LINXD-2180: Added recordings

README.md

@@ -1,11 +1,5 @@
 # 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
 

app.js

@@ -1,38 +1,31 @@
-require('dotenv').config()
+import 'dotenv/config'
 
-const express = require('express');
-const app = express();
-const Server = require('socket.io');
-const path = require('node:path');
-const fs = require('node:fs');
-let https = require('https');
-try {
-  https = require('node:https');
-} catch (err) {
-  console.log('https support is disabled!');
-}
-const mediasoup = require('mediasoup');
+/**
+ * integrating mediasoup server with a node.js application
+ */
+
+/* Please follow mediasoup installation requirements */
+/* https://mediasoup.org/documentation/v3/mediasoup/installation/ */
+import express from 'express'
+const app = express()
+
+import https from 'httpolyglot'
+import fs from 'fs'
+import path from 'path'
+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
-/**
- * videoCalls
- * |-> Router
- * |-> Producer
- * |-> Consumer 
- * |-> Producer Transport
- * |-> Consumer Transport
- * 
- * '<callId>': {
- *  router: Router,
- *  producer: Producer,
- *  producerTransport: Producer Transport,
- *  consumer: Consumer,
- *  consumerTransport: Consumer Transport
- * }
- * 
- **/
-let videoCalls = {}
-let socketDetails = {}
+let router = {}
+let producerTransport
+let consumerTransport
+let producer
+let consumer
 
 app.get('/', (_req, res) => {
   res.send('Hello from mediasoup app!')
@@ -43,44 +36,294 @@ 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 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 });
+const httpsServer = https.createServer(options, app)
 
 httpsServer.listen(process.env.PORT, () => {
-  console.log('Video server listening on port:', process.env.PORT)
+  console.log('Listening on port:', process.env.PORT)
 })
 
-const peers = io.of('/')
+const startRecordingFfmpeg = () => {
+  // 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 () => {
-  try {
   worker = await mediasoup.createWorker({
-      rtcMinPort: 2000,
-      rtcMaxPort: 2020,
+    rtcMinPort: 32256,
+    rtcMaxPort: 65535,
   })
-    console.log(`[createWorker] worker pid ${worker.pid}`);
+  console.log(`[createWorker] worker pid ${worker.pid}`)
 
   worker.on('died', error => {
     // This implies something serious happened, so kill the application
-      console.error('mediasoup worker has died', error);
-      setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
+    console.error('mediasoup worker has died', error)
+    setTimeout(() => process.exit(1), 2000) // exit in 2 seconds
   })
-    return worker;
-  } catch (error) {
-    console.log(`ERROR | createWorker | ${error.message}`);
-  }
+
+  return worker
 }
 
 // We create a Worker as soon as our application starts
@@ -92,243 +335,222 @@ worker = createWorker()
 // https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
 const mediaCodecs = [
   {
-    kind: 'audio',
-    mimeType: 'audio/opus',
+    kind: "audio",
+    mimeType: "audio/opus",
+    preferredPayloadType: 111,
     clockRate: 48000,
     channels: 2,
+    parameters: {
+      minptime: 10,
+      useinbandfec: 1,
+    },
   },
   {
-    kind: 'video',
-    mimeType: 'video/VP8',
+    kind: "video",
+    mimeType: "video/VP8",
+    preferredPayloadType: 96,
+    clockRate: 90000,
+  },
+  {
+    kind: "video",
+    mimeType: "video/H264",
+    preferredPayloadType: 125,
     clockRate: 90000,
     parameters: {
-      'x-google-start-bitrate': 1000,
+      "level-asymmetry-allowed": 1,
+      "packetization-mode": 1,
+      "profile-level-id": "42e01f",
     },
   },
 ]
 
-const closeCall = (callId) => {
-  try {
-    if (videoCalls[callId]) {
-      videoCalls[callId].producer?.close();
-      videoCalls[callId].consumer?.close();
-      videoCalls[callId]?.consumerTransport.close();
-      videoCalls[callId]?.producerTransport.close();
-      videoCalls[callId].router.close();
-      delete videoCalls[callId];
-    }
-  } 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
-*/
 peers.on('connection', async socket => {
-  console.log('[connection] socketId:', socket.id);
-
-  // After making the connection successfully, we send the client a 'connection-success' event
+  console.log('[connection] socketId:', socket.id)
   socket.emit('connection-success', {
-    socketId: socket.id
-  });
+    socketId: socket.id,
+    existsProducer: producer ? true : false,
+  })
 
-  // It is triggered when the peer is disconnected
   socket.on('disconnect', () => {
-    console.log('peer disconnected | socket.id', socket.id);
-    delete socketDetails[socket.id];
-  });
+    // do some cleanup
+    console.log('peer disconnected')
+  })
 
-  /*
-    - This event creates a room with the roomId and the callId sent
-    - It will return the rtpCapabilities of that room
-    - If the room already exists, it will not create it, but will only return rtpCapabilities
-  */
   socket.on('createRoom', async ({ callId }, callback) => {
-    try {
-      if (callId) {
-        console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
-        if (!videoCalls[callId]) {
     console.log('[createRoom] callId', callId);
-          videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) }
-          console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
+    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}`)
     }
-        socketDetails[socket.id] = callId;
-        getRtpCapabilities(callId, callback);
-      } else {
-        console.log(`[createRoom] missing callId ${callId}`);
-      }
-    } catch (error) {
-      console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
-    }
-  });
     
-  /*
-    - 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) {
-        videoCalls[callId].producerTransport = await createWebRtcTransportLayer(callId, callback);
-      } else {
-        videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
-      }
-    } catch (error) {
-      console.log(`ERROR | createWebRtcTransport | callId ${callId} | sender ${sender} | ${error.message}`);
-    }
-  });
+    getRtpCapabilities(callId, callback)
+  })
 
