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

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

.dockerignore

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

.gitignore

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

Dockerfile

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

README.md

@@ -22,18 +22,20 @@
 2. Run the `npm start:prod` command to start the server in production mode.
 (To connect to the terminal, use `pm2 log video-server`)
 
----
+
+### Web client
 
 - The server will start by default on port 3000, and the ssl certificates will have to be configured
 - The web client can be accessed using the /sfu path
-ex: http://localhost:3000/sfu/?assetId=1&&accountId=1&producer=true&assetName=Adi&assetType=linx
+ex: https://HOST/sfu/?assetId=1&&accountId=1&producer=true&dest_asset_id=75&assetName=Adi
 assetId = asset id of the unit on which you are doing the test
 accountId = account id of the unit on which you are doing the test
 producer = it will always be true because you are the producer
 (it's possible to put false, but then you have to have another client with producer true)
 assetName = asset name of the unit on which you are doing the test
-assetType = asset type of the unit on which you are doing the test
+dest_asset_id= the addressee with whom the call is made
+- To make a call using this client, you need a microphone and permission to use it
+- For any changes related to the client, the command `npm run watch' will have to be used to generate the bundle.js used by the web client
 
 ### Demo project
   The demo project used initially and then modified for our needs `https://github.com/jamalag/mediasoup2`
-

app.js

@@ -1,83 +1,84 @@
-require('dotenv').config()
+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');
-const https = require('https');
+let https;
+try {
+  https = require('node:https');
+} catch (err) {
+  console.log('https support is disabled!');
+}
 const mediasoup = require('mediasoup');
 
-let worker
+let worker;
 /**
- * videoCalls
- * |-> Router
- * |-> Producer
- * |-> Consumer 
- * |-> Producer Transport
- * |-> Consumer Transport
  *
+ * videoCalls - Dictionary of Object(s)
  * '<callId>': {
  *  router: Router,
- *  producer: Producer,
+ *  initiatorAudioProducer: Producer,
- *  producerTransport: Producer Transport,
+ *  initiatorVideoProducer: Producer,
- *  consumer: Consumer,
+ *  receiverVideoProducer: Producer,
- *  consumerTransport: Consumer Transport
+ *  receiverAudioProducer: Producer,
+ *  initiatorProducerTransport: Producer Transport,
+ *  receiverProducerTransport: Producer Transport,
+ *  initiatorConsumerVideo: Consumer,
+ *  initiatorConsumerAudio: Consumer,
+ *  initiatorConsumerTransport: Consumer Transport
+ *  initiatorSocket
+ *  receiverSocket
  * }
  *
  **/
-let videoCalls = {}
+let videoCalls = {};
-let socketDetails = {}
+let socketDetails = {};
 
 app.get('/', (_req, res) => {
-  res.send('Hello from mediasoup app!')
+  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 = {
-  key: fs.readFileSync('./server/ssl/key.pem', 'utf-8'),
+  key: fs.readFileSync(process.env.SERVER_KEY, 'utf-8'),
-  cert: fs.readFileSync('./server/ssl/cert.pem', '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: ["*:*"],
+  origins: ['*:*'],
-  // allowRequest: (req, next) => {
-  //   console.log('req', req);
-  //   next(null, true)
-  // }
 });
-// const io = new Server(server, { origins: '*:*', allowEIO3: true });
 
 httpsServer.listen(process.env.PORT, () => {
   console.log('Video server listening on port:', process.env.PORT);
 });
 
 const peers = io.of('/');
-console.log('process.env.RTC_MIN_PORT', process.env.RTC_MIN_PORT);
+
-console.log('process.env.RTC_MAX_PORT', process.env.RTC_MAX_PORT, process.env.RTC_MAX_PORT.length);
 const createWorker = async () => {
   try {
     worker = await mediasoup.createWorker({
-      rtcMinPort: process.env.RTC_MIN_PORT,
+      rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
-      rtcMaxPort: process.env.RTC_MAX_PORT,
+      rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
-    })
+    });
     console.log(`[createWorker] worker pid ${worker.pid}`);
 
