added log for dtls transport-connect

Reviewed-on: #8

README.md

@@ -1,5 +1,11 @@
 # Video server
 
+### Generating certificates
+
+##### To generate SSL certificates you must:
+1. Go to `/server/ssl`
+2. Execute `openssl req -newkey rsa:2048 -new -nodes -x509 -days 3650 -keyout key.pem -out cert.pem`
+
 
 ### Development
 
@@ -27,3 +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

@@ -1,31 +1,38 @@
-import 'dotenv/config'
+require('dotenv').config()
 
-/**
+const express = require('express');
- * integrating mediasoup server with a node.js application
+const app = express();
- */
+const Server = require('socket.io');
-
+const path = require('node:path');
-/* Please follow mediasoup installation requirements */
+const fs = require('node:fs');
-/* https://mediasoup.org/documentation/v3/mediasoup/installation/ */
+let https = require('https');
-import express from 'express'
+try {
-const app = express()
+  https = require('node:https');
-
+} catch (err) {
-import https from 'httpolyglot'
+  console.log('https support is disabled!');
-import fs from 'fs'
+}
-import path from 'path'
+const mediasoup = require('mediasoup');
-const __dirname = path.resolve()
-
-// const FFmpegStatic = require("ffmpeg-static")
-import FFmpegStatic from 'ffmpeg-static'
-import Server from 'socket.io'
-import mediasoup, { getSupportedRtpCapabilities } from 'mediasoup'
-import Process from 'child_process'
 
 let worker
-let router = {}
+/**
-let producerTransport
+ * videoCalls
-let consumerTransport
+ * |-> Router
-let producer
+ * |-> Producer
-let consumer
+ * |-> Consumer 
+ * |-> Producer Transport
+ * |-> Consumer Transport
+ * 
+ * '<callId>': {
+ *  router: Router,
+ *  producer: Producer,
+ *  producerTransport: Producer Transport,
+ *  consumer: Consumer,
+ *  consumerTransport: Consumer Transport
+ * }
+ * 
+ **/
+let videoCalls = {}
+let socketDetails = {}
 
 app.get('/', (_req, res) => {
   res.send('Hello from mediasoup app!')
@@ -36,294 +43,44 @@ 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')
+  cert: fs.readFileSync('./server/ssl/cert.pem', 'utf-8'),
 }
 
-const httpsServer = https.createServer(options, app)
+const httpsServer = https.createServer(options, app);
+
+const io = new Server(httpsServer, {
+  allowEIO3: true,
+  origins: ["*:*"],
+  // allowRequest: (req, next) => {
+  //   console.log('req', req);
+  //   next(null, true)
+  // }
+});
+// const io = new Server(server, { origins: '*:*', allowEIO3: true });
 
 httpsServer.listen(process.env.PORT, () => {
-  console.log('Listening on port:', process.env.PORT)
+  console.log('Video server listening on port:', process.env.PORT)
 })
 
