Merge pull request 'Update 'app.js'' (#33) from sergiu-patch-1 into develop

Reviewed-on: #33
Reviewed-by: bmamihai <mihai.bozieru@safemobile.com>

.dockerignore

@@ -0,0 +1,2 @@
+node_modules
+doc

.env

@@ -1,3 +1,7 @@
 PORT=3000
 IP=0.0.0.0 #	Listening IPv4 or IPv6.
 ANNOUNCED_IP=185.8.154.190 # Announced IPv4 or IPv6 (useful when running mediasoup behind NAT with private IP).
+RTC_MIN_PORT=2000
+RTC_MAX_PORT=2020
+SERVER_CERT="./server/ssl/cert.pem"
+SERVER_KEY="./server/ssl/key.pem"

.gitignore

@@ -1 +1,2 @@
 /node_modules
+/dist

Dockerfile

@@ -1,11 +1,25 @@
-FROM ubuntu
+FROM ubuntu:22.04
+
+WORKDIR /app
+
 RUN apt-get update && \
     apt-get install -y build-essential pip net-tools iputils-ping iproute2 curl
 
-RUN curl -fsSL https://deb.nodesource.com/setup_16.x | bash -
+RUN curl -fsSL https://deb.nodesource.com/setup_18.x | bash -
 RUN apt-get install -y nodejs
-RUN npm install -g watchify
 
-EXPOSE 3000
+COPY . /app/
-EXPOSE 2000-2020
+
-EXPOSE 10000-10100
+RUN npm install
+
+EXPOSE 3000/tcp
+EXPOSE 2000-2200/udp
+
+CMD node app.js
+
+#docker build -t linx-video .
+# docker run -it -d --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video
+#Run under host network
+# docker run -it -d --network host --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video
+#https://docs.docker.com/config/containers/resource_constraints/
+#docker run -it -d --network host --cpus="0.25" --memory="512m" --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video

README.md

@@ -1,5 +1,11 @@
 # Video server
 
+### Generating certificates
+
+##### To generate SSL certificates you must:
+1. Go to `/server/ssl`
+2. Execute `openssl req -newkey rsa:2048 -new -nodes -x509 -days 3650 -keyout key.pem -out cert.pem`
+
 
 ### Development
 
@@ -16,14 +22,20 @@
 2. Run the `npm start:prod` command to start the server in production mode.
 (To connect to the terminal, use `pm2 log video-server`)
 
----
+
+### Web client
 
 - The server will start by default on port 3000, and the ssl certificates will have to be configured
 - The web client can be accessed using the /sfu path
-ex: http://localhost:3000/sfu/?assetId=1&&accountId=1&producer=true&assetName=Adi&assetType=linx
+ex: https://HOST/sfu/?assetId=1&&accountId=1&producer=true&dest_asset_id=75&assetName=Adi
 assetId = asset id of the unit on which you are doing the test
 accountId = account id of the unit on which you are doing the test
 producer = it will always be true because you are the producer
 (it's possible to put false, but then you have to have another client with producer true)
 assetName = asset name of the unit on which you are doing the test
-assetType = asset type of the unit on which you are doing the test
+dest_asset_id= the addressee with whom the call is made
+- To make a call using this client, you need a microphone and permission to use it
+- For any changes related to the client, the command `npm run watch' will have to be used to generate the bundle.js used by the web client
+
+### Demo project
+  The demo project used initially and then modified for our needs `https://github.com/jamalag/mediasoup2`

