mediasoup/app.js

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require('dotenv').config()
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const express = require('express');
const app = express();
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const Server = require('socket.io');
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const path = require('node:path');
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const fs = require('node:fs');
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let https;
try {
https = require('node:https');
} catch (err) {
console.log('https support is disabled!');
}
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const mediasoup = require('mediasoup');
let worker
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/**
* videoCalls
* |-> Router
* |-> Producer
* |-> Consumer
* |-> Producer Transport
* |-> Consumer Transport
*
* '<callId>': {
* router: Router,
* producer: Producer,
* producerTransport: Producer Transport,
* consumer: Consumer,
* consumerTransport: Consumer Transport
* }
*
**/
let videoCalls = {}
let socketDetails = {}
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app.get('/', (_req, res) => {
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res.send('Hello from mediasoup app!')
})
app.use('/sfu', express.static(path.join(__dirname, 'public')))
// SSL cert for HTTPS access
const options = {
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key: fs.readFileSync(process.env.SERVER_KEY, 'utf-8'),
cert: fs.readFileSync(process.env.SERVER_CERT, 'utf-8'),
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}
const httpsServer = https.createServer(options, app);
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const io = new Server(httpsServer, {
allowEIO3: true,
origins: ["*:*"],
// allowRequest: (req, next) => {
// console.log('req', req);
// next(null, true)
// }
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});
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// const io = new Server(server, { origins: '*:*', allowEIO3: true });
httpsServer.listen(process.env.PORT, () => {
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console.log('Video server listening on port:', process.env.PORT);
});
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const peers = io.of('/');
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const createWorker = async () => {
try {
worker = await mediasoup.createWorker({
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rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
})
console.log(`[createWorker] worker pid ${worker.pid}`);
worker.on('died', error => {
// This implies something serious happened, so kill the application
console.error('mediasoup worker has died', error);
setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
})
return worker;
} catch (error) {
console.log(`ERROR | createWorker | ${error.message}`);
}
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}
// We create a Worker as soon as our application starts
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worker = createWorker();
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// This is an Array of RtpCapabilities
// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
// list of media codecs supported by mediasoup ...
// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
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const mediaCodecs = [
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{
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kind : 'audio',
mimeType : 'audio/opus',
clockRate : 48000,
channels : 2
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},
{
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kind : 'video',
mimeType : 'video/VP8',
clockRate : 90000,
parameters :
{
'x-google-start-bitrate' : 1000
}
},
{
kind : 'video',
mimeType : 'video/VP9',
clockRate : 90000,
parameters :
{
'profile-id' : 2,
'x-google-start-bitrate' : 1000
}
},
{
kind : 'video',
mimeType : 'video/h264',
clockRate : 90000,
parameters :
{
'packetization-mode' : 1,
'profile-level-id' : '4d0032',
'level-asymmetry-allowed' : 1,
'x-google-start-bitrate' : 1000
}
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},
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{
kind : 'video',
mimeType : 'video/h264',
clockRate : 90000,
parameters :
{
'packetization-mode' : 1,
'profile-level-id' : '42e01f',
'level-asymmetry-allowed' : 1,
'x-google-start-bitrate' : 1000
}
}
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];
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const closeCall = (callId) => {
try {
if (callId && videoCalls[callId]) {
videoCalls[callId].producer?.close();
videoCalls[callId].consumer?.close();
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videoCalls[callId]?.consumerTransport?.close();
videoCalls[callId]?.producerTransport?.close();
videoCalls[callId]?.router?.close();
delete videoCalls[callId];
} else {
console.log(`The call with id ${callId} has already been deleted`);
}
} catch (error) {
console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
}
}
/*
- Handlers for WS events
- These are created only when we have a connection with a peer
*/
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peers.on('connection', async socket => {
console.log('[connection] socketId:', socket.id);
// After making the connection successfully, we send the client a 'connection-success' event
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socket.emit('connection-success', {
socketId: socket.id
});
// It is triggered when the peer is disconnected
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socket.on('disconnect', () => {
const callId = socketDetails[socket.id];
console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
delete socketDetails[socket.id];
closeCall(callId);
});
/*
- This event creates a room with the roomId and the callId sent
- It will return the rtpCapabilities of that room
- If the room already exists, it will not create it, but will only return rtpCapabilities
*/
socket.on('createRoom', async ({ callId }, callback) => {
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let callbackResponse = null;
try {
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// We can continue with the room creation process only if we have a callId
if (callId) {
console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
if (!videoCalls[callId]) {
console.log('[createRoom] callId', callId);
videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) }
console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
}
socketDetails[socket.id] = callId;
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// rtpCapabilities is set for callback
console.log('[getRtpCapabilities] callId', callId);
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callbackResponse = {
rtpCapabilities :videoCalls[callId].router.rtpCapabilities
};
} else {
console.log(`[createRoom] missing callId ${callId}`);
}
} catch (error) {
console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
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} finally {
callback(callbackResponse);
}
});
/*
- Client emits a request to create server side Transport
- Depending on the sender, producerTransport or consumerTransport is created on that router
- It will return parameters, these are required for the client to create the RecvTransport
from the client.
