Update server

README.md

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

app.js

@@ -90,81 +90,39 @@ worker = createWorker();
 // https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
 // list of media codecs supported by mediasoup ...
 // https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
-const mediaCodecs = 			[
+const mediaCodecs = [
   {
-    kind      : 'audio',
+    // kind: 'audio',
-    mimeType  : 'audio/opus',
+    // mimeType: 'audio/opus',
-    clockRate : 48000,
+    // clockRate: 48000,
-    channels  : 2
+    // channels: 2,
+    clockRate: 48000,
+		mimeType: "audio/opus",
+		parameters: {
+			minptime: 10,
+			useinbandfec: 1
+		},
+		payloadType: 111,
+		rtcpFeedback: [{
+      parameter: "",
+      type: "transport-cc"
+		}]
   },
   {
-    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
-    }
-  }
-//   {
-//     kind: 'audio',
-//     mimeType: 'audio/opus',
-//     clockRate: 48000,
-//     channels: 2,
-//   },
-//   {
-//     kind: 'video',
-//     mimeType: 'video/VP8',
-//     clockRate: 90000,
-//     parameters: {
-//       'x-google-start-bitrate': 1000,
-//     },
-//   },
 ];
 
 const closeCall = (callId) => {
   try {
     if (callId && videoCalls[callId]) {
-      videoCalls[callId].producerVideo?.close();
+      videoCalls[callId].producer?.close();
-      videoCalls[callId].producerAudio?.close();
+      videoCalls[callId].consumer?.close();
-      videoCalls[callId].consumerVideo?.close();
-      videoCalls[callId].consumerAudio?.close();
-
       videoCalls[callId]?.consumerTransport?.close();
       videoCalls[callId]?.producerTransport?.close();
       videoCalls[callId]?.router?.close();
@@ -283,54 +241,29 @@ peers.on('connection', async socket => {
     - The event sent by the client (PRODUCER) after successfully connecting to producerTransport
     - For the router with the id callId, we make produce on producerTransport
     - Create the handler on producer at the 'transportclose' event
-    */
+  */
-   socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
+  socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
-     try {
+    try {
-       const callId = socketDetails[socket.id];
+      const callId = socketDetails[socket.id];
-       if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
+      if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
-       
-       console.log(`[transport-produce] kind: ${kind} | socket.id: ${socket.id} | callId: ${callId}`);
-       console.log('kind', kind);
-       console.log('rtpParameters', rtpParameters);
 
-      if (kind === 'video') {
+      console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
-        videoCalls[callId].producerVideo = await videoCalls[callId].producerTransport.produce({
+      videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
-          kind,
+        kind,
-          rtpParameters,
+        rtpParameters,
-        });
+      });
-      
+      console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producer.id} | kind: ${videoCalls[callId].producer.kind}`);
+  
+      videoCalls[callId].producer.on('transportclose', () => {
+        const callId = socketDetails[socket.id];
+        console.log('transport for this producer closed', callId)
+        closeCall(callId);
+      });
 
-        console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producerVideo.id} | kind: ${videoCalls[callId].producerVideo.kind}`);
+      // Send back to the client the Producer's id
-    
+      callback && callback({
-        videoCalls[callId].producerVideo.on('transportclose', () => {
+        id: videoCalls[callId].producer.id
-          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].producerVideo.id
-        });
-      } else if (kind === 'audio') {
-        videoCalls[callId].producerAudio = await videoCalls[callId].producerTransport.produce({
-          kind,
-          rtpParameters,
-        });
-
-        console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producerAudio.id} | kind: ${videoCalls[callId].producerAudio.kind}`);
-    
-        videoCalls[callId].producerAudio.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].producerAudio.id
-        });
-      }
     } catch (error) {
       console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
     }
@@ -359,36 +292,48 @@ peers.on('connection', async socket => {
   */
   socket.on('consume', async ({ rtpCapabilities }, callback) => {
     try {
-      console.log(`[consume] rtpCapabilities: ${JSON.stringify(rtpCapabilities)}`);
-
       const callId = socketDetails[socket.id];
       console.log('[consume] callId', callId);
 