app.js

@@ -1,299 +1,597 @@
-import 'dotenv/config'
+require('dotenv').config();
-
+
-/**
+const express = require('express');
- * integrating mediasoup server with a node.js application
+const app = express();
- */
+const Server = require('socket.io');
-
+const path = require('node:path');
-/* Please follow mediasoup installation requirements */
+const fs = require('node:fs');
-/* https://mediasoup.org/documentation/v3/mediasoup/installation/ */
+let https;
-import express from 'express'
+try {
-const app = express()
+  https = require('node:https');
-
+} catch (err) {
-import https from 'httpolyglot'
+  console.log('https support is disabled!');
-import fs from 'fs'
+}
-import path from 'path'
+const mediasoup = require('mediasoup');
-const __dirname = path.resolve()
+
-
+let worker;
-import Server from 'socket.io'
+/**
-import mediasoup, { getSupportedRtpCapabilities } from 'mediasoup'
+ *
-
+ * videoCalls - Dictionary of Object(s)
-let worker
+ * '<callId>': {
-let router = {}
+ *  router: Router,
-let producerTransport
+ *  initiatorAudioProducer: Producer,
-let consumerTransport
+ *  initiatorVideoProducer: Producer,
-let producer
+ *  receiverVideoProducer: Producer,
-let consumer
+ *  receiverAudioProducer: Producer,
-
+ *  initiatorProducerTransport: Producer Transport,
-app.get('/', (_req, res) => {
+ *  receiverProducerTransport: Producer Transport,
-  res.send('Hello from mediasoup app!')
+ *  initiatorConsumerVideo: Consumer,
-})
+ *  initiatorConsumerAudio: Consumer,
-
+ *  initiatorConsumerTransport: Consumer Transport
-app.use('/sfu', express.static(path.join(__dirname, 'public')))
+ *  initiatorSocket
-
+ *  receiverSocket
-// SSL cert for HTTPS access
+ * }
-const options = {
+ *
-  key: fs.readFileSync('./server/ssl/key.pem', 'utf-8'),
+ **/
-  cert: fs.readFileSync('./server/ssl/cert.pem', 'utf-8')
+let videoCalls = {};
-}
+let socketDetails = {};
-
+
-const httpsServer = https.createServer(options, app)
+app.get('/', (_req, res) => {
-
+  res.send('OK');
-httpsServer.listen(process.env.PORT, () => {
+});
-  console.log('Listening on port:', process.env.PORT)
+
-})
+app.use('/sfu', express.static(path.join(__dirname, 'public')));
-
+
-const io = new Server(httpsServer)
+// SSL cert for HTTPS access
-
+const options = {
-// socket.io namespace (could represent a room?)
+  key: fs.readFileSync(process.env.SERVER_KEY, 'utf-8'),
-const peers = io.of('/mediasoup')
+  cert: fs.readFileSync(process.env.SERVER_CERT, 'utf-8'),
-
+};
-/**
+
- * Worker
+const httpsServer = https.createServer(options, app);
- * |-> Router(s)
+
- *     |-> Producer Transport(s)
+const io = new Server(httpsServer, {
- *         |-> Producer
+  allowEIO3: true,
- *     |-> Consumer Transport(s)
+  origins: ['*:*'],
- *         |-> Consumer 
+});
- **/
+
-
+httpsServer.listen(process.env.PORT, () => {
-const createWorker = async () => {
+  console.log('Video server listening on port:', process.env.PORT);
-  worker = await mediasoup.createWorker({
+});
-    rtcMinPort: 2000,
+
-    rtcMaxPort: 2020,
+const peers = io.of('/');
-  })
+
-  console.log(`[createWorker] worker pid ${worker.pid}`)
+const createWorker = async () => {
-
+  try {
-  worker.on('died', error => {
+    worker = await mediasoup.createWorker({
-    // This implies something serious happened, so kill the application
+      rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
-    console.error('mediasoup worker has died', error)
+      rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
-    setTimeout(() => process.exit(1), 2000) // exit in 2 seconds
+    });
-  })
+    console.log(`[createWorker] worker pid ${worker.pid}`);
-
+
-  return worker
+    worker.on('died', (error) => {
-}
+      // This implies something serious happened, so kill the application
-
+      console.error('mediasoup worker has died', error);
-// We create a Worker as soon as our application starts
+      setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
-worker = createWorker()
+    });
-
+    return worker;
-// This is an Array of RtpCapabilities
+  } catch (error) {
-// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
+    console.error(`[createWorker] | ERROR | error: ${error.message}`);
-// list of media codecs supported by mediasoup ...
+  }
-// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
+};
-const mediaCodecs = [
+
-  {
+// We create a Worker as soon as our application starts
-    kind: 'audio',
+worker = createWorker();
-    mimeType: 'audio/opus',
+
-    clockRate: 48000,
+// This is an Array of RtpCapabilities
-    channels: 2,
+// 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
-    kind: 'video',
+const mediaCodecs = [
-    mimeType: 'video/VP8',
+  {
-    clockRate: 90000,
+    kind: 'audio',
-    parameters: {
+    mimeType: 'audio/opus',
-      'x-google-start-bitrate': 1000,
+    clockRate: 48000,
-    },
+    channels: 2,
-  },
+  },
-]
+  {
-
+    kind: 'video',
-peers.on('connection', async socket => {
+    mimeType: 'video/VP8',
-  console.log('[connection] socketId:', socket.id)
+    clockRate: 90000,
-  socket.emit('connection-success', {
+    parameters: {
-    socketId: socket.id,
+      'x-google-start-bitrate': 1000,
-    existsProducer: producer ? true : false,
+    },
-  })
+    channels: 2,
-
+  },
-  socket.on('disconnect', () => {
+  {
-    // do some cleanup
+    kind: 'video',
-    console.log('peer disconnected')
+    mimeType: 'video/VP9',
-  })
+    clockRate: 90000,
-
+    parameters: {
-  socket.on('createRoom', async ({ callId }, callback) => {
+      'profile-id': 2,
-    console.log('[createRoom] callId', callId);
+      'x-google-start-bitrate': 1000,
-    console.log('Router length:', Object.keys(router).length);
+    },
-    if (router[callId] === undefined) {
+  },
-      // worker.createRouter(options)
+  {
-      // options = { mediaCodecs, appData }
+    kind: 'video',
-      // mediaCodecs -> defined above
+    mimeType: 'video/h264',
-      // appData -> custom application data - we are not supplying any
+    clockRate: 90000,
-      // none of the two are required
+    parameters: {
-      router[callId] = await worker.createRouter({ mediaCodecs })
+      'packetization-mode': 1,
-      console.log(`[createRoom] Router ID: ${router[callId].id}`)
+      'profile-level-id': '4d0032',
-    }
+      'level-asymmetry-allowed': 1,
-    
+      'x-google-start-bitrate': 1000,
-    getRtpCapabilities(callId, callback)
+    },
-  })
+  },
-
+  {
-  const getRtpCapabilities = (callId, callback) => {
+    kind: 'video',
-    const rtpCapabilities = router[callId].rtpCapabilities
+    mimeType: 'video/h264',
-
+    clockRate: 90000,
-    callback({ rtpCapabilities })
+    parameters: {
-  }
+      'packetization-mode': 1,
-
+      'profile-level-id': '42e01f',
-  // Client emits a request to create server side Transport
+      'level-asymmetry-allowed': 1,
-  // We need to differentiate between the producer and consumer transports
+      'x-google-start-bitrate': 1000,
-  socket.on('createWebRtcTransport', async ({ sender, callId }, callback) => {
+    },
-    console.log(`[createWebRtcTransport] Is this a sender request? ${sender} | callId ${callId}`)
+  },
-    // The client indicates if it is a producer or a consumer
+];
-    // if sender is true, indicates a producer else a consumer
+
-    if (sender)
+const closeCall = (callId) => {
-      producerTransport = await createWebRtcTransportLayer(callId, callback)
+  try {
-    else
+    if (callId && videoCalls[callId]) {
-      consumerTransport = await createWebRtcTransportLayer(callId, callback)
+      videoCalls[callId].receiverVideoProducer?.close();
-  })
+      videoCalls[callId].receiverAudioProducer?.close();
-
+      videoCalls[callId].initiatorConsumerVideo?.close();
-  // see client's socket.emit('transport-connect', ...)
+      videoCalls[callId].initiatorConsumerAudio?.close();
-  socket.on('transport-connect', async ({ dtlsParameters }) => {
+
-    console.log('[transport-connect] DTLS PARAMS... ', { dtlsParameters })
+      videoCalls[callId]?.initiatorConsumerTransport?.close();
-    await producerTransport.connect({ dtlsParameters })
+      videoCalls[callId]?.receiverProducerTransport?.close();
-  })
+      videoCalls[callId]?.router?.close();
-
+      delete videoCalls[callId];
-  // see client's socket.emit('transport-produce', ...)
+      console.log(`[closeCall] | callId: ${callId}`);
-  socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
+    }
-    // call produce based on the prameters from the client
+  } catch (error) {
-    producer = await producerTransport.produce({
+    console.error(`[closeCall] | ERROR | callId: ${callId} | error: ${error.message}`);
-      kind,
+  }
-      rtpParameters,
+};
-    })
+
-
+/*
-    console.log(`[transport-produce] Producer ID: ${producer.id} | kind: ${producer.kind}`)
+  - Handlers for WS events
-
+  - These are created only when we have a connection with a peer
-    producer.on('transportclose', () => {
+*/
-      console.log('transport for this producer closed', callId)
+peers.on('connection', async (socket) => {
-      
+  console.log('[connection] socketId:', socket.id);
-      // https://mediasoup.org/documentation/v3/mediasoup/api/#producer-close
+
-      producer.close()
+  // After making the connection successfully, we send the client a 'connection-success' event
-
+  socket.emit('connection-success', {
-      // https://mediasoup.org/documentation/v3/mediasoup/api/#router-close
+    socketId: socket.id,
-      router[callId].close()
+  });
-      delete router[callId]
+
-    })
+  // It is triggered when the peer is disconnected
-
+  socket.on('disconnect', () => {
-    // Send back to the client the Producer's id
+    const callId = socketDetails[socket.id];
-    callback({
+    console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
-      id: producer.id
+    delete socketDetails[socket.id];
-    })
+    closeCall(callId);
-  })
+  });
-
+
-  // see client's socket.emit('transport-recv-connect', ...)
+  /*
-  socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
+    - This event creates a room with the roomId and the callId sent
-    console.log(`[transport-recv-connect] DTLS PARAMS: ${dtlsParameters}`)
+    - It will return the rtpCapabilities of that room
-    await consumerTransport.connect({ dtlsParameters })
+    - If the room already exists, it will not create it, but will only return rtpCapabilities
-  })
+  */
-
+  socket.on('createRoom', async ({ callId }, callback) => {
-  socket.on('consume', async ({ rtpCapabilities, callId }, callback) => {
+    let callbackResponse = null;
-    try {
+    try {
-      console.log('consume', rtpCapabilities, callId);
+      // We can continue with the room creation process only if we have a callId
-      // check if the router can consume the specified producer
+      if (callId) {
-      if (router[callId].canConsume({
+        console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
-        producerId: producer.id,
+        if (!videoCalls[callId]) {
-        rtpCapabilities
+          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) };
-      })) {
+          console.log(`[createRoom] Generate Router ID: ${videoCalls[callId].router.id}`);
-        // transport can now consume and return a consumer
+          videoCalls[callId].receiverSocket = socket;
-        consumer = await consumerTransport.consume({
+        } else {
-          producerId: producer.id,
+          videoCalls[callId].initiatorSocket = socket;
-          rtpCapabilities,
+        }
-          paused: true,
+        socketDetails[socket.id] = callId;
-        })
+        // rtpCapabilities is set for callback
-
+        callbackResponse = {
-        consumer.on('transportclose', () => {
+          rtpCapabilities: videoCalls[callId].router.rtpCapabilities,
-          console.log('transport close from consumer', callId)
+        };
-          // closeRoom(callId)
+      } else {
-          delete router[callId]
+        console.log(`[createRoom] missing callId: ${callId}`);
-        })
+      }
-
+    } catch (error) {
-        consumer.on('producerclose', () => {
+      console.error(`[createRoom] | ERROR | callId: ${callId} | error: ${error.message}`);
-          console.log('producer of consumer closed', callId)
+    } finally {
-          
+      callback(callbackResponse);
-          // https://mediasoup.org/documentation/v3/mediasoup/api/#router-close
+    }
-          router[callId].close()
+  });
-          delete router[callId]
+
-        })
+  /*
-
+    - Client emits a request to create server side Transport
-        // from the consumer extract the following params
+    - Depending on the sender, a producer or consumer is created is created on that router
-        // to send back to the Client
+    - It will return parameters, these are required for the client to create the RecvTransport 
-        const params = {
+    from the client.
-          id: consumer.id,
+      - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
-          producerId: producer.id,
+      - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
-          kind: consumer.kind,
+  */
-          rtpParameters: consumer.rtpParameters,
+  socket.on('createWebRtcTransport', async ({ sender }, callback) => {
-        }
+    try {
-
+      const callId = socketDetails[socket.id];
-        // send the parameters to the client
+      console.log(`[createWebRtcTransport] socket ${socket.id} | sender ${sender} | callId ${callId}`);
-        callback({ params })
+      if (sender) {
-      }
+        if (!videoCalls[callId].receiverProducerTransport && !isInitiator(callId, socket.id)) {
-    } catch (error) {
+          videoCalls[callId].receiverProducerTransport = await createWebRtcTransportLayer(callId, callback);
-      console.log(error.message)
+        } else if (!videoCalls[callId].initiatorProducerTransport && isInitiator(callId, socket.id)) {
-      callback({
+          videoCalls[callId].initiatorProducerTransport = await createWebRtcTransportLayer(callId, callback);
-        params: {
+        } else {
-          error: error
+          console.log(`producerTransport has already been defined | callId ${callId}`);
-        }
+          callback(null);
