WebRTC：一个视频聊天的简单例子

作者：程序猿小卡
阅读时间： 25~30min
WebRTC通信相关的API非常多，为避免篇幅过长，文中部分采用了伪代码进行讲解。详细代码例子可以在笔者的Github上找到，有问题欢迎留言交流

前文链接：

• WebRTC：会话描述协议
• WebRTC：连接建立过程的网络穿透

相关API简介

在前面的章节中，已经对WebRTC相关的重要知识点进行了介绍，包括涉及的网络协议、会话描述协议、如何进行网络穿透等，剩下的就是WebRTC的API了。 WebRTC通信相关的API非常多，主要完成了如下功能：

• 信令交换

• 通信候选地址交换

• 音视频采集

• 音视频发送、接收

相关API太多，为避免篇幅过长，文中部分采用了伪代码进行讲解。详细代码参考文章末尾，也可以在笔者的Github上找到，有问题欢迎留言交流。

1、信令交换

是WebRTC通信中的关键环节，交换的信息包括编解码器、网络协议、候选地址等。对于如何进行信令交换，WebRTC并没有明确说明，而是交给应用自己来决定，比如可以采用WebSocket。 发送方伪代码如下：

const pc = new RTCPeerConnection(iceConfig);
const offer = await pc.createOffer();
await pc.setLocalDescription(offer);
sendToPeerViaSignalingServer(SIGNALING_OFFER, offer); // 发送方发送信令消息
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接收方伪代码如下：

const pc = new RTCPeerConnection(iceConfig);
await pc.setRemoteDescription(offer);
const answer = await pc.createAnswer();
await pc.setLocalDescription(answer);
sendToPeerViaSignalingServer(SIGNALING_ANSWER, answer); // 接收方发送信令消息
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2、候选地址交换服务

当本地设置了会话描述信息，并添加了媒体流的情况下，ICE框架就会开始收集候选地址。两边收集到候选地址后，需要交换候选地址，并从中知道合适的候选地址对。

候选地址的交换，同样采用前面提到的信令服务，伪代码如下：

// 设置本地会话描述信息
const localPeer = new RTCPeerConnection(iceConfig);
const offer = await pc.createOffer();
await localPeer.setLocalDescription(offer);

// 本地采集音视频
const localVideo = document.getElementById('local-video');
const mediaStream = await navigator.mediaDevices.getUserMedia({ 
    video: true, 
    audio: true
});
localVideo.srcObject = mediaStream;

// 添加音视频流
mediaStream.getTracks().forEach(track => {
    localPeer.addTrack(track, mediaStream);
});

// 交换候选地址
localPeer.onicecandidate = function(evt) {
    if (evt.candidate) {
        sendToPeerViaSignalingServer(SIGNALING_CANDIDATE, evt.candidate);
    }
}
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3、音视频采集

可以使用浏览器提供的getUserMedia接口，采集本地的音视频。

const localVideo = document.getElementById('local-video');
const mediaStream = await navigator.mediaDevices.getUserMedia({ 
    video: true, 
    audio: true
});
localVideo.srcObject = mediaStream;
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4、音视频发送、接收

将采集到的音视频轨道，通过addTrack进行添加，发送给远端。

mediaStream.getTracks().forEach(track => {
    localPeer.addTrack(track, mediaStream);
});
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远端可以通过监听ontrack来监听音视频的到达，并进行播放。

remotePeer.ontrack = function(evt) {
    const remoteVideo = document.getElementById('remote-video');
    remoteVideo.srcObject = evt.streams[0];
}
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完整代码