-    worker.on('died', error => {
+    worker.on('died', (error) => {
       // This implies something serious happened, so kill the application
       console.error('mediasoup worker has died', error);
       setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
-    })
+    });
     return worker;
   } catch (error) {
-    console.log(`ERROR | createWorker | ${error.message}`);
+    console.error(`[createWorker] | ERROR | error: ${error.message}`);
-  }
   }
+};
 
 // We create a Worker as soon as our application starts
 worker = createWorker();
@@ -100,46 +101,70 @@ const mediaCodecs = [
     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].producer?.close();
+      videoCalls[callId].receiverVideoProducer?.close();
-      videoCalls[callId].consumer?.close();
+      videoCalls[callId].receiverAudioProducer?.close();
-      videoCalls[callId]?.consumerTransport?.close();
+      videoCalls[callId].initiatorConsumerVideo?.close();
-      videoCalls[callId]?.producerTransport?.close();
+      videoCalls[callId].initiatorConsumerAudio?.close();
+
+      videoCalls[callId]?.initiatorConsumerTransport?.close();
+      videoCalls[callId]?.receiverProducerTransport?.close();
       videoCalls[callId]?.router?.close();
       delete videoCalls[callId];
-    } else {
+      console.log(`[closeCall] | callId: ${callId}`);
-      console.log(`The call with id ${callId} has already been deleted`);
     }
   } catch (error) {
-    console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
+    console.error(`[closeCall] | ERROR | callId: ${callId} | error: ${error.message}`);
-  }
-}
-
-const getRtpCapabilities = (callId, callback) => {
-  try {
-    console.log('[getRtpCapabilities] callId', callId);
-    const rtpCapabilities = videoCalls[callId].router.rtpCapabilities;
-    callback({ rtpCapabilities });
-  } catch (error) {
-    console.log(`ERROR | getRtpCapabilities | callId ${callId} | ${error.message}`);
-  }
   }
+};
 
 /*
   - Handlers for WS events
   - These are created only when we have a connection with a peer
 */
-peers.on('connection', async socket => {
+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
+    socketId: socket.id,
   });
 
   // It is triggered when the peer is disconnected
@@ -156,27 +181,36 @@ peers.on('connection', async socket => {
     - 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]) {
-          console.log('[createRoom] callId', callId);
+          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) };
-          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) }
+          console.log(`[createRoom] Generate Router ID: ${videoCalls[callId].router.id}`);
-          console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
+          videoCalls[callId].receiverSocket = socket;
+        } else {
+          videoCalls[callId].initiatorSocket = socket;
         }
         socketDetails[socket.id] = callId;
-        getRtpCapabilities(callId, callback);
+        // rtpCapabilities is set for callback
+        callbackResponse = {
+          rtpCapabilities: videoCalls[callId].router.rtpCapabilities,
+        };
       } else {
-        console.log(`[createRoom] missing callId ${callId}`);
+        console.log(`[createRoom] missing callId: ${callId}`);
       }
     } catch (error) {
-      console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
+      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, producerTransport or consumerTransport is created on that router
+    - 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)
@@ -185,22 +219,30 @@ peers.on('connection', async socket => {
   socket.on('createWebRtcTransport', async ({ sender }, callback) => {
     try {
       const callId = socketDetails[socket.id];
-      console.log(`[createWebRtcTransport] sender ${sender} | callId ${callId}`);
+      console.log(`[createWebRtcTransport] socket ${socket.id} | sender ${sender} | callId ${callId}`);
       if (sender) {
-        if (!videoCalls[callId].producerTransport) {
+        if (!videoCalls[callId].receiverProducerTransport && !isInitiator(callId, socket.id)) {
-          videoCalls[callId].producerTransport = await createWebRtcTransportLayer(callId, callback);
+          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].consumerTransport) {
+        if (!videoCalls[callId].receiverConsumerTransport && !isInitiator(callId, socket.id)) {
-          videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
+          videoCalls[callId].receiverConsumerTransport = await createWebRtcTransportLayer(callId, callback);
-        } else {
+        } else if (!videoCalls[callId].initiatorConsumerTransport && isInitiator(callId, socket.id)) {
-          console.log(`consumerTransport has already been defined | callId ${callId}`);
+          videoCalls[callId].initiatorConsumerTransport = await createWebRtcTransportLayer(callId, callback);
         }
       }
     } catch (error) {
-      console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
+      console.error(
+        `[createWebRtcTransport] | ERROR | callId: ${socketDetails[socket.id]} | sender: ${sender} | error: ${
+          error.message
+        }`
+      );
+      callback(error);
     }
   });
 