-      })
+        }
-    }
+      } else if (!sender) {
-  })
+        if (!videoCalls[callId].receiverConsumerTransport && !isInitiator(callId, socket.id)) {
-
+          videoCalls[callId].receiverConsumerTransport = await createWebRtcTransportLayer(callId, callback);
-  socket.on('consumer-resume', async () => {
+        } else if (!videoCalls[callId].initiatorConsumerTransport && isInitiator(callId, socket.id)) {
-    console.log(`[consumer-resume]`)
+          videoCalls[callId].initiatorConsumerTransport = await createWebRtcTransportLayer(callId, callback);
-    await consumer.resume()
+        }
-  })
+      }
-})
+    } catch (error) {
-
+      console.error(
-const createWebRtcTransportLayer = async (callId, callback) => {
+        `[createWebRtcTransport] | ERROR | callId: ${socketDetails[socket.id]} | sender: ${sender} | error: ${
-  try {
+          error.message
-    console.log('[createWebRtcTransportLayer] callId', callId);
+        }`
-    // https://mediasoup.org/documentation/v3/mediasoup/api/#WebRtcTransportOptions
+      );
-    const webRtcTransport_options = {
+      callback(error);
-      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).
+  /*
-        }
+    - The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
-      ],
+    - The connection is made to the created transport
-      enableUdp: true,
+  */
-      enableTcp: true,
+  socket.on('transport-connect', async ({ dtlsParameters }) => {
-      preferUdp: true,
+    try {
-    }
+      const callId = socketDetails[socket.id];
-
+      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
-    // console.log('webRtcTransport_options', webRtcTransport_options);
+
-    // console.log('router', router, '| router[callId]', router[callId]);
+      console.log(`[transport-connect] socket ${socket.id} | callId ${callId}`);
-    
+
-    // https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
+      isInitiator(callId, socket.id)
-    let transport = await router[callId].createWebRtcTransport(webRtcTransport_options)
+        ? await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters })
-    console.log(`callId: ${callId} | transport id: ${transport.id}`)
+        : await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
-
+    } catch (error) {
-    transport.on('dtlsstatechange', dtlsState => {
+      console.error(`[transport-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
-      if (dtlsState === 'closed') {
+    }
-        transport.close()
+  });
-      }
+
-    })
+  /*
-
+    - The event sent by the client (PRODUCER) after successfully connecting to receiverProducerTransport/initiatorProducerTransport
-    transport.on('close', () => {
+    - For the router with the id callId, we make produce on receiverProducerTransport/initiatorProducerTransport
-      console.log('transport closed')
+    - Create the handler on producer at the 'transportclose' event
-    })
+    */
-
+  socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
-    const params = {
+    try {
-      id: transport.id,
+      const callId = socketDetails[socket.id];
-      iceParameters: transport.iceParameters,
+      if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
-      iceCandidates: transport.iceCandidates,
+
-      dtlsParameters: transport.dtlsParameters,
+      console.log(`[transport-produce] callId: ${callId} | kind: ${kind} | socket: ${socket.id}`);
-    }
+
-
+      if (kind === 'video') {
-    console.log('params', params);
+        if (!isInitiator(callId, socket.id)) {
-
+          videoCalls[callId].receiverVideoProducer = await videoCalls[callId].receiverProducerTransport.produce({
-    // send back to the client the following prameters
+            kind,
-    callback({
+            rtpParameters,
-      // https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
+          });
-      params
+
-    })
+          videoCalls[callId].receiverVideoProducer.on('transportclose', () => {
-
+            console.log('transport for this producer closed', callId);
-    return transport
+            closeCall(callId);
-
+          });
-  } catch (error) {
+
-    console.log('[createWebRtcTransportLayer] ERROR', JSON.stringify(error));
+          // Send back to the client the Producer's id
-    callback({
+          callback &&
-      params: {
+            callback({
-        error: error
+              id: videoCalls[callId].receiverVideoProducer.id,
-      }
+            });
-    })
+        } else {
-  }
+          videoCalls[callId].initiatorVideoProducer = await videoCalls[callId].initiatorProducerTransport.produce({
-}
+            kind,
+            rtpParameters,
+          });
+
+          videoCalls[callId].initiatorVideoProducer.on('transportclose', () => {
+            console.log('transport for this producer closed', callId);
+            closeCall(callId);
+          });
+
+          callback &&
+            callback({
+              id: videoCalls[callId].initiatorVideoProducer.id,
+            });
+        }
+      } else if (kind === 'audio') {
+        if (!isInitiator(callId, socket.id)) {
+          videoCalls[callId].receiverAudioProducer = await videoCalls[callId].receiverProducerTransport.produce({
+            kind,
+            rtpParameters,
+          });
+
+          videoCalls[callId].receiverAudioProducer.on('transportclose', () => {
+            console.log('transport for this producer closed', callId);
+            closeCall(callId);
+          });
+
+          // Send back to the client the Producer's id
+          callback &&
+            callback({
+              id: videoCalls[callId].receiverAudioProducer.id,
+            });
+        } else {
+          videoCalls[callId].initiatorAudioProducer = await videoCalls[callId].initiatorProducerTransport.produce({
+            kind,
+            rtpParameters,
+          });
+
+          videoCalls[callId].initiatorAudioProducer.on('transportclose', () => {
+            console.log('transport for this producer closed', callId);
+            closeCall(callId);
+          });
+
+          // Send back to the client the Producer's id
+          callback &&
+            callback({
+              id: videoCalls[callId].initiatorAudioProducer.id,
+            });
+        }
+      }
+
+      const socketToEmit = isInitiator(callId, socket.id)
+        ? videoCalls[callId].receiverSocket
+        : videoCalls[callId].initiatorSocket;
+
+      // callId - Id of the call
+      // kind - producer type: audio/video
+      socketToEmit?.emit('new-producer', { callId, kind });
+    } catch (error) {
+      console.error(`[transport-produce] | ERROR | callId: ${socketDetails[socket.id]} | error: ${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 ${socket.id} | callId ${callId}`);
+      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
+      // await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
+      if (!isInitiator(callId, socket.id)) {
+        await videoCalls[callId].receiverConsumerTransport.connect({ dtlsParameters });
+      } else if (isInitiator(callId, socket.id)) {
+        await videoCalls[callId].initiatorConsumerTransport.connect({ dtlsParameters });
+      }
+    } catch (error) {
+      console.error(`[transport-recv-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${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) => {
+    const callId = socketDetails[socket.id];
+    const socketId = socket.id;
+
+    console.log(`[consume] socket ${socketId} | callId: ${callId}`);
+
+    if (typeof rtpCapabilities === 'string') rtpCapabilities = JSON.parse(rtpCapabilities);
+
+    callback({
+      videoParams: await consumeVideo({ callId, socketId, rtpCapabilities }),
+      audioParams: await consumeAudio({ callId, socketId, rtpCapabilities }),
+    });
+  });
+
+  /*
+    - 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
+    - For the initiator we resume the initiatorConsumerAUDIO/VIDEO and for receiver the receiverConsumerAUDIO/VIDEO
+  */
+  socket.on('consumer-resume', () => {
+    try {
+      const callId = socketDetails[socket.id];
+      const isInitiatorValue = isInitiator(callId, socket.id);
+      console.log(`[consumer-resume] callId: ${callId} | isInitiator: ${isInitiatorValue}`);
+
+      const consumerVideo = isInitiatorValue
+        ? videoCalls[callId].initiatorConsumerVideo
+        : videoCalls[callId].receiverConsumerVideo;
+
+      const consumerAudio = isInitiatorValue
+        ? videoCalls[callId].initiatorConsumerAudio
+        : videoCalls[callId].receiverConsumerAudio;
+
+      consumerVideo?.resume();
+      consumerAudio?.resume();
+    } catch (error) {
+      console.error(
+        `[consumer-resume] | ERROR | callId: ${socketDetails[socket.id]} | isInitiator: ${isInitiator} | error: ${
+          error.message
+        }`
+      );
+    }
+  });
+
+  socket.on('close-producer', ({ callId, kind }) => {
+    try {
+      if (isInitiator(callId, socket.id)) {
+        console.log(`[close-producer] initiator --EMIT--> receiver | callId: ${callId} | kind: ${kind}`);
+        videoCalls[callId].receiverSocket.emit('close-producer', { callId, kind });
+      } else {
+        console.log(`[close-producer] receiver --EMIT--> initiator | callId: ${callId} | kind: ${kind}`);
+        videoCalls[callId].initiatorSocket.emit('close-producer', { callId, kind });
+      }
+    } catch (error) {
+      console.error(`[close-producer] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
+    }
+  });
+});
+
+const canConsume = ({ callId, producerId, rtpCapabilities }) => {
+  return !!videoCalls[callId].router.canConsume({
+    producerId,
+    rtpCapabilities,
+  });
+};
+
+const consumeVideo = async ({ callId, socketId, rtpCapabilities }) => {
+  // Handlers for consumer transport https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
+  if (isInitiator(callId, socketId) && videoCalls[callId].receiverVideoProducer) {
+    const producerId = videoCalls[callId].receiverVideoProducer.id;
+    if (!canConsume({ callId, producerId, rtpCapabilities })) return null;
+
+    videoCalls[callId].initiatorConsumerVideo = await videoCalls[callId].initiatorConsumerTransport.consume({
+      producerId,
+      rtpCapabilities,
+      paused: true,
+    });
+
+    return {
+      id: videoCalls[callId].initiatorConsumerVideo.id,
+      producerId,
+      kind: 'video',
+      rtpParameters: videoCalls[callId].initiatorConsumerVideo.rtpParameters,
+    };
+  } else if (videoCalls[callId].initiatorVideoProducer) {
+    const producerId = videoCalls[callId].initiatorVideoProducer.id;
+    if (!canConsume({ callId, producerId, rtpCapabilities })) return null;
+
+    videoCalls[callId].receiverConsumerVideo = await videoCalls[callId].receiverConsumerTransport.consume({
+      producerId,
+      rtpCapabilities,
+      paused: true,
+    });
+
+    return {
+      id: videoCalls[callId].receiverConsumerVideo.id,
+      producerId,
+      kind: 'video',
+      rtpParameters: videoCalls[callId].receiverConsumerVideo.rtpParameters,
+    };
+  } else {
+    return null;
+  }
+};
+
+const consumeAudio = async ({ callId, socketId, rtpCapabilities }) => {
+  try {
+    // Handlers for consumer transport https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
+    if (isInitiator(callId, socketId) && videoCalls[callId].receiverAudioProducer) {
+      const producerId = videoCalls[callId].receiverAudioProducer.id;
+      if (!canConsume({ callId, producerId, rtpCapabilities })) return null;
+
+      videoCalls[callId].initiatorConsumerAudio = await videoCalls[callId].initiatorConsumerTransport.consume({
+        producerId,
+        rtpCapabilities,
+        paused: true,
+      });
+
+      return {
+        id: videoCalls[callId].initiatorConsumerAudio.id,
+        producerId,
+        kind: 'audio',
+        rtpParameters: videoCalls[callId].initiatorConsumerAudio.rtpParameters,
+      };
+    } else if (videoCalls[callId].initiatorAudioProducer) {
+      const producerId = videoCalls[callId].initiatorAudioProducer.id;
+      if (!canConsume({ callId, producerId, rtpCapabilities })) return null;
+
+      videoCalls[callId].receiverConsumerAudio = await videoCalls[callId].receiverConsumerTransport.consume({
+        producerId,
+        rtpCapabilities,
+        paused: true,
+      });
+
+      return {
+        id: videoCalls[callId].receiverConsumerAudio.id,
+        producerId,
+        kind: 'audio',
+        rtpParameters: videoCalls[callId].receiverConsumerAudio.rtpParameters,
+      };
+    } else {
+      return null;
+    }
+  } catch (error) {
+    console.error(`[consumeAudio] | ERROR | error: ${error}`);
+  }
+};
+
+const isInitiator = (callId, socketId) => {
+  return videoCalls[callId]?.initiatorSocket?.id === socketId;
+};
+
+/*
+  - 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) => {
+  try {
+    console.log(`[createWebRtcTransportLayer] callId: ${callId}`);
+    // 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).
+        },
+      ],
+      enableUdp: true,
+      enableTcp: true,
+      preferUdp: true,
+    };
+
+    // https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
+    let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options);
+
+    // Handler for when DTLS(Datagram Transport Layer Security) changes
+    transport.on('dtlsstatechange', (dtlsState) => {
+      console.log(`transport | dtlsstatechange | calldId ${callId} | dtlsState ${dtlsState}`);
+      if (dtlsState === 'closed') {
+        transport.close();
+      }
+    });
+
+    // Handler if the transport layer has closed (for various reasons)
+    transport.on('close', () => {
+      console.log(`transport | closed | calldId ${callId}`);
+    });
+
+    const params = {
+      id: transport.id,
+      iceParameters: transport.iceParameters,
+      iceCandidates: transport.iceCandidates,
+      dtlsParameters: transport.dtlsParameters,
+    };
+
+    // Send back to the client the params
+    callback({ params });
+
+    // Set transport to producerTransport or consumerTransport
+    return transport;
+  } catch (error) {
+    console.error(
+      `[createWebRtcTransportLayer] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`
+    );
+    callback({ params: { error } });
+  }
+};

