diff --git a/app.js b/app.js
index cf9ddee..adbf45e 100644
--- a/app.js
+++ b/app.js
@@ -1,597 +1,597 @@
-require('dotenv').config();
-
-const express = require('express');
-const app = express();
-const Server = require('socket.io');
-const path = require('node:path');
-const fs = require('node:fs');
-let https;
-try {
-  https = require('node:https');
-} catch (err) {
-  console.log('https support is disabled!');
-}
-const mediasoup = require('mediasoup');
-
-let worker;
-/**
- *
- * videoCalls - Dictionary of Object(s)
- * '<callId>': {
- *  router: Router,
- *  initiatorAudioProducer: Producer,
- *  initiatorVideoProducer: Producer,
- *  receiverVideoProducer: Producer,
- *  receiverAudioProducer: Producer,
- *  initiatorProducerTransport: Producer Transport,
- *  receiverProducerTransport: Producer Transport,
- *  initiatorConsumerVideo: Consumer,
- *  initiatorConsumerAudio: Consumer,
- *  initiatorConsumerTransport: Consumer Transport
- *  initiatorSocket
- *  receiverSocket
- * }
- *
- **/
-let videoCalls = {};
-let socketDetails = {};
-
-app.get('/', (_req, res) => {
-  res.send('Hello from mediasoup app!');
-});
-
-app.use('/sfu', express.static(path.join(__dirname, 'public')));
-
-// SSL cert for HTTPS access
-const options = {
-  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 io = new Server(httpsServer, {
-  allowEIO3: true,
-  origins: ['*:*'],
-});
-
-httpsServer.listen(process.env.PORT, () => {
-  console.log('Video server listening on port:', process.env.PORT);
-});
-
-const peers = io.of('/');
-
-const createWorker = async () => {
-  try {
-    worker = await mediasoup.createWorker({
-      rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
-      rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
-    });
-    console.log(`[createWorker] worker pid ${worker.pid}`);
-
-    worker.on('died', (error) => {
-      // This implies something serious happened, so kill the application
-      console.error('mediasoup worker has died', error);
-      setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
-    });
-    return worker;
-  } catch (error) {
-    console.error(`[createWorker] | ERROR | error: ${error.message}`);
-  }
-};
-
-// We create a Worker as soon as our application starts
-worker = createWorker();
-
-// This is an Array of RtpCapabilities
-// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
-// list of media codecs supported by mediasoup ...
-// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
-const mediaCodecs = [
-  {
-    kind: 'audio',
-    mimeType: 'audio/opus',
-    clockRate: 48000,
-    channels: 2,
-  },
-  {
-    kind: 'video',
-    mimeType: 'video/VP8',
-    clockRate: 90000,
-    parameters: {
-      'x-google-start-bitrate': 1000,
-    },
-    channels: 2,
-  },
-  {
-    kind: 'video',
-    mimeType: 'video/VP9',
-    clockRate: 90000,
-    parameters: {
-      'profile-id': 2,
-      'x-google-start-bitrate': 1000,
-    },
-  },
-  {
-    kind: 'video',
-    mimeType: 'video/h264',
-    clockRate: 90000,
-    parameters: {
-      'packetization-mode': 1,
-      'profile-level-id': '4d0032',
-      'level-asymmetry-allowed': 1,
-      'x-google-start-bitrate': 1000,
-    },
-  },
-  {
-    kind: 'video',
-    mimeType: 'video/h264',
-    clockRate: 90000,
-    parameters: {
-      'packetization-mode': 1,
-      'profile-level-id': '42e01f',
-      'level-asymmetry-allowed': 1,
-      'x-google-start-bitrate': 1000,
-    },
-  },
-];
-
-const closeCall = (callId) => {
-  try {
-    if (callId && videoCalls[callId]) {
-      videoCalls[callId].receiverVideoProducer?.close();
-      videoCalls[callId].receiverAudioProducer?.close();
-      videoCalls[callId].initiatorConsumerVideo?.close();
-      videoCalls[callId].initiatorConsumerAudio?.close();
-
-      videoCalls[callId]?.initiatorConsumerTransport?.close();
-      videoCalls[callId]?.receiverProducerTransport?.close();
-      videoCalls[callId]?.router?.close();
-      delete videoCalls[callId];
-      console.log(`[closeCall] | callId: ${callId}`);
-    }
-  } catch (error) {
-    console.error(`[closeCall] | ERROR | callId: ${callId} | error: ${error.message}`);
-  }
-};
-
-/*
-  - Handlers for WS events
-  - These are created only when we have a connection with a peer
-*/
-peers.on('connection', async (socket) => {
-  console.log('[connection] socketId:', socket.id);
-
-  // After making the connection successfully, we send the client a 'connection-success' event
-  socket.emit('connection-success', {
-    socketId: socket.id,
-  });
-
-  // It is triggered when the peer is disconnected
-  socket.on('disconnect', () => {
-    const callId = socketDetails[socket.id];
-    console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