@@ -213,16 +255,19 @@ peers.on('connection', async socket => {
       const callId = socketDetails[socket.id];
       if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
 
-      console.log(`[transport-connect] socket.id ${socket.id} | callId ${callId}`);
+      console.log(`[transport-connect] socket ${socket.id} | callId ${callId}`);
-      await videoCalls[callId].producerTransport.connect({ dtlsParameters });
+
+      isInitiator(callId, socket.id)
+        ? await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters })
+        : await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
     } catch (error) {
-      console.log(`ERROR | transport-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
+      console.error(`[transport-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
     }
   });
 
   /*
-    - The event sent by the client (PRODUCER) after successfully connecting to producerTransport
+    - The event sent by the client (PRODUCER) after successfully connecting to receiverProducerTransport/initiatorProducerTransport
-    - For the router with the id callId, we make produce on producerTransport
+    - 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) => {
@@ -230,25 +275,86 @@ peers.on('connection', async socket => {
       const callId = socketDetails[socket.id];
       if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
 
-      console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
+      console.log(`[transport-produce] callId: ${callId} | kind: ${kind} | socket: ${socket.id}`);
-      videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
+
+      if (kind === 'video') {
+        if (!isInitiator(callId, socket.id)) {
+          videoCalls[callId].receiverVideoProducer = await videoCalls[callId].receiverProducerTransport.produce({
             kind,
             rtpParameters,
           });
-      console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producer.id} | kind: ${videoCalls[callId].producer.kind}`);
 
-      videoCalls[callId].producer.on('transportclose', () => {
+          videoCalls[callId].receiverVideoProducer.on('transportclose', () => {
-        const callId = socketDetails[socket.id];
+            console.log('transport for this producer closed', callId);
-        console.log('transport for this producer closed', callId)
             closeCall(callId);
           });
 
           // Send back to the client the Producer's id
-      // callback({
+          callback &&
-      //   id: videoCalls[callId].producer.id
+            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.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
+      console.error(`[transport-produce] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
     }
   });
 
@@ -259,12 +365,18 @@ peers.on('connection', async socket => {
   socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
     try {
       const callId = socketDetails[socket.id];
-      console.log(`[transport-recv-connect] socket.id ${socket.id} | callId ${callId}`);
+      console.log(`[transport-recv-connect] socket ${socket.id} | callId ${callId}`);
-      await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
+      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
-    } catch (error) {
+      // await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
-      console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
+      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
@@ -274,71 +386,157 @@ peers.on('connection', async socket => {
       - The consumer does consumerTransport.consume(params)
   */
   socket.on('consume', async ({ rtpCapabilities }, callback) => {
-    try {
     const callId = socketDetails[socket.id];
-      console.log('[consume] callId', callId);
+    const socketId = socket.id;
 
-      // Check if the router can consume the specified producer
+    console.log(`[consume] socket ${socketId} | callId: ${callId}`);
-      if (videoCalls[callId].router.canConsume({
+
-        producerId: videoCalls[callId].producer.id,
+    if (typeof rtpCapabilities === 'string') rtpCapabilities = JSON.parse(rtpCapabilities);
-        rtpCapabilities
+
-      })) {
+    callback({
-        console.log('[consume] Can consume', callId);
+      videoParams: await consumeVideo({ callId, socketId, rtpCapabilities }),
-        // Transport can now consume and return a consumer
+      audioParams: await consumeAudio({ callId, socketId, rtpCapabilities }),
-        videoCalls[callId].consumer = await videoCalls[callId].consumerTransport.consume({
-          producerId: videoCalls[callId].producer.id,
-          rtpCapabilities,
-          paused: true,
     });
-
-        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
-        videoCalls[callId].consumer.on('transportclose', () => {
-          const callId = socketDetails[socket.id];
-          console.log('transport close from consumer', callId);
-          closeCall(callId);
-        });
-
-        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
-        videoCalls[callId].consumer.on('producerclose', () => {
-          const callId = socketDetails[socket.id];
-          console.log('producer of consumer closed', callId);
-          closeCall(callId);
-        });
-
-        // From the consumer extract the following params to send back to the Client
-        const params = {
-          id: videoCalls[callId].consumer.id,
-          producerId: videoCalls[callId].producer.id,
-          kind: videoCalls[callId].consumer.kind,
-          rtpParameters: videoCalls[callId].consumer.rtpParameters,
-        };
-
-        // Send the parameters to the client
-        callback({ params });
-      } else {
-        console.log(`[canConsume] Can't consume | callId ${callId}`);
-      }
-    } catch (error) {
-      console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
-      callback({ params: { error } });
-    }
   });
 