-      const canConsumeVideo = !!videoCalls[callId].producerVideo && !!videoCalls[callId].router.canConsume({
+      // Check if the router can consume the specified producer
-        producerId: videoCalls[callId].producerVideo.id,
+      if (videoCalls[callId].router.canConsume({
+        producerId: videoCalls[callId].producer.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,
+        });
 
-      const canConsumeAudio = !!videoCalls[callId].producerAudio && !!videoCalls[callId].router.canConsume({
+        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
-        producerId: videoCalls[callId].producerAudio.id,
+        videoCalls[callId].consumer.on('transportclose', () => {
-        rtpCapabilities
+          const callId = socketDetails[socket.id];
-      })
+          console.log('transport close from consumer', callId);
+          closeCall(callId);
+        });
 
-      console.log('[consume] canConsumeVideo', canConsumeVideo);
+        // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
-      console.log('[consume] canConsumeAudio', canConsumeAudio);
+        videoCalls[callId].consumer.on('producerclose', () => {
+          const callId = socketDetails[socket.id];
+          console.log('producer of consumer closed', callId);
+          closeCall(callId);
+        });
 
-      if (canConsumeVideo && !canConsumeAudio) {
+        // From the consumer extract the following params to send back to the Client
-        console.log('1');
+        const params = {
-        const videoParams = await consumeVideo(callId, rtpCapabilities)
+          id: videoCalls[callId].consumer.id,
-        console.log('videoParams', videoParams);
+          producerId: videoCalls[callId].producer.id,
-        callback({ videoParams, audioParams: null });
+          kind: videoCalls[callId].consumer.kind,
-      } else if (canConsumeVideo && canConsumeAudio) {
+          rtpParameters: videoCalls[callId].consumer.rtpParameters,
-        console.log('2');
+        };
-        const videoParams = await consumeVideo(callId, rtpCapabilities)
+
-        const audioParams = await consumeAudio(callId, rtpCapabilities)
+        // Send the parameters to the client
-        callback({ videoParams, audioParams });
+        callback({ params });
       } else {
-        console.log(`[consume] Can't consume | callId ${callId}`);
+        console.log(`[canConsume] Can't consume | callId ${callId}`);
         callback(null);
       }
     } catch (error) {
@@ -405,71 +350,13 @@ peers.on('connection', async socket => {
     try {
       const callId = socketDetails[socket.id];
       console.log(`[consumer-resume] callId ${callId}`)
-      await videoCalls[callId].consumerVideo.resume();
+      await videoCalls[callId].consumer.resume();
-      await videoCalls[callId].consumerAudio.resume();
     } catch (error) {
       console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
     }
   });
 });
 
-const consumeVideo = async (callId, rtpCapabilities) => {
-  videoCalls[callId].consumerVideo = await videoCalls[callId].consumerTransport.consume({
-    producerId: videoCalls[callId].producerVideo.id,
-    rtpCapabilities,
-    paused: true,
-  });
-
-  // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
-  videoCalls[callId].consumerVideo.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].consumerVideo.on('producerclose', () => {
-    const callId = socketDetails[socket.id];
-    console.log('producer of consumer closed', callId);
-    closeCall(callId);
-  });
-
-  return {
-    id: videoCalls[callId].consumerVideo.id,
-    producerId: videoCalls[callId].producerVideo.id,
-    kind: 'video',
-    rtpParameters: videoCalls[callId].consumerVideo.rtpParameters,
-  }
-}
-
-const consumeAudio = async (callId, rtpCapabilities) => {
-  videoCalls[callId].consumerAudio = await videoCalls[callId].consumerTransport.consume({
-    producerId: videoCalls[callId].producerAudio.id,
-    rtpCapabilities,
-    paused: true,
-  });
-
-  // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
-  videoCalls[callId].consumerAudio.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].consumerAudio.on('producerclose', () => {
-    const callId = socketDetails[socket.id];
-    console.log('producer of consumer closed', callId);
-    closeCall(callId);
-  });
-  return {
-    id: videoCalls[callId].consumerAudio.id,
-    producerId: videoCalls[callId].producerAudio.id,
-    kind: 'audio',
-    rtpParameters: videoCalls[callId].consumerAudio.rtpParameters,
-  }
-}
-
 /*
   - 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 
@@ -517,7 +404,6 @@ 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