-    delete socketDetails[socket.id];
-    closeCall(callId);
-  });
-
-  /*
-    - This event creates a room with the roomId and the callId sent
-    - It will return the rtpCapabilities of that room
-    - If the room already exists, it will not create it, but will only return rtpCapabilities
-  */
-  socket.on('createRoom', async ({ callId }, callback) => {
-    let callbackResponse = null;
-    try {
-      // We can continue with the room creation process only if we have a callId
-      if (callId) {
-        console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
-        if (!videoCalls[callId]) {
-          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) };
-          console.log(`[createRoom] Generate Router ID: ${videoCalls[callId].router.id}`);
-          videoCalls[callId].receiverSocket = socket;
-        } else {
-          videoCalls[callId].initiatorSocket = socket;
-        }
-        socketDetails[socket.id] = callId;
-        // rtpCapabilities is set for callback
-        callbackResponse = {
-          rtpCapabilities: videoCalls[callId].router.rtpCapabilities,
-        };
-      } else {
-        console.log(`[createRoom] missing callId: ${callId}`);
-      }
-    } catch (error) {
-      console.error(`[createRoom] | ERROR | callId: ${callId} | error: ${error.message}`);
-    } finally {
-      callback(callbackResponse);
-    }
-  });
-
-  /*
-    - Client emits a request to create server side Transport
-    - Depending on the sender, a producer or consumer is created is created on that router
-    - It will return parameters, these are required for the client to create the RecvTransport 
-    from the client.
-      - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
-      - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
-  */
-  socket.on('createWebRtcTransport', async ({ sender }, callback) => {
-    try {
-      const callId = socketDetails[socket.id];
-      console.log(`[createWebRtcTransport] socket ${socket.id} | sender ${sender} | callId ${callId}`);
-      if (sender) {
-        if (!videoCalls[callId].receiverProducerTransport && !isInitiator(callId, socket.id)) {
-          videoCalls[callId].receiverProducerTransport = await createWebRtcTransportLayer(callId, callback);
-        } 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);
-        }
-      } else if (!sender) {
-        if (!videoCalls[callId].receiverConsumerTransport && !isInitiator(callId, socket.id)) {
-          videoCalls[callId].receiverConsumerTransport = await createWebRtcTransportLayer(callId, callback);
-        } else if (!videoCalls[callId].initiatorConsumerTransport && isInitiator(callId, socket.id)) {
-          videoCalls[callId].initiatorConsumerTransport = await createWebRtcTransportLayer(callId, callback);
-        }
-      }
-    } catch (error) {
-      console.error(
-        `[createWebRtcTransport] | ERROR | callId: ${socketDetails[socket.id]} | sender: ${sender} | error: ${
-          error.message
-        }`
-      );
-      callback(error);
-    }
-  });
-
-  /*
-    - The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
-    - The connection is made to the created transport
-  */
-  socket.on('transport-connect', async ({ dtlsParameters }) => {
-    try {
-      const callId = socketDetails[socket.id];
-      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
-
-      console.log(`[transport-connect] socket ${socket.id} | callId ${callId}`);
-
-      isInitiator(callId, socket.id)
-        ? await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters })
-        : await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
-    } catch (error) {
-      console.error(`[transport-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
-    }
-  });
-
-  /*
-    - The event sent by the client (PRODUCER) after successfully connecting to receiverProducerTransport/initiatorProducerTransport
-    - For the router with the id callId, we make produce on receiverProducerTransport/initiatorProducerTransport
-    - Create the handler on producer at the 'transportclose' event
-    */
-  socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
-    try {
-      const callId = socketDetails[socket.id];
-      if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
-
-      console.log(`[transport-produce] callId: ${callId} | kind: ${kind} | socket: ${socket.id}`);
-
-      if (kind === 'video') {
-        if (!isInitiator(callId, socket.id)) {
-          videoCalls[callId].receiverVideoProducer = await videoCalls[callId].receiverProducerTransport.produce({
-            kind,
-            rtpParameters,
-          });
-
-          videoCalls[callId].receiverVideoProducer.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].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 } });