build.sh

@@ -0,0 +1,75 @@
+#/!bin/bash
+## FUNCTIONS
+function getGitVersion(){
+  version=$(git describe)
+  count=$(echo ${version%%-*} | grep -o "\." | wc -l)
+  if (( $count > 1 )); then
+    version=${version%%-*}
+  elif (( $count == 0 ));then
+    echo -e "Error: Git version \"${version%%-*}\" not respecting Safemobile standard.\n Must be like 4.xx or 4.xx.xx"
+    version="0.0.0"
+  else
+    if [[ "$1" == "dev" ]];then
+      cleanprefix=${version#*-} # remove everything before `-` including `-`
+      cleansuffix=${cleanprefix%-*} # remove everything after `-` including `-`
+      version="${version%%-*}.${cleansuffix}"
+    else
+      version="${version%%-*}.0" # one `%` remove everything after last `-`, two `%%` remove everything after all `-`
+    fi
+  fi
+}
+
+function addVersionPm2(){
+  file_pkg="package.json"
+  key="  \"version\": \""
+
+  if [ -f "$file_pkg" ] && [ ! -z "$version" ]; then
+    versionApp="  \"version\": \"$version\","
+    sed -i "s|^.*$key.*|${versionApp//\//\\/}|g" $file_pkg
+    text=$(cat $file_pkg | grep -c "$version")
+    if [ $text -eq 0 ]; then
+      echo "Version couldn't be set"
+    else
+      echo "Version $version successfully applied to App"
+    fi
+  fi
+}
+
+## PREBUILD PROCESS
+# check dist dir to be present and empty
+if [ ! -d "dist" ]; then
+  ## MAKE DIR
+  mkdir "dist"
+  echo "Directory dist created."
+else
+  ## CLEANUP
+  rm -fr dist/*
+fi
+
+if [ -d "node_modules" ]; then
+  rm -fr node_modules
+fi
+
+# Install dependencies
+#npm install
+
+## PROJECT NEEDS
+echo "Building app... from $(git rev-parse --abbrev-ref HEAD)"
+#npm run-script build
+cp -r {.env,app.js,package.json,server,public,doc,Dockerfile} dist/
+#cp -r ./* dist/
+
+# Generate Git log
+dateString=$(date +"%Y%m%d-%H%M%S")
+git log --pretty=format:"%ad%x09%an%x09%s" --no-merges -20 > "dist/git-$dateString.log"
+
+# Get Git version control
+getGitVersion $1
+
+# Add version control for pm2
+cd dist
+addVersionPm2
+
+## POST BUILD
+
+cd -

