LH-265: Update client

README.md

@@ -22,17 +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

@@ -92,26 +92,79 @@ worker = createWorker();
 // https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
 const mediaCodecs = 			[
   {
-    kind: 'audio',
-    mimeType: 'audio/opus',
-    clockRate: 48000,
-    channels: 2,
+    kind      : 'audio',
+    mimeType  : 'audio/opus',
+    clockRate : 48000,
+    channels  : 2
   },
   {
-    kind: 'video',
-    mimeType: 'video/VP8',
-    clockRate: 90000,
-    parameters: {
-      'x-google-start-bitrate': 1000,
+    kind       : 'video',
+    mimeType   : 'video/VP8',
+    clockRate  : 90000,
+    parameters :
+    {
+      'x-google-start-bitrate' : 1000
     },
+    channels  : 2
   },
+  {
+    kind       : 'video',
+    mimeType   : 'video/VP9',
+    clockRate  : 90000,
+    parameters :
+    {
+      'profile-id'             : 2,
+      'x-google-start-bitrate' : 1000
+    }
+  },
+  {
+    kind       : 'video',
+    mimeType   : 'video/h264',
+    clockRate  : 90000,
+    parameters :
+    {
+      'packetization-mode'      : 1,
+      'profile-level-id'        : '4d0032',
+      'level-asymmetry-allowed' : 1,
+      'x-google-start-bitrate'  : 1000
+    }
+  },
+  {
+    kind       : 'video',
+    mimeType   : 'video/h264',
+    clockRate  : 90000,
+    parameters :
+    {
+      'packetization-mode'      : 1,
+      'profile-level-id'        : '42e01f',
+      'level-asymmetry-allowed' : 1,
+      'x-google-start-bitrate'  : 1000
+    }
+  }
+//   {
+//     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].producer?.close();
-      videoCalls[callId].consumer?.close();
+      videoCalls[callId].producerVideo?.close();
+      videoCalls[callId].producerAudio?.close();
+      videoCalls[callId].consumerVideo?.close();
+      videoCalls[callId].consumerAudio?.close();
+
       videoCalls[callId]?.consumerTransport?.close();
       videoCalls[callId]?.producerTransport?.close();
       videoCalls[callId]?.router?.close();
@@ -236,14 +289,20 @@ 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);
-      videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
+       console.log(`[transport-produce] kind: ${kind} | socket.id: ${socket.id} | callId: ${callId}`);
+       console.log('kind', kind);
+       console.log('rtpParameters', rtpParameters);
+
+      if (kind === 'video') {
+        videoCalls[callId].producerVideo = await videoCalls[callId].producerTransport.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] Producer ID: ${videoCalls[callId].producerVideo.id} | kind: ${videoCalls[callId].producerVideo.kind}`);
+    
+        videoCalls[callId].producerVideo.on('transportclose', () => {
           const callId = socketDetails[socket.id];
           console.log('transport for this producer closed', callId)
           closeCall(callId);
@@ -251,8 +310,27 @@ peers.on('connection', async socket => {
 
         // Send back to the client the Producer's id
         callback && callback({
-        id: videoCalls[callId].producer.id
+          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}`);
     }
@@ -281,48 +359,36 @@ 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);
 
-      // Check if the router can consume the specified producer
-      if (videoCalls[callId].router.canConsume({
-        producerId: videoCalls[callId].producer.id,
+      const canConsumeVideo = !!videoCalls[callId].producerVideo && !!videoCalls[callId].router.canConsume({
+        producerId: videoCalls[callId].producerVideo.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(callId);
-        });
+      const canConsumeAudio = !!videoCalls[callId].producerAudio && !!videoCalls[callId].router.canConsume({
+        producerId: videoCalls[callId].producerAudio.id,
+        rtpCapabilities
+      })
 
-        // 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);
-        });
+      console.log('[consume] canConsumeVideo', canConsumeVideo);
+      console.log('[consume] canConsumeAudio', canConsumeAudio);
 