-const startRecordingFfmpeg = () => {
+const peers = io.of('/')
-  // Return a Promise that can be awaited
-  let recResolve;
-  const promise = new Promise((res, _rej) => {
-    recResolve = res;
-  });
-
-  // const useAudio = audioEnabled();
-  // const useVideo = videoEnabled();
-  // const useH264 = h264Enabled();
-
-  // const cmdProgram = "ffmpeg"; // Found through $PATH
-  const cmdProgram = FFmpegStatic; // From package "ffmpeg-static"
-
-  let cmdInputPath = `${__dirname}/recording/input-vp8.sdp`;
-  let cmdOutputPath = `${__dirname}/recording/output-ffmpeg-vp8.webm`;
-  let cmdCodec = "";
-  let cmdFormat = "-f webm -flags +global_header";
-
-  // Ensure correct FFmpeg version is installed
-  const ffmpegOut = Process.execSync(cmdProgram + " -version", {
-    encoding: "utf8",
-  });
-  const ffmpegVerMatch = /ffmpeg version (\d+)\.(\d+)\.(\d+)/.exec(ffmpegOut);
-  let ffmpegOk = false;
-  if (ffmpegOut.startsWith("ffmpeg version git")) {
-    // Accept any Git build (it's up to the developer to ensure that a recent
-    // enough version of the FFmpeg source code has been built)
-    ffmpegOk = true;
-  } else if (ffmpegVerMatch) {
-    const ffmpegVerMajor = parseInt(ffmpegVerMatch[1], 10);
-    if (ffmpegVerMajor >= 4) {
-      ffmpegOk = true;
-    }
-  }
-
-  if (!ffmpegOk) {
-    console.error("FFmpeg >= 4.0.0 not found in $PATH; please install it");
-    process.exit(1);
-  }
-
-  // if (useAudio) {
-    // cmdCodec += " -map 0:a:0 -c:a copy";
-  // }
-  // if (useVideo) {
-    cmdCodec += " -map 0:v:0 -c:v copy";
-
-    // if (useH264) {
-      cmdInputPath = `${__dirname}/recording/input-h264.sdp`;
-      cmdOutputPath = `${__dirname}/recording/output-ffmpeg-h264.mp4`;
-
-      // "-strict experimental" is required to allow storing
-      // OPUS audio into MP4 container
-      cmdFormat = "-f mp4 -strict experimental";
-  //   }
-  // }
-
-  // Run process
-  const cmdArgStr = [
-    "-nostdin",
-    "-protocol_whitelist file,rtp,udp",
-    "-loglevel debug",
-    "-analyzeduration 5M",
-    "-probesize 5M",
-    "-fflags +genpts",
-    `-i ${cmdInputPath}`,
-    cmdCodec,
-    cmdFormat,
-    `-y ${cmdOutputPath}`,
-  ]
-    .join(" ")
-    .trim();
-
-  console.log('💗', cmdCodec);
-  console.log(`Run command: ${cmdProgram} ${cmdArgStr}`);
-
-  let recProcess = Process.spawn(cmdProgram, cmdArgStr.split(/\s+/));
-  global.recProcess = recProcess;
-
-  recProcess.on("error", (err) => {
-    console.error("Recording process error:", err);
-  });
-
-  recProcess.on("exit", (code, signal) => {
-    console.log("Recording process exit, code: %d, signal: %s", code, signal);
-
-    global.recProcess = null;
-    stopMediasoupRtp();
-
-    if (!signal || signal === "SIGINT") {
-      console.log("Recording stopped");
-    } else {
-      console.warn(
-        "Recording process didn't exit cleanly, output file might be corrupt"
-      );
-    }
-  });
-
-  // FFmpeg writes its logs to stderr
-  recProcess.stderr.on("data", (chunk) => {
-    chunk
-      .toString()
-      .split(/\r?\n/g)
-      .filter(Boolean) // Filter out empty strings
-      .forEach((line) => {
-        console.log(line);
-        if (line.startsWith("ffmpeg version")) {
-          setTimeout(() => {
-            recResolve();
-          }, 1000);
-        }
-      });
-  });
-
-  return promise;
-}
-
-const startRecordingGstreamer = () => {
-  // Return a Promise that can be awaited
-  let recResolve;
-  const promise = new Promise((res, _rej) => {
-    recResolve = res;
-  });
-
-  // const useAudio = audioEnabled();
-  // const useVideo = videoEnabled();
-  // const useH264 = h264Enabled();
-
-  let cmdInputPath = `${__dirname}/recording/input-vp8.sdp`;
-  let cmdOutputPath = `${__dirname}/recording/output-gstreamer-vp8.webm`;
-  let cmdMux = "webmmux";
-  let cmdAudioBranch = "";
-  let cmdVideoBranch = "";
-
-  // if (useAudio) {
-  //   // prettier-ignore
-  //   cmdAudioBranch =
-  //     "demux. ! queue \
-  //     ! rtpopusdepay \
-  //     ! opusparse \
-  //     ! mux.";
-  // }
-
-  // if (useVideo) {
-  //   if (useH264) {
-      cmdInputPath = `${__dirname}/recording/input-h264.sdp`;
-      cmdOutputPath = `${__dirname}/recording/output-gstreamer-h264.mp4`;
-      cmdMux = `mp4mux faststart=true faststart-file=${cmdOutputPath}.tmp`;
-
-      // prettier-ignore
-      cmdVideoBranch =
-        "demux. ! queue \
-        ! rtph264depay \
-        ! h264parse \
-        ! mux.";
-  //   } else {
-  //     // prettier-ignore
-  //     cmdVideoBranch =
-  //       "demux. ! queue \
-  //       ! rtpvp8depay \
-  //       ! mux.";
-  //   }
-  // }
-
-  // Run process
-  const cmdProgram = "gst-launch-1.0"; // Found through $PATH
-  const cmdArgStr = [
-    "--eos-on-shutdown",
-    `filesrc location=${cmdInputPath}`,
-    "! sdpdemux timeout=0 name=demux",
-    `${cmdMux} name=mux`,
-    `! filesink location=${cmdOutputPath}`,
-    cmdAudioBranch,
-    cmdVideoBranch,
-  ]
-    .join(" ")
-    .trim();
-
-  console.log(
-    `Run command: ${cmdProgram} ${cmdArgStr}`
-  );
-
-  let recProcess = Process.spawn(cmdProgram, cmdArgStr.split(/\s+/));
-  global.recProcess = recProcess;
-
-  recProcess.on("error", (err) => {
-    console.error("Recording process error:", err);
-  });
-
-  recProcess.on("exit", (code, signal) => {
-    console.log("Recording process exit, code: %d, signal: %s", code, signal);
-
-    global.recProcess = null;
-    stopMediasoupRtp();
-
-    if (!signal || signal === "SIGINT") {
-      console.log("Recording stopped");
-    } else {
-      console.warn(
-        "Recording process didn't exit cleanly, output file might be corrupt"
-      );
-    }
-  });
-
-  // GStreamer writes some initial logs to stdout
-  recProcess.stdout.on("data", (chunk) => {
-    chunk
-      .toString()
-      .split(/\r?\n/g)
-      .filter(Boolean) // Filter out empty strings
-      .forEach((line) => {
-        console.log(line);
-        if (line.startsWith("Setting pipeline to PLAYING")) {
-          setTimeout(() => {
-            recResolve();
-          }, 1000);
-        }
-      });
-  });
-
-  // GStreamer writes its progress logs to stderr
-  recProcess.stderr.on("data", (chunk) => {
-    chunk
-      .toString()
-      .split(/\r?\n/g)
-      .filter(Boolean) // Filter out empty strings
-      .forEach((line) => {
-        console.log(line);
-      });
-  });
-
-  return promise;
-}
-
-function stopMediasoupRtp() {
-  console.log("Stop mediasoup RTP transport and consumer");
-
-  // const useAudio = audioEnabled();
-  // const useVideo = videoEnabled();
-
-  // if (useAudio) {
-  //   global.mediasoup.rtp.audioConsumer.close();
-  //   global.mediasoup.rtp.audioTransport.close();
-  // }
-
-  // if (useVideo) {
-  //   global.mediasoup.rtp.videoConsumer.close();
-  //   global.mediasoup.rtp.videoTransport.close();
-  // }
-}
-
-const io = new Server(httpsServer)
-
-// socket.io namespace (could represent a room?)
-const peers = io.of('/mediasoup')
-
-/**
- * Worker
- * |-> Router(s)
- *     |-> Producer Transport(s)
- *         |-> Producer
- *     |-> Consumer Transport(s)
- *         |-> Consumer 
- **/
 