-  }
-};
+require('dotenv').config();
+
+const express = require('express');
+const app = express();
+const Server = require('socket.io');
+const path = require('node:path');
+const fs = require('node:fs');
+let https;
+try {
+  https = require('node:https');
+} catch (err) {
+  console.log('https support is disabled!');
+}
+const mediasoup = require('mediasoup');
+
+let worker;
+/**
+ *
+ * videoCalls - Dictionary of Object(s)
+ * '<callId>': {
+ *  router: Router,
+ *  initiatorAudioProducer: Producer,
+ *  initiatorVideoProducer: Producer,
+ *  receiverVideoProducer: Producer,
+ *  receiverAudioProducer: Producer,
+ *  initiatorProducerTransport: Producer Transport,
+ *  receiverProducerTransport: Producer Transport,
+ *  initiatorConsumerVideo: Consumer,
+ *  initiatorConsumerAudio: Consumer,
+ *  initiatorConsumerTransport: Consumer Transport
+ *  initiatorSocket
+ *  receiverSocket
+ * }
+ *
+ **/
+let videoCalls = {};
+let socketDetails = {};
+
+app.get('/', (_req, res) => {
+  res.send('OK');
+});
+
+app.use('/sfu', express.static(path.join(__dirname, 'public')));
+
+// SSL cert for HTTPS access
+const options = {
+  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 io = new Server(httpsServer, {
+  allowEIO3: true,
+  origins: ['*:*'],
+});
+
+httpsServer.listen(process.env.PORT, () => {
+  console.log('Video server listening on port:', process.env.PORT);
+});
+
+const peers = io.of('/');
+
+const createWorker = async () => {
+  try {
+    worker = await mediasoup.createWorker({
+      rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
+      rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
+    });
+    console.log(`[createWorker] worker pid ${worker.pid}`);
+
+    worker.on('died', (error) => {
+      // This implies something serious happened, so kill the application
+      console.error('mediasoup worker has died', error);
+      setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
+    });
+    return worker;
+  } catch (error) {
+    console.error(`[createWorker] | ERROR | error: ${error.message}`);
+  }
+};
+
+// We create a Worker as soon as our application starts
+worker = createWorker();
+
+// This is an Array of RtpCapabilities
+// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
+// list of media codecs supported by mediasoup ...
+// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
+const mediaCodecs = [
+  {
+    kind: 'audio',
+    mimeType: 'audio/opus',
+    clockRate: 48000,
+    channels: 2,
+  },
+  {
+    kind: 'video',
+    mimeType: 'video/VP8',
+    clockRate: 90000,
+    parameters: {
+      'x-google-start-bitrate': 1000,
+    },
+    channels: 2,
+  },
+  {
+    kind: 'video',
+    mimeType: 'video/VP9',
+    clockRate: 90000,
+    parameters: {
+      'profile-id': 2,
+      'x-google-start-bitrate': 1000,
+    },
+  },
+  {
+    kind: 'video',
+    mimeType: 'video/h264',
+    clockRate: 90000,
+    parameters: {
+      'packetization-mode': 1,
+      'profile-level-id': '4d0032',
+      'level-asymmetry-allowed': 1,
+      'x-google-start-bitrate': 1000,
+    },
+  },
+  {
+    kind: 'video',
+    mimeType: 'video/h264',
+    clockRate: 90000,
+    parameters: {
+      'packetization-mode': 1,
+      'profile-level-id': '42e01f',
+      'level-asymmetry-allowed': 1,
+      'x-google-start-bitrate': 1000,
+    },
+  },
+];
+
+const closeCall = (callId) => {
+  try {
+    if (callId && videoCalls[callId]) {
+      videoCalls[callId].receiverVideoProducer?.close();
+      videoCalls[callId].receiverAudioProducer?.close();
+      videoCalls[callId].initiatorConsumerVideo?.close();
+      videoCalls[callId].initiatorConsumerAudio?.close();
+
+      videoCalls[callId]?.initiatorConsumerTransport?.close();
+      videoCalls[callId]?.receiverProducerTransport?.close();
+      videoCalls[callId]?.router?.close();
+      delete videoCalls[callId];
+      console.log(`[closeCall] | callId: ${callId}`);
+    }
+  } catch (error) {
+    console.error(`[closeCall] | ERROR | callId: ${callId} | error: ${error.message}`);
+  }
+};
+
+/*
+  - Handlers for WS events
+  - These are created only when we have a connection with a peer
+*/
+peers.on('connection', async (socket) => {
+  console.log('[connection] socketId:', socket.id);
+
+  // After making the connection successfully, we send the client a 'connection-success' event
+  socket.emit('connection-success', {
+    socketId: socket.id,
+  });
+
+  // It is triggered when the peer is disconnected
+  socket.on('disconnect', () => {
+    const callId = socketDetails[socket.id];
+    console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
+    delete socketDetails[socket.id];
+    closeCall(callId);
+  });
+
