Added logs

.dockerignore

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

.env

@@ -1,7 +1,3 @@
 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

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

Dockerfile

@@ -1,25 +1,11 @@
-FROM ubuntu:22.04
+FROM ubuntu
-
-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_18.x | bash -
+RUN curl -fsSL https://deb.nodesource.com/setup_16.x | bash -
 RUN apt-get install -y nodejs
+RUN npm install -g watchify
 
-COPY . /app/
+EXPOSE 3000
-
+EXPOSE 2000-2020
-RUN npm install
+EXPOSE 10000-10100
-
-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

@@ -22,20 +22,18 @@
 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: https://HOST/sfu/?assetId=1&&accountId=1&producer=true&dest_asset_id=75&assetName=Adi
+ex: http://localhost:3000/sfu/?assetId=1&&accountId=1&producer=true&assetName=Adi&assetType=linx
 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
-dest_asset_id= the addressee with whom the call is made
+assetType = asset type of the unit on which you are doing the test
-- 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

@@ -1,597 +1,399 @@
-require('dotenv').config();
+require('dotenv').config()
-
+
-const express = require('express');
+const express = require('express');
-const app = express();
+const app = express();
-const Server = require('socket.io');
+const Server = require('socket.io');
-const path = require('node:path');
+const path = require('node:path');
-const fs = require('node:fs');
+const fs = require('node:fs');
-let https;
+let https = require('https');
-try {
+try {
-  https = require('node:https');
+  https = require('node:https');
-} catch (err) {
+} catch (err) {
-  console.log('https support is disabled!');
+  console.log('https support is disabled!');
-}
+}
-const mediasoup = require('mediasoup');
+const mediasoup = require('mediasoup');
-
+
-let worker;
+let worker
-/**
+/**
- *
+ * videoCalls
- * videoCalls - Dictionary of Object(s)
+ * |-> Router
- * '<callId>': {
+ * |-> Producer
- *  router: Router,
+ * |-> Consumer 
- *  initiatorAudioProducer: Producer,
+ * |-> Producer Transport
- *  initiatorVideoProducer: Producer,
+ * |-> Consumer Transport
- *  receiverVideoProducer: Producer,
+ * 
- *  receiverAudioProducer: Producer,
+ * '<callId>': {
- *  initiatorProducerTransport: Producer Transport,
+ *  router: Router,
- *  receiverProducerTransport: Producer Transport,
+ *  producer: Producer,
- *  initiatorConsumerVideo: Consumer,
+ *  producerTransport: Producer Transport,
- *  initiatorConsumerAudio: Consumer,
+ *  consumer: Consumer,
- *  initiatorConsumerTransport: Consumer Transport
+ *  consumerTransport: Consumer Transport
- *  initiatorSocket
+ * }
- *  receiverSocket
+ * 
- * }
+ **/
- *
+let videoCalls = {}
- **/
+let socketDetails = {}
-let videoCalls = {};
+
-let socketDetails = {};
+app.get('/', (_req, res) => {
-
+  res.send('Hello from mediasoup app!')
-app.get('/', (_req, res) => {
+})
-  res.send('OK');
+
-});
+app.use('/sfu', express.static(path.join(__dirname, 'public')))
-
+
-app.use('/sfu', express.static(path.join(__dirname, 'public')));
+// SSL cert for HTTPS access
-
+const options = {
-// SSL cert for HTTPS access
+  key: fs.readFileSync('./server/ssl/key.pem', 'utf-8'),
-const options = {
+  cert: fs.readFileSync('./server/ssl/cert.pem', 'utf-8'),
-  key: fs.readFileSync(process.env.SERVER_KEY, 'utf-8'),
+}
-  cert: fs.readFileSync(process.env.SERVER_CERT, 'utf-8'),
+
-};
+const httpsServer = https.createServer(options, app);
-
+
-const httpsServer = https.createServer(options, app);
+const io = new Server(httpsServer, {
-
+  allowEIO3: true,
-const io = new Server(httpsServer, {
+  origins: ["*:*"],
-  allowEIO3: true,
+  // allowRequest: (req, next) => {
-  origins: ['*:*'],
+  //   console.log('req', req);
-});
+  //   next(null, true)
-
+  // }
-httpsServer.listen(process.env.PORT, () => {
+});
-  console.log('Video server listening on port:', process.env.PORT);
+// const io = new Server(server, { origins: '*:*', allowEIO3: true });
-});
+
-
+httpsServer.listen(process.env.PORT, () => {
-const peers = io.of('/');
+  console.log('Video server listening on port:', process.env.PORT)
-
+})
-const createWorker = async () => {
+
-  try {
+const peers = io.of('/')
-    worker = await mediasoup.createWorker({
+
-      rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
+const createWorker = async () => {
-      rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
+  try {
-    });
+    worker = await mediasoup.createWorker({
-    console.log(`[createWorker] worker pid ${worker.pid}`);
+      rtcMinPort: 2000,
-
+      rtcMaxPort: 2020,
-    worker.on('died', (error) => {
+    })
-      // This implies something serious happened, so kill the application