 const createWorker = async () => {
-  worker = await mediasoup.createWorker({
+  try {
-    rtcMinPort: 32256,
+    worker = await mediasoup.createWorker({
-    rtcMaxPort: 65535,
+      rtcMinPort: 2000,
-  })
+      rtcMaxPort: 2020,
-  console.log(`[createWorker] worker pid ${worker.pid}`)
+    })
+    console.log(`[createWorker] worker pid ${worker.pid}`);
   
-  worker.on('died', error => {
+    worker.on('died', error => {
-    // This implies something serious happened, so kill the application
+      // This implies something serious happened, so kill the application
-    console.error('mediasoup worker has died', error)
+      console.error('mediasoup worker has died', error);
-    setTimeout(() => process.exit(1), 2000) // exit in 2 seconds
+      setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
-  })
+    })
-
+    return worker;
-  return worker
+  } catch (error) {
+    console.log(`ERROR | createWorker | ${error.message}`);
+  }
 }
 
 // We create a Worker as soon as our application starts
@@ -335,222 +92,264 @@ worker = createWorker()
 // https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
 const mediaCodecs = [
   {
-    kind: "audio",
+    kind: 'audio',
-    mimeType: "audio/opus",
+    mimeType: 'audio/opus',
-    preferredPayloadType: 111,
     clockRate: 48000,
     channels: 2,
-    parameters: {
-      minptime: 10,
-      useinbandfec: 1,
-    },
   },
   {
-    kind: "video",
+    kind: 'video',
-    mimeType: "video/VP8",
+    mimeType: 'video/VP8',
-    preferredPayloadType: 96,
-    clockRate: 90000,
-  },
-  {
-    kind: "video",
-    mimeType: "video/H264",
-    preferredPayloadType: 125,
     clockRate: 90000,
     parameters: {
-      "level-asymmetry-allowed": 1,
+      'x-google-start-bitrate': 1000,
-      "packetization-mode": 1,
-      "profile-level-id": "42e01f",
     },
   },
 ]
 