package-lock.json

@@ -12,7 +12,6 @@
         "@types/express": "^4.17.13",
         "dotenv": "^16.0.1",
         "express": "^4.18.1",
-        "httpolyglot": "^0.1.2",
         "mediasoup": "^3.10.4",
         "mediasoup-client": "^3.6.54",
         "parcel": "^2.7.0",
@@ -1792,27 +1791,6 @@
         "node": ">=10"
       }
     },
-    "node_modules/@pm2/agent/node_modules/ws": {
-      "version": "7.4.6",
-      "resolved": "https://registry.npmjs.org/ws/-/ws-7.4.6.tgz",
-      "integrity": "sha512-YmhHDO4MzaDLB+M9ym/mDA5z0naX8j7SIlT8f8z+I0VtzsRbekxEutHSme7NPS2qE8StCYQNUnfWdXta/Yu85A==",
-      "dev": true,
-      "engines": {
-        "node": ">=8.3.0"
-      },
-      "peerDependencies": {
-        "bufferutil": "^4.0.1",
-        "utf-8-validate": "^5.0.2"
-      },
-      "peerDependenciesMeta": {
-        "bufferutil": {
-          "optional": true
-        },
-        "utf-8-validate": {
-          "optional": true
-        }
-      }
-    },
     "node_modules/@pm2/io": {
       "version": "5.0.0",
       "resolved": "https://registry.npmjs.org/@pm2/io/-/io-5.0.0.tgz",
@@ -2388,11 +2366,6 @@
       "integrity": "sha512-iAB+JbDEGXhyIUavoDl9WP/Jj106Kz9DEn1DPgYw5ruDn0e3Wgi3sKFm55sASdGBNOQB8F59d9qQ7deqrHA8wQ==",
       "dev": true
     },
-    "node_modules/async-limiter": {
-      "version": "1.0.1",
-      "resolved": "https://registry.npmjs.org/async-limiter/-/async-limiter-1.0.1.tgz",
-      "integrity": "sha512-csOlWGAcRFJaI6m+F2WKdnMKr4HhdhFVBk0H/QbJFMCr+uO2kwohwXQPxw/9OCxp05r5ghVBFSyioixx3gfkNQ=="
-    },
     "node_modules/async-listener": {
       "version": "0.6.10",
       "resolved": "https://registry.npmjs.org/async-listener/-/async-listener-0.6.10.tgz",
@@ -2455,9 +2428,9 @@
       }
     },
     "node_modules/base64-arraybuffer": {
-      "version": "0.1.5",
+      "version": "0.1.4",
-      "resolved": "https://registry.npmjs.org/base64-arraybuffer/-/base64-arraybuffer-0.1.5.tgz",
+      "resolved": "https://registry.npmjs.org/base64-arraybuffer/-/base64-arraybuffer-0.1.4.tgz",
-      "integrity": "sha512-437oANT9tP582zZMwSvZGy2nmSeAb8DW2me3y+Uv1Wp2Rulr8Mqlyrv3E7MLxmsiaPSMMDmiDVzgE+e8zlMx9g==",
+      "integrity": "sha512-a1eIFi4R9ySrbiMuyTGx5e92uRH5tQY6kArNcFaKBUleIoLjdjBg7Zxm3Mqm3Kmkf27HLR/1fnxX9q8GQ7Iavg==",
       "engines": {
         "node": ">= 0.6.0"
       }
@@ -2483,22 +2456,11 @@
       ]
     },
     "node_modules/base64id": {
-      "version": "1.0.0",
+      "version": "2.0.0",
-      "resolved": "https://registry.npmjs.org/base64id/-/base64id-1.0.0.tgz",
+      "resolved": "https://registry.npmjs.org/base64id/-/base64id-2.0.0.tgz",
-      "integrity": "sha512-rz8L+d/xByiB/vLVftPkyY215fqNrmasrcJsYkVcm4TgJNz+YXKrFaFAWibSaHkiKoSgMDCb+lipOIRQNGYesw==",
+      "integrity": "sha512-lGe34o6EHj9y3Kts9R4ZYs/Gr+6N7MCaMlIFA3F1R2O5/m7K06AxfSeO5530PEERE6/WyEg3lsuyw4GHlPZHog==",
       "engines": {
-        "node": ">= 0.4.0"
+        "node": "^4.5.0 || >= 5.9"
-      }
-    },
-    "node_modules/better-assert": {
-      "version": "1.0.2",
-      "resolved": "https://registry.npmjs.org/better-assert/-/better-assert-1.0.2.tgz",
-      "integrity": "sha512-bYeph2DFlpK1XmGs6fvlLRUN29QISM3GBuUwSFsMY2XRx4AvC0WNCS57j4c/xGrK2RS24C1w3YoBOsw9fT46tQ==",
-      "dependencies": {
-        "callsite": "1.0.0"
-      },
-      "engines": {
-        "node": "*"
       }
     },
     "node_modules/binary-extensions": {
@@ -2875,14 +2837,6 @@
         "url": "https://github.com/sponsors/ljharb"
       }
     },
-    "node_modules/callsite": {
-      "version": "1.0.0",
-      "resolved": "https://registry.npmjs.org/callsite/-/callsite-1.0.0.tgz",
-      "integrity": "sha512-0vdNRFXn5q+dtOqjfFtmtlI9N2eVZ7LMyEV2iKC5mEEFvSg/69Ml6b/WU2qF8W1nLRa0wiSrDT3Y5jOHZCwKPQ==",
-      "engines": {
-        "node": "*"
-      }
-    },
     "node_modules/callsites": {
       "version": "3.1.0",
       "resolved": "https://registry.npmjs.org/callsites/-/callsites-3.1.0.tgz",
@@ -3051,9 +3005,9 @@
       "integrity": "sha512-WZveuKPeKAG9qY+FkYDeADzdHyTYdIboXS59ixDeRJL5ZhxpqUnxSOwop4FQjMsiYm3/Or8cegVbpAHNA7pHxw=="
     },
     "node_modules/component-emitter": {
-      "version": "1.2.1",
+      "version": "1.3.0",
-      "resolved": "https://registry.npmjs.org/component-emitter/-/component-emitter-1.2.1.tgz",
+      "resolved": "https://registry.npmjs.org/component-emitter/-/component-emitter-1.3.0.tgz",
-      "integrity": "sha512-jPatnhd33viNplKjqXKRkGU345p263OIWzDL2wH3LGIGp5Kojo+uXizHmOADRvhGFFTnJqX3jBAKP6vvmSDKcA=="
+      "integrity": "sha512-Rd3se6QB+sO1TwqZjscQrurpEPIfO0/yYnSin6Q/rD3mOutHvUrCAhJub3r90uNb+SESBuE0QYoB90YdfatsRg=="
     },
     "node_modules/component-inherit": {
       "version": "0.0.3",
@@ -3629,36 +3583,36 @@
       }
     },
     "node_modules/engine.io": {
-      "version": "3.1.5",
+      "version": "3.6.0",
-      "resolved": "https://registry.npmjs.org/engine.io/-/engine.io-3.1.5.tgz",
+      "resolved": "https://registry.npmjs.org/engine.io/-/engine.io-3.6.0.tgz",
-      "integrity": "sha512-D06ivJkYxyRrcEe0bTpNnBQNgP9d3xog+qZlLbui8EsMr/DouQpf5o9FzJnWYHEYE0YsFHllUv2R1dkgYZXHcA==",
+      "integrity": "sha512-Kc8fo5bbg8F4a2f3HPHTEpGyq/IRIQpyeHu3H1ThR14XDD7VrLcsGBo16HUpahgp8YkHJDaU5gNxJZbuGcuueg==",
       "dependencies": {
         "accepts": "~1.3.4",
-        "base64id": "1.0.0",
+        "base64id": "2.0.0",
-        "cookie": "0.3.1",
+        "cookie": "~0.4.1",
-        "debug": "~3.1.0",
+        "debug": "~4.1.0",
-        "engine.io-parser": "~2.1.0",
+        "engine.io-parser": "~2.2.0",
-        "ws": "~3.3.1"
+        "ws": "~7.4.2"
       },
-      "optionalDependencies": {
+      "engines": {
-        "uws": "~9.14.0"
+        "node": ">=8.0.0"
       }
     },
     "node_modules/engine.io-client": {
-      "version": "3.1.6",
+      "version": "3.5.3",
-      "resolved": "https://registry.npmjs.org/engine.io-client/-/engine.io-client-3.1.6.tgz",
+      "resolved": "https://registry.npmjs.org/engine.io-client/-/engine.io-client-3.5.3.tgz",
-      "integrity": "sha512-hnuHsFluXnsKOndS4Hv6SvUrgdYx1pk2NqfaDMW+GWdgfU3+/V25Cj7I8a0x92idSpa5PIhJRKxPvp9mnoLsfg==",
+      "integrity": "sha512-qsgyc/CEhJ6cgMUwxRRtOndGVhIu5hpL5tR4umSpmX/MvkFoIxUTM7oFMDQumHNzlNLwSVy6qhstFPoWTf7dOw==",
       "dependencies": {
-        "component-emitter": "1.2.1",
+        "component-emitter": "~1.3.0",
         "component-inherit": "0.0.3",
         "debug": "~3.1.0",
-        "engine.io-parser": "~2.1.1",
+        "engine.io-parser": "~2.2.0",
         "has-cors": "1.1.0",
         "indexof": "0.0.1",
-        "parseqs": "0.0.5",
+        "parseqs": "0.0.6",
-        "parseuri": "0.0.5",
+        "parseuri": "0.0.6",
-        "ws": "~3.3.1",
+        "ws": "~7.4.2",
-        "xmlhttprequest-ssl": "~1.5.4",
+        "xmlhttprequest-ssl": "~1.6.2",
         "yeast": "0.1.2"
       }
     },
@@ -3671,33 +3625,39 @@
       }
     },
     "node_modules/engine.io-parser": {
-      "version": "2.1.3",
+      "version": "2.2.1",
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+      "version": "1.6.3",
-      "resolved": "https://registry.npmjs.org/xmlhttprequest-ssl/-/xmlhttprequest-ssl-1.5.5.tgz",
+      "resolved": "https://registry.npmjs.org/xmlhttprequest-ssl/-/xmlhttprequest-ssl-1.6.3.tgz",
-      "integrity": "sha512-/bFPLUgJrfGUL10AIv4Y7/CUt6so9CLtB/oFxQSHseSDNNCdC6vwwKEqwLN6wNPBg9YWXAiMu8jkf6RPRS/75Q=="
+      "integrity": "sha512-3XfeQE/wNkvrIktn2Kf0869fC0BN6UpydVasGIeSm2B1Llihf7/0UfZM+eCkOw3P7bP4+qPgqhm7ZoxuJtFU0Q=="
     },
     "xregexp": {
       "version": "2.0.0",