+    console.log(`[createWorker] worker pid ${worker.pid}`);
-      console.error('mediasoup worker has died', error);
+  
-      setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
+    worker.on('died', error => {
-    });
+      // This implies something serious happened, so kill the application
-    return worker;
+      console.error('mediasoup worker has died', error);
-  } catch (error) {
+      setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
-    console.error(`[createWorker] | ERROR | error: ${error.message}`);
+    })
-  }
+    return worker;
-};
+  } catch (error) {
-
+    console.log(`ERROR | createWorker | ${error.message}`);
-// We create a Worker as soon as our application starts
+  }
-worker = createWorker();
+}
-
+
-// This is an Array of RtpCapabilities
+// We create a Worker as soon as our application starts
-// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
+worker = createWorker()
-// list of media codecs supported by mediasoup ...
+
-// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
+// This is an Array of RtpCapabilities
-const mediaCodecs = [
+// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
-  {
+// list of media codecs supported by mediasoup ...
-    kind: 'audio',
+// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
-    mimeType: 'audio/opus',
+const mediaCodecs = [
-    clockRate: 48000,
+  {
-    channels: 2,
+    kind: 'audio',
-  },
+    mimeType: 'audio/opus',
-  {
+    clockRate: 48000,
-    kind: 'video',
+    channels: 2,
-    mimeType: 'video/VP8',
+  },
-    clockRate: 90000,
+  {
-    parameters: {
+    kind: 'video',
-      'x-google-start-bitrate': 1000,
+    mimeType: 'video/VP8',
-    },
+    clockRate: 90000,
-    channels: 2,
+    parameters: {
-  },
+      'x-google-start-bitrate': 1000,
-  {
+    },
-    kind: 'video',
+  },
-    mimeType: 'video/VP9',
+]
-    clockRate: 90000,
+
-    parameters: {
+const closeCall = (callId) => {
-      'profile-id': 2,
+  try {
-      'x-google-start-bitrate': 1000,
+    if (videoCalls[callId]) {
-    },
+      videoCalls[callId].producer?.close();
-  },
+      videoCalls[callId].consumer?.close();
-  {
+      videoCalls[callId]?.consumerTransport?.close();
-    kind: 'video',
+      videoCalls[callId]?.producerTransport?.close();
-    mimeType: 'video/h264',
+      videoCalls[callId]?.router?.close();
-    clockRate: 90000,
+      delete videoCalls[callId];
-    parameters: {
+    } else {
-      'packetization-mode': 1,
+      console.log(`The call with id ${callId} has already been deleted`);
-      'profile-level-id': '4d0032',
+    }
-      'level-asymmetry-allowed': 1,
+  } catch (error) {
-      'x-google-start-bitrate': 1000,
+    console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
-    },
+  }
-  },
+}
-  {
+
-    kind: 'video',
+const getRtpCapabilities = (callId, callback) => {
-    mimeType: 'video/h264',
+  try {
-    clockRate: 90000,
+    console.log('[getRtpCapabilities] callId', callId);
-    parameters: {
+    const rtpCapabilities = videoCalls[callId].router.rtpCapabilities;
-      'packetization-mode': 1,
+    callback({ rtpCapabilities });
-      'profile-level-id': '42e01f',
+  } catch (error) {
-      'level-asymmetry-allowed': 1,
+    console.log(`ERROR | getRtpCapabilities | callId ${callId} | ${error.message}`);
-      'x-google-start-bitrate': 1000,
+  }
-    },
+}
-  },
+
-];
+/*
-
+  - Handlers for WS events
-const closeCall = (callId) => {
+  - These are created only when we have a connection with a peer
-  try {
+*/
-    if (callId && videoCalls[callId]) {
+peers.on('connection', async socket => {
-      videoCalls[callId].receiverVideoProducer?.close();
+  console.log('[connection] socketId:', socket.id);
-      videoCalls[callId].receiverAudioProducer?.close();
+
-      videoCalls[callId].initiatorConsumerVideo?.close();
+  // After making the connection successfully, we send the client a 'connection-success' event
-      videoCalls[callId].initiatorConsumerAudio?.close();
+  socket.emit('connection-success', {
-
+    socketId: socket.id
-      videoCalls[callId]?.initiatorConsumerTransport?.close();
+  });
-      videoCalls[callId]?.receiverProducerTransport?.close();
+
-      videoCalls[callId]?.router?.close();
+  // It is triggered when the peer is disconnected
-      delete videoCalls[callId];
+  socket.on('disconnect', () => {
-      console.log(`[closeCall] | callId: ${callId}`);
+    const callId = socketDetails[socket.id];
-    }
+    console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
-  } catch (error) {
+    delete socketDetails[socket.id];
-    console.error(`[closeCall] | ERROR | callId: ${callId} | error: ${error.message}`);
+    closeCall(callId);
-  }
+  });
-};
+
-
+  /*
-/*
+    - This event creates a room with the roomId and the callId sent
