LINXD-2270: Removed console logs; Update commented code

Reviewed-on: #17
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>

.env

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

.gitignore

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

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

@@ -5,7 +5,7 @@ const app = express();
 const Server = require('socket.io');
 const path = require('node:path');
 const fs = require('node:fs');
-let https = require('https');
+let https;
 try {
   https = require('node:https');
 } catch (err) {
@@ -13,62 +13,59 @@ try {
 }
 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
+ *  initiatorSockerId
+ *  receiverSocketId
  * }
  * 
  **/
-let videoCalls = {}
+let videoCalls = {};
-let socketDetails = {}
+let socketDetails = {};
 
 app.get('/', (_req, res) => {
   res.send('Hello from mediasoup app!')
-})
+});
 
-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)
+  console.log('Video server listening on port:', process.env.PORT);
-})
+});
 
-const peers = io.of('/')
+const peers = io.of('/');
 
 const createWorker = async () => {
   try {
     worker = await mediasoup.createWorker({
-      rtcMinPort: 2000,
+      rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
-      rtcMaxPort: 2020,
+      rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
     })
     console.log(`[createWorker] worker pid ${worker.pid}`);
   
@@ -84,7 +81,7 @@ const createWorker = async () => {
 }
 
 // We create a Worker as soon as our application starts
-worker = createWorker()
+worker = createWorker();
 
 // This is an Array of RtpCapabilities
 // https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
@@ -92,28 +89,67 @@ worker = createWorker()
 // https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
 const mediaCodecs = 			[
   {
-    kind: 'audio',
+    kind      : 'audio',
-    mimeType: 'audio/opus',
+    mimeType  : 'audio/opus',
-    clockRate: 48000,
+    clockRate : 48000,
-    channels: 2,
+    channels  : 2
   },
   {
-    kind: 'video',
+    kind       : 'video',
-    mimeType: 'video/VP8',
+    mimeType   : 'video/VP8',
-    clockRate: 90000,
+    clockRate  : 90000,
-    parameters: {
+    parameters :
-      'x-google-start-bitrate': 1000,
+    {
+      '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 (videoCalls[callId]) {
+    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 {
@@ -124,16 +160,6 @@ const closeCall = (callId) => {
   }
 }
 
-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
@@ -160,27 +186,38 @@ 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 }) }
           console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
+          videoCalls[callId].receiverSocketId = socket.id
+        } else {
+          videoCalls[callId].initiatorSockerId = socket.id
         }
         socketDetails[socket.id] = callId;
-        getRtpCapabilities(callId, callback);
+        // rtpCapabilities is set for callback
+        console.log('[getRtpCapabilities] callId', callId);
+        callbackResponse = { 
+          rtpCapabilities :videoCalls[callId].router.rtpCapabilities
+        };
       } else {
         console.log(`[createRoom] missing callId ${callId}`);
       }
     } catch (error) {
       console.log(`ERROR | createRoom | callId ${callId} | ${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)
@@ -189,22 +226,26 @@ 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}`);
+      callback(error);
     }
   });
 
@@ -217,16 +258,20 @@ 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 });
+      if (!isInitiator(callId, socket.id)) {
+        await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
+      } else {
+        await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters });
+      }
     } catch (error) {
       console.log(`ERROR | transport-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
     }
   });
 
   /*
-    - The event sent by the client (PRODUCER) after successfully connecting to producerTransport
+    - 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) => {
@@ -234,23 +279,83 @@ 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] kind: ${kind} | socket: ${socket.id} | callId: ${callId}`);
-      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', () => {
+          console.log(`[transport-produce] receiverVideoProducer Producer ID: ${videoCalls[callId].receiverVideoProducer.id} | kind: ${videoCalls[callId].receiverVideoProducer.kind}`);
+
+          videoCalls[callId].receiverVideoProducer.on('transportclose', () => {
             const callId = socketDetails[socket.id];
             console.log('transport for this producer closed', callId)
             closeCall(callId);
           });
 
