最简单的webrtc p2p demo – 源码集中营

最简单的webrtc p2p demo
之前没看过js的服务，nodejs 看起来做demo好方便
github
nodej启动服务
需要express 和 express-ws
启动websocket 协议的服务，监听8080端口
router有 / 和 /p2p
首先访问 / ，触发遍历每个client，向每个client发送 data
/ 的 data 是./client/index.html
./client/index.html 内容
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<meta http-equiv="X-UA-Compatible" content="ie=edge">
<title>p2p webrtc</title>
<style>
.container {
width: 250px;
margin: 100px auto;
padding: 10px 30px;
border-radius: 4px;
border: 1px solid #ebeef5;
box-shadow: 0 2px 12px 0 rgba(0,0,0,.1);
color: #303133;
}
</style>
</head>
<body>
<div class="container">
<p>流程：</p>
<ul>
<li>打开<a href="/go.html?url=/p2p?type=answer" target="_blank">接收方页面</a>；</li>
<li>打开<a href="/go.html?url=/p2p?type=offer" target="_blank">发起方页面</a>；</li>
<li>确认双方都已建立ws连接；</li>
<li>发起方点击 start 按钮。</li>
</ul>
</div>
</body>
</html>
主要是有俩链接
接收是answer
发送是 offer
点击链接后 会做为一个客户端，向nodejs发送请求
nodejs会发 ./client/p2p.html 给客户端
<li>打开<a href="/go.html?url=/p2p?type=answer" target="_blank">接收方页面</a>；</li>
<li>打开<a href="/go.html?url=/p2p?type=offer" target="_blank">发起方页面</a>；</li>
<li>确认双方都已建立ws连接；</li>
<li>发起方点击 start 按钮。</li>
code
const app = require('express')();
const wsInstance = require('express-ws')(app);
app.ws('/', ws => {
ws.on('message', data => {
// 未做业务处理，收到消息后直接广播
wsInstance.getWss().clients.forEach(server => {
if (server !== ws) {
server.send(data);
}
});
});
});
app.get('/', (req, res) => {
res.sendFile('./client/index.html', { root: __dirname });
});
app.get('/p2p', (req, res) => {
res.sendFile('./client/p2p.html', { root: __dirname });
});
app.listen(8080);
客户端收到 ./client/p2p.html
code
<!DOCTYPE html>
<html lang="en">
<head>
<meta charset="UTF-8">
<meta name="viewport" content="width=device-width, initial-scale=1.0">
<meta http-equiv="X-UA-Compatible" content="ie=edge">
<title></title>
<style>
* {
padding: 0;
margin: 0;
box-sizing: border-box;
}
.container {
width: 100%;
display: flex;
display: -webkit-flex;
justify-content: space-around;
padding-top: 20px;
}
.video-box {
position: relative;
width: 800px;
height: 400px;
}
#remote-video {
width: 100%;
height: 100%;
display: block;
object-fit: cover;
border: 1px solid #eee;
background-color: #F2F6FC;
}
#local-video {
position: absolute;
right: 0;
bottom: 0;
width: 240px;
height: 120px;
object-fit: cover;
border: 1px solid #eee;
background-color: #EBEEF5;
}
.start-button {
position: absolute;
left: 50%;
top: 50%;
width: 100px;
display: none;
line-height: 40px;
outline: none;
color: #fff;
background-color: #409eff;
border: none;
border-radius: 4px;
cursor: pointer;
transform: translate(-50%, -50%);
}
.logger {
width: 40%;
padding: 14px;
line-height: 1.5;
color: #4fbf40;
border-radius: 6px;
background-color: #272727;
}
.logger .error {
color: #DD4A68;
}
</style>
</head>
<body>
<div class="container">
<div class="video-box">
<video id="remote-video"></video>
<video id="local-video" muted></video>
<button class="start-button" onclick="startLive()">start</button>
</div>
<div class="logger"></div>
</div>
<script>
const message = {
el: document.querySelector('.logger'),
log (msg) {
this.el.innerHTML += `<span>${new Date().toLocaleTimeString()}：${msg}</span><br/>`;
},
error (msg) {
this.el.innerHTML += `<span class="error">${new Date().toLocaleTimeString()}：${msg}</span><br/>`;
}
};
const target = location.search.slice(6);
const localVideo = document.querySelector('#local-video');
const remoteVideo = document.querySelector('#remote-video');
