LAPI-675: Refactor transport-connect new-producer
This commit is contained in:
parent
0f4342777a
commit
fb99fd57f8
263
app.js
263
app.js
@ -1,4 +1,4 @@
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require('dotenv').config()
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require('dotenv').config();
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const express = require('express');
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const app = express();
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@ -37,7 +37,7 @@ let videoCalls = {};
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let socketDetails = {};
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app.get('/', (_req, res) => {
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res.send('Hello from mediasoup app!')
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res.send('Hello from mediasoup app!');
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});
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app.use('/sfu', express.static(path.join(__dirname, 'public')));
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@ -52,7 +52,7 @@ const httpsServer = https.createServer(options, app);
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const io = new Server(httpsServer, {
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allowEIO3: true,
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origins: ["*:*"]
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origins: ['*:*'],
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});
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httpsServer.listen(process.env.PORT, () => {
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@ -66,19 +66,19 @@ const createWorker = async () => {
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worker = await mediasoup.createWorker({
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rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
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rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
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})
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});
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console.log(`[createWorker] worker pid ${worker.pid}`);
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worker.on('died', error => {
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worker.on('died', (error) => {
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// This implies something serious happened, so kill the application
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console.error('mediasoup worker has died', error);
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setTimeout(() => process.exit(1), 2000); // exit in 2 seconds
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})
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});
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return worker;
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} catch (error) {
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console.log(`ERROR | createWorker | ${error.message}`);
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}
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}
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};
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// We create a Worker as soon as our application starts
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worker = createWorker();
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@ -89,55 +89,51 @@ worker = createWorker();
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// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
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const mediaCodecs = [
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{
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kind : 'audio',
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mimeType : 'audio/opus',
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clockRate : 48000,
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channels : 2
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kind: 'audio',
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mimeType: 'audio/opus',
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clockRate: 48000,
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channels: 2,
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},
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{
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kind : 'video',
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mimeType : 'video/VP8',
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clockRate : 90000,
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parameters :
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{
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'x-google-start-bitrate' : 1000
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kind: 'video',
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mimeType: 'video/VP8',
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clockRate: 90000,
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parameters: {
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'x-google-start-bitrate': 1000,
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},
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channels : 2
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channels: 2,
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},
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{
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kind : 'video',
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mimeType : 'video/VP9',
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clockRate : 90000,
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parameters :
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{
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'profile-id' : 2,
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'x-google-start-bitrate' : 1000
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}
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kind: 'video',
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mimeType: 'video/VP9',
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clockRate: 90000,
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parameters: {
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'profile-id': 2,
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'x-google-start-bitrate': 1000,
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},
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},
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{
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kind : 'video',
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mimeType : 'video/h264',
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clockRate : 90000,
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parameters :
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{
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'packetization-mode' : 1,
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'profile-level-id' : '4d0032',
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'level-asymmetry-allowed' : 1,
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'x-google-start-bitrate' : 1000
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}
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kind: 'video',
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mimeType: 'video/h264',
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clockRate: 90000,
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parameters: {
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'packetization-mode': 1,
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'profile-level-id': '4d0032',