   /*
     - 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', async () => {
+  socket.on('consumer-resume', () => {
     try {
       const callId = socketDetails[socket.id];
-      console.log(`[consumer-resume] callId ${callId}`)
+      const isInitiatorValue = isInitiator(callId, socket.id);
-      await videoCalls[callId].consumer.resume();
+      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.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
+      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 
@@ -348,14 +546,14 @@ peers.on('connection', async socket => {
 */
 const createWebRtcTransportLayer = async (callId, callback) => {
   try {
-    console.log('[createWebRtcTransportLayer] callId', callId);
+    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,
@@ -363,11 +561,10 @@ const createWebRtcTransportLayer = async (callId, callback) => {
     };
 
     // https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
-    let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options)
+    let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options);
-    console.log(`callId: ${callId} | transport id: ${transport.id}`)
 
     // Handler for when DTLS(Datagram Transport Layer Security) changes
-    transport.on('dtlsstatechange', dtlsState => {
+    transport.on('dtlsstatechange', (dtlsState) => {
       console.log(`transport | dtlsstatechange | calldId ${callId} | dtlsState ${dtlsState}`);
       if (dtlsState === 'closed') {
         transport.close();
@@ -391,9 +588,10 @@ const createWebRtcTransportLayer = async (callId, callback) => {
 
     // Set transport to producerTransport or consumerTransport
     return transport;
-
   } catch (error) {
-    console.log(`ERROR | createWebRtcTransportLayer | callId ${socketDetails[socket.id]} | ${error.message}`);
+    console.error(
+      `[createWebRtcTransportLayer] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`
+    );
     callback({ params: { error } });
   }
-}
+};

build.sh

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

public/config.js

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

public/index.html

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

public/index.js

@@ -10,9 +10,72 @@ const ASSET_NAME = urlParams.get('assetName') || null;
 const ASSET_TYPE = urlParams.get('assetType') || null;
 let callId = parseInt(urlParams.get('callId')) || null;
 const IS_PRODUCER = urlParams.get('producer') === 'true' ? true : false
+let remoteVideo = document.getElementById('remoteVideo')
+remoteVideo.defaultMuted = true
+let produceAudio = false
+
 console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId', callId, ' | IS_PRODUCER', IS_PRODUCER)
 
-let socket 
+console.log('🟩 config', config)
+
+produceAudioSelector = document.getElementById('produceAudio');
+produceAudioSelector.addEventListener('change', e => {
+    if(e.target.checked) {
+      produceAudio = true
+      console.log('produce audio');
+    } else {
+      produceAudio = false
+    }
+});
+
+let socket, hub
+let device
+let rtpCapabilities
+let producerTransport
+let consumerTransport
+let producerVideo
+let producerAudio
+let consumer
+let originAssetId
+let consumerVideo // local consumer video(consumer not transport)
+let consumerAudio // local consumer audio(consumer not transport)
+
+const remoteSoundControl = document.getElementById('remoteSoundControl');
+
+remoteSoundControl.addEventListener('click', function handleClick() {
+  console.log('remoteSoundControl.textContent', remoteSoundControl.textContent);
+  if (remoteSoundControl.textContent === 'Unmute') {
+    remoteVideo.muted = false
+    remoteSoundControl.textContent = 'Mute';
+  } else {
+    remoteVideo.muted = true
+    remoteSoundControl.textContent = 'Unmute';
+  }
+});
+
+// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerOptions
+// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
+let videoParams = {
+  encodings: [
+    { scaleResolutionDownBy: 4, maxBitrate: 500000 },
+    { scaleResolutionDownBy: 2, maxBitrate: 1000000 },
+    { scaleResolutionDownBy: 1, maxBitrate: 5000000 },
+    { scalabilityMode: 'S3T3_KEY' }
+  ],
+  codecOptions: {
+    videoGoogleStartBitrate: 1000
+  }
+}
+
+let audioParams = {
+  codecOptions :
+  {
+    opusStereo : true,
+    opusDtx    : true
+  }
+}
+
+setTimeout(() => {
   hub = io(config.hubAddress)
   