const button = document.querySelector('.start-button');
localVideo.onloadeddata = () => {
message.log('播放本地视频');
localVideo.play();
}
remoteVideo.onloadeddata = () => {
message.log('播放对方视频');
remoteVideo.play();
}
document.title = target === 'offer' ? '发起方' : '接收方';
message.log('信令通道（WebSocket）创建中......');
const socket = new WebSocket('ws://localhost:8080');
socket.onopen = () => {
message.log('信令通道创建成功！');
target === 'offer' && (button.style.display = 'block');
}
socket.onerror = () => message.error('信令通道创建失败！');
socket.onmessage = e => {
const { type, sdp, iceCandidate } = JSON.parse(e.data)
if (type === 'answer') {
peer.setRemoteDescription(new RTCSessionDescription({ type, sdp }));
} else if (type === 'answer_ice') {
peer.addIceCandidate(iceCandidate);
} else if (type === 'offer') {
startLive(new RTCSessionDescription({ type, sdp }));
} else if (type === 'offer_ice') {
peer.addIceCandidate(iceCandidate);
}
};
const PeerConnection = window.RTCPeerConnection || window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
!PeerConnection && message.error('浏览器不支持WebRTC！');
const peer = new PeerConnection();
peer.ontrack = e => {
if (e && e.streams) {
message.log('收到对方音频/视频流数据...');
remoteVideo.srcObject = e.streams[0];
}
};
peer.onicecandidate = e => {
if (e.candidate) {
message.log('搜集并发送候选人');
socket.send(JSON.stringify({
type: `${target}_ice`,
iceCandidate: e.candidate
}));
} else {
message.log('候选人收集完成！');
}
};
async function startLive (offerSdp) {
target === 'offer' && (button.style.display = 'none');
let stream;
try {
message.log('尝试调取本地摄像头/麦克风');
stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
message.log('摄像头/麦克风获取成功！');
localVideo.srcObject = stream;
} catch {
message.error('摄像头/麦克风获取失败！');
return;
}
message.log(`------ WebRTC ${target === 'offer' ? '发起方' : '接收方'}流程开始 ------`);
message.log('将媒体轨道添加到轨道集');
stream.getTracks().forEach(track => {
peer.addTrack(track, stream);
});
if (!offerSdp) {
message.log('创建本地SDP');
const offer = await peer.createOffer();
await peer.setLocalDescription(offer);
message.log(`传输发起方本地SDP`);
socket.send(JSON.stringify(offer));
} else {
message.log('接收到发送方SDP');
await peer.setRemoteDescription(offerSdp);
message.log('创建接收方（应答）SDP');
const answer = await peer.createAnswer();
message.log(`传输接收方（应答）SDP`);
socket.send(JSON.stringify(answer));
await peer.setLocalDescription(answer);
}
}
</script>
</body>
</html>
p2p
作为websocket的客户端，向服务端8080端口建立websocket连接
收到socket的消息，要做解析，json格式？
消息类型 sdk文件 icecandidate
收到answer ，自己就设置对端的sdp
收到answer 活着 offer 的ice，自己就设置ice
收到offer，自己就开始使用对端的sdp做播放？？
socket.onmessage = e => {
const { type, sdp, iceCandidate } = JSON.parse(e.data)
if (type === 'answer') {
peer.setRemoteDescription(new RTCSessionDescription({ type, sdp }));
} else if (type === 'answer_ice') {
peer.addIceCandidate(iceCandidate);
} else if (type === 'offer') {
startLive(new RTCSessionDescription({ type, sdp }));
} else if (type === 'offer_ice') {
peer.addIceCandidate(iceCandidate);
}
};
打印,带显示时间的
const message = {
el: document.querySelector('.logger'),
log (msg) {
this.el.innerHTML += `<span>${new Date().toLocaleTimeString()}：${msg}</span><br/>`;
},
error (msg) {
this.el.innerHTML += `<span class="error">${new Date().toLocaleTimeString()}：${msg}</span><br/>`;
}
};
这好像是获取页面上对应的元素对象
本地
远程
开始按钮
const target = location.search.slice(6);
const localVideo = document.querySelector('#local-video');
const remoteVideo = document.querySelector('#remote-video');
const button = document.querySelector('.start-button');