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'level-asymmetry-allowed': 1,
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'x-google-start-bitrate': 1000,
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},
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},
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{
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kind : 'video',
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mimeType : 'video/h264',
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clockRate : 90000,
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parameters :
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{
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'packetization-mode' : 1,
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'profile-level-id' : '42e01f',
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'level-asymmetry-allowed' : 1,
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'x-google-start-bitrate' : 1000
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}
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}
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kind: 'video',
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mimeType: 'video/h264',
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clockRate: 90000,
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parameters: {
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'packetization-mode': 1,
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'profile-level-id': '42e01f',
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'level-asymmetry-allowed': 1,
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'x-google-start-bitrate': 1000,
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},
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},
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];
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const closeCall = (callId) => {
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@ -158,18 +154,18 @@ const closeCall = (callId) => {
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} catch (error) {
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console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
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}
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}
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};
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/*
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- Handlers for WS events
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- These are created only when we have a connection with a peer
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*/
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peers.on('connection', async socket => {
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peers.on('connection', async (socket) => {
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console.log('[connection] socketId:', socket.id);
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// After making the connection successfully, we send the client a 'connection-success' event
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socket.emit('connection-success', {
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socketId: socket.id
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socketId: socket.id,
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});
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// It is triggered when the peer is disconnected
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@ -202,7 +198,7 @@ peers.on('connection', async socket => {
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// rtpCapabilities is set for callback
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console.log('[getRtpCapabilities] callId', callId);
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callbackResponse = {
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rtpCapabilities :videoCalls[callId].router.rtpCapabilities
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rtpCapabilities: videoCalls[callId].router.rtpCapabilities,
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};
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} else {
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console.log(`[createRoom] missing callId ${callId}`);
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@ -227,23 +223,25 @@ peers.on('connection', async socket => {
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const callId = socketDetails[socket.id];
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console.log(`[createWebRtcTransport] socket ${socket.id} | sender ${sender} | callId ${callId}`);
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if (sender) {
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if(!videoCalls[callId].receiverProducerTransport && !isInitiator(callId, socket.id)) {
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if (!videoCalls[callId].receiverProducerTransport && !isInitiator(callId, socket.id)) {
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videoCalls[callId].receiverProducerTransport = await createWebRtcTransportLayer(callId, callback);
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} else if(!videoCalls[callId].initiatorProducerTransport && isInitiator(callId, socket.id)) {
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} else if (!videoCalls[callId].initiatorProducerTransport && isInitiator(callId, socket.id)) {
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videoCalls[callId].initiatorProducerTransport = await createWebRtcTransportLayer(callId, callback);
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} else {
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console.log(`producerTransport has already been defined | callId ${callId}`);
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callback(null);
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}
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} else if (!sender) {
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if(!videoCalls[callId].receiverConsumerTransport && !isInitiator(callId, socket.id)) {
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if (!videoCalls[callId].receiverConsumerTransport && !isInitiator(callId, socket.id)) {
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videoCalls[callId].receiverConsumerTransport = await createWebRtcTransportLayer(callId, callback);
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} else if(!videoCalls[callId].initiatorConsumerTransport && isInitiator(callId, socket.id)) {
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} else if (!videoCalls[callId].initiatorConsumerTransport && isInitiator(callId, socket.id)) {
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videoCalls[callId].initiatorConsumerTransport = await createWebRtcTransportLayer(callId, callback);
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}
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}
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} catch (error) {
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console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
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console.log(
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`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`
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);
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callback(error);
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}
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});
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@ -259,18 +257,26 @@ peers.on('connection', async socket => {
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console.log(`[transport-connect] socket ${socket.id} | callId ${callId}`);
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if (!isInitiator(callId, socket.id)) {
