流媒体协议之WebRTC实现p2p视频通话（二）

最新推荐文章于 2024-05-14 21:57:51 发布

2401_84092542

最新推荐文章于 2024-05-14 21:57:51 发布

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分类专栏：程序员文章标签： webrtc p2p 音视频

本文链接：https://blog.csdn.net/2401_84092542/article/details/138639287

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}
// 被叫响应
called(callingInfo) {
this.calledHandle && this.calledHandle(callingInfo)
}
// 创建RTC
createLoacalPeer() {
this.peer = new RTCPeerConnection()
return this
}
// 将媒体流加入通信
addTrack() {
if (!this.peer || !this.localMedia) return
//this.localMedia.getTracks().forEach(track => this.peer.addTrack(track, this.localMedia));
this.peer.addStream(this.localMedia)
return this
}
// 创建 SDP offer
async createOffer(cb) {
if (!this.peer) return
let offer = await this.peer.createOffer({ OfferToReceiveAudio: true, OfferToReceiveVideo: true })
this.peer.setLocalDescription(offer)
cb && cb(offer)
return this
}
async createAnswer(offer, cb) {
if (!this.peer) return
this.peer.setRemoteDescription(offer)
let answer = await this.peer.createAnswer({ OfferToReceiveAudio: true, OfferToReceiveVideo: true })
this.peer.setLocalDescription(answer)
cb && cb(answer)
return this

}
listenerAddStream(cb) {
this.peer.addEventListener(‘addstream’, event => {
console.log(‘addstream事件触发’, event.stream);
cb && cb(event.stream);
})
return this
}
// 监听候选加入
listenerCandidateAdd(cb) {
this.peer.addEventListener(‘icecandidate’, event => {
let iceCandidate = event.candidate;
if (iceCandidate) {
console.log(‘发送candidate给远端’);
cb && cb(iceCandidate);
}

})
return this
}
// 检测ice协商过程
listenerGatheringstatechange () {
this.peer.addEventListener(‘icegatheringstatechange’, e => {
console.log('ice协商中: ', e.target.iceGatheringState);
})
return this
}
// 关闭RTC
closeRTC() {
// …
}
}

后端

const SocketIO = require(‘socket.io’)
const socketIO = new SocketIO({
path: ‘/websocket’
})

let userRoom = {
list: [],
add(user) {
this.list.push(user)
return this
},
del(id) {
this.list = this.list.filter(u => u.id !== id)
return this
},
sendAllUser(name, data) {
this.list.forEach(({ id }) => {
console.log(‘>>>>>’, id)
socketIO.to(id).emit(name, data)
})
return this
},
sendTo(id) {
return (eventName, data) => {
socketIO.to(id).emit(eventName, data)
}

2.后端

const SocketIO = require(‘socket.io’)
const socketIO = new SocketIO({
path: ‘/websocket’
})

socketIO.on(‘connection’, function(socket) {
console.log(‘连接加入.’, socket.id)

socket.on(‘addUser’, function(data) {
console.log(data.name, ‘加入房间’)
let user = {
id: socket.id,
name: data.name,
calling: false
}
userRoom.add(user).sendAllUser(‘updateUserList’, userRoom.list)
})

socket.on(‘sendMessage’, ({ content }) => {
console.log(‘转发消息：’, content)
userRoom.sendAllUser(‘updateMessageList’, { userId: socket.id, content, user: userRoom.findName(socket.id) })
})

socket.on(‘iceCandidate’, ({ id, iceCandidate }) => {
console.log(‘转发信道’)
userRoom.sendTo(id)(‘iceCandidate’, { iceCandidate, id: socket.id })
})

socket.on(‘offer’, ({id, offer}) => {
console.log(‘转发offer’)
userRoom.sendTo(id)(‘called’, { offer, id: socket.id, name: userRoom.findName(socket.id)})
})

socket.on(‘answer’, ({id, answer}) => {
console.log(‘接受视频’);
userRoom.sendTo(id)(‘answer’, {answer})
})

socket.on(‘rejectCall’, id => {
console.log(‘转发拒接视频’)
userRoom.sendTo(id)(‘callRejected’)
})

socket.on(‘disconnect’, () => {
// 断开删除
console.log(‘连接断开’, socket.id)
userRoom.del(socket.id).sendAllUser(‘updateUserList’, userRoom.list)
})
})

module.exports = socketIO

// www.js 这就不关键了
const http = require(‘http’)
const app = require(‘…/app’)
const socketIO = require(‘…/socket.js’)
const server = http.createServer(app.callback())
socketIO.attach(server)
server.listen(3003, () => {
console.log(‘server start on 127.0.0.1:3003’)
})

搭建 STUN/TURN

因为没有钱买服务器没试过

coturn 据说使用它搭建 STUN/TURN 服务非常的方便

编译

cd coturn
./configure --prefix=/usr/local/coturn
sudo make -j 4 && make install

配置

listening-port=3478 #指定侦听的端口
external-ip=39.105.185.198 #指定云主机的公网IP地址
user=aaaaaa:bbbbbb #访问 stun/turn服务的用户名和密码
realm=stun.xxx.cn #域名，这个一定要设置

#启动
cd /usr/local/coturn/bin
turnserver -c …/etc/turnserver.conf

trickle-ice https://webrtc.github.io/samples/src/content/peerconnection/trickle-ice 按里面的要求输入 stun/turn 地址、用户和密码
输入的信息分别是：
STUN or TURN URI 的值为： turn:stun.xxx.cn
用户名为： aaaaaa
密码为： bbbbbb

STUN参数传递

let ice = {“iceServers”: [
{“url”: “stun:stun.l.google.com:19302”}, // 无需密码的
// TURN 一般需要自己去定义
{
‘url’: ‘turn:192.158.29.39:3478?transport=udp’,
‘credential’: ‘JZEOEt2V3Qb0y27GRntt2u2PAYA=’, // 密码
‘username’: ‘28224511:1379330808’ // 用户名
},
{
‘url’: ‘turn:192.158.29.39:3478?transport=tcp’,
‘credential’: ‘JZEOEt2V3Qb0y27GRntt2u2PAYA=’,
‘username’: ‘28224511:1379330808’
}
]}
// 可以提供多iceServers地址，但RTC追选择一个进行协商

// 实例化的是给上参数 RTC会在合适的时候去获取本地墙后IP
let pc = new RTCPeerConnection(ice);

/*
// 据说这些免费的地址都可以用
stun:stun1.l.google.com:19302
stun:stun2.l.google.com:19302
stun:stun3.l.google.com:19302
stun:stun4.l.google.com:19302
stun:23.21.150.121
stun:stun01.sipphone.com
stun:stun.ekiga.net
stun:stun.fwdnet.net
stun:stun.ideasip.com
stun:stun.iptel.org
stun:stun.rixtelecom.se
stun:stun.schlund.de
stun:stunserver.org
stun:stun.softjoys.com
stun:stun.voiparound.com
stun:stun.voipbuster.com
stun:stun.voipstunt.com
stun:stun.voxgratia.org
stun:stun.xten.com
*/