-        // 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 });
+      if (canConsumeVideo && !canConsumeAudio) {
+        console.log('1');
+        const videoParams = await consumeVideo(callId, rtpCapabilities)
+        console.log('videoParams', videoParams);
+        callback({ videoParams, audioParams: null });
+      } else if (canConsumeVideo && canConsumeAudio) {
+        console.log('2');
+        const videoParams = await consumeVideo(callId, rtpCapabilities)
+        const audioParams = await consumeAudio(callId, rtpCapabilities)
+        callback({ videoParams, audioParams });
       } else {
-        console.log(`[canConsume] Can't consume | callId ${callId}`);
+        console.log(`[consume] Can't consume | callId ${callId}`);
         callback(null);
       }
     } catch (error) {
@@ -339,13 +405,71 @@ peers.on('connection', async socket => {
     try {
       const callId = socketDetails[socket.id];
       console.log(`[consumer-resume] callId ${callId}`)
-      await videoCalls[callId].consumer.resume();
+      await videoCalls[callId].consumerVideo.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 
@@ -393,6 +517,7 @@ const createWebRtcTransportLayer = async (callId, callback) => {
       dtlsParameters: transport.dtlsParameters,
     };
 
+    console.log('[createWebRtcTransportLayer] callback params', params);
     // Send back to the client the params
     callback({ params });
 

public/config.js

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

public/index.html

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

public/index.js

@@ -12,10 +12,44 @@ let callId = parseInt(urlParams.get('callId')) || null;
 const IS_PRODUCER = urlParams.get('producer') === 'true' ? true : false
 console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId', callId, ' | IS_PRODUCER', IS_PRODUCER)
 
-let socket 
-hub = io(config.hubAddress)
+console.log('🟩 config', config)
 
-const connectToMediasoup = () => {
+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)
+  
+  const connectToMediasoup = () => {
   
     socket = io(config.mediasoupAddress, {
       reconnection: true,
@@ -32,11 +66,11 @@ 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 +114,55 @@ 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]
-  params = {
-    track,
-    ...params
+  console.log('stream', stream);
+  const videoTrack = stream.getVideoTracks()[0]
+  const audioTrack = stream.getAudioTracks()[0]
+
+  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: true,
     video: {
-      width: {
-        min: 640,
-        max: 1920,
-      },
-      height: {
-        min: 400,
-        max: 1080,
-      }
+      qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
+      vga  : { width: { ideal: 640 }, height: { ideal: 480 } },
+      hd   : { width: { ideal: 1280 }, height: { ideal: 720 } }
     }
   })
   .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 +179,6 @@ 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 +189,7 @@ const createDevice = async () => {
     })
 
     console.log('Device RTP Capabilities', device.rtpCapabilities)
+    console.log('[createDevice] device', device);
     
     // once the device loads, create transport
     goCreateTransport()
@@ -207,18 +219,20 @@ 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, callId }, (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 +258,10 @@ 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 +284,44 @@ 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
+  producerVideo = await producerTransport.produce(videoParams)
+  console.log('videoParams', videoParams);
+  console.log('producerVideo', producerVideo);
+
+  producerVideo.on('trackended', () => {
     console.log('track ended')
     // close video track
   })  
 
-  producer.on('transportclose', () => {
+  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', () => {
+    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 +330,7 @@ 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 +346,7 @@ 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 +356,15 @@ 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 +389,8 @@ const resetCallSettings = () => {
   localVideo.srcObject = null
   remoteVideo.srcObject = null
   consumer = null
-  producer = null
+  producerVideo = null
+  producerAudio = null
   producerTransport = null
   consumerTransport = null
   device = undefined
@@ -361,7 +398,7 @@ 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', {
@@ -373,7 +410,7 @@ 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,
@@ -416,6 +453,7 @@ const closeCall = () => {
   resetCallSettings()
 }
 
+
 btnLocalVideo.addEventListener('click', getLocalStream)
 btnRecvSendTransport.addEventListener('click', goConnect)
 btnCloseCall.addEventListener('click', closeCall)