包含两部分：客户端代码、服务端代码。

1、客户端代码

const socket = io.connect('http://localhost:3000');

const CLIENT_RTC_EVENT = 'CLIENT_RTC_EVENT';
const SERVER_RTC_EVENT = 'SERVER_RTC_EVENT';

const CLIENT_USER_EVENT = 'CLIENT_USER_EVENT';
const SERVER_USER_EVENT = 'SERVER_USER_EVENT';

const CLIENT_USER_EVENT_LOGIN = 'CLIENT_USER_EVENT_LOGIN'; // 登录

const SERVER_USER_EVENT_UPDATE_USERS = 'SERVER_USER_EVENT_UPDATE_USERS';

const SIGNALING_OFFER = 'SIGNALING_OFFER';
const SIGNALING_ANSWER = 'SIGNALING_ANSWER';
const SIGNALING_CANDIDATE = 'SIGNALING_CANDIDATE';

let remoteUser = ''; // 远端用户
let localUser = ''; // 本地登录用户

function log(msg) {
    console.log(`[client] ${msg}`);
}

socket.on('connect', function() {
    log('ws connect.');
});

socket.on('connect_error', function() {
    log('ws connect_error.');
});

socket.on('error', function(errorMessage) {
    log('ws error, ' + errorMessage);
});

socket.on(SERVER_USER_EVENT, function(msg) {
    const type = msg.type;
    const payload = msg.payload;

    switch(type) {
        case SERVER_USER_EVENT_UPDATE_USERS:
            updateUserList(payload);
            break;
    }
    log(`[${SERVER_USER_EVENT}] [${type}], ${JSON.stringify(msg)}`);
});

socket.on(SERVER_RTC_EVENT, function(msg) {
    const {type} = msg;

    switch(type) {
        case SIGNALING_OFFER:
            handleReceiveOffer(msg);
            break;
        case SIGNALING_ANSWER:
            handleReceiveAnswer(msg);
            break;
        case SIGNALING_CANDIDATE:
            handleReceiveCandidate(msg);
            break;
    }
});

async function handleReceiveOffer(msg) {
    log(`receive remote description from ${msg.payload.from}`);

    // 设置远端描述
    const remoteDescription = new RTCSessionDescription(msg.payload.sdp);
    remoteUser = msg.payload.from;
    createPeerConnection();
    await pc.setRemoteDescription(remoteDescription); // TODO 错误处理

    // 本地音视频采集
    const localVideo = document.getElementById('local-video');
    const mediaStream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
    localVideo.srcObject = mediaStream;
    mediaStream.getTracks().forEach(track => {
        pc.addTrack(track, mediaStream);
        // pc.addTransceiver(track, {streams: [mediaStream]}); // 这个也可以
    });
    // pc.addStream(mediaStream); // 目前这个也可以，不过接口后续会废弃

    const answer = await pc.createAnswer(); // TODO 错误处理
    await pc.setLocalDescription(answer);
    sendRTCEvent({
        type: SIGNALING_ANSWER,
        payload: {
            sdp: answer,
            from: localUser,
            target: remoteUser
        }
    });
}

async function handleReceiveAnswer(msg) {
    log(`receive remote answer from ${msg.payload.from}`);

    const remoteDescription = new RTCSessionDescription(msg.payload.sdp);
    remoteUser = msg.payload.from;

    await pc.setRemoteDescription(remoteDescription); // TODO 错误处理
}

async function handleReceiveCandidate(msg){
    log(`receive candidate from ${msg.payload.from}`);
    await pc.addIceCandidate(msg.payload.candidate); // TODO 错误处理
}

/**
 * 发送用户相关消息给服务器
 * @param {Object} msg 格式如 { type: 'xx', payload: {} }
 */
function sendUserEvent(msg) {
    socket.emit(CLIENT_USER_EVENT, JSON.stringify(msg));
}

/**
 * 发送RTC相关消息给服务器
 * @param {Object} msg 格式如{ type: 'xx', payload: {} }
 */
function sendRTCEvent(msg) {
    socket.emit(CLIENT_RTC_EVENT, JSON.stringify(msg));
}

let pc = null;

/**
 * 邀请用户加入视频聊天
 *  1、本地启动视频采集
 *  2、交换信令
 */
async function startVideoTalk() {
    // 开启本地视频
    const localVideo = document.getElementById('local-video');
    const mediaStream = await navigator.mediaDevices.getUserMedia({
        video: true, 
        audio: true
    });
    localVideo.srcObject = mediaStream;

