Implementing online audio conference system using WebMan technology

Title: Using WebRTC technology to implement an online audio conferencing system

Introduction:
With the development of globalization and the popularity of remote work, online audio conferencing systems have become An important tool for modern enterprise communication and collaboration. This article will introduce how to use WebRTC technology to build a Web-based audio conferencing system, and show the implementation details through code examples.

Part One: WebRTC Technology Overview
WebRTC (Web Real-Time Communication) is an open standard that can implement real-time audio and video communication in the browser. It enables peer-to-peer communication between browsers via a JavaScript API, without the need for any plugins or extensions.

Before starting development, we must first ensure that the browser supports WebRTC technology. Currently, mainstream browsers (such as Chrome, Firefox, etc.) already natively support WebRTC.

Part 2: Building the basic architecture of the audio conference system
First, we need to create a server to manage each participant in the audio conference. This server uses WebRTC technology middleware, such as WebMan, to handle signaling exchange and stream transmission.

The code to implement the server is as follows (using Node.js and Express framework):

const express = require('express');
const http = require('http');
const WebSocket = require('ws');

const app = express();
const server = http.createServer(app);
const wss = new WebSocket.Server({server});

wss.on('connection', ws => {
  // 处理信令交换和流传输
});

server.listen(8080, () => {
  console.log('Server is running on port 8080');
});

Part 3: Implement the audio conferencing function on the participant side
On the participant side, we need to use WebRTC API to realize the collection, processing and transmission of audio streams. The following is a simple code example:

const configuration = { iceServers: [{ urls: 'stun:stun.l.google.com:19302' }] };
const peerConnection = new RTCPeerConnection(configuration);
const localStream = await navigator.mediaDevices.getUserMedia({ audio: true });

localStream.getTracks().forEach(track => {
  peerConnection.addTrack(track, localStream);
});

peerConnection.addEventListener('icecandidate', event => {
  if (event.candidate) {
    // 将候选者传递给服务器
  }
});

peerConnection.addEventListener('negotiationneeded', async () => {
  // 创建发起通话的信令
  await peerConnection.setLocalDescription();
  // 将本地描述SDP发送给其他参与者
});

peerConnection.addEventListener('track', event => {
  // 处理对方的音频流
});

// 加入音频会议
async function joinConference() {
  // 从服务器获取其他参与者的信息
  const remoteDescription = await fetch('https://example.com/remoteDescription');
  
  await peerConnection.setRemoteDescription(new RTCSessionDescription(remoteDescription));
}

Part 4: Implementing room management and user interface of the audio conferencing system
In order to manage multiple audio conferencing rooms and the joining of users, we can use a database (such as MongoDB) to save room and user information, and a web interface can also be used to display the status of the audio conference and provide user operations.

In terms of user interface, we can use HTML, CSS and JavaScript to implement functions such as room selection, joining meetings and leaving meetings.

Conclusion:
By leveraging WebRTC technology and using middleware such as WebMan, we successfully built a Web-based audio conferencing system. This system can effectively realize remote collaboration and communication, providing enterprises with a more efficient and convenient working environment. Through the code examples provided in this article, readers can learn how to use the WebRTC API and basic framework to implement various communication functions.

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