package.json

@@ -5,19 +5,17 @@
   "main": "app.js",
   "scripts": {
     "test": "echo \"Error: no test specified\" && exit 1",
-    "start:dev": "nodemon app.ts",
+    "start:dev": "nodemon app.js",
     "start:prod": "pm2 start ./app.js -n video-server",
     "watch": "watchify public/index.js -o public/bundle.js -v"
   },
   "keywords": [],
   "author": "",
   "license": "ISC",
-  "type": "module",
   "dependencies": {
     "@types/express": "^4.17.13",
     "dotenv": "^16.0.1",
     "express": "^4.18.1",
-    "httpolyglot": "^0.1.2",
     "mediasoup": "^3.10.4",
     "mediasoup-client": "^3.6.54",
     "parcel": "^2.7.0",

public/config.js

@@ -1,5 +1,4 @@
 module.exports = {
   hubAddress: 'https://hub.dev.linx.safemobile.com/',
-  mediasoupAddress: 'https://video.safemobile.org/mediasoup',
+  mediasoupAddress: 'https://testing.video.safemobile.org/',
-  // mediasoupAddress: 'http://localhost:3000/mediasoup',
 }

public/index.html

@@ -34,6 +34,9 @@
   <body>
     <body>
         <div id="video">
+            <legend>Client options:</legend>
+            <input type="checkbox" id="produceAudio" name="produceAudio">
+            <label for="produceAudio">Produce audio</label><br>
             <table>
                 <thead>
                     <th>Local Video</th>
@@ -43,12 +46,24 @@
                     <tr>
                         <td>
                             <div id="sharedBtns">
-                                <video id="localVideo" autoplay class="video" ></video>
+                                <video
+                                    id="localVideo"
+                                    class="video"
+                                    autoplay
+                                    muted
+                                    playsinline
+                                ></video>
                             </div>
                         </td>
                         <td>
                             <div id="sharedBtns">
-                                <video id="remoteVideo" autoplay class="video" ></video>
+                                <video
+                                    id="remoteVideo"
+                                    class="video"
+                                    autoplay
+                                    muted
+                                    playsinline
+                                ></video>
                             </div>
                         </td>
                     </tr>
@@ -60,34 +75,11 @@
                         </td>
                         <td>
                             <div id="sharedBtns">
-                                <button id="btnRecvSendTransport">Consume</button>
+                                <!-- <button id="btnRecvSendTransport">Consume</button> -->
+                                <button id="remoteSoundControl">Unmute</button>
                             </div>
                         </td>
                     </tr>
-                    <!-- <tr>
-                        <td colspan="2">
-                            <div id="sharedBtns">
-                                <button id="btnRtpCapabilities">2. Get Rtp Capabilities</button>
-                                <br />
-                                <button id="btnDevice">3. Create Device</button>
-                            </div>
-                        </td>
-                    </tr>
-                    <tr>
-                        <td>
-                            <div id="sharedBtns">
-                                <button id="btnCreateSendTransport">4. Create Send Transport</button>
-                                <br />
-                                <button id="btnConnectSendTransport">5. Connect Send Transport & Produce</button></td>
-                            </div>
-                        <td>
-                            <div id="sharedBtns">
-                                <button id="btnRecvSendTransport">6. Create Recv Transport</button>
-                                <br />
-                                <button id="btnConnectRecvTransport">7. Connect Recv Transport & Consume</button>
-                            </div>
-                        </td>
-                    </tr> -->
                 </tbody>
             </table>
             <div id="closeCallBtn">

public/index.js

@@ -10,147 +10,201 @@ const ASSET_NAME = urlParams.get('assetName') || null;
 const ASSET_TYPE = urlParams.get('assetType') || null;
 let callId = parseInt(urlParams.get('callId')) || null;
 const IS_PRODUCER = urlParams.get('producer') === 'true' ? true : false
+let remoteVideo = document.getElementById('remoteVideo')
+remoteVideo.defaultMuted = true
+let produceAudio = false
+
 console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId', callId, ' | IS_PRODUCER', IS_PRODUCER)
 
-let socket 
+console.log('🟩 config', config)
-hub = io(config.hubAddress)
 
-const connectToMediasoup = () => {
+produceAudioSelector = document.getElementById('produceAudio');
-
+produceAudioSelector.addEventListener('change', e => {
-  socket = io(config.mediasoupAddress, {
+    if(e.target.checked) {
-    reconnection: true,
+      produceAudio = true
-    reconnectionDelay: 1000,
+      console.log('produce audio');
-    reconnectionDelayMax : 5000,
+    } else {
-    reconnectionAttempts: Infinity
+      produceAudio = false
-  })
-
-  socket.on('connection-success', ({ _socketId, existsProducer }) => {
-    console.log(`[MEDIA] ${config.mediasoupAddress} | connected: ${socket.connected} | existsProducer: ${existsProducer}`)
-    if (!IS_PRODUCER && existsProducer && consumer === undefined) { 
-      goConnect()
-      // document.getElementById('btnRecvSendTransport').click();
     }
-    if (IS_PRODUCER && urlParams.get('testing') === 'true') { getLocalStream() }
+});
-  })
-}
-
-if (IS_PRODUCER === true) {
-  hub.on('connect', async () => {
-    console.log(`[HUB] ${config.hubAddress} | connected: ${hub.connected}`)
-    connectToMediasoup()
-
-    hub.emit(
-      'ars',
-      JSON.stringify({
-        ars: true,
-        asset_id: ASSET_ID,
-        account_id: ACCOUNT_ID,
-      })
-    )
-  
-    hub.on('video', (data) => {
-      const parsedData = JSON.parse(data);
-
-      if (parsedData.type === 'notify-request') {
-      console.log('video', parsedData)
-      originAssetId = parsedData.origin_asset_id;
-      // originAssetName = parsedData.origin_asset_name;
-      // originAssetTypeName = parsedData.origin_asset_type_name;
-      callId = parsedData.video_call_id;
-  
-        console.log('[VIDEO] notify-request | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
-        getLocalStream()
-      }
-
-      if (parsedData.type === 'notify-end') {
-        console.log('[VIDEO] notify-end | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
-        resetCallSettings()
-      }
-    })
-  })
-
-  hub.on('connect_error', (error) => {
-    console.log('connect_error', error);
-  });
-  
-  hub.on('connection', () => {
-    console.log('connection')
-  })
-  
-  hub.on('disconnect', () => {
-    console.log('disconnect')
-  })
-} else {
-  connectToMediasoup()
-}
 