   const connectToMediasoup = () => {
@@ -28,15 +91,28 @@ const connectToMediasoup = () => {
       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() }
     })
+
+    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}`)
+      console.log(`[HUB]! ${config.hubAddress} | connected: ${hub.connected}`)
       connectToMediasoup()
   
       hub.emit(
@@ -84,73 +160,51 @@ if (IS_PRODUCER === true) {
     connectToMediasoup()
   }
 
-let device
+}, 1600);
-let rtpCapabilities
-let producerTransport
-let consumerTransport
-let producer
-let consumer
-let originAssetId
-// let originAssetName = 'Adi'
-// let originAssetTypeName = 'linx'
-
-// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerOptions
-// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
-let params = {
-  // mediasoup params
-  encodings: [
-    {
-      rid: 'r0',
-      maxBitrate: 100000,
-      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
-  }
-}
 
 const streamSuccess = (stream) => {
-  console.log('[streamSuccess]');
+  console.log('[streamSuccess] device', device);
   localVideo.srcObject = stream
-  const track = stream.getVideoTracks()[0]
+  console.log('stream', stream);
-  params = {
+  const videoTrack = stream.getVideoTracks()[0]
-    track,
+  const audioTrack = stream.getAudioTracks()[0]
-    ...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: false,
+    audio: produceAudio ? true : false,
     video: {
-      width: {
+      qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
-        min: 640,
+      vga  : { width: { ideal: 640 }, height: { ideal: 480 } },
-        max: 1920,
+      hd   : { width: { ideal: 1280 }, height: { ideal: 720 } }
-      },
-      height: {
-        min: 400,
-        max: 1080,
-      }
     }
   })
   .then(streamSuccess)
   .catch(error => {
     console.log(error.message)
   })
+
+  navigator.permissions.query(
+    { name: 'microphone' }
+  ).then((permissionStatus) =>{
+    console.log('🟨 [PERMISSION] permissionStatus', permissionStatus); // granted, denied, prompt
+    // It will block the code from execution and display "Permission denied" if we don't have microphone permissions
+  })
+
 }
 
 const goConnect = () => {
@@ -167,7 +221,6 @@ const goCreateTransport = () => {
 // server side to send/recive media
 const createDevice = async () => {
   try {
-    console.log('[createDevice]');
     device = new mediasoupClient.Device()
 
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@@ -178,6 +231,7 @@ const createDevice = async () => {
     })
 
     console.log('Device RTP Capabilities', device.rtpCapabilities)
+    console.log('[createDevice] device', device);
     
     // once the device loads, create transport
     goCreateTransport()
@@ -207,18 +261,20 @@ const getRtpCapabilities = () => {
 }
 
 const createSendTransport = () => {
+  console.log('[createSendTransport');
   // see server's socket.on('createWebRtcTransport', sender?, ...)
   // this is a call from Producer, so sender = true
-  socket.emit('createWebRtcTransport', { sender: true, callId }, ({ params }) => {
+  socket.emit('createWebRtcTransport', { sender: true }, (value) => {
+
+    console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
+
+    const params = value.params;
     // The server sends back params needed 
     // to create Send Transport on the client side
     if (params.error) {
       console.log(params.error)
       return
     }
-
-    console.log(params)
-
     // creates a new WebRTC Transport to send media
     // based on the server's producer transport params
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
@@ -244,10 +300,10 @@ const createSendTransport = () => {
     })
 
     producerTransport.on('produce', async (parameters, callback, errback) => {
-      console.log(parameters)
+      console.log('[produce] parameters', parameters)
 
       try {
-        // tell the server to create a Producer
+        // Tell the server to create a Producer
         // with the following parameters and produce
         // and expect back a server side producer id
         // see server's socket.on('transport-produce', ...)
@@ -270,22 +326,46 @@ 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)
   