+  /*
+    - This event creates a room with the roomId and the callId sent
+    - It will return the rtpCapabilities of that room
+    - If the room already exists, it will not create it, but will only return rtpCapabilities
+  */
+  socket.on('createRoom', async ({ callId }, callback) => {
+    let callbackResponse = null;
+    try {
+      // We can continue with the room creation process only if we have a callId
+      if (callId) {
+        console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
+        if (!videoCalls[callId]) {
+          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) };
+          console.log(`[createRoom] Generate Router ID: ${videoCalls[callId].router.id}`);
+          videoCalls[callId].receiverSocket = socket;
+        } else {
+          videoCalls[callId].initiatorSocket = socket;
+        }
+        socketDetails[socket.id] = callId;
+        // rtpCapabilities is set for callback
+        callbackResponse = {
+          rtpCapabilities: videoCalls[callId].router.rtpCapabilities,
+        };
+      } else {
+        console.log(`[createRoom] missing callId: ${callId}`);
+      }
+    } catch (error) {
+      console.error(`[createRoom] | ERROR | callId: ${callId} | error: ${error.message}`);
+    } finally {
+      callback(callbackResponse);
+    }
+  });
+
+  /*
+    - Client emits a request to create server side Transport
+    - Depending on the sender, a producer or consumer is created is created on that router
+    - It will return parameters, these are required for the client to create the RecvTransport 
+    from the client.
+      - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
+      - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
+  */
+  socket.on('createWebRtcTransport', async ({ sender }, callback) => {
+    try {
+      const callId = socketDetails[socket.id];
+      console.log(`[createWebRtcTransport] socket ${socket.id} | sender ${sender} | callId ${callId}`);
+      if (sender) {
+        if (!videoCalls[callId].receiverProducerTransport && !isInitiator(callId, socket.id)) {
+          videoCalls[callId].receiverProducerTransport = await createWebRtcTransportLayer(callId, callback);
+        } 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);
+        }
+      } else if (!sender) {
+        if (!videoCalls[callId].receiverConsumerTransport && !isInitiator(callId, socket.id)) {
+          videoCalls[callId].receiverConsumerTransport = await createWebRtcTransportLayer(callId, callback);
+        } else if (!videoCalls[callId].initiatorConsumerTransport && isInitiator(callId, socket.id)) {
+          videoCalls[callId].initiatorConsumerTransport = await createWebRtcTransportLayer(callId, callback);
+        }
+      }
+    } catch (error) {
+      console.error(
+        `[createWebRtcTransport] | ERROR | callId: ${socketDetails[socket.id]} | sender: ${sender} | error: ${
+          error.message
+        }`
+      );
+      callback(error);
+    }
+  });
+
+  /*
+    - The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
+    - The connection is made to the created transport
+  */
+  socket.on('transport-connect', async ({ dtlsParameters }) => {
+    try {
+      const callId = socketDetails[socket.id];
+      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
+
+      console.log(`[transport-connect] socket ${socket.id} | callId ${callId}`);
+
+      isInitiator(callId, socket.id)
+        ? await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters })
+        : await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
+    } catch (error) {
+      console.error(`[transport-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
+    }
+  });
+
+  /*
+    - The event sent by the client (PRODUCER) after successfully connecting to receiverProducerTransport/initiatorProducerTransport
+    - For the router with the id callId, we make produce on receiverProducerTransport/initiatorProducerTransport
+    - Create the handler on producer at the 'transportclose' event
+    */
+  socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
+    try {
+      const callId = socketDetails[socket.id];
+      if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
+
+      console.log(`[transport-produce] callId: ${callId} | kind: ${kind} | socket: ${socket.id}`);
+
+      if (kind === 'video') {
+        if (!isInitiator(callId, socket.id)) {
+          videoCalls[callId].receiverVideoProducer = await videoCalls[callId].receiverProducerTransport.produce({
+            kind,
+            rtpParameters,
+          });
+
+          videoCalls[callId].receiverVideoProducer.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].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 } });
+  }
+};