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await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
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videoCalls[callId].initiatorSocket.emit('new-producer', {
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callId
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});
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} else {
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await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters });
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videoCalls[callId].receiverSocket.emit('new-producer', {
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callId
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});
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}
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isInitiator(callId, socket.id)
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? await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters })
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: await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
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const socketToEmit = isInitiator(callId, socket.id)
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? videoCalls[callId].receiverSocket
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: videoCalls[callId].initiatorSocket;
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socketToEmit.emit('new-producer', { callId });
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// if (!isInitiator(callId, socket.id)) {
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// await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
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// videoCalls[callId].initiatorSocket.emit('new-producer', {
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// callId,
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// });
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// } else {
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// await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters });
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// videoCalls[callId].receiverSocket.emit('new-producer', {
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// callId,
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// });
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// }
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} catch (error) {
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console.log(`ERROR | transport-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
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}
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@ -295,17 +301,20 @@ peers.on('connection', async socket => {
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rtpParameters,
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});
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console.log(`[transport-produce] receiverVideoProducer Producer ID: ${videoCalls[callId].receiverVideoProducer.id} | kind: ${videoCalls[callId].receiverVideoProducer.kind}`);
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console.log(
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`[transport-produce] receiverVideoProducer Producer ID: ${videoCalls[callId].receiverVideoProducer.id} | kind: ${videoCalls[callId].receiverVideoProducer.kind}`
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);
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videoCalls[callId].receiverVideoProducer.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport for this producer closed', callId)
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console.log('transport for this producer closed', callId);
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closeCall(callId);
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});
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// Send back to the client the Producer's id
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callback && callback({
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id: videoCalls[callId].receiverVideoProducer.id
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callback &&
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callback({
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id: videoCalls[callId].receiverVideoProducer.id,
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});
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} else {
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videoCalls[callId].initiatorVideoProducer = await videoCalls[callId].initiatorProducerTransport.produce({
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@ -313,16 +322,19 @@ peers.on('connection', async socket => {
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rtpParameters,
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});
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console.log(`[transport-produce] initiatorVideoProducer Producer ID: ${videoCalls[callId].initiatorVideoProducer.id} | kind: ${videoCalls[callId].initiatorVideoProducer.kind}`);
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console.log(
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`[transport-produce] initiatorVideoProducer Producer ID: ${videoCalls[callId].initiatorVideoProducer.id} | kind: ${videoCalls[callId].initiatorVideoProducer.kind}`
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);
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videoCalls[callId].initiatorVideoProducer.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport for this producer closed', callId)
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console.log('transport for this producer closed', callId);
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closeCall(callId);
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});
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callback && callback({
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id: videoCalls[callId].initiatorVideoProducer.id
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callback &&
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callback({
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id: videoCalls[callId].initiatorVideoProducer.id,
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});
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}
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} else if (kind === 'audio') {
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@ -332,17 +344,20 @@ peers.on('connection', async socket => {
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rtpParameters,
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});
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console.log(`[transport-produce] receiverAudioProducer Producer ID: ${videoCalls[callId].receiverAudioProducer.id} | kind: ${videoCalls[callId].receiverAudioProducer.kind}`);
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console.log(
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`[transport-produce] receiverAudioProducer Producer ID: ${videoCalls[callId].receiverAudioProducer.id} | kind: ${videoCalls[callId].receiverAudioProducer.kind}`
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);
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videoCalls[callId].receiverAudioProducer.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport for this producer closed', callId)
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console.log('transport for this producer closed', callId);
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closeCall(callId);
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});
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// Send back to the client the Producer's id
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callback && callback({