-  /*
-    - The client sends this event after successfully creating a createSendTransport(AS PRODUCER)
-    - The connection is made to the created transport
-  */
+  const getRtpCapabilities = (callId, callback) => {
+    const rtpCapabilities = router[callId].rtpCapabilities
+
+    callback({ rtpCapabilities })
+  }
+
+  // Client emits a request to create server side Transport
+  // We need to differentiate between the producer and consumer transports
+  socket.on('createWebRtcTransport', async ({ sender, callId }, callback) => {
+    console.log(`[createWebRtcTransport] Is this a sender request? ${sender} | callId ${callId}`)
+    // The client indicates if it is a producer or a consumer
+    // if sender is true, indicates a producer else a consumer
+    if (sender)
+      producerTransport = await createWebRtcTransportLayer(callId, callback)
+    else
+      consumerTransport = await createWebRtcTransportLayer(callId, callback)
+  })
+
+  // see client's socket.emit('transport-connect', ...)
   socket.on('transport-connect', async ({ dtlsParameters }) => {
-    try {
-      const callId = socketDetails[socket.id];
-      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}`);
-    }
-  });
+    console.log('[transport-connect] DTLS PARAMS... ', { dtlsParameters })
+    await producerTransport.connect({ dtlsParameters })
+  })
 
-  /*
-    - 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 }) => {
-    try {
-      const callId = socketDetails[socket.id];
-      console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
-      videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
+  // see client's socket.emit('transport-produce', ...)
+  socket.on('transport-produce', async ({ kind, rtpParameters, callId }, callback) => {
+    // call produce based on the prameters from the client
+    producer = await 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-produce] Producer ID: ${producer.id} | kind: ${producer.kind}`)
+
+    producer.on('transportclose', () => {
       console.log('transport for this producer closed', callId)
-        closeCall(callId);
+      
+      // https://mediasoup.org/documentation/v3/mediasoup/api/#producer-close
+      producer.close()
+
+      // 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 }
     });
-    } catch (error) {
-      console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
-    }
+    await rtpTransport.connect({
+      ip: "127.0.0.1",
+      port: 5006,
+      rtcpPort: 5007,
     });
 
-  /*
-    - The client sends this event after successfully creating a createRecvTransport(AS CONSUMER)
-    - The connection is made to the created consumerTransport
-  */
+    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 {
-      const callId = socketDetails[socket.id];
-      console.log(`[transport-recv-connect] socket.id ${socket.id} | callId ${callId}`);
-      await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
+      console.log('consume', rtpCapabilities, callId);
+      // check if the router can consume the specified producer
+      if (router[callId].canConsume({
+        producerId: producer.id,
+        rtpCapabilities
+      })) {
+        // 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,
+        }
+
+        // send the parameters to the client
+        callback({ params })
+      }
     } catch (error) {
-      console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ERROR`);
+      console.log(error.message)
+      callback({
+        params: {
+          error: error
+        }
+      })
     }
   })
 
-  /*
-    - The customer consumes after successfully connecting to consumerTransport
-    - 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 ${callId} | ${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}`);
-    }
-  });
-});
+    console.log(`[consumer-resume]`)
+    await consumer.resume()
+  })
+})
 
-/*
-  - 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);
@@ -343,40 +565,49 @@ 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 videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options)
+    let transport = await router[callId].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 | calldId ${callId}`);
-    });
+      console.log('transport closed')
+    })
 
     const params = {
       id: transport.id,
       iceParameters: transport.iceParameters,
       iceCandidates: transport.iceCandidates,
       dtlsParameters: transport.dtlsParameters,
-    };
+    }
 
-    // Send back to the client the params
-    callback({ params });
+    console.log('params', params);
 
-    // Set transport to producerTransport or consumerTransport
-    return transport;
+    // send back to the client the following prameters
+    callback({
+      // https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
+      params
+    })
+
+    return transport
 
   } catch (error) {
-    console.log(`ERROR | createWebRtcTransportLayer | callId ${callId} | ${error.message}`);
-    callback({ params: { error } });
+    console.log('[createWebRtcTransportLayer] ERROR', JSON.stringify(error));
+    callback({
+      params: {
+        error: error
+      }
+    })
   }
 }

package.json

@@ -5,17 +5,20 @@
   "main": "app.js",
   "scripts": {
     "test": "echo \"Error: no test specified\" && exit 1",
-    "start:dev": "nodemon app.js",
+    "start:dev": "nodemon app.ts",
     "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(error.message)
+    console.log('getLocalStream', error)
   })
 }
 
@@ -20903,7 +20903,7 @@ const createSendTransport = () => {
     })
 
     producerTransport.on('produce', async (parameters, callback, errback) => {
-      console.log(parameters)
+      console.log('produce', 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,
-          appData: parameters.appData,
+          callId: callId
         }, ({ id }) => {
           // Tell the transport that parameters were transmitted and provide it with the
           // server side producer's id.
@@ -21009,7 +21009,6 @@ const createRecvTransport = async () => {
 }
 
 const resetCallSettings = () => {
-  socket.emit('transportclose', { callId })
   localVideo.srcObject = null
   remoteVideo.srcObject = null
   consumer = null
@@ -21072,7 +21071,7 @@ const closeCall = () => {
   const closeCallBtn = document.getElementById('btnCloseCall')
   closeCallBtn.setAttribute('disabled', '')
 
-  // Reset settings and send closeTransport to video server
+  // Reset settings
   resetCallSettings()
 }
 

public/config.js

@@ -1,5 +1,5 @@
 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/mediasoup',
+  mediasoupAddress: 'http://localhost:3000/mediasoup',
 }

public/index.js

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