@@ -43,4 +43,6 @@ fi
 
 ## POST BUILD
 
-cd -
+cd -
+
+

public/bundle.js

@@ -20373,36 +20373,6 @@ console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId',
 console.log('🟩 config', config)
 
 let socket, hub
-let device
-let rtpCapabilities
-let producerTransport
-let consumerTransport
-let producerVideo
-let producerAudio
-let consumer
-let originAssetId
-
-// 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)
@@ -20476,10 +20446,66 @@ setTimeout(() => {
     connectToMediasoup()
   }
 
-}, 1600);
+}, 2000);
+
+
+let device
+let rtpCapabilities
+let producerTransport
+let consumerTransport
+let producerVideo
+let producerAudio
+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 videoParams = {
+  // mediasoup params
+  encodings: [
+    {
+      maxBitrate: 100000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 300000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 900000,
+      scalabilityMode: 'S1T3',
+    },
+  ],
+  // https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerCodecOptions
+  codecOptions: {
+    videoGoogleStartBitrate: 1000
+  }
+}
+
+let audioParams = {
+  // mediasoup params
+  encodings: [
+    {
+      maxBitrate: 100000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 300000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 900000,
+      scalabilityMode: 'S1T3',
+    },
+  ],
+  codecOptions: {
+    opusStereo: true
+  }
+}
 
 const streamSuccess = (stream) => {
-  console.log('[streamSuccess] device', device);
   localVideo.srcObject = stream
   console.log('stream', stream);
   const videoTrack = stream.getVideoTracks()[0]
@@ -20504,23 +20530,20 @@ const getLocalStream = () => {
   navigator.mediaDevices.getUserMedia({
     audio: true,
     video: {
-      qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
+      width: {
-      vga  : { width: { ideal: 640 }, height: { ideal: 480 } },
+        min: 640,
-      hd   : { width: { ideal: 1280 }, height: { ideal: 720 } }
+        max: 1920,
+      },
+      height: {
+        min: 400,
+        max: 1080,
+      }
     }
   })
   .then(streamSuccess)
   .catch(error => {
     console.log(error.message)
   })
-
-  navigator.permissions.query(
-    { name: 'microphone' }
-  ).then((permissionStatus) =>{
-    console.log('🟨 [PERMISSION] permissionStatus', permissionStatus); // granted, denied, prompt
-    // It will block the code from execution and display "Permission denied" if we don't have microphone permissions
-  })
-
 }
 
 const goConnect = () => {
@@ -20537,6 +20560,7 @@ const goCreateTransport = () => {
 // server side to send/recive media
 const createDevice = async () => {
   try {
+    console.log('[createDevice]');
     device = new mediasoupClient.Device()
 
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@@ -20547,8 +20571,7 @@ const createDevice = async () => {
     })
 
     console.log('Device RTP Capabilities', device.rtpCapabilities)
-    console.log('[createDevice] device', device);
+
-    
     // once the device loads, create transport
     goCreateTransport()
 
@@ -20580,17 +20603,16 @@ 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 }, (value) => {
+  socket.emit('createWebRtcTransport', { sender: true, callId }, ({ params }) => {
-
-    console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
-
-    const params = value.params;
     // The server sends back params needed 
     // to create Send Transport on the client side
     if (params.error) {
       console.log(params.error)
       return
     }
+
+    console.log('[createWebRtcTransport] params', 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
@@ -20619,7 +20641,7 @@ const createSendTransport = () => {
       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', ...)
@@ -20645,40 +20667,33 @@ const connectSendTransport = async () => {
 
   console.log('[connectSendTransport] producerTransport');
 