           // Send back to the client the Producer's id
-      // callback({
+          callback && callback({
-      //   id: videoCalls[callId].producer.id
+            id: videoCalls[callId].receiverVideoProducer.id
-      // });
+          });
+        } else {
+          videoCalls[callId].initiatorVideoProducer = await videoCalls[callId].initiatorProducerTransport.produce({
+            kind,
+            rtpParameters,
+          });
+
+          console.log(`[transport-produce] initiatorVideoProducer Producer ID: ${videoCalls[callId].initiatorVideoProducer.id} | kind: ${videoCalls[callId].initiatorVideoProducer.kind}`);
+
+          videoCalls[callId].initiatorVideoProducer.on('transportclose', () => {
+            const callId = socketDetails[socket.id];
+            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,
+          });
+
+          console.log(`[transport-produce] receiverAudioProducer Producer ID: ${videoCalls[callId].receiverAudioProducer.id} | kind: ${videoCalls[callId].receiverAudioProducer.kind}`);
+      
+          videoCalls[callId].receiverAudioProducer.on('transportclose', () => {
+            const callId = socketDetails[socket.id];
+            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,
+          });
+
+          console.log(`[transport-produce] initiatorAudioProducer Producer ID: ${videoCalls[callId].initiatorAudioProducer.id} | kind: ${videoCalls[callId].initiatorAudioProducer.kind}`);
+      
+          videoCalls[callId].initiatorAudioProducer.on('transportclose', () => {
+            const callId = socketDetails[socket.id];
+            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
+          });
+        }
+      }
     } catch (error) {
       console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
     }
@@ -263,8 +368,13 @@ 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 });
+      // 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.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
     }
@@ -280,47 +390,48 @@ peers.on('connection', async socket => {
   socket.on('consume', async ({ rtpCapabilities }, callback) => {
     try {
       const callId = socketDetails[socket.id];
+      console.log(`[consume] socket ${socket.id} | callId ${callId} | rtpCapabilities: ${JSON.stringify(rtpCapabilities)}`);
+
       console.log('[consume] callId', callId);
-
+      let canConsumeVideo, canConsumeAudio;
-      // Check if the router can consume the specified producer
+      if (isInitiator(callId, socket.id)) {
-      if (videoCalls[callId].router.canConsume({
+        canConsumeVideo = !!videoCalls[callId].receiverVideoProducer && !!videoCalls[callId].router.canConsume({
-        producerId: videoCalls[callId].producer.id,
+          producerId: videoCalls[callId].receiverVideoProducer.id,
+          rtpCapabilities
+        });
+        canConsumeAudio = !!videoCalls[callId].receiverAudioProducer && !!videoCalls[callId].router.canConsume({
+          producerId: videoCalls[callId].receiverAudioProducer.id,
           rtpCapabilities
-      })) {
-        console.log('[consume] Can consume', callId);
-        // Transport can now consume and return a consumer
-        videoCalls[callId].consumer = await videoCalls[callId].consumerTransport.consume({
-          producerId: videoCalls[callId].producer.id,
-          rtpCapabilities,
-          paused: true,
         });
 
-        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
-        videoCalls[callId].consumer.on('transportclose', () => {
-          const callId = socketDetails[socket.id];
-          console.log('transport close from consumer', callId);
-          closeCall();
-        });
-
-        // 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();
-        });
-
-        // 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}`);
+        canConsumeVideo = !!videoCalls[callId].initiatorVideoProducer && !!videoCalls[callId].router.canConsume({
+          producerId: videoCalls[callId].initiatorVideoProducer.id,
+          rtpCapabilities
+        });
+
+        canConsumeAudio = !!videoCalls[callId].initiatorAudioProducer && !!videoCalls[callId].router.canConsume({
+          producerId: videoCalls[callId].initiatorAudioProducer.id,
+          rtpCapabilities
+        });
+      }
+      console.log('[consume] canConsumeVideo', canConsumeVideo);
+      console.log('[consume] canConsumeAudio', canConsumeAudio);
+
+      if (canConsumeVideo && !canConsumeAudio) {
+        const videoParams = await consumeVideo(callId, socket.id, rtpCapabilities)
+        callback({ videoParams, audioParams: null });
+      } else if (canConsumeVideo && canConsumeAudio) {
+        const videoParams = await consumeVideo(callId, socket.id, rtpCapabilities)
+        const audioParams = await consumeAudio(callId, socket.id, rtpCapabilities)
+        callback({ videoParams, audioParams });
+      } else if (!canConsumeVideo && canConsumeAudio) {
+        const audioParams = await consumeAudio(callId, socket.id, rtpCapabilities)
+        const data = { videoParams: null, audioParams };
+        callback(data);
+      } else {
+        console.log(`[consume] Can't consume | callId ${callId}`);
+        callback(null);
       }
     } catch (error) {
       console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
@@ -336,13 +447,139 @@ peers.on('connection', async socket => {
     try {
       const callId = socketDetails[socket.id];
       console.log(`[consumer-resume] callId ${callId}`)
-      await videoCalls[callId].consumer.resume();
+
+      if (isInitiator(callId, socket.id)) {
+        console.log(`[consumer-resume] isInitiator true`);
+        await videoCalls[callId].initiatorConsumerVideo.resume();
+        await videoCalls[callId].initiatorConsumerAudio.resume();
+      } else {
+        console.log(`[consumer-resume] isInitiator false`);
+        (videoCalls[callId].receiverConsumerVideo) && await videoCalls[callId].receiverConsumerVideo.resume();
+        (videoCalls[callId].receiverConsumerVideo) && await videoCalls[callId].receiverConsumerAudio.resume();
+      }
     } catch (error) {
       console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
     }
   });
 });
 