-peers.on('connection', async socket => {
+const closeCall = (callId) => {
-  console.log('[connection] socketId:', socket.id)
+  try {
-  socket.emit('connection-success', {
+    if (videoCalls[callId]) {
-    socketId: socket.id,
+      videoCalls[callId].producer?.close();
-    existsProducer: producer ? true : false,
+      videoCalls[callId].consumer?.close();
-  })
+      videoCalls[callId]?.consumerTransport?.close();
-
+      videoCalls[callId]?.producerTransport?.close();
-  socket.on('disconnect', () => {
+      videoCalls[callId]?.router?.close();
-    // do some cleanup
+      delete videoCalls[callId];
-    console.log('peer disconnected')
+    } else {
-  })
+      console.log(`The call with id ${callId} has already been deleted`);
-
-  socket.on('createRoom', async ({ callId }, callback) => {
-    console.log('[createRoom] callId', callId);
-    console.log('Router length:', Object.keys(router).length);
-    if (router[callId] === undefined) {
-      // worker.createRouter(options)
-      // options = { mediaCodecs, appData }
-      // mediaCodecs -> defined above
-      // appData -> custom application data - we are not supplying any
-      // none of the two are required
-      router[callId] = await worker.createRouter({ mediaCodecs })
-      console.log(`[createRoom] Router ID: ${router[callId].id}`)
     }
-    
+  } catch (error) {
-    getRtpCapabilities(callId, callback)
+    console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
-  })
-
-  const getRtpCapabilities = (callId, callback) => {
-    const rtpCapabilities = router[callId].rtpCapabilities
-
-    callback({ rtpCapabilities })
   }
+}
 
-  // Client emits a request to create server side Transport
+const getRtpCapabilities = (callId, callback) => {
-  // We need to differentiate between the producer and consumer transports
+  try {
-  socket.on('createWebRtcTransport', async ({ sender, callId }, callback) => {
+    console.log('[getRtpCapabilities] callId', callId);
-    console.log(`[createWebRtcTransport] Is this a sender request? ${sender} | callId ${callId}`)
+    const rtpCapabilities = videoCalls[callId].router.rtpCapabilities;
-    // The client indicates if it is a producer or a consumer
+    callback({ rtpCapabilities });
-    // if sender is true, indicates a producer else a consumer
+  } catch (error) {
-    if (sender)
+    console.log(`ERROR | getRtpCapabilities | callId ${callId} | ${error.message}`);
-      producerTransport = await createWebRtcTransportLayer(callId, callback)
+  }
-    else
+}
-      consumerTransport = await createWebRtcTransportLayer(callId, callback)
-  })
 
-  // see client's socket.emit('transport-connect', ...)
+/*
-  socket.on('transport-connect', async ({ dtlsParameters }) => {
+  - Handlers for WS events
-    console.log('[transport-connect] DTLS PARAMS... ', { dtlsParameters })
+  - These are created only when we have a connection with a peer
-    await producerTransport.connect({ dtlsParameters })
+*/
-  })
+peers.on('connection', async socket => {
+  console.log('[connection] socketId:', socket.id);
 
-  // see client's socket.emit('transport-produce', ...)
+  // After making the connection successfully, we send the client a 'connection-success' event
-  socket.on('transport-produce', async ({ kind, rtpParameters, callId }, callback) => {
+  socket.emit('connection-success', {
-    // call produce based on the prameters from the client
+    socketId: socket.id
-    producer = await producerTransport.produce({
+  });
-      kind,
-      rtpParameters,
-    })
 
-    console.log(`[transport-produce] Producer ID: ${producer.id} | kind: ${producer.kind}`)
+  // It is triggered when the peer is disconnected
+  socket.on('disconnect', () => {
+    const callId = socketDetails[socket.id];
+    console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
+    delete socketDetails[socket.id];
+    closeCall(callId);
+  });
 