-  - Handlers for WS events
+    - It will return the rtpCapabilities of that room
-  - These are created only when we have a connection with a peer
+    - If the room already exists, it will not create it, but will only return rtpCapabilities
-*/
+  */
-peers.on('connection', async (socket) => {
+  socket.on('createRoom', async ({ callId }, callback) => {
-  console.log('[connection] socketId:', socket.id);
+    try {
-
+      if (callId) {
-  // After making the connection successfully, we send the client a 'connection-success' event
+        console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
-  socket.emit('connection-success', {
+        if (!videoCalls[callId]) {
-    socketId: socket.id,
+          console.log('[createRoom] callId', callId);
-  });
+          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) }
-
+          console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
-  // It is triggered when the peer is disconnected
+        }
-  socket.on('disconnect', () => {
+        socketDetails[socket.id] = callId;
-    const callId = socketDetails[socket.id];
+        getRtpCapabilities(callId, callback);
-    console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
+      } else {
-    delete socketDetails[socket.id];
+        console.log(`[createRoom] missing callId ${callId}`);
-    closeCall(callId);
+      }
-  });
+    } catch (error) {
-
+      console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
-  /*
+    }
-    - 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
+  /*
-  */
+    - Client emits a request to create server side Transport
-  socket.on('createRoom', async ({ callId }, callback) => {
+    - Depending on the sender, producerTransport or consumerTransport is created on that router
-    let callbackResponse = null;
+    - It will return parameters, these are required for the client to create the RecvTransport 
-    try {
+    from the client.
-      // We can continue with the room creation process only if we have a callId
+      - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
-      if (callId) {
+      - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
-        console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
+  */
-        if (!videoCalls[callId]) {
+  socket.on('createWebRtcTransport', async ({ sender }, callback) => {
-          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) };
+    try {
-          console.log(`[createRoom] Generate Router ID: ${videoCalls[callId].router.id}`);
+      const callId = socketDetails[socket.id];
-          videoCalls[callId].receiverSocket = socket;
+      console.log(`[createWebRtcTransport] sender ${sender} | callId ${callId}`);
-        } else {
+      if (sender) {
-          videoCalls[callId].initiatorSocket = socket;
+        if (!videoCalls[callId].producerTransport) {
-        }
+          videoCalls[callId].producerTransport = await createWebRtcTransportLayer(callId, callback);
-        socketDetails[socket.id] = callId;
+        } else {
-        // rtpCapabilities is set for callback
+          console.log(`producerTransport has already been defined | callId ${callId}`);
-        callbackResponse = {
+        }
-          rtpCapabilities: videoCalls[callId].router.rtpCapabilities,
+      } else if (!sender) {
-        };
+        if (!videoCalls[callId].consumerTransport) {
-      } else {
+          videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
-        console.log(`[createRoom] missing callId: ${callId}`);
+        } else {
-      }
+          console.log(`consumerTransport has already been defined | callId ${callId}`);
-    } catch (error) {
+        }
-      console.error(`[createRoom] | ERROR | callId: ${callId} | error: ${error.message}`);
+      }
-    } finally {
+    } catch (error) {
-      callback(callbackResponse);
+      console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
-    }
+    }
-  });
+  });
-
+
-  /*
+  /*
-    - Client emits a request to create server side Transport
+    - The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
-    - Depending on the sender, a producer or consumer is created is created on that router
+    - The connection is made to the created transport
-    - It will return parameters, these are required for the client to create the RecvTransport 
+  */
-    from the client.
+  socket.on('transport-connect', async ({ dtlsParameters }) => {
-      - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
+    try {
-      - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
+      const callId = socketDetails[socket.id];
-  */
+      // console.log('🔴 typeof dtlsParameters', typeof dtlsParameters);
-  socket.on('createWebRtcTransport', async ({ sender }, callback) => {