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id: videoCalls[callId].receiverAudioProducer.id
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callback &&
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callback({
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id: videoCalls[callId].receiverAudioProducer.id,
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});
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} else {
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videoCalls[callId].initiatorAudioProducer = await videoCalls[callId].initiatorProducerTransport.produce({
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@ -350,17 +365,20 @@ peers.on('connection', async socket => {
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rtpParameters,
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});
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console.log(`[transport-produce] initiatorAudioProducer Producer ID: ${videoCalls[callId].initiatorAudioProducer.id} | kind: ${videoCalls[callId].initiatorAudioProducer.kind}`);
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console.log(
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`[transport-produce] initiatorAudioProducer Producer ID: ${videoCalls[callId].initiatorAudioProducer.id} | kind: ${videoCalls[callId].initiatorAudioProducer.kind}`
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);
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videoCalls[callId].initiatorAudioProducer.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport for this producer closed', callId)
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console.log('transport for this producer closed', callId);
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closeCall(callId);
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});
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// Send back to the client the Producer's id
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callback && callback({
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id: videoCalls[callId].initiatorAudioProducer.id
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callback &&
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callback({
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id: videoCalls[callId].initiatorAudioProducer.id,
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});
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}
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}
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@ -379,15 +397,15 @@ peers.on('connection', async socket => {
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console.log(`[transport-recv-connect] socket ${socket.id} | callId ${callId}`);
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if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
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// await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
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if(!isInitiator(callId, socket.id)) {
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if (!isInitiator(callId, socket.id)) {
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await videoCalls[callId].receiverConsumerTransport.connect({ dtlsParameters });
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} else if(isInitiator(callId, socket.id)) {
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} else if (isInitiator(callId, socket.id)) {
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await videoCalls[callId].initiatorConsumerTransport.connect({ dtlsParameters });
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}
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} catch (error) {
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console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
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}
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})
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});
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/*
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- The customer consumes after successfully connecting to consumerTransport
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@ -399,45 +417,54 @@ peers.on('connection', async socket => {
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socket.on('consume', async ({ rtpCapabilities }, callback) => {
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try {
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const callId = socketDetails[socket.id];
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console.log(`[consume] socket ${socket.id} | callId ${callId} | rtpCapabilities: ${JSON.stringify(rtpCapabilities)}`);
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console.log(
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`[consume] socket ${socket.id} | callId ${callId} | rtpCapabilities: ${JSON.stringify(rtpCapabilities)}`
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);
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if (typeof rtpCapabilities === 'string') rtpCapabilities = JSON.parse(rtpCapabilities);
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console.log('[consume] callId', callId);
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let canConsumeVideo, canConsumeAudio;
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if (isInitiator(callId, socket.id)) {
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canConsumeVideo = !!videoCalls[callId].receiverVideoProducer && !!videoCalls[callId].router.canConsume({
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canConsumeVideo =
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!!videoCalls[callId].receiverVideoProducer &&
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!!videoCalls[callId].router.canConsume({
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producerId: videoCalls[callId].receiverVideoProducer.id,
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rtpCapabilities
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rtpCapabilities,
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});
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canConsumeAudio = !!videoCalls[callId].receiverAudioProducer && !!videoCalls[callId].router.canConsume({
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canConsumeAudio =
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!!videoCalls[callId].receiverAudioProducer &&
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!!videoCalls[callId].router.canConsume({
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producerId: videoCalls[callId].receiverAudioProducer.id,
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rtpCapabilities
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rtpCapabilities,
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});
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} else {
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canConsumeVideo = !!videoCalls[callId].initiatorVideoProducer && !!videoCalls[callId].router.canConsume({
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canConsumeVideo =
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!!videoCalls[callId].initiatorVideoProducer &&
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!!videoCalls[callId].router.canConsume({
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producerId: videoCalls[callId].initiatorVideoProducer.id,
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rtpCapabilities
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rtpCapabilities,
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});
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canConsumeAudio = !!videoCalls[callId].initiatorAudioProducer && !!videoCalls[callId].router.canConsume({
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canConsumeAudio =