-    producer.on('transportclose', () => {
+  /*
-      console.log('transport for this producer closed', callId)
+    - This event creates a room with the roomId and the callId sent
-      
+    - It will return the rtpCapabilities of that room
-      // https://mediasoup.org/documentation/v3/mediasoup/api/#producer-close
+    - If the room already exists, it will not create it, but will only return rtpCapabilities
-      producer.close()
+  */
-
+  socket.on('createRoom', async ({ callId }, callback) => {
-      // https://mediasoup.org/documentation/v3/mediasoup/api/#router-close
-      router[callId].close()
-      delete router[callId]
-    })
-
-    // Send back to the client the Producer's id
-    callback({
-      id: producer.id
-    })
-
-
-    console.log('🔴', callId);
-
-    const rtpTransport = await router[callId].createPlainTransport({
-      comedia: false,
-      rtcpMux: false,
-      listenIp: { ip: "127.0.0.1", announcedIp: null }
-    });
-    await rtpTransport.connect({
-      ip: "127.0.0.1",
-      port: 5006,
-      rtcpPort: 5007,
-    });
-
-    console.log(
-      "mediasoup VIDEO RTP SEND transport connected: %s:%d <--> %s:%d (%s)",
-      rtpTransport.tuple.localIp,
-      rtpTransport.tuple.localPort,
-      rtpTransport.tuple.remoteIp,
-      rtpTransport.tuple.remotePort,
-      rtpTransport.tuple.protocol
-    );
-
-    console.log(
-      "mediasoup VIDEO RTCP SEND transport connected: %s:%d <--> %s:%d (%s)",
-      rtpTransport.rtcpTuple.localIp,
-      rtpTransport.rtcpTuple.localPort,
-      rtpTransport.rtcpTuple.remoteIp,
-      rtpTransport.rtcpTuple.remotePort,
-      rtpTransport.rtcpTuple.protocol
-    );
-
-    const rtpConsumer = await rtpTransport.consume({
-      // producerId: global.mediasoup.webrtc.videoProducer.id,
-      producerId: producer.id,
-      // rtpCapabilities: router.rtpCapabilities,
-      rtpCapabilities: router[callId].rtpCapabilities,
-      paused: true,
-    });
-    // console.log('🟡 producerId:', producer.id, 'rtpCapabilities:', router[callId].rtpCapabilities, 'paused:', true);
-    await startRecordingFfmpeg();
-    // await startRecordingGstreamer();
-    rtpConsumer.resume();
-
-  })
-
-  // see client's socket.emit('transport-recv-connect', ...)
-  socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
-    console.log(`[transport-recv-connect] DTLS PARAMS: ${dtlsParameters}`)
-    await consumerTransport.connect({ dtlsParameters })
-  })
-
-  socket.on('consume', async ({ rtpCapabilities, callId }, callback) => {
     try {
-      console.log('consume', rtpCapabilities, callId);
+      if (callId) {
-      // check if the router can consume the specified producer
+        console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
-      if (router[callId].canConsume({
+        if (!videoCalls[callId]) {
-        producerId: producer.id,
+          console.log('[createRoom] callId', callId);
-        rtpCapabilities
+          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) }
-      })) {
+          console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
-        // transport can now consume and return a consumer
-        consumer = await consumerTransport.consume({
-          producerId: producer.id,
-          rtpCapabilities,
-          paused: true,
-        })
-
-        consumer.on('transportclose', () => {
-          console.log('transport close from consumer', callId)
-          // closeRoom(callId)
-          delete router[callId]
-        })
-
-        consumer.on('producerclose', () => {
-          console.log('producer of consumer closed', callId)
-          
-          // https://mediasoup.org/documentation/v3/mediasoup/api/#router-close
-          router[callId].close()
-          delete router[callId]
-        })
-
-        // from the consumer extract the following params
-        // to send back to the Client
-        const params = {
-          id: consumer.id,
-          producerId: producer.id,
-          kind: consumer.kind,
-          rtpParameters: consumer.rtpParameters,
         }
-
+        socketDetails[socket.id] = callId;
-        // send the parameters to the client
+        getRtpCapabilities(callId, callback);
-        callback({ params })
+      } else {
+        console.log(`[createRoom] missing callId ${callId}`);
       }
     } catch (error) {
-      console.log(error.message)
+      console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
-      callback({
+    }
-        params: {
+  });
-          error: error
+
+  /*
+    - Client emits a request to create server side Transport
+    - Depending on the sender, producerTransport or consumerTransport is created on that router
+    - It will return parameters, these are required for the client to create the RecvTransport 
+    from the client.
+      - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
+      - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
+  */
+  socket.on('createWebRtcTransport', async ({ sender }, callback) => {
+    try {
+      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}`);
         }
-      })
+      } else if (!sender) {
+        if (!videoCalls[callId].consumerTransport) {
+          videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
+        } else {
+          console.log(`consumerTransport has already been defined | callId ${callId}`);
+        }
+      }
+    } catch (error) {
+      console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
+    }
+  });
+
+  /*
+    - The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
+    - The connection is made to the created transport
+  */
+  socket.on('transport-connect', async ({ dtlsParameters }) => {
+    try {
+      const callId = socketDetails[socket.id];
+      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}`);
+    }
+  });
+
+  /*
+    - 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) => {
+    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,
+        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);
+      });
+
+      // Send back to the client the Producer's id
+      // callback({
+      //   id: videoCalls[callId].producer.id
+      // });
+    } catch (error) {
+      console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
+    }
+  });
+
+  /*
+    - The client sends this event after successfully creating a createRecvTransport(AS CONSUMER)
+    - The connection is made to the created consumerTransport
+  */
+  socket.on('transport-recv-connect', async ({ dtlsParameters }) => {
+    try {
+      const callId = socketDetails[socket.id];
+      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.message}`);
     }
   })
 