+      // console.log('🟢 dtlsParameters', JSON.parse(dtlsParameters));
-    try {
+      // console.log('🟡 dtlsParameters', dtlsParameters);
-      const callId = socketDetails[socket.id];
+      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
-      console.log(`[createWebRtcTransport] socket ${socket.id} | sender ${sender} | callId ${callId}`);
+      console.log(`[transport-connect] socket.id ${socket.id} | callId ${callId}`);
-      if (sender) {
+      await videoCalls[callId].producerTransport.connect({ dtlsParameters });
-        if (!videoCalls[callId].receiverProducerTransport && !isInitiator(callId, socket.id)) {
+    } catch (error) {
-          videoCalls[callId].receiverProducerTransport = await createWebRtcTransportLayer(callId, callback);
+      console.log(`ERROR | transport-connect | callId ${socketDetails[socket.id]} | ${error.stack}`);
-        } else if (!videoCalls[callId].initiatorProducerTransport && isInitiator(callId, socket.id)) {
+    }
-          videoCalls[callId].initiatorProducerTransport = await createWebRtcTransportLayer(callId, callback);
+  });
-        } else {
+
-          console.log(`producerTransport has already been defined | callId ${callId}`);
+  /*
-          callback(null);
+    - The event sent by the client (PRODUCER) after successfully connecting to producerTransport
-        }
+    - For the router with the id callId, we make produce on producerTransport
-      } else if (!sender) {
+    - Create the handler on producer at the 'transportclose' event
-        if (!videoCalls[callId].receiverConsumerTransport && !isInitiator(callId, socket.id)) {
+  */
-          videoCalls[callId].receiverConsumerTransport = await createWebRtcTransportLayer(callId, callback);
+  socket.on('transport-produce', async ({ kind, rtpParameters, appData }) => {
-        } else if (!videoCalls[callId].initiatorConsumerTransport && isInitiator(callId, socket.id)) {
+    try {
-          videoCalls[callId].initiatorConsumerTransport = await createWebRtcTransportLayer(callId, callback);
+      const callId = socketDetails[socket.id];
-        }
+      console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
-      }
+      videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
-    } catch (error) {
+        kind,
-      console.error(
+        rtpParameters,
-        `[createWebRtcTransport] | ERROR | callId: ${socketDetails[socket.id]} | sender: ${sender} | error: ${
+      });
-          error.message
+      console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producer.id} | kind: ${videoCalls[callId].producer.kind}`);
-        }`
+  
-      );
+      videoCalls[callId].producer.on('transportclose', () => {
-      callback(error);
+        const callId = socketDetails[socket.id];
-    }
+        console.log('transport for this producer closed', callId)
-  });
+        closeCall(callId);
-
+      });
-  /*
+    } catch (error) {
-    - The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
+      console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
-    - The connection is made to the created transport
+    }
-  */
+  });
-  socket.on('transport-connect', async ({ dtlsParameters }) => {
+
-    try {
+  /*
-      const callId = socketDetails[socket.id];
+    - The client sends this event after successfully creating a createRecvTransport(AS CONSUMER)
-      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
+    - The connection is made to the created consumerTransport
-
+  */
-      console.log(`[transport-connect] socket ${socket.id} | callId ${callId}`);
+  socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
-
+    try {
-      isInitiator(callId, socket.id)
+      const callId = socketDetails[socket.id];
-        ? await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters })
+      console.log(`[transport-recv-connect] socket.id ${socket.id} | callId ${callId}`);
-        : await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
+      await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
-    } catch (error) {
+    } catch (error) {
-      console.error(`[transport-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
+      console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
-    }
+    }
-  });
+  })
-
+
-  /*
+  /*
-    - The event sent by the client (PRODUCER) after successfully connecting to receiverProducerTransport/initiatorProducerTransport
+    - The customer consumes after successfully connecting to consumerTransport
-    - For the router with the id callId, we make produce on receiverProducerTransport/initiatorProducerTransport
+    - The previous step was 'transport-recv-connect', and before that 'createWebRtcTransport'
-    - Create the handler on producer at the 'transportclose' event
+    - This event is only sent by the consumer
-    */
