Update server

app.js

@@ -92,10 +92,21 @@ worker = createWorker();
 // https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
 const mediaCodecs = [
   {
-    kind: 'audio',
-    mimeType: 'audio/opus',
+    // kind: 'audio',
+    // mimeType: 'audio/opus',
+    // clockRate: 48000,
+    // channels: 2,
     clockRate: 48000,
-    channels: 2,
+		mimeType: "audio/opus",
+		parameters: {
+			minptime: 10,
+			useinbandfec: 1
+		},
+		payloadType: 111,
+		rtcpFeedback: [{
+      parameter: "",
+      type: "transport-cc"
+		}]
   },
   {
     kind: 'video',

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.js

@@ -12,10 +12,14 @@ 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
+
+setTimeout(() => {
+  hub = io(config.hubAddress)
+  
+  const connectToMediasoup = () => {
   
     socket = io(config.mediasoupAddress, {
       reconnection: true,
@@ -32,11 +36,11 @@ 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(
@@ -80,15 +84,19 @@ if (IS_PRODUCER === true) {
     hub.on('disconnect', () => {
       console.log('disconnect')
     })
-} else {
+  } else {
     connectToMediasoup()
-}
+  }
+
+}, 2000);
+
 
 let device
 let rtpCapabilities
 let producerTransport
 let consumerTransport
-let producer
+let producerVideo
+let producerAudio
 let consumer
 let originAssetId
 // let originAssetName = 'Adi'
@@ -96,21 +104,18 @@ let originAssetId
 
 // 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: [
     {
-      rid: 'r0',
       maxBitrate: 100000,
       scalabilityMode: 'S1T3',
     },
     {
-      rid: 'r1',
       maxBitrate: 300000,
       scalabilityMode: 'S1T3',
     },
     {
-      rid: 'r2',
       maxBitrate: 900000,
       scalabilityMode: 'S1T3',
     },
@@ -121,21 +126,51 @@ let params = {
   }
 }
 
-const streamSuccess = (stream) => {
-  console.log('[streamSuccess]');
-  localVideo.srcObject = stream
-  const track = stream.getVideoTracks()[0]
-  params = {
-    track,
-    ...params
+let audioParams = {
+  // mediasoup params
+  encodings: [
+    {
+      maxBitrate: 100000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 300000,
+      scalabilityMode: 'S1T3',
+    },
+    {
+      maxBitrate: 900000,
+      scalabilityMode: 'S1T3',
+    },
+  ],
+  codecOptions: {
+    opusStereo: true
   }
+}
+
+const streamSuccess = (stream) => {
+  localVideo.srcObject = stream
+  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,
@@ -207,6 +242,7 @@ 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 +253,7 @@ 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 +280,7 @@ 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 +306,37 @@ 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)
-
-  producer.on('trackended', () => {
+  producerVideo = await producerTransport.produce(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
   })
 
+  // 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 +371,7 @@ 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 +404,8 @@ const resetCallSettings = () => {
   localVideo.srcObject = null
   remoteVideo.srcObject = null
   consumer = null
-  producer = null
+  producerVideo = null
+  producerAudio = null
   producerTransport = null
   consumerTransport = null
   device = undefined