+const consumeVideo = async (callId, socketId, rtpCapabilities) => {
+
+  if(isInitiator(callId, socketId)) {
+    videoCalls[callId].initiatorConsumerVideo = await videoCalls[callId].initiatorConsumerTransport.consume({
+      producerId: videoCalls[callId].receiverVideoProducer.id,
+      rtpCapabilities,
+      paused: true,
+    });
+
+    // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
+    videoCalls[callId].initiatorConsumerVideo.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].initiatorConsumerVideo.on('producerclose', () => {
+      const callId = socketDetails[socket.id];
+      console.log('producer of consumer closed', callId);
+      closeCall(callId);
+    });
+
+    return {
+      id: videoCalls[callId].initiatorConsumerVideo.id,
+      producerId: videoCalls[callId].receiverVideoProducer.id,
+      kind: 'video',
+      rtpParameters: videoCalls[callId].initiatorConsumerVideo.rtpParameters,
+    }
+  } else {
+    videoCalls[callId].receiverConsumerVideo = await videoCalls[callId].receiverConsumerTransport.consume({
+      producerId: videoCalls[callId].initiatorVideoProducer.id,
+      rtpCapabilities,
+      paused: true,
+    });
+
+    videoCalls[callId].receiverConsumerVideo.on('transportclose', () => {
+      const callId = socketDetails[socket.id];
+      console.log('transport close from consumer', callId);
+      closeCall(callId);
+    });
+
+    videoCalls[callId].receiverConsumerVideo.on('producerclose', () => {
+      const callId = socketDetails[socket.id];
+      console.log('producer of consumer closed', callId);
+      closeCall(callId);
+    });
+
+    return {
+      id: videoCalls[callId].receiverConsumerVideo.id,
+      producerId: videoCalls[callId].initiatorVideoProducer.id,
+      kind: 'video',
+      rtpParameters: videoCalls[callId].receiverConsumerVideo.rtpParameters,
+    }
+  }
+}
+
+const consumeAudio = async (callId, socketId, rtpCapabilities) => {
+  if(isInitiator(callId, socketId)) {
+    videoCalls[callId].initiatorConsumerAudio = await videoCalls[callId].initiatorConsumerTransport.consume({
+      producerId: videoCalls[callId].receiverAudioProducer.id,
+      rtpCapabilities,
+      paused: true,
+    });
+
+    // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
+    videoCalls[callId].initiatorConsumerAudio.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].initiatorConsumerAudio.on('producerclose', () => {
+      const callId = socketDetails[socket.id];
+      console.log('producer of consumer closed', callId);
+      closeCall(callId);
+    });
+
+    return {
+      id: videoCalls[callId].initiatorConsumerAudio.id,
+      producerId: videoCalls[callId].receiverAudioProducer.id,
+      kind: 'audio',
+      rtpParameters: videoCalls[callId].initiatorConsumerAudio.rtpParameters,
+    }
+  } else {
+    videoCalls[callId].receiverConsumerAudio = await videoCalls[callId].receiverConsumerTransport.consume({
+      producerId: videoCalls[callId].initiatorAudioProducer.id,
+      rtpCapabilities,
+      paused: true,
+    });
+
+    videoCalls[callId].receiverConsumerAudio.on('transportclose', () => {
+      const callId = socketDetails[socket.id];
+      console.log('transport close from consumer', callId);
+      closeCall(callId);
+    });
+
+    videoCalls[callId].receiverConsumerAudio.on('producerclose', () => {
+      const callId = socketDetails[socket.id];
+      console.log('producer of consumer closed', callId);
+      closeCall(callId);
+    });
+
+    return {
+      id: videoCalls[callId].receiverConsumerAudio.id,
+      producerId: videoCalls[callId].initiatorAudioProducer.id,
+      kind: 'audio',
+      rtpParameters: videoCalls[callId].receiverConsumerAudio.rtpParameters,
+    }
+  }
+}
+
+const isInitiator = (callId, socketId) => {
+  return (videoCalls[callId].initiatorSockerId === 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 
@@ -390,6 +627,7 @@ const createWebRtcTransportLayer = async (callId, callback) => {
       dtlsParameters: transport.dtlsParameters,
     };
 