+    - The parameters that the consumer consumes are returned
-  socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
+      - The consumer does consumerTransport.consume(params)
-    try {
+  */
-      const callId = socketDetails[socket.id];
+  socket.on('consume', async ({ rtpCapabilities }, callback) => {
-      if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
+    try {
-
+      const callId = socketDetails[socket.id];
-      console.log(`[transport-produce] callId: ${callId} | kind: ${kind} | socket: ${socket.id}`);
+      console.log('[consume] callId', callId);
-
+
-      if (kind === 'video') {
+      // Check if the router can consume the specified producer
-        if (!isInitiator(callId, socket.id)) {
+      if (videoCalls[callId].router.canConsume({
-          videoCalls[callId].receiverVideoProducer = await videoCalls[callId].receiverProducerTransport.produce({
+        producerId: videoCalls[callId].producer.id,
-            kind,
+        rtpCapabilities
-            rtpParameters,
+      })) {
-          });
+        console.log('[consume] Can consume', callId);
-
+        // Transport can now consume and return a consumer
-          videoCalls[callId].receiverVideoProducer.on('transportclose', () => {
+        videoCalls[callId].consumer = await videoCalls[callId].consumerTransport.consume({
-            console.log('transport for this producer closed', callId);
+          producerId: videoCalls[callId].producer.id,
-            closeCall(callId);
+          rtpCapabilities,
-          });
+          paused: true,
-
+        });
-          // Send back to the client the Producer's id
+
-          callback &&
+        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
-            callback({
+        videoCalls[callId].consumer.on('transportclose', () => {
-              id: videoCalls[callId].receiverVideoProducer.id,
+          const callId = socketDetails[socket.id];
-            });
+          console.log('transport close from consumer', callId);
-        } else {
+          closeCall();
-          videoCalls[callId].initiatorVideoProducer = await videoCalls[callId].initiatorProducerTransport.produce({
+        });
-            kind,
+
-            rtpParameters,
+        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
-          });
+        videoCalls[callId].consumer.on('producerclose', () => {
-
+          const callId = socketDetails[socket.id];
-          videoCalls[callId].initiatorVideoProducer.on('transportclose', () => {
+          console.log('producer of consumer closed', callId);
-            console.log('transport for this producer closed', callId);
+          closeCall();
-            closeCall(callId);
+        });
-          });
+
-
+        // From the consumer extract the following params to send back to the Client
-          callback &&
+        const params = {
-            callback({
+          id: videoCalls[callId].consumer.id,
-              id: videoCalls[callId].initiatorVideoProducer.id,
+          producerId: videoCalls[callId].producer.id,
-            });
+          kind: videoCalls[callId].consumer.kind,
-        }
+          rtpParameters: videoCalls[callId].consumer.rtpParameters,
-      } else if (kind === 'audio') {
+        };
-        if (!isInitiator(callId, socket.id)) {
+
-          videoCalls[callId].receiverAudioProducer = await videoCalls[callId].receiverProducerTransport.produce({
+        // Send the parameters to the client
-            kind,
+        callback({ params });
-            rtpParameters,
+      } else {
-          });
+        console.log(`[canConsume] Can't consume | callId ${callId}`);
-
+      }
-          videoCalls[callId].receiverAudioProducer.on('transportclose', () => {
+    } catch (error) {
-            console.log('transport for this producer closed', callId);
+      console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
-            closeCall(callId);
+      callback({ params: { error } });
-          });
+    }
-
+  });
-          // Send back to the client the Producer's id
+
-          callback &&
+  /*
-            callback({
+    - Event sent by the consumer after consuming to resume the pause
-              id: videoCalls[callId].receiverAudioProducer.id,
+    - When consuming on consumerTransport, it is initially done with paused: true, here we will resume 
-            });
+  */
-        } else {
+  socket.on('consumer-resume', async () => {
-          videoCalls[callId].initiatorAudioProducer = await videoCalls[callId].initiatorProducerTransport.produce({
+    try {
-            kind,
+      const callId = socketDetails[socket.id];
-            rtpParameters,
+      console.log(`[consumer-resume] callId ${callId}`)
-          });
+      await videoCalls[callId].consumer.resume();
