 11);
logger.info("connection");
//转发信息
socket.on('message', (room, data) => {
console.log('message消息推送', 33,room,data);
logger.info("message data " + socket.id + " " + room, data);
socket.emit('message', room, socket.id, '回复链接')//房间内所有人,除自己外
});
//用户加入
socket.on('join', (room) => {
console.log('用户加入', room, 22);
logger.info("join", ",room = ", room, ",socket.id = ", socket.id);
var myRoom = io.sockets.adapter.rooms[room];
console.log('===========', myRoom);
var users = (myRoom) ? Object.keys(myRoom.sockets).length : 0;
logger.debug('the number of user in room is:' + users)
//处理一对一通信
if (users < USERCOUNT) {
//给本人回信息
socket.emit('joined', room, socket.id);
if (users > 1) {
socket.to(room).emit('otherjoin', room, socket.id);
}
} else {
socket.leave(room);
socket.emit('full', room, socket.id);
}
});
//用户离开
socket.on('leave', (room) => {
console.log('用户离开', 44);
var myRoom = io.sockets.adapter.rooms[room];
var users = (myRoom) ? Object.keys(myRoom.sockets).length : 0;
//users-1
logger.debug('the number of user in room is:' + (users - 1));
socket.to(room).emit('exit', room, socket.id);
socket.emit('leaved', room, socket.id);
});
});
serve.on('error', (error) => {
console.log('启动失败', error)
})
serve.on('listening', () => {
logger.debug('监听3100端口启动-启动成功')
console.log('3100端口启动成功');
})
使用了express框架，没有使用路由中间件，使用log4js进行日志记录，使用scoketIo进行数据通讯
聊天室代码
<html>
<head>
<title>
Chat Room
</title>
</head>
<body>
<table align="center">
<tr>
<td>
<label>UserName:</label>
<input type="text" id="uesrname"/>
</td>
</tr>
<tr>
<td>
<label>room:</label>
<input type="text" id="room"/>
<button id="connect">加入</button>
</td>
</tr>
<tr>
<td>
<label>Content:</label><br>
<textarea disabled id="output" rows="10" cols="50"></textarea>
</td>
</tr>
<tr>
<td>
<label>Input:</label><br>
<textarea disabled id="input" rows="3" cols="50"></textarea>
</td>
</tr>
<tr>
<td>
<button id="send">Send</button>
</td>
</tr>
</table>
<script src="https://cdnjs.cloudflare.com/ajax/libs/socket.io/4.5.1/socket.io.js"></script>
<script src="./js/client.js"></script>
</body>
</html>
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<title>视频通话</title>
</head>
<body>
<div>
<h4>我的视频
<button type="button" id="startvideo">发起视频</button>
<button type="button" id="endvideo">结束视频</button>
</h4>
<video id="myvideo" autoplay playsinline></video>
</div>
<div>
<h4>对方的视频：</h4>
<video id="usersvideo" width="" height="" controls autoplay></video>
</div>
</body>
<script src="https://webrtc.github.io/adapter/adapter-latest.js"></script>
<script>
console.log(adapter.browserDetails.browser);
var myvideo = document.querySelector('#myvideo');
var usersvideo = document.querySelector('#usersvideo');
var startvideo = document.querySelector('#startvideo')
var endvideo = document.querySelector('#endvideo')
var videostream = ''
console.log(startvideo);
var constraints = {
audio: true,
video: true
};
//发起视频
startvideo.onclick = () => {
console.log('发起视频');
function handleSuccess (stream) {
var videoTracks = stream.getVideoTracks();
console.log('Using video device: ' + videoTracks[0].label);
myvideo.srcObject = stream;
}
function handleError (error) {
console.log('getUserMedia error: ' + error.name, error);
}
videostream = navigator.mediaDevices.getUserMedia(constraints).then(handleSuccess).catch(handleError);
}
//结束视频
endvideo.onclick = () => {
myvideo.src = ''
console.log('结束视频');
videostream.then(item => {
console.log(item);
})
// videostream.then(item => {
// console.log(item);
// })
// videostream.getTracks().forEach(function (track) {
// track.stop();
// console.log('关闭视频通道不成功', track);
// myvideo.src = ''
// })
}
</script>
</html>
对于不懂的api及流程可以看看webrtc开发入门教程
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