    // 创建 peerConnection
    createPeerConnection();

    // 将媒体流添加到webrtc的音视频收发器
    mediaStream.getTracks().forEach(track => {
        pc.addTrack(track, mediaStream);
        // pc.addTransceiver(track, {streams: [mediaStream]});
    });
    // pc.addStream(mediaStream); // 目前这个也可以，不过接口后续会废弃
}

function createPeerConnection() {
    const iceConfig = {"iceServers": [
        {url: 'stun:stun.ekiga.net'},
        {url: 'turn:turnserver.com', username: 'user', credential: 'pass'}
    ]};

    pc = new RTCPeerConnection(iceConfig);

    pc.onnegotiationneeded = onnegotiationneeded;
    pc.onicecandidate = onicecandidate;
    pc.onicegatheringstatechange = onicegatheringstatechange;
    pc.oniceconnectionstatechange = oniceconnectionstatechange;
    pc.onsignalingstatechange = onsignalingstatechange;
    pc.ontrack = ontrack;

    return pc;
}

async function onnegotiationneeded() {
    log(`onnegotiationneeded.`);

    const offer = await pc.createOffer();
    await pc.setLocalDescription(offer); // TODO 错误处理

    sendRTCEvent({
        type: SIGNALING_OFFER,
        payload: {
            from: localUser,
            target: remoteUser,
            sdp: pc.localDescription // TODO 直接用offer？
        }
    });
}

function onicecandidate(evt) {
    if (evt.candidate) {
        log(`onicecandidate.`);

        sendRTCEvent({
            type: SIGNALING_CANDIDATE,            
            payload: {
                from: localUser,
                target: remoteUser,
                candidate: evt.candidate
            }
        });
    }
}

function onicegatheringstatechange(evt) {
    log(`onicegatheringstatechange, pc.iceGatheringState is ${pc.iceGatheringState}.`);
}

function oniceconnectionstatechange(evt) {
    log(`oniceconnectionstatechange, pc.iceConnectionState is ${pc.iceConnectionState}.`);
}

function onsignalingstatechange(evt) {
    log(`onsignalingstatechange, pc.signalingstate is ${pc.signalingstate}.`);
}

// 调用 pc.addTrack(track, mediaStream)，remote peer的 onTrack 会触发两次
// 实际上两次触发时，evt.streams[0] 指向同一个mediaStream引用
// 这个行为有点奇怪，github issue 也有提到 https://github.com/meetecho/janus-gateway/issues/1313
let stream;
function ontrack(evt) {
    // if (!stream) {
    //     stream = evt.streams[0];
    // } else {
    //     console.log(`${stream === evt.streams[0]}`); // 这里为true
    // }
    log(`ontrack.`);
    const remoteVideo = document.getElementById('remote-video');
    remoteVideo.srcObject = evt.streams[0];
}

// 点击用户列表
async function handleUserClick(evt) {
    const target = evt.target;
    const userName = target.getAttribute('data-name').trim();

    if (userName === localUser) {
        alert('不能跟自己进行视频会话');
        return;
    }

    log(`online user selected: ${userName}`);

    remoteUser = userName;
    await startVideoTalk(remoteUser);
}

/**
 * 更新用户列表
 * @param {Array} users 用户列表，比如 [{name: '小明', name: '小强'}]
 */
function updateUserList(users) {
    const fragment = document.createDocumentFragment();
    const userList = document.getElementById('login-users');
    userList.innerHTML = '';

    users.forEach(user => {
        const li = document.createElement('li');
        li.innerHTML = user.userName;
        li.setAttribute('data-name', user.userName);
        li.addEventListener('click', handleUserClick);
        fragment.appendChild(li);
    });    

    userList.appendChild(fragment);
}

/**
 * 用户登录
 * @param {String} loginName 用户名
 */
function login(loginName) {
    localUser = loginName;
    sendUserEvent({
        type: CLIENT_USER_EVENT_LOGIN,
        payload: {
            loginName: loginName
        }
    });
}