-  // We now call produce() to instruct the producer transport
+  // 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
-  
-  // Produce video
   producerVideo = await producerTransport.produce(videoParams)
-  console.log('videoParams', videoParams);
   console.log('producerVideo', producerVideo);
-
   producerVideo.on('trackended', () => {
     console.log('track ended')
     // close video track
-  })  
+  })
 
   producerVideo.on('transportclose', () => {
     console.log('transport ended')
     // close video track
-  })    
-  
-  // Produce audio
-  producerAudio = await producerTransport.produce(audioParams)
-  console.log('audioParams', audioParams);
-  console.log('producerAudio', producerAudio);
-
-  producerAudio.on('trackended', () => {
-    console.log('track ended')
-    // close audio track
   })
 
-  producerAudio.on('transportclose', () => {
+  // producerAudio = await producerTransport.produce(audioParams)
-    console.log('transport ended')
+  // console.log('producerAudio', producerAudio);
-    // close audio track
+  // producerAudio.on('trackended', () => {
-  })
+  //   console.log('track ended')
+  //   // close video track
+  // })
+
+  // producerAudio.on('transportclose', () => {
+  //   console.log('transport ended')
+  //   // close video track
+  // })
 
   const answer = {
     origin_asset_id: ASSET_ID,
@@ -20688,7 +20703,7 @@ const connectSendTransport = async () => {
     origin_asset_type_name: ASSET_TYPE,
     origin_asset_name: ASSET_NAME,
     video_call_id: callId,
-    answer: 'accepted', // answer: accepted/rejected
+    answer: 'accepted', // answer: 'rejected'
   };
   console.log('SEND answer', answer);
 
@@ -20704,7 +20719,7 @@ 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 
@@ -20716,13 +20731,13 @@ const createRecvTransport = async () => {
 
     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 {
@@ -20756,7 +20771,7 @@ 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', {
@@ -20768,7 +20783,7 @@ const connectRecvTransport = async () => {
       return
     }
 
-    // Then consume with the local consumer transport
+    // then consume with the local consumer transport
     // which creates a consumer
     consumer = await consumerTransport.consume({
       id: params.id,
@@ -20811,7 +20826,6 @@ const closeCall = () => {
   resetCallSettings()
 }
 
-
 btnLocalVideo.addEventListener('click', getLocalStream)
 btnRecvSendTransport.addEventListener('click', goConnect)
 btnCloseCall.addEventListener('click', closeCall)

public/index.html

@@ -43,7 +43,7 @@
                     <tr>
                         <td>
                             <div id="sharedBtns">
-                                <video id="localVideo" autoplay class="video" muted></video>
+                                <video id="localVideo" autoplay class="video" ></video>
                             </div>
                         </td>
                         <td>

public/index.js

@@ -15,36 +15,6 @@ console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId',
 console.log('🟩 config', config)
 
 let socket, hub
-let device
-let rtpCapabilities
-let producerTransport
-let consumerTransport
-let producerVideo
-let producerAudio
-let consumer
-let originAssetId
-
-// 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)
@@ -118,10 +88,66 @@ setTimeout(() => {
     connectToMediasoup()
   }
 
-}, 1600);
+}, 2000);
+
+
+let device
+let rtpCapabilities
+let producerTransport
+let consumerTransport
+let producerVideo
+let producerAudio
+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 videoParams = {
+  // mediasoup params
+  encodings: [
+    {
+      maxBitrate: 100000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 300000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 900000,
+      scalabilityMode: 'S1T3',
+    },
+  ],
+  // https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerCodecOptions
+  codecOptions: {
+    videoGoogleStartBitrate: 1000
+  }
+}
+
+let audioParams = {
+  // mediasoup params
+  encodings: [
+    {
+      maxBitrate: 100000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 300000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 900000,
+      scalabilityMode: 'S1T3',
+    },
+  ],
+  codecOptions: {
+    opusStereo: true
+  }
+}
 
