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。
信令交换
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
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
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);
}
}
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;
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);
});
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];
}
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();
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);
});
});
});
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
https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection
onremotestream called twice for each remote stream