-  producer.on('trackended', () => {
+  // Produce video
+  let producerVideoHandler = await producerTransport.produce(videoParams)
+  console.log('videoParams', videoParams);
+  console.log('producerVideo', producerVideo);
+
+  producerVideoHandler.on('trackended', () => {
     console.log('track ended')
     // close video track
   })  
 
-  producer.on('transportclose', () => {
+  producerVideoHandler.on('transportclose', () => {
     console.log('transport ended')
     // close video track
   })    
   
+  // Produce audio
+  if (produceAudio) {
+    let producerAudioHandler = await producerTransport.produce(audioParams)
+    console.log('audioParams', audioParams);
+    console.log('producerAudio', producerAudio);
+  
+    producerAudioHandler.on('trackended', () => {
+      console.log('track ended')
+      // close audio track
+    })
+  
+    producerAudioHandler.on('transportclose', () => {
+      console.log('transport ended')
+      // close audio track
+    })
+  }
+
   const answer = {
     origin_asset_id: ASSET_ID,
     dest_asset_id: originAssetId || parseInt(urlParams.get('dest_asset_id')),
@@ -294,7 +374,7 @@ const connectSendTransport = async () => {
     origin_asset_type_name: ASSET_TYPE,
     origin_asset_name: ASSET_NAME,
     video_call_id: callId,
-    answer: 'accepted', // answer: 'rejected'
+    answer: 'accepted', // answer: accepted/rejected
   };
   console.log('SEND answer', answer);
 
@@ -306,11 +386,13 @@ const connectSendTransport = async () => {
   // Enable Close call button
   const closeCallBtn = document.getElementById('btnCloseCall');
   closeCallBtn.removeAttribute('disabled');
+
+  createRecvTransport();
 }
 
 const createRecvTransport = async () => {
   console.log('createRecvTransport');
-  // see server's socket.on('consume', sender?, ...)
+  // See server's socket.on('consume', sender?, ...)
   // this is a call from Consumer, so sender = false
   await socket.emit('createWebRtcTransport', { sender: false, callId }, ({ params }) => {
     // The server sends back params needed 
@@ -320,15 +402,15 @@ const createRecvTransport = async () => {
       return
     }
 
-    console.log(params)
+    console.log('[createRecvTransport] params', params)
 
-    // creates a new WebRTC Transport to receive media
+    // Creates a new WebRTC Transport to receive media
     // based on server's consumer transport params
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-createRecvTransport
     consumerTransport = device.createRecvTransport(params)
 
     // https://mediasoup.org/documentation/v3/communication-between-client-and-server/#producing-media
-    // this event is raised when a first call to transport.produce() is made
+    // This event is raised when a first call to transport.produce() is made
     // see connectRecvTransport() below
     consumerTransport.on('connect', async ({ dtlsParameters }, callback, errback) => {
       try {
@@ -345,7 +427,8 @@ const createRecvTransport = async () => {
         errback(error)
       }
     })
-    connectRecvTransport()
+    // We call it in new-rpoducer, we don't need it here anymore
+    // connectRecvTransport()
   })
 }
 
@@ -353,7 +436,8 @@ const resetCallSettings = () => {
   localVideo.srcObject = null
   remoteVideo.srcObject = null
   consumer = null
-  producer = null
+  producerVideo = null
+  producerAudio = null
   producerTransport = null
   consumerTransport = null
   device = undefined
@@ -361,38 +445,95 @@ const resetCallSettings = () => {
 