-
+    } catch (error) {
-          videoCalls[callId].initiatorAudioProducer.on('transportclose', () => {
+      console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
-            console.log('transport for this producer closed', callId);
+    }
-            closeCall(callId);
+  });
-          });
+});
-
+
-          // Send back to the client the Producer's id
+/*
-          callback &&
+  - Called from at event 'createWebRtcTransport' and assigned to the consumer or producer transport
-            callback({
+  - It will return parameters, these are required for the client to create the RecvTransport 
-              id: videoCalls[callId].initiatorAudioProducer.id,
+    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) => {
-      const socketToEmit = isInitiator(callId, socket.id)
+  try {
-        ? videoCalls[callId].receiverSocket
+    console.log('[createWebRtcTransportLayer] callId', callId);
-        : videoCalls[callId].initiatorSocket;
+    // https://mediasoup.org/documentation/v3/mediasoup/api/#WebRtcTransportOptions
-
+    const webRtcTransport_options = {
-      // callId - Id of the call
+      listenIps: [
-      // kind - producer type: audio/video
+        {
-      socketToEmit?.emit('new-producer', { callId, kind });
+          ip: process.env.IP, // Listening IPv4 or IPv6.
-    } catch (error) {
+          announcedIp: process.env.ANNOUNCED_IP, // Announced IPv4 or IPv6 (useful when running mediasoup behind NAT with private IP).
-      console.error(`[transport-produce] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
+        }
-    }
+      ],
-  });
+      enableUdp: true,
-
+      enableTcp: true,
-  /*
+      preferUdp: true,
-    - The client sends this event after successfully creating a createRecvTransport(AS CONSUMER)
+    };
-    - The connection is made to the created consumerTransport
+    
-  */
+    // https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
-  socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
+    let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options)
-    try {
+    console.log(`callId: ${callId} | transport id: ${transport.id}`)
-      const callId = socketDetails[socket.id];
+
-      console.log(`[transport-recv-connect] socket ${socket.id} | callId ${callId}`);
+    // Handler for when DTLS(Datagram Transport Layer Security) changes
-      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
+    transport.on('dtlsstatechange', dtlsState => {
-      // await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
+      console.log(`transport | dtlsstatechange | calldId ${callId} | dtlsState ${dtlsState}`);
-      if (!isInitiator(callId, socket.id)) {
+      if (dtlsState === 'closed') {
-        await videoCalls[callId].receiverConsumerTransport.connect({ dtlsParameters });
+        transport.close();
-      } else if (isInitiator(callId, socket.id)) {
+      }
-        await videoCalls[callId].initiatorConsumerTransport.connect({ dtlsParameters });
+    });
-      }
+
-    } catch (error) {
+    // Handler if the transport layer has closed (for various reasons)
-      console.error(`[transport-recv-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
+    transport.on('close', () => {
-    }
+      console.log(`transport | closed | calldId ${callId}`);
-  });
+    });
-
+
-  /*
+    const params = {
-    - The customer consumes after successfully connecting to consumerTransport
+      id: transport.id,
-    - The previous step was 'transport-recv-connect', and before that 'createWebRtcTransport'
+      iceParameters: transport.iceParameters,
-    - This event is only sent by the consumer
+      iceCandidates: transport.iceCandidates,
-    - The parameters that the consumer consumes are returned
+      dtlsParameters: transport.dtlsParameters,
-      - The consumer does consumerTransport.consume(params)
+    };
-  */
+
-  socket.on('consume', async ({ rtpCapabilities }, callback) => {
+    console.log(`createWebRtcTransportLayer | params.dtlsParameters ${params.dtlsParameters}`);
-    const callId = socketDetails[socket.id];
+    // Send back to the client the params
-    const socketId = socket.id;
+    callback({ params });
-
+
-    console.log(`[consume] socket ${socketId} | callId: ${callId}`);
+    // Set transport to producerTransport or consumerTransport
-
+    return transport;
-    if (typeof rtpCapabilities === 'string') rtpCapabilities = JSON.parse(rtpCapabilities);
+
-
+  } catch (error) {
-    callback({
+    console.log(`ERROR | createWebRtcTransportLayer | callId ${socketDetails[socket.id]} | ${error.message}`);
-      videoParams: await consumeVideo({ callId, socketId, rtpCapabilities }),
+    callback({ params: { error } });
-      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