+let socket, hub
 let device
 let rtpCapabilities
 let producerTransport
 let consumerTransport
-let producer
+let producerVideo
+let producerAudio
 let consumer
 let originAssetId
-// let originAssetName = 'Adi'
+let consumerVideo // local consumer video(consumer not transport)
-// let originAssetTypeName = 'linx'
+let consumerAudio // local consumer audio(consumer not transport)
+
+const remoteSoundControl = document.getElementById('remoteSoundControl');
+
+remoteSoundControl.addEventListener('click', function handleClick() {
+  console.log('remoteSoundControl.textContent', remoteSoundControl.textContent);
+  if (remoteSoundControl.textContent === 'Unmute') {
+    remoteVideo.muted = false
+    remoteSoundControl.textContent = 'Mute';
+  } else {
+    remoteVideo.muted = true
+    remoteSoundControl.textContent = 'Unmute';
+  }
+});
 
 // https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerOptions
 // https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
-let params = {
+let videoParams = {
-  // mediasoup params
   encodings: [
-    {
+    { scaleResolutionDownBy: 4, maxBitrate: 500000 },
-      rid: 'r0',
+    { scaleResolutionDownBy: 2, maxBitrate: 1000000 },
-      maxBitrate: 100000,
+    { scaleResolutionDownBy: 1, maxBitrate: 5000000 },
-      scalabilityMode: 'S1T3',
+    { scalabilityMode: 'S3T3_KEY' }
-    },
-    {
-      rid: 'r1',
-      maxBitrate: 300000,
-      scalabilityMode: 'S1T3',
-    },
-    {
-      rid: 'r2',
-      maxBitrate: 900000,
-      scalabilityMode: 'S1T3',
-    },
   ],
-  // https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerCodecOptions
   codecOptions: {
     videoGoogleStartBitrate: 1000
   }
 }
 
-const streamSuccess = (stream) => {
+let audioParams = {
-  console.log('[streamSuccess]');
+  codecOptions :
-  localVideo.srcObject = stream
+  {
-  const track = stream.getVideoTracks()[0]
+    opusStereo : true,
-  params = {
+    opusDtx    : true
-    track,
-    ...params
   }
+}
+
+setTimeout(() => {
+  hub = io(config.hubAddress)
+  
+  const connectToMediasoup = () => {
+  
+    socket = io(config.mediasoupAddress, {
+      reconnection: true,
+      reconnectionDelay: 1000,
+      reconnectionDelayMax : 5000,
+      reconnectionAttempts: Infinity
+    })
+  
+    socket.on('connection-success', ({ _socketId, existsProducer }) => {
+      console.log(`[MEDIA] ${config.mediasoupAddress} | connected: ${socket.connected} | existsProducer: ${existsProducer}`)
+      if (!IS_PRODUCER && existsProducer && consumer === undefined) { 
+        goConnect()
+      }
+      if (IS_PRODUCER && urlParams.get('testing') === 'true') { getLocalStream() }
+    })
+
+    socket.on('new-producer', ({ callId, kind }) => {
+      console.log(`🟢 new-producer | callId: ${callId} | kind: ${kind} | Ready to consume`);
+      connectRecvTransport();
+    })
+
+    socket.on('close-producer', ({ callId, kind }) => {
+      console.log(`🔴 close-producer | callId: ${callId} | kind: ${kind}`);
+      if (kind === 'video') {
+        consumerVideo.close()
+        remoteVideo.srcObject = null
+      }
+      else if (kind === 'audio') consumerAudio.close()
+    })
+  }
+  
+  if (IS_PRODUCER === true) {
+    hub.on('connect', async () => {
+      console.log(`[HUB]! ${config.hubAddress} | connected: ${hub.connected}`)
+      connectToMediasoup()
+  
+      hub.emit(
+        'ars',
+        JSON.stringify({
+          ars: true,
+          asset_id: ASSET_ID,
+          account_id: ACCOUNT_ID,
+        })
+      )
+    
+      hub.on('video', (data) => {
+        const parsedData = JSON.parse(data);
+  
+        if (parsedData.type === 'notify-request') {
+        console.log('video', parsedData)
+        originAssetId = parsedData.origin_asset_id;
+        // originAssetName = parsedData.origin_asset_name;
+        // originAssetTypeName = parsedData.origin_asset_type_name;
+        callId = parsedData.video_call_id;
+    
+          console.log('[VIDEO] notify-request | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
+          getLocalStream()
+        }
+  
+        if (parsedData.type === 'notify-end') {
+          console.log('[VIDEO] notify-end | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
+          resetCallSettings()
+        }
+      })
+    })
+  
+    hub.on('connect_error', (error) => {
+      console.log('connect_error', error);
+    });
+    
+    hub.on('connection', () => {
+      console.log('connection')
+    })
+    
+    hub.on('disconnect', () => {
+      console.log('disconnect')
+    })
+  } else {
+    connectToMediasoup()
+  }
+
+}, 1600);
+
+const streamSuccess = (stream) => {
+  console.log('[streamSuccess] device', device);
+  localVideo.srcObject = stream
+  console.log('stream', stream);
+  const videoTrack = stream.getVideoTracks()[0]
+  const audioTrack = stream.getAudioTracks()[0]
+
+  videoParams = {
+    track: videoTrack,
+    ...videoParams
+  }
+
+  audioParams = {
+    track: audioTrack,
+    ...audioParams
+  }
+
+  console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
   goConnect()
 }
 
 const getLocalStream = () => {
   console.log('[getLocalStream]');
   navigator.mediaDevices.getUserMedia({
-    audio: false,
+    audio: produceAudio ? true : false,
     video: {
-      width: {
+      qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
-        min: 640,
+      vga  : { width: { ideal: 640 }, height: { ideal: 480 } },
-        max: 1920,
+      hd   : { width: { ideal: 1280 }, height: { ideal: 720 } }
-      },
-      height: {
-        min: 400,
-        max: 1080,
-      }
     }
   })
   .then(streamSuccess)
   .catch(error => {
     console.log(error.message)
   })
+
+  navigator.permissions.query(
+    { name: 'microphone' }
+  ).then((permissionStatus) =>{
+    console.log('🟨 [PERMISSION] permissionStatus', permissionStatus); // granted, denied, prompt
+    // It will block the code from execution and display "Permission denied" if we don't have microphone permissions
+  })
+
 }
 
 const goConnect = () => {
@@ -167,7 +221,6 @@ const goCreateTransport = () => {
 // server side to send/recive media
 const createDevice = async () => {
   try {
-    console.log('[createDevice]');
     device = new mediasoupClient.Device()
 
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@@ -178,7 +231,8 @@ const createDevice = async () => {
     })
 
     console.log('Device RTP Capabilities', device.rtpCapabilities)
-
+    console.log('[createDevice] device', device);
+    
     // once the device loads, create transport
     goCreateTransport()
 
@@ -207,18 +261,20 @@ const getRtpCapabilities = () => {
 }
 
 const createSendTransport = () => {
+  console.log('[createSendTransport');
   // see server's socket.on('createWebRtcTransport', sender?, ...)
   // this is a call from Producer, so sender = true
-  socket.emit('createWebRtcTransport', { sender: true, callId }, ({ params }) => {
+  socket.emit('createWebRtcTransport', { sender: true }, (value) => {
+
+    console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
+
+    const params = value.params;
     // The server sends back params needed 
     // to create Send Transport on the client side
     if (params.error) {
       console.log(params.error)
       return
     }
-
-    console.log(params)
-
     // creates a new WebRTC Transport to send media
     // based on the server's producer transport params
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
@@ -244,10 +300,10 @@ const createSendTransport = () => {
     })
 
     producerTransport.on('produce', async (parameters, callback, errback) => {
-      console.log(parameters)
+      console.log('[produce] parameters', parameters)
 
       try {
-        // tell the server to create a Producer
+        // Tell the server to create a Producer
         // with the following parameters and produce
         // and expect back a server side producer id
         // see server's socket.on('transport-produce', ...)
@@ -270,21 +326,45 @@ const createSendTransport = () => {
 }
 
 const connectSendTransport = async () => {
-  // we now call produce() to instruct the producer transport
+
+  console.log('[connectSendTransport] producerTransport');
+
+  // We now call produce() to instruct the producer transport
   // to send media to the Router
   // https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
   // this action will trigger the 'connect' and 'produce' events above
-  producer = await producerTransport.produce(params)
+  
+  // Produce video
+  let producerVideoHandler = await producerTransport.produce(videoParams)
+  console.log('videoParams', videoParams);
+  console.log('producerVideo', producerVideo);
 