+    console.log('[createWebRtcTransportLayer] callback params', params);
     // Send back to the client the params
     callback({ params });
 

build.sh

@@ -0,0 +1,46 @@
+#/!bin/bash
+## 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
+# Install dependencies
+#npm install
+
+## PROJECT NEEDS
+echo "Building app... from $(git rev-parse --abbrev-ref HEAD)"
+#npm run-script build
+cp -r {.env,app.js,package.json,server,public} dist/
+
+#Add version control for pm2
+cd dist
+#Add version control for pm2
+version=$(git describe)
+file_pkg="package.json"
+key="  \"version\": \""
+
+count=$(echo ${version%%-*} | grep -o "\." | wc -l)
+if (( $count > 1 )); then
+        version=${version%%-*}
+else
+        version="${version%%-*}.0"
+fi
+if [ -f "$file_pkg" ] && [ ! -z "$version" ]; then
+  version="  \"version\": \"$version\","
+  sed -i "s|^.*$key.*|${version//\//\\/}|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
+
+## 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>
@@ -61,33 +76,10 @@
                         <td>
                             <div id="sharedBtns">
                                 <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

@@ -10,12 +10,75 @@ const ASSET_NAME = urlParams.get('assetName') || null;
 const ASSET_TYPE = urlParams.get('assetType') || null;
 let callId = parseInt(urlParams.get('callId')) || null;
 const IS_PRODUCER = urlParams.get('producer') === 'true' ? true : false
+let remoteVideo = document.getElementById('remoteVideo')
+remoteVideo.defaultMuted = true
+let produceAudio = false
+
 console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId', callId, ' | IS_PRODUCER', IS_PRODUCER)
 
-let socket 
+console.log('🟩 config', config)
-hub = io(config.hubAddress)
 
-const connectToMediasoup = () => {
+produceAudioSelector = document.getElementById('produceAudio');
+produceAudioSelector.addEventListener('change', e => {
+    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 = () => {
   
     socket = io(config.mediasoupAddress, {
       reconnection: true,
@@ -32,11 +95,11 @@ const connectToMediasoup = () => {
       }
       if (IS_PRODUCER && urlParams.get('testing') === 'true') { getLocalStream() }
     })
-}
+  }
   
-if (IS_PRODUCER === true) {
+  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(
@@ -80,77 +143,55 @@ if (IS_PRODUCER === true) {
     hub.on('disconnect', () => {
       console.log('disconnect')
     })
-} else {
+  } else {
     connectToMediasoup()
-}
-
-let device
-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
   }
-}
+
+}, 1600);
 
 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 +208,6 @@ 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 +218,7 @@ const createDevice = async () => {
     })
 
     console.log('Device RTP Capabilities', device.rtpCapabilities)
+    console.log('[createDevice] device', device);
     
     // once the device loads, create transport
     goCreateTransport()
@@ -207,18 +248,20 @@ 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 +287,10 @@ 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 +313,46 @@ 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 +361,7 @@ 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);
 
@@ -310,7 +377,7 @@ const connectSendTransport = async () => {
 
 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 +387,15 @@ 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 {
@@ -353,7 +420,8 @@ const resetCallSettings = () => {
   localVideo.srcObject = null
   remoteVideo.srcObject = null
   consumer = null
-  producer = null
+  producerVideo = null
+  producerAudio = null
   producerTransport = null
   consumerTransport = null
   device = undefined
@@ -361,38 +429,99 @@ 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) => {
+      displayError(`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
+  }) 
+
+  consumerVideo.on('transportclose', () => {
+    console.log('transport closed so consumer closed')
+  })
+
+  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 +545,30 @@ const closeCall = () => {
   resetCallSettings()
 }
 
+const consume = async () => {
+  console.log('[consume]')
+  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)