// 处理登录
function handleLogin(evt) {
    let loginName = document.getElementById('login-name').value.trim();
    if (loginName === '') {
        alert('用户名为空！');
        return;
    }
    login(loginName);
}

function init() {
    document.getElementById('login-btn').addEventListener('click', handleLogin);
}

init();
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2、服务端代码

// 添加ws服务
const io = require('socket.io')(server);
let connectionList = [];

const CLIENT_RTC_EVENT = 'CLIENT_RTC_EVENT';
const SERVER_RTC_EVENT = 'SERVER_RTC_EVENT';

const CLIENT_USER_EVENT = 'CLIENT_USER_EVENT';
const SERVER_USER_EVENT = 'SERVER_USER_EVENT';

const CLIENT_USER_EVENT_LOGIN = 'CLIENT_USER_EVENT_LOGIN';
const SERVER_USER_EVENT_UPDATE_USERS = 'SERVER_USER_EVENT_UPDATE_USERS';

function getOnlineUser() {
  return connectionList
  .filter(item => {
    return item.userName !== '';
  })
  .map(item => {
    return {
      userName: item.userName
    };
  });
}

function setUserName(connection, userName) {
  connectionList.forEach(item => {
    if (item.connection.id === connection.id) {
      item.userName = userName;
    }
  });
}

function updateUsers(connection) {
  connection.emit(SERVER_USER_EVENT, { type: SERVER_USER_EVENT_UPDATE_USERS, payload: getOnlineUser()});  
}

io.on('connection', function (connection) {

  connectionList.push({
    connection: connection,
    userName: ''
  });

  // 连接上的用户，推送在线用户列表
  // connection.emit(SERVER_USER_EVENT, { type: SERVER_USER_EVENT_UPDATE_USERS, payload: getOnlineUser()});
  updateUsers(connection);

  connection.on(CLIENT_USER_EVENT, function(jsonString) {
    const msg = JSON.parse(jsonString);
    const {type, payload} = msg;

    if (type === CLIENT_USER_EVENT_LOGIN) {
      setUserName(connection, payload.loginName);
      connectionList.forEach(item => {
        // item.connection.emit(SERVER_USER_EVENT, { type: SERVER_USER_EVENT_UPDATE_USERS, payload: getOnlineUser()});
        updateUsers(item.connection);
      });
    }
  });

  connection.on(CLIENT_RTC_EVENT, function(jsonString) {
    const msg = JSON.parse(jsonString);
    const {payload} = msg;
    const target = payload.target;

    const targetConn = connectionList.find(item => {
      return item.userName === target;
    });
    if (targetConn) {
      targetConn.connection.emit(SERVER_RTC_EVENT, msg);
    }
  });

  connection.on('disconnect', function () {
    connectionList = connectionList.filter(item => {
      return item.connection.id !== connection.id;
    });
    connectionList.forEach(item => {
      // item.connection.emit(SERVER_USER_EVENT, { type: SERVER_USER_EVENT_UPDATE_USERS, payload: getOnlineUser()});
      updateUsers(item.connection);
    });    
  });
});
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写在后面

WebRTC的API非常多，因为WebRTC本身就比较复杂，随着时间的推移，WebRTC的某些API(包括某些协议细节)也在改动或被废弃，这其中也有向后兼容带来的复杂性，比如本地视频采集后加入传输流，可以采用 addStream 或 addTrack 或 addTransceiver，再比如会话描述版本从plan-b迁移到unified-plan。

建议亲自动手撸一遍代码，加深了解。

相关链接

• 2019.08.02-video-talk-using-webrtc
• developer.mozilla.org/en-US/docs/…
• onremotestream called twice for each remote stream

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转载于:https://juejin.im/post/5d4b84a6f265da03bf0f2de8