@@ -1,75 +0,0 @@
-#/!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 -

public/config.js

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

public/index.html

@@ -34,9 +34,6 @@
   <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>
@@ -46,24 +43,12 @@
                     <tr>
                         <td>
                             <div id="sharedBtns">
-                                <video
+                                <video id="localVideo" autoplay class="video" ></video>
-                                    id="localVideo"
-                                    class="video"
-                                    autoplay
-                                    muted
-                                    playsinline
-                                ></video>
                             </div>
                         </td>
                         <td>
                             <div id="sharedBtns">
-                                <video
+                                <video id="remoteVideo" autoplay class="video" ></video>
-                                    id="remoteVideo"
-                                    class="video"
-                                    autoplay
-                                    muted
-                                    playsinline
-                                ></video>
                             </div>
                         </td>
                     </tr>
@@ -75,11 +60,34 @@
                         </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

@@ -10,201 +10,147 @@ 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)
 
-console.log('🟩 config', config)
+let socket 
+hub = io(config.hubAddress)
 
-produceAudioSelector = document.getElementById('produceAudio');
+const connectToMediasoup = () => {
-produceAudioSelector.addEventListener('change', e => {
+
-    if(e.target.checked) {
+  socket = io(config.mediasoupAddress, {
-      produceAudio = true
+    reconnection: true,
-      console.log('produce audio');
+    reconnectionDelay: 1000,
-    } else {
+    reconnectionDelayMax : 5000,
-      produceAudio = false
+    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()
+      // 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 producerVideo
+let producer
-let producerAudio
 let consumer
 let originAssetId
-let consumerVideo // local consumer video(consumer not transport)
+// let originAssetName = 'Adi'
-let consumerAudio // local consumer audio(consumer not transport)
+// let originAssetTypeName = 'linx'
-
-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 videoParams = {
+let params = {
+  // mediasoup params
   encodings: [
-    { scaleResolutionDownBy: 4, maxBitrate: 500000 },
+    {
-    { scaleResolutionDownBy: 2, maxBitrate: 1000000 },
+      rid: 'r0',
-    { scaleResolutionDownBy: 1, maxBitrate: 5000000 },
+      maxBitrate: 100000,
-    { scalabilityMode: 'S3T3_KEY' }
+      scalabilityMode: 'S1T3',
+    },
+    {
+      rid: 'r1',
+      maxBitrate: 300000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      rid: 'r2',
+      maxBitrate: 900000,
+      scalabilityMode: 'S1T3',
+    },
   ],
+  // https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerCodecOptions
   codecOptions: {
     videoGoogleStartBitrate: 1000
   }
 }
 