-  socket.on('consumer-resume', async () => {
+  /*
-    console.log(`[consumer-resume]`)
+    - The customer consumes after successfully connecting to consumerTransport
-    await consumer.resume()
+    - The previous step was 'transport-recv-connect', and before that 'createWebRtcTransport'
-  })
+    - This event is only sent by the consumer
-})
+    - The parameters that the consumer consumes are returned
+      - The consumer does consumerTransport.consume(params)
+  */
+  socket.on('consume', async ({ rtpCapabilities }, callback) => {
+    try {
+      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,
+        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();
+        });
+
+        // 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();
+        });
+
+        // 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 });
+      } else {
+        console.log(`[canConsume] Can't consume | callId ${callId}`);
+      }
+    } catch (error) {
+      console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
+      callback({ params: { error } });
+    }
+  });
+
+  /*
+    - Event sent by the consumer after consuming to resume the pause
+    - When consuming on consumerTransport, it is initially done with paused: true, here we will resume 
+  */
+  socket.on('consumer-resume', async () => {
+    try {
+      const callId = socketDetails[socket.id];
+      console.log(`[consumer-resume] callId ${callId}`)
+      await videoCalls[callId].consumer.resume();
+    } catch (error) {
+      console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
+    }
+  });
+});
+
+/*
+  - 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 
+    from the client.
+    - If the client is producer(sender: true) then it will use parameters for device.createSendTransport(params)
+    - If the client is a consumer(sender: false) then it will use parameters for device.createRecvTransport(params)
+*/
 const createWebRtcTransportLayer = async (callId, callback) => {
   try {
     console.log('[createWebRtcTransportLayer] callId', callId);
@@ -565,49 +364,40 @@ const createWebRtcTransportLayer = async (callId, callback) => {
       enableUdp: true,
       enableTcp: true,
       preferUdp: true,
-      initialAvailableOutgoingBitrate: 300000
+    };
-    }
-
-    // console.log('webRtcTransport_options', webRtcTransport_options);
-    // console.log('router', router, '| router[callId]', router[callId]);
     
     // https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
-    let transport = await router[callId].createWebRtcTransport(webRtcTransport_options)
+    let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options)
     console.log(`callId: ${callId} | transport id: ${transport.id}`)
 
+    // Handler for when DTLS(Datagram Transport Layer Security) changes
     transport.on('dtlsstatechange', dtlsState => {
+      console.log(`transport | dtlsstatechange | calldId ${callId} | dtlsState ${dtlsState}`);
       if (dtlsState === 'closed') {
-        transport.close()
+        transport.close();
       }
-    })
+    });
 
+    // Handler if the transport layer has closed (for various reasons)
     transport.on('close', () => {
-      console.log('transport closed')
+      console.log(`transport | closed | calldId ${callId}`);
-    })
+    });
 
     const params = {
       id: transport.id,
       iceParameters: transport.iceParameters,
       iceCandidates: transport.iceCandidates,
       dtlsParameters: transport.dtlsParameters,
-    }
+    };
 