 const streamSuccess = (stream) => {
-  console.log('[streamSuccess] device', device);
   localVideo.srcObject = stream
   console.log('stream', stream);
   const videoTrack = stream.getVideoTracks()[0]
@@ -146,23 +172,20 @@ const getLocalStream = () => {
   navigator.mediaDevices.getUserMedia({
     audio: true,
     video: {
-      qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
+      width: {
-      vga  : { width: { ideal: 640 }, height: { ideal: 480 } },
+        min: 640,
-      hd   : { width: { ideal: 1280 }, height: { ideal: 720 } }
+        max: 1920,
+      },
+      height: {
+        min: 400,
+        max: 1080,
+      }
     }
   })
   .then(streamSuccess)
   .catch(error => {
     console.log(error.message)
   })
-
-  navigator.permissions.query(
-    { name: 'microphone' }
-  ).then((permissionStatus) =>{
-    console.log('🟨 [PERMISSION] permissionStatus', permissionStatus); // granted, denied, prompt
-    // It will block the code from execution and display "Permission denied" if we don't have microphone permissions
-  })
-
 }
 
 const goConnect = () => {
@@ -179,6 +202,7 @@ const goCreateTransport = () => {
 // server side to send/recive media
 const createDevice = async () => {
   try {
+    console.log('[createDevice]');
     device = new mediasoupClient.Device()
 
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@@ -189,8 +213,7 @@ const createDevice = async () => {
     })
 
     console.log('Device RTP Capabilities', device.rtpCapabilities)
-    console.log('[createDevice] device', device);
+
-    
     // once the device loads, create transport
     goCreateTransport()
 
@@ -222,17 +245,16 @@ 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 }, (value) => {
+  socket.emit('createWebRtcTransport', { sender: true, callId }, ({ params }) => {
-
-    console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
-
-    const params = value.params;
     // The server sends back params needed 
     // to create Send Transport on the client side
     if (params.error) {
       console.log(params.error)
       return
     }
+
+    console.log('[createWebRtcTransport] params', 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
@@ -261,7 +283,7 @@ const createSendTransport = () => {
       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', ...)
@@ -287,40 +309,33 @@ const connectSendTransport = async () => {
 
   console.log('[connectSendTransport] producerTransport');
 
-  // We now call produce() to instruct the producer transport
+  // 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
-  
-  // Produce video
   producerVideo = await producerTransport.produce(videoParams)
-  console.log('videoParams', videoParams);
   console.log('producerVideo', producerVideo);
-
   producerVideo.on('trackended', () => {
     console.log('track ended')
     // close video track
-  })  
+  })
 
   producerVideo.on('transportclose', () => {
     console.log('transport ended')
     // close video track
-  })    
-  
-  // Produce audio
-  producerAudio = await producerTransport.produce(audioParams)
-  console.log('audioParams', audioParams);
-  console.log('producerAudio', producerAudio);
-
-  producerAudio.on('trackended', () => {
-    console.log('track ended')
-    // close audio track
   })
 
-  producerAudio.on('transportclose', () => {
+  // producerAudio = await producerTransport.produce(audioParams)
-    console.log('transport ended')
+  // console.log('producerAudio', producerAudio);
-    // close audio track
+  // producerAudio.on('trackended', () => {
-  })
+  //   console.log('track ended')
+  //   // close video track
+  // })
+
+  // producerAudio.on('transportclose', () => {
+  //   console.log('transport ended')
+  //   // close video track
+  // })
 
   const answer = {
     origin_asset_id: ASSET_ID,
@@ -330,7 +345,7 @@ const connectSendTransport = async () => {
     origin_asset_type_name: ASSET_TYPE,
     origin_asset_name: ASSET_NAME,
     video_call_id: callId,
-    answer: 'accepted', // answer: accepted/rejected
+    answer: 'accepted', // answer: 'rejected'
   };
   console.log('SEND answer', answer);
 
@@ -346,7 +361,7 @@ 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 
@@ -358,13 +373,13 @@ const createRecvTransport = async () => {
 
     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 {
@@ -398,7 +413,7 @@ 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', {
@@ -410,7 +425,7 @@ const connectRecvTransport = async () => {
       return
     }
 
-    // Then consume with the local consumer transport
+    // then consume with the local consumer transport
     // which creates a consumer
     consumer = await consumerTransport.consume({
       id: params.id,
@@ -453,7 +468,6 @@ const closeCall = () => {
   resetCallSettings()
 }
 
-
 btnLocalVideo.addEventListener('click', getLocalStream)
 btnRecvSendTransport.addEventListener('click', goConnect)
 btnCloseCall.addEventListener('click', closeCall)