 const connectRecvTransport = async () => {
   console.log('connectRecvTransport');
-  // for consumer, we need to tell the server first
+  // For consumer, we need to tell the server first
   // to create a consumer based on the rtpCapabilities and consume
   // if the router can consume, it will send back a set of params as below
   await socket.emit('consume', {
     rtpCapabilities: device.rtpCapabilities,
     callId
-  }, async ({ params }) => {
+  }, async ({videoParams, audioParams}) => {
-    if (params.error) {
+    console.log(`[consume] 🟩 videoParams`, videoParams)
-      console.log('Cannot Consume')
+    console.log(`[consume] 🟩 audioParams`, audioParams)
-      return
+    console.log('[consume] 🟩 consumerTransport', consumerTransport)
-    }
-
-    // then consume with the local consumer transport
-    // which creates a consumer
-    consumer = await consumerTransport.consume({
-      id: params.id,
-      producerId: params.producerId,
-      kind: params.kind,
-      rtpParameters: params.rtpParameters
-    })
-
-    // destructure and retrieve the video track from the producer
-    const { track } = consumer
 
     let stream = new MediaStream()
-    stream.addTrack(track)
-    // stream.removeTrack(track)
-    remoteVideo.srcObject = stream
-    socket.emit('consumer-resume')
-    console.log('consumer', consumer);
 
+    // Maybe the unit does not produce video or audio, so we must only consume what is produced
+    if (videoParams) {
+      console.log('❗ Have VIDEO stream to consume');
+      stream.addTrack(await getVideoTrask(videoParams))
+    } else {
+      console.log('❗ Don\'t have VIDEO stream to consume');
+    }
+    
+    if (audioParams) {
+      console.log('❗ Have AUDIO stream to consume');
+      let audioTrack = await getAudioTrask(audioParams)
+      stream.addTrack(audioTrack)
+    } else {
+      console.log('❗ Don\'t have AUDIO stream to consume');
+    }
+
+    socket.emit('consumer-resume')
+
+    remoteVideo.srcObject = stream
+    remoteVideo.setAttribute('autoplay', true)
+
+    remoteVideo.play()
+    .then(() => {
+      console.log('remoteVideo PLAY')
     })
+    .catch((error) => {
+      console.error(`remoteVideo PLAY ERROR | ${error.message}`)
+    })
+  })
+}
+
+const getVideoTrask = async (videoParams) => {
+  consumerVideo = await consumerTransport.consume({
+    id: videoParams.id,
+    producerId: videoParams.producerId,
+    kind: videoParams.kind,
+    rtpParameters: videoParams.rtpParameters
+  }) 
+
+  return consumerVideo.track
+}
+
+const getAudioTrask = async (audioParams) => {
+  consumerAudio = await consumerTransport.consume({
+    id: audioParams.id,
+    producerId: audioParams.producerId,
+    kind: audioParams.kind,
+    rtpParameters: audioParams.rtpParameters
+  })
+
+  consumerAudio.on('transportclose', () => {
+    console.log('transport closed so consumer closed')
+  })
+
+  const audioTrack = consumerAudio.track
+
+  audioTrack.applyConstraints({
+    audio: {
+      advanced: [
+        {
+          echoCancellation: {exact: true}
+        },
+        {
+          autoGainControl: {exact: true}
+        },
+        {
+          noiseSuppression: {exact: true}
+        }, 
+        {
+          highpassFilter: {exact: true}
+        }
+      ]
+    }
+  })
+
+  return audioTrack
 }
 
 const closeCall = () => {
@@ -416,6 +557,30 @@ const closeCall = () => {
   resetCallSettings()
 }
 
+// const consume = async (kind) => {
+//   console.log(`[consume] kind: ${kind}`)
+//   console.log('createRecvTransport Consumer')
+//   await socket.emit('createWebRtcTransport', { sender: false, callId, dispatcher: true }, ({ params }) => {
+//     if (params.error) {
+//       console.log('createRecvTransport | createWebRtcTransport | Error', params.error)
+//       return
+//     }
+//     consumerTransport = device.createRecvTransport(params)
+//     consumerTransport.on('connect', async ({ dtlsParameters }, callback, errback) => {
+//       try {
+//         await socket.emit('transport-recv-connect', {
+//           dtlsParameters,
+//         })
+//         callback()
+//       } catch (error) {
+//         errback(error)
+//       }
+//     })
+    
+//     connectRecvTransport()
+//   })
+// }
+
 btnLocalVideo.addEventListener('click', getLocalStream)
-btnRecvSendTransport.addEventListener('click', goConnect)
+// btnRecvSendTransport.addEventListener('click', consume)
 btnCloseCall.addEventListener('click', closeCall)