-let audioParams = {
-  codecOptions :
-  {
-    opusStereo : true,
-    opusDtx    : true
-  }
-}
-
-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);
+  console.log('[streamSuccess]');
   localVideo.srcObject = stream
-  console.log('stream', stream);
+  const track = stream.getVideoTracks()[0]
-  const videoTrack = stream.getVideoTracks()[0]
+  params = {
-  const audioTrack = stream.getAudioTracks()[0]
+    track,
-
+    ...params
-  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: produceAudio ? true : false,
+    audio: false,
     video: {
-      qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
+      width: {
-      vga  : { width: { ideal: 640 }, height: { ideal: 480 } },
+        min: 640,
-      hd   : { width: { ideal: 1280 }, height: { ideal: 720 } }
+        max: 1920,
+      },
+      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 = () => {
@@ -221,6 +167,7 @@ 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
@@ -231,8 +178,7 @@ const createDevice = async () => {
     })
 
     console.log('Device RTP Capabilities', device.rtpCapabilities)
-    console.log('[createDevice] device', device);
+
-    
     // once the device loads, create transport
     goCreateTransport()
 
@@ -261,20 +207,18 @@ 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 }, (value) => {
+  socket.emit('createWebRtcTransport', { sender: true, callId }, ({ params }) => {
-
-    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
@@ -300,10 +244,10 @@ const createSendTransport = () => {
     })
 
     producerTransport.on('produce', async (parameters, callback, errback) => {
-      console.log('[produce] parameters', parameters)
+      console.log(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', ...)
@@ -326,45 +270,21 @@ 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);
 
-  producerVideoHandler.on('trackended', () => {
+  producer.on('trackended', () => {
     console.log('track ended')
     // close video track
-  })  
+  })
 
-  producerVideoHandler.on('transportclose', () => {
+  producer.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,
@@ -374,7 +294,7 @@ const connectSendTransport = async () => {
     origin_asset_type_name: ASSET_TYPE,
     origin_asset_name: ASSET_NAME,
     video_call_id: callId,
-    answer: 'accepted', // answer: accepted/rejected
+    answer: 'accepted', // answer: 'rejected'
   };
   console.log('SEND answer', answer);
 
@@ -386,13 +306,11 @@ 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 
@@ -402,15 +320,15 @@ const createRecvTransport = async () => {
       return
     }
 
-    console.log('[createRecvTransport] params', params)
+    console.log(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 {
@@ -427,8 +345,7 @@ const createRecvTransport = async () => {
         errback(error)
       }
     })
-    // We call it in new-rpoducer, we don't need it here anymore
+    connectRecvTransport()
-    // connectRecvTransport()
   })
 }
 
@@ -436,8 +353,7 @@ const resetCallSettings = () => {
   localVideo.srcObject = null
   remoteVideo.srcObject = null
   consumer = null
-  producerVideo = null
+  producer = null
-  producerAudio = null
   producerTransport = null
   consumerTransport = null
   device = undefined
@@ -445,97 +361,40 @@ 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 ({videoParams, audioParams}) => {
+  }, async ({ params }) => {
-    console.log(`[consume] 🟩 videoParams`, videoParams)
+    if (params.error) {
-    console.log(`[consume] 🟩 audioParams`, audioParams)
+      console.log('Cannot Consume')
-    console.log('[consume] 🟩 consumerTransport', consumerTransport)
+      return
+    }
+
+    // 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)
-    // Maybe the unit does not produce video or audio, so we must only consume what is produced
+    // stream.removeTrack(track)
-    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)
+    socket.emit('consumer-resume')
+    console.log('consumer', consumer);
 
-    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');
   
@@ -557,30 +416,6 @@ 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', consume)
+btnRecvSendTransport.addEventListener('click', goConnect)
 btnCloseCall.addEventListener('click', closeCall)