Update server

Reviewed-on: #14

.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

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) {
@@ -42,8 +42,8 @@ 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);
@@ -59,16 +59,16 @@ const io = new Server(httpsServer, {
 // 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 +84,7 @@ 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
@@ -95,21 +95,70 @@ const mediaCodecs = [
     kind      : 'audio',
     mimeType  : 'audio/opus',
     clockRate : 48000,
-    channels: 2,
+    channels  : 2
   },
   {
     kind       : 'video',
     mimeType   : 'video/VP8',
     clockRate  : 90000,
-    parameters: {
+    parameters :
-      'x-google-start-bitrate': 1000,
+    {
+      'x-google-start-bitrate' : 1000
+    }
   },
+  {
+    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 (videoCalls[callId]) {
+    if (callId && videoCalls[callId]) {
       videoCalls[callId].producer?.close();
       videoCalls[callId].consumer?.close();
       videoCalls[callId]?.consumerTransport?.close();
@@ -124,16 +173,6 @@ 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
@@ -160,7 +199,9 @@ 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]) {
@@ -169,12 +210,19 @@ peers.on('connection', async socket => {
           console.log(`[createRoom] Router ID: ${videoCalls[callId].router.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);
     }
   });
 
@@ -195,16 +243,19 @@ peers.on('connection', async socket => {
           videoCalls[callId].producerTransport = await createWebRtcTransportLayer(callId, callback);
         } else {
           console.log(`producerTransport has already been defined | callId ${callId}`);
+          callback(null);
         }
       } else if (!sender) {
         if (!videoCalls[callId].consumerTransport) {
           videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
         } else {
           console.log(`consumerTransport has already been defined | callId ${callId}`);
+          callback(null);
         }
       }
     } catch (error) {
       console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
+      callback(error);
     }
   });
 
@@ -215,7 +266,9 @@ peers.on('connection', async socket => {
   socket.on('transport-connect', async ({ dtlsParameters }) => {
     try {
       const callId = socketDetails[socket.id];
-      console.log(`[transport-connect] socket.id ${socket.id} | callId ${callId}`)
+      if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
+
+      console.log(`[transport-connect] socket.id ${socket.id} | callId ${callId}`);
       await videoCalls[callId].producerTransport.connect({ dtlsParameters });
     } catch (error) {
       console.log(`ERROR | transport-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
@@ -227,9 +280,11 @@ peers.on('connection', async socket => {
     - 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 }) => {
+  socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
     try {
       const callId = socketDetails[socket.id];
+      if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
+
       console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
       videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
         kind,
@@ -242,6 +297,11 @@ peers.on('connection', async socket => {
         console.log('transport for this producer closed', callId)
         closeCall(callId);
       });
+
+      // Send back to the client the Producer's id
+      callback && callback({
+        id: videoCalls[callId].producer.id
+      });
     } catch (error) {
       console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
     }
@@ -290,14 +350,14 @@ peers.on('connection', async socket => {
         videoCalls[callId].consumer.on('transportclose', () => {
           const callId = socketDetails[socket.id];
           console.log('transport close from consumer', callId);
-          closeCall();
+          closeCall(callId);
         });
 
         // https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
         videoCalls[callId].consumer.on('producerclose', () => {
           const callId = socketDetails[socket.id];
           console.log('producer of consumer closed', callId);
-          closeCall();
+          closeCall(callId);
         });
 
         // From the consumer extract the following params to send back to the Client
@@ -312,6 +372,7 @@ peers.on('connection', async socket => {
         callback({ params });
       } else {
         console.log(`[canConsume] Can't consume | callId ${callId}`);
+        callback(null);
       }
     } catch (error) {
       console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)

build.sh

@@ -0,0 +1,48 @@
+#/!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://video.safemobile.org',
-  // mediasoupAddress: 'http://localhost:3000/mediasoup',
 }

public/index.js

@@ -12,7 +12,11 @@ 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 
+console.log('🟩 config', config)
+
+let socket, hub
+
+setTimeout(() => {
   hub = io(config.hubAddress)
   
   const connectToMediasoup = () => {
@@ -36,7 +40,7 @@ const connectToMediasoup = () => {
   
   if (IS_PRODUCER === true) {
     hub.on('connect', async () => {
-    console.log(`[HUB] ${config.hubAddress} | connected: ${hub.connected}`)
+      console.log(`[HUB]! ${config.hubAddress} | connected: ${hub.connected}`)
       connectToMediasoup()
   
       hub.emit(
@@ -84,20 +88,30 @@ if (IS_PRODUCER === true) {
     connectToMediasoup()
   }
 