网页标题
document.title = target === 'offer' ? '发起方' : '接收方';
收到本地的数据就play本地，收到对方的数据就play对端
localVideo.onloadeddata = () => {
message.log('播放本地视频');
localVideo.play();
}
remoteVideo.onloadeddata = () => {
message.log('播放对方视频');
remoteVideo.play();
}
创建peerconnection
peer 应该指的p2p里面的对端
peer里处理的是对端的音视频
比如 remoteVideo 拿到对端的stream 进行播放
const PeerConnection = window.RTCPeerConnection || window.mozRTCPeerConnection || window.webkitRTCPeerConnection;
!PeerConnection && message.error('浏览器不支持WebRTC！');
const peer = new PeerConnection();
在peer做处理
track 记录track里的每个流
ice ，会通过socket发送到对端？？？
peer.ontrack = e => {
if (e && e.streams) {
message.log('收到对方音频/视频流数据...');
remoteVideo.srcObject = e.streams[0];
}
};
peer.onicecandidate = e => {
if (e.candidate) {
message.log('搜集并发送候选人');
socket.send(JSON.stringify({
type: `${target}_ice`,
iceCandidate: e.candidate
}));
} else {
message.log('候选人收集完成！');
}
};
使用 SDP 直播 ： startLive
1 getUserMedia 获取用户的音视频设备，这样拿到stream
2 从流里面拿到track
3 非offer的sdp 就是本地自己的sdp
TODO 回头再来看把，这里还要发送sdp到对端 然后才设置
async function startLive (offerSdp) {
target === 'offer' && (button.style.display = 'none');
let stream;
try {
message.log('尝试调取本地摄像头/麦克风');
stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true });
message.log('摄像头/麦克风获取成功！');
localVideo.srcObject = stream;
} catch {
message.error('摄像头/麦克风获取失败！');
return;
}
message.log(`------ WebRTC ${target === 'offer' ? '发起方' : '接收方'}流程开始 ------`);
message.log('将媒体轨道添加到轨道集');
stream.getTracks().forEach(track => {
peer.addTrack(track, stream);
});
if (!offerSdp) {
message.log('创建本地SDP');
const offer = await peer.createOffer();
await peer.setLocalDescription(offer);
message.log(`传输发起方本地SDP`);
socket.send(JSON.stringify(offer));
} else {
message.log('接收到发送方SDP');
await peer.setRemoteDescription(offerSdp);
message.log('创建接收方（应答）SDP');
const answer = await peer.createAnswer();
message.log(`传输接收方（应答）SDP`);
socket.send(JSON.stringify(answer));
await peer.setLocalDescription(answer);
}
}
综上
里面有信令的交互
TODO sdp 是通过信令通道来的？ice好像也是信令做的？
通过peerconnection拿到stream
mac 安装
npm run dev 才能打开8080 服务
npm install 只是安装依赖库
✘ zhangbin@pb6a80114  ~/tet/licodelllcode  git clone https://github.com/shushushv/webrtc-p2p.git
Cloning into 'webrtc-p2p'...
remote: Enumerating objects: 12, done.
remote: Counting objects: 100% (12/12), done.
remote: Compressing objects: 100% (9/9), done.
remote: Total 12 (delta 0), reused 9 (delta 0), pack-reused 0
Unpacking objects: 100% (12/12), done.
zhangbin@pb6a80114  ~/tet/licodelllcode  cd webrtc-p2p
zhangbin@pb6a80114  ~/tet/licodelllcode/webrtc-p2p   master  npm install
npm notice created a lockfile as package-lock.json. You should commit this file.
npm WARN [email protected] No repository field.
added 53 packages from 45 contributors and audited 129 packages in 4.953s
found 0 vulnerabilities
╭────────────────────────────────────────────────────────────────╮
│                                                                │
│       New minor version of npm available! 6.9.0 → 6.13.1       │
│   Changelog: https://github.com/npm/cli/releases/tag/v6.13.1   │
│               Run npm install -g npm to update!                │
│                                                                │
╰────────────────────────────────────────────────────────────────╯
zhangbin@pb6a80114  ~/tet/licodelllcode/webrtc-p2p   master  npm run dev
> [email protected] dev /Users/zhangbin/tet/licodelllcode/webrtc-p2p
> node index.js

最简单的webrtc p2p demo
之前没看过js的服务，nodejs 看起来做demo好方便