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!!videoCalls[callId].initiatorAudioProducer &&
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!!videoCalls[callId].router.canConsume({
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producerId: videoCalls[callId].initiatorAudioProducer.id,
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rtpCapabilities
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rtpCapabilities,
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});
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}
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console.log('[consume] canConsumeVideo', canConsumeVideo);
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console.log('[consume] canConsumeAudio', canConsumeAudio);
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if (canConsumeVideo && !canConsumeAudio) {
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const videoParams = await consumeVideo(callId, socket.id, rtpCapabilities)
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const videoParams = await consumeVideo(callId, socket.id, rtpCapabilities);
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callback({ videoParams, audioParams: null });
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} else if (canConsumeVideo && canConsumeAudio) {
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const videoParams = await consumeVideo(callId, socket.id, rtpCapabilities)
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const audioParams = await consumeAudio(callId, socket.id, rtpCapabilities)
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const videoParams = await consumeVideo(callId, socket.id, rtpCapabilities);
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const audioParams = await consumeAudio(callId, socket.id, rtpCapabilities);
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callback({ videoParams, audioParams });
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} else if (!canConsumeVideo && canConsumeAudio) {
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const audioParams = await consumeAudio(callId, socket.id, rtpCapabilities)
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const audioParams = await consumeAudio(callId, socket.id, rtpCapabilities);
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const data = { videoParams: null, audioParams };
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callback(data);
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} else {
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@ -445,7 +472,7 @@ peers.on('connection', async socket => {
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callback(null);
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}
|
||||
} catch (error) {
|
||||
console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
|
||||
console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`);
|
||||
callback({ params: { error } });
|
||||
}
|
||||
});
|
||||
@ -455,10 +482,10 @@ peers.on('connection', async socket => {
|
||||
- When consuming on consumerTransport, it is initially done with paused: true, here we will resume
|
||||
- For the initiator we resume the initiatorConsumerAUDIO/VIDEO and for receiver the receiverConsumerAUDIO/VIDEO
|
||||
*/
|
||||
socket.on('consumer-resume',() => {
|
||||
socket.on('consumer-resume', () => {
|
||||
try {
|
||||
const callId = socketDetails[socket.id];
|
||||
console.log(`[consumer-resume] callId ${callId}`)
|
||||
console.log(`[consumer-resume] callId ${callId}`);
|
||||
if (isInitiator(callId, socket.id)) {
|
||||
videoCalls[callId]?.initiatorConsumerVideo?.resume();
|
||||
videoCalls[callId]?.initiatorConsumerAudio?.resume();
|
||||
@ -473,8 +500,7 @@ peers.on('connection', async socket => {
|
||||
});
|
||||
|
||||
const consumeVideo = async (callId, socketId, rtpCapabilities) => {
|
||||
|
||||
if(isInitiator(callId, socketId)) {
|
||||
if (isInitiator(callId, socketId)) {
|
||||
videoCalls[callId].initiatorConsumerVideo = await videoCalls[callId].initiatorConsumerTransport.consume({
|
||||
producerId: videoCalls[callId].receiverVideoProducer.id,
|
||||
rtpCapabilities,
|
||||
@ -500,7 +526,7 @@ const consumeVideo = async (callId, socketId, rtpCapabilities) => {
|
||||
producerId: videoCalls[callId].receiverVideoProducer.id,
|
||||
kind: 'video',
|
||||
rtpParameters: videoCalls[callId].initiatorConsumerVideo.rtpParameters,
|
||||
}
|
||||
};
|
||||
} else {
|
||||
videoCalls[callId].receiverConsumerVideo = await videoCalls[callId].receiverConsumerTransport.consume({
|
||||
producerId: videoCalls[callId].initiatorVideoProducer.id,
|
||||
@ -525,12 +551,12 @@ const consumeVideo = async (callId, socketId, rtpCapabilities) => {
|
||||
producerId: videoCalls[callId].initiatorVideoProducer.id,
|
||||
kind: 'video',
|
||||
rtpParameters: videoCalls[callId].receiverConsumerVideo.rtpParameters,
|
||||
};
|
||||
}
|
||||
}
|
||||
}
|
||||
};
|
||||
|
||||
const consumeAudio = async (callId, socketId, rtpCapabilities) => {
|
||||
if(isInitiator(callId, socketId)) {
|
||||
if (isInitiator(callId, socketId)) {
|
||||
videoCalls[callId].initiatorConsumerAudio = await videoCalls[callId].initiatorConsumerTransport.consume({
|
||||
producerId: videoCalls[callId].receiverAudioProducer.id,
|
||||
rtpCapabilities,
|
||||
@ -556,7 +582,7 @@ const consumeAudio = async (callId, socketId, rtpCapabilities) => {
|
||||
producerId: videoCalls[callId].receiverAudioProducer.id,
|
||||
kind: 'audio',
|
||||
rtpParameters: videoCalls[callId].initiatorConsumerAudio.rtpParameters,
|
||||
}
|
||||
};
|
||||
} else {
|
||||
videoCalls[callId].receiverConsumerAudio = await videoCalls[callId].receiverConsumerTransport.consume({
|
||||
producerId: videoCalls[callId].initiatorAudioProducer.id,
|
||||
@ -581,13 +607,13 @@ const consumeAudio = async (callId, socketId, rtpCapabilities) => {
|
||||
producerId: videoCalls[callId].initiatorAudioProducer.id,
|
||||
kind: 'audio',
|
||||
rtpParameters: videoCalls[callId].receiverConsumerAudio.rtpParameters,
|
||||
};
|
||||
}
|
||||
}
|
||||
}
|
||||
};
|
||||
|
||||
const isInitiator = (callId, socketId) => {
|
||||
return (videoCalls[callId]?.initiatorSocket?.id === socketId);
|
||||
}
|
||||
return videoCalls[callId]?.initiatorSocket?.id === socketId;
|
||||
};
|
||||
|
||||
/*
|
||||
- Called from at event 'createWebRtcTransport' and assigned to the consumer or producer transport
|
||||
@ -605,7 +631,7 @@ const createWebRtcTransportLayer = async (callId, callback) => {
|
||||
{
|
||||
ip: process.env.IP, // Listening IPv4 or IPv6.
|
||||
announcedIp: process.env.ANNOUNCED_IP, // Announced IPv4 or IPv6 (useful when running mediasoup behind NAT with private IP).
|
||||
}
|
||||
},
|
||||
],
|
||||
enableUdp: true,
|
||||
enableTcp: true,
|
||||
@ -613,10 +639,10 @@ const createWebRtcTransportLayer = async (callId, callback) => {
|
||||
};
|
||||
|
||||
// https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
|
||||
let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options)
|
||||
let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options);
|
||||
|
||||
// Handler for when DTLS(Datagram Transport Layer Security) changes
|
||||
transport.on('dtlsstatechange', dtlsState => {
|
||||
transport.on('dtlsstatechange', (dtlsState) => {
|
||||
console.log(`transport | dtlsstatechange | calldId ${callId} | dtlsState ${dtlsState}`);
|
||||
if (dtlsState === 'closed') {
|
||||
transport.close();
|
||||
@ -641,9 +667,8 @@ const createWebRtcTransportLayer = async (callId, callback) => {
|
||||
|
||||
// Set transport to producerTransport or consumerTransport
|
||||
return transport;
|
||||
|
||||
} catch (error) {
|
||||
console.log(`ERROR | createWebRtcTransportLayer | callId ${socketDetails[socket.id]} | ${error.message}`);
|
||||
callback({ params: { error } });
|
||||
}
|
||||
}
|
||||
};
|
||||
|
||||
Loading…
Reference in New Issue
Block a user