-    console.log('params', params);
+    // Send back to the client the params
+    callback({ params });
 
-    // send back to the client the following prameters
+    // Set transport to producerTransport or consumerTransport
-    callback({
+    return transport;
-      // https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
-      params
-    })
-
-    return transport
 
   } catch (error) {
-    console.log('[createWebRtcTransportLayer] ERROR', JSON.stringify(error));
+    console.log(`ERROR | createWebRtcTransportLayer | callId ${socketDetails[socket.id]} | ${error.message}`);
-    callback({
+    callback({ params: { error } });
-      params: {
-        error: error
-      }
-    })
   }
 }

package.json

@@ -5,20 +5,17 @@
   "main": "app.js",
   "scripts": {
     "test": "echo \"Error: no test specified\" && exit 1",
-    "start:dev": "nodemon app.ts",
+    "start:dev": "nodemon app.js",
     "start:prod": "pm2 start ./app.js -n video-server",
     "watch": "watchify public/index.js -o public/bundle.js -v"
   },
   "keywords": [],
   "author": "",
   "license": "ISC",
-  "type": "module",
   "dependencies": {
     "@types/express": "^4.17.13",
     "dotenv": "^16.0.1",
     "express": "^4.18.1",
-    "ffmpeg-static": "^5.0.2",
-    "httpolyglot": "^0.1.2",
     "mediasoup": "^3.10.4",
     "mediasoup-client": "^3.6.54",
     "parcel": "^2.7.0",

public/bundle.js

@@ -20808,7 +20808,7 @@ const getLocalStream = () => {
   })
   .then(streamSuccess)
   .catch(error => {
-    console.log('getLocalStream', error)
+    console.log(error.message)
   })
 }
 
@@ -20903,7 +20903,7 @@ const createSendTransport = () => {
     })
 
     producerTransport.on('produce', async (parameters, callback, errback) => {
-      console.log('produce', parameters)
+      console.log(parameters)
 
       try {
         // tell the server to create a Producer
@@ -20913,7 +20913,7 @@ const createSendTransport = () => {
         await socket.emit('transport-produce', {
           kind: parameters.kind,
           rtpParameters: parameters.rtpParameters,
-          callId: callId
+          appData: parameters.appData,
         }, ({ id }) => {
           // Tell the transport that parameters were transmitted and provide it with the
           // server side producer's id.
@@ -21009,6 +21009,7 @@ const createRecvTransport = async () => {
 }
 
 const resetCallSettings = () => {
+  socket.emit('transportclose', { callId })
   localVideo.srcObject = null
   remoteVideo.srcObject = null
   consumer = null
@@ -21059,9 +21060,9 @@ const closeCall = () => {
 
   // Emit 'notify-end' to Hub so the consumer will know to close the video
   const notifyEnd = {
-      origin_asset_id: ASSET_ID,
+    origin_asset_id: ASSET_ID,
-      dest_asset_id: originAssetId || parseInt(urlParams.get('dest_asset_id')),
+    dest_asset_id: originAssetId || parseInt(urlParams.get('dest_asset_id')),
-      type: 'notify-end', 
+    type: 'notify-end', 
     video_call_id: callId
   }
   console.log('notifyEnd', notifyEnd)
@@ -21071,7 +21072,7 @@ const closeCall = () => {
   const closeCallBtn = document.getElementById('btnCloseCall')
   closeCallBtn.setAttribute('disabled', '')
 
-  // Reset settings
+  // Reset settings and send closeTransport to video server
   resetCallSettings()
 }
 

public/config.js

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

public/index.js

@@ -149,7 +149,7 @@ const getLocalStream = () => {
   })
   .then(streamSuccess)
   .catch(error => {
-    console.log('getLocalStream', error)
+    console.log(error.message)
   })
 }
 
@@ -244,7 +244,7 @@ const createSendTransport = () => {
     })
 
     producerTransport.on('produce', async (parameters, callback, errback) => {
-      console.log('produce', parameters)
+      console.log(parameters)
 
       try {
         // tell the server to create a Producer
@@ -254,7 +254,7 @@ const createSendTransport = () => {
         await socket.emit('transport-produce', {
           kind: parameters.kind,
           rtpParameters: parameters.rtpParameters,
-          callId: callId
+          appData: parameters.appData,
         }, ({ id }) => {
           // Tell the transport that parameters were transmitted and provide it with the
           // server side producer's id.