-  producer.on('trackended', () => {
+  producerVideoHandler.on('trackended', () => {
     console.log('track ended')
     // close video track
-  })
+  })  
 
-  producer.on('transportclose', () => {
+  producerVideoHandler.on('transportclose', () => {
     console.log('transport ended')
     // close video track
-  })
+  })    
+  
+  // Produce audio
+  if (produceAudio) {
+    let producerAudioHandler = await producerTransport.produce(audioParams)
+    console.log('audioParams', audioParams);
+    console.log('producerAudio', producerAudio);
+  
+    producerAudioHandler.on('trackended', () => {
+      console.log('track ended')
+      // close audio track
+    })
+  
+    producerAudioHandler.on('transportclose', () => {
+      console.log('transport ended')
+      // close audio track
+    })
+  }
 
   const answer = {
     origin_asset_id: ASSET_ID,
@@ -294,7 +374,7 @@ const connectSendTransport = async () => {
     origin_asset_type_name: ASSET_TYPE,
     origin_asset_name: ASSET_NAME,
     video_call_id: callId,
-    answer: 'accepted', // answer: 'rejected'
+    answer: 'accepted', // answer: accepted/rejected
   };
   console.log('SEND answer', answer);
 
@@ -306,11 +386,13 @@ const connectSendTransport = async () => {
   // Enable Close call button
   const closeCallBtn = document.getElementById('btnCloseCall');
   closeCallBtn.removeAttribute('disabled');
+
+  createRecvTransport();
 }
 
 const createRecvTransport = async () => {
   console.log('createRecvTransport');
-  // see server's socket.on('consume', sender?, ...)
+  // See server's socket.on('consume', sender?, ...)
   // this is a call from Consumer, so sender = false
   await socket.emit('createWebRtcTransport', { sender: false, callId }, ({ params }) => {
     // The server sends back params needed 
@@ -320,15 +402,15 @@ const createRecvTransport = async () => {
       return
     }
 
-    console.log(params)
+    console.log('[createRecvTransport] params', params)
 
-    // creates a new WebRTC Transport to receive media
+    // Creates a new WebRTC Transport to receive media
     // based on server's consumer transport params
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-createRecvTransport
     consumerTransport = device.createRecvTransport(params)
 
     // https://mediasoup.org/documentation/v3/communication-between-client-and-server/#producing-media
-    // this event is raised when a first call to transport.produce() is made
+    // This event is raised when a first call to transport.produce() is made
     // see connectRecvTransport() below
     consumerTransport.on('connect', async ({ dtlsParameters }, callback, errback) => {
       try {
@@ -345,7 +427,8 @@ const createRecvTransport = async () => {
         errback(error)
       }
     })
-    connectRecvTransport()
+    // We call it in new-rpoducer, we don't need it here anymore
+    // connectRecvTransport()
   })
 }
 
@@ -353,7 +436,8 @@ const resetCallSettings = () => {
   localVideo.srcObject = null
   remoteVideo.srcObject = null
   consumer = null
-  producer = null
+  producerVideo = null
+  producerAudio = null
   producerTransport = null
   consumerTransport = null
   device = undefined
@@ -361,40 +445,97 @@ const resetCallSettings = () => {
 
 const connectRecvTransport = async () => {
   console.log('connectRecvTransport');
-  // for consumer, we need to tell the server first
+  // For consumer, we need to tell the server first
   // to create a consumer based on the rtpCapabilities and consume
   // if the router can consume, it will send back a set of params as below
   await socket.emit('consume', {
     rtpCapabilities: device.rtpCapabilities,
     callId
-  }, async ({ params }) => {
+  }, async ({videoParams, audioParams}) => {
-    if (params.error) {
+    console.log(`[consume] 🟩 videoParams`, videoParams)
-      console.log('Cannot Consume')
+    console.log(`[consume] 🟩 audioParams`, audioParams)
-      return
+    console.log('[consume] 🟩 consumerTransport', consumerTransport)
-    }
-
-    // then consume with the local consumer transport
-    // which creates a consumer
-    consumer = await consumerTransport.consume({
-      id: params.id,
-      producerId: params.producerId,
-      kind: params.kind,
-      rtpParameters: params.rtpParameters
-    })
-
-    // destructure and retrieve the video track from the producer
-    const { track } = consumer
 
     let stream = new MediaStream()
-    stream.addTrack(track)
-    // stream.removeTrack(track)
-    remoteVideo.srcObject = stream
-    socket.emit('consumer-resume')
-    console.log('consumer', consumer);
 
+    // Maybe the unit does not produce video or audio, so we must only consume what is produced
+    if (videoParams) {
+      console.log('❗ Have VIDEO stream to consume');
+      stream.addTrack(await getVideoTrask(videoParams))
+    } else {
+      console.log('❗ Don\'t have VIDEO stream to consume');
+    }
+    
+    if (audioParams) {
+      console.log('❗ Have AUDIO stream to consume');
+      let audioTrack = await getAudioTrask(audioParams)
+      stream.addTrack(audioTrack)
+    } else {
+      console.log('❗ Don\'t have AUDIO stream to consume');
+    }
+
+    socket.emit('consumer-resume')
+
+    remoteVideo.srcObject = stream
+    remoteVideo.setAttribute('autoplay', true)
+
+    remoteVideo.play()
+    .then(() => {
+      console.log('remoteVideo PLAY')
+    })
+    .catch((error) => {
+      console.error(`remoteVideo PLAY ERROR | ${error.message}`)
+    })
   })
 }
 
+const getVideoTrask = async (videoParams) => {
+  consumerVideo = await consumerTransport.consume({
+    id: videoParams.id,
+    producerId: videoParams.producerId,
+    kind: videoParams.kind,
+    rtpParameters: videoParams.rtpParameters
+  }) 
+
+  return consumerVideo.track
+}
+
+const getAudioTrask = async (audioParams) => {
+  consumerAudio = await consumerTransport.consume({
+    id: audioParams.id,
+    producerId: audioParams.producerId,
+    kind: audioParams.kind,
+    rtpParameters: audioParams.rtpParameters
+  })
+
+  consumerAudio.on('transportclose', () => {
+    console.log('transport closed so consumer closed')
+  })
+
+  const audioTrack = consumerAudio.track
+
+  audioTrack.applyConstraints({
+    audio: {
+      advanced: [
+        {
+          echoCancellation: {exact: true}
+        },
+        {
+          autoGainControl: {exact: true}
+        },
+        {
+          noiseSuppression: {exact: true}
+        }, 
+        {
+          highpassFilter: {exact: true}
+        }
+      ]
+    }
+  })
+
+  return audioTrack
+}
+
 const closeCall = () => {
   console.log('closeCall');
   
@@ -416,6 +557,30 @@ const closeCall = () => {
   resetCallSettings()
 }
 
+// const consume = async (kind) => {
+//   console.log(`[consume] kind: ${kind}`)
+//   console.log('createRecvTransport Consumer')
+//   await socket.emit('createWebRtcTransport', { sender: false, callId, dispatcher: true }, ({ params }) => {
+//     if (params.error) {
+//       console.log('createRecvTransport | createWebRtcTransport | Error', params.error)
+//       return
+//     }
+//     consumerTransport = device.createRecvTransport(params)
+//     consumerTransport.on('connect', async ({ dtlsParameters }, callback, errback) => {
+//       try {
+//         await socket.emit('transport-recv-connect', {
+//           dtlsParameters,
+//         })
+//         callback()
+//       } catch (error) {
+//         errback(error)
+//       }
+//     })
+    
+//     connectRecvTransport()
+//   })
+// }
+
 btnLocalVideo.addEventListener('click', getLocalStream)
-btnRecvSendTransport.addEventListener('click', goConnect)
+// btnRecvSendTransport.addEventListener('click', consume)
 btnCloseCall.addEventListener('click', closeCall)