+}, 1600);
+
+
 let device
 let rtpCapabilities
 let producerTransport
 let consumerTransport
-let producer
+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 params = {
+let videoParams = {
-  // mediasoup params
+  // encodings: [
+  //   { scaleResolutionDownBy: 4, maxBitrate: 500000 },
+  //   { scaleResolutionDownBy: 2, maxBitrate: 1000000 },
+  //   { scaleResolutionDownBy: 1, maxBitrate: 5000000 },
+  //   { scalabilityMode: 'S3T3_KEY' }
+  // ],
+  // codecOptions: {
+  //   videoGoogleStartBitrate: 1000
+  // }
   encodings: [
     {
       rid: 'r0',
@@ -121,19 +135,55 @@ let params = {
   }
 }
 
+let audioParams = {
+  codecOptions :
+  {
+    opusStereo : true,
+    opusDtx    : true
+  }
+}
+
 const streamSuccess = (stream) => {
+  // console.log('[streamSuccess] device', device);
+  // localVideo.srcObject = stream
+  // console.log('stream', stream);
+  // const videoTrack = stream.getVideoTracks()[0]
+  // const audioTrack = stream.getAudioTracks()[0]
+
+  // videoParams = {
+  //   track: videoTrack,
+  //   // codec : device.rtpCapabilities.codecs.find((codec) => codec.mimeType.toLowerCase() === 'video/vp9'),
+  //   // codec : 'video/vp9',
+  //   ...videoParams
+  // }
+
+  // audioParams = {
+  //   track: audioTrack,
+  //   ...audioParams
+  // }
+
+  // console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
+  // goConnect()
   console.log('[streamSuccess]');
   localVideo.srcObject = stream
   const track = stream.getVideoTracks()[0]
-  params = {
+  videoParams = {
     track,
-    ...params
+    ...videoParams
   }
   goConnect()
 }
 
 const getLocalStream = () => {
   console.log('[getLocalStream]');
+  // navigator.mediaDevices.getUserMedia({
+  //   audio: false,
+  //   video: {
+  //     qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
+  //     vga  : { width: { ideal: 640 }, height: { ideal: 480 } },
+  //     hd   : { width: { ideal: 1280 }, height: { ideal: 720 } }
+  //   }
+  // })
   navigator.mediaDevices.getUserMedia({
     audio: false,
     video: {
@@ -167,7 +217,7 @@ const goCreateTransport = () => {
 // server side to send/recive media
 const createDevice = async () => {
   try {
-    console.log('[createDevice]');
+    console.log('[createDevice] 1 device', device);
     device = new mediasoupClient.Device()
 
     // https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@@ -178,6 +228,7 @@ const createDevice = async () => {
     })
 
     console.log('Device RTP Capabilities', device.rtpCapabilities)
+    console.log('[createDevice] 2 device', device);
     
     // once the device loads, create transport
     goCreateTransport()
@@ -207,6 +258,7 @@ 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 }) => {
@@ -217,7 +269,7 @@ const createSendTransport = () => {
       return
     }
 
-    console.log(params)
+    console.log('[createWebRtcTransport] params', params)
 
     // creates a new WebRTC Transport to send media
     // based on the server's producer transport params
@@ -244,7 +296,7 @@ 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
@@ -270,22 +322,37 @@ const createSendTransport = () => {
 }
 
 const connectSendTransport = async () => {
+
+  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)
+  producerVideo = await producerTransport.produce(videoParams)
-
+  console.log('producerVideo', producerVideo);
-  producer.on('trackended', () => {
+  producerVideo.on('trackended', () => {
     console.log('track ended')
     // close video track
   })
 
-  producer.on('transportclose', () => {
+  producerVideo.on('transportclose', () => {
     console.log('transport ended')
     // close video track
   })
 
+  // producerAudio = await producerTransport.produce(audioParams)
+  // console.log('producerAudio', producerAudio);
+  // 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,
     dest_asset_id: originAssetId || parseInt(urlParams.get('dest_asset_id')),
@@ -320,7 +387,7 @@ const createRecvTransport = async () => {
       return
     }
 
-    console.log(params)
+    console.log('[createRecvTransport] params', params)
 
     // creates a new WebRTC Transport to receive media
     // based on server's consumer transport params
@@ -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