LH-265: Enable audio on video 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

README.md

@@ -33,3 +33,7 @@ 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
+
+### 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) {
@@ -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'),
-  cert: fs.readFileSync('./server/ssl/cert.pem', 'utf-8'),
+  key: fs.readFileSync(process.env.SERVER_KEY, '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,
-      rtcMaxPort: 2020,
+      rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
+      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
@@ -105,33 +105,25 @@ const mediaCodecs = [
       '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();
-      videoCalls[callId]?.producerTransport.close();
-      videoCalls[callId].router.close();
+      videoCalls[callId]?.consumerTransport?.close();
+      videoCalls[callId]?.producerTransport?.close();
+      videoCalls[callId]?.router?.close();
       delete videoCalls[callId];
+    } else {
+      console.log(`The call with id ${callId} has already been deleted`);
     }
   } catch (error) {
     console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
   }
 }
 
-const getRtpCapabilities = (callId, callback) => {
-  try {
-    console.log('[getRtpCapabilities] callId', callId);
-    const rtpCapabilities = videoCalls[callId].router.rtpCapabilities;
-    callback({ rtpCapabilities });
-  } catch (error) {
-    console.log(`ERROR | getRtpCapabilities | callId ${callId} | ${error.message}`);
-  }
-}
-
 /*
   - Handlers for WS events
   - These are created only when we have a connection with a peer
@@ -146,8 +138,10 @@ peers.on('connection', async socket => {
 
   // It is triggered when the peer is disconnected
   socket.on('disconnect', () => {
-    console.log('peer disconnected | socket.id', socket.id);
+    const callId = socketDetails[socket.id];
+    console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
     delete socketDetails[socket.id];
+    closeCall(callId);
   });
 
   /*
@@ -156,7 +150,9 @@ 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]) {
@@ -165,12 +161,19 @@ 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);
     }
   });
 
@@ -187,12 +190,23 @@ peers.on('connection', async socket => {
       const callId = socketDetails[socket.id];
       console.log(`[createWebRtcTransport] sender ${sender} | callId ${callId}`);
       if (sender) {
+        if (!videoCalls[callId].producerTransport) {
           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 ${callId} | sender ${sender} | ${error.message}`);
+      console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
+      callback(error);
     }
   });
 
@@ -203,7 +217,9 @@ 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}`);
@@ -215,9 +231,11 @@ 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,
@@ -230,6 +248,11 @@ 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}`);
     }
@@ -245,7 +268,7 @@ peers.on('connection', async socket => {
       console.log(`[transport-recv-connect] socket.id ${socket.id} | callId ${callId}`);
       await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
     } catch (error) {
-      console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ERROR`);
+      console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
     }
   })
 
@@ -278,14 +301,14 @@ 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
@@ -300,9 +323,10 @@ 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 ${callId} | ${error.message}`)
+      console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
       callback({ params: { error } });
     }
   });
@@ -376,7 +400,7 @@ const createWebRtcTransportLayer = async (callId, callback) => {
     return transport;
 
   } catch (error) {
-    console.log(`ERROR | createWebRtcTransportLayer | callId ${callId} | ${error.message}`);
+    console.log(`ERROR | createWebRtcTransportLayer | callId ${socketDetails[socket.id]} | ${error.message}`);
     callback({ params: { error } });
   }
 }

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/index.js

@@ -135,7 +135,7 @@ const streamSuccess = (stream) => {
 const getLocalStream = () => {
   console.log('[getLocalStream]');
   navigator.mediaDevices.getUserMedia({
-    audio: false,
+    audio: true,
     video: {
       width: {
         min: 640,