- If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
- If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
*/
socket.on('createWebRtcTransport', async ({ sender }, callback) => {
try {
const callId = socketDetails[socket.id];
console.log(`[createWebRtcTransport] sender ${sender} | callId ${callId}`);
if (sender) {
if (!videoCalls[callId].producerTransport) {
videoCalls[callId].producerTransport = await createWebRtcTransportLayer(callId, callback);
} else {
console.log(`producerTransport has already been defined | callId ${callId}`);
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callback(null);
}
} else if (!sender) {
if (!videoCalls[callId].consumerTransport) {
videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
} else {
console.log(`consumerTransport has already been defined | callId ${callId}`);
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callback(null);
}
}
} catch (error) {
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console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
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callback(error);
}
});
/*
- The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
- The connection is made to the created transport
*/
socket.on('transport-connect', async ({ dtlsParameters }) => {
try {
const callId = socketDetails[socket.id];
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if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
console.log(`[transport-connect] socket.id ${socket.id} | callId ${callId}`);
await videoCalls[callId].producerTransport.connect({ dtlsParameters });
} catch (error) {
console.log(`ERROR | transport-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
}
});
/*
- The event sent by the client (PRODUCER) after successfully connecting to producerTransport
- For the router with the id callId, we make produce on producerTransport
- Create the handler on producer at the 'transportclose' event
*/
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socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
try {
const callId = socketDetails[socket.id];
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if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
kind,
rtpParameters,
});
console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producer.id} | kind: ${videoCalls[callId].producer.kind}`);
videoCalls[callId].producer.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport for this producer closed', callId)
closeCall(callId);
});
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// Send back to the client the Producer's id
callback && callback({
id: videoCalls[callId].producer.id
});
} catch (error) {
console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
}
});
/*
- The client sends this event after successfully creating a createRecvTransport(AS CONSUMER)
- The connection is made to the created consumerTransport
*/
socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
try {
const callId = socketDetails[socket.id];
console.log(`[transport-recv-connect] socket.id ${socket.id} | callId ${callId}`);
await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
} catch (error) {
console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
}
})
/*
- The customer consumes after successfully connecting to consumerTransport
- The previous step was 'transport-recv-connect', and before that 'createWebRtcTransport'
- This event is only sent by the consumer
- The parameters that the consumer consumes are returned
- The consumer does consumerTransport.consume(params)
*/
socket.on('consume', async ({ rtpCapabilities }, callback) => {
try {
const callId = socketDetails[socket.id];
console.log('[consume] callId', callId);
// Check if the router can consume the specified producer
if (videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producer.id,
rtpCapabilities
})) {
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console.log('[consume] Can consume', callId);
// Transport can now consume and return a consumer
videoCalls[callId].consumer = await videoCalls[callId].consumerTransport.consume({
producerId: videoCalls[callId].producer.id,
rtpCapabilities,
paused: true,
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
videoCalls[callId].consumer.on('transportclose', () => {
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const callId = socketDetails[socket.id];
console.log('transport close from consumer', callId);
closeCall(callId);
});
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// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
videoCalls[callId].consumer.on('producerclose', () => {
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const callId = socketDetails[socket.id];
console.log('producer of consumer closed', callId);
closeCall(callId);
});
// From the consumer extract the following params to send back to the Client
const params = {
id: videoCalls[callId].consumer.id,
producerId: videoCalls[callId].producer.id,
kind: videoCalls[callId].consumer.kind,
rtpParameters: videoCalls[callId].consumer.rtpParameters,
};
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// Send the parameters to the client
callback({ params });
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} else {
console.log(`[canConsume] Can't consume | callId ${callId}`);
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callback(null);
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}
} catch (error) {
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console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
callback({ params: { error } });
}
});
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/*
- Event sent by the consumer after consuming to resume the pause
- When consuming on consumerTransport, it is initially done with paused: true, here we will resume
*/
socket.on('consumer-resume', async () => {
try {
const callId = socketDetails[socket.id];
console.log(`[consumer-resume] callId ${callId}`)
await videoCalls[callId].consumer.resume();
} catch (error) {
console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
}
});
});
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/*
- Called from at event 'createWebRtcTransport' and assigned to the consumer or producer transport
- It will return parameters, these are required for the client to create the RecvTransport
from the client.
- If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
- If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
*/
const createWebRtcTransportLayer = async (callId, callback) => {
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try {
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console.log('[createWebRtcTransportLayer] callId', callId);
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// https://mediasoup.org/documentation/v3/mediasoup/api/#WebRtcTransportOptions
const webRtcTransport_options = {
listenIps: [
{
ip: process.env.IP, // Listening IPv4 or IPv6.
announcedIp: process.env.ANNOUNCED_IP, // Announced IPv4 or IPv6 (useful when running mediasoup behind NAT with private IP).
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}
],
enableUdp: true,
enableTcp: true,
preferUdp: true,
};
// https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options)
console.log(`callId: ${callId} | transport id: ${transport.id}`)
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// Handler for when DTLS(Datagram Transport Layer Security) changes
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transport.on('dtlsstatechange', dtlsState => {
console.log(`transport | dtlsstatechange | calldId ${callId} | dtlsState ${dtlsState}`);
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if (dtlsState === 'closed') {
transport.close();
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}
});
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// Handler if the transport layer has closed (for various reasons)
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transport.on('close', () => {
console.log(`transport | closed | calldId ${callId}`);
});
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const params = {
id: transport.id,
iceParameters: transport.iceParameters,
iceCandidates: transport.iceCandidates,
dtlsParameters: transport.dtlsParameters,
};
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// Send back to the client the params
callback({ params });
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// Set transport to producerTransport or consumerTransport
return transport;
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} catch (error) {
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console.log(`ERROR | createWebRtcTransportLayer | callId ${socketDetails[socket.id]} | ${error.message}`);
callback({ params: { error } });
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}
}