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Author SHA1 Message Date
451fff0a6b Update 2022-12-23 11:39:49 +02:00
a568e542a7 Update 2022-12-22 09:34:31 +02:00
14af825eab Update 2022-12-22 09:26:37 +02:00
f3ba6d37c2 Update 2022-12-22 09:22:23 +02:00
a4356f06be Update 2022-12-22 09:16:52 +02:00
c4d6cdd209 Merge branch 'LINXD-2270-p2' of https://git.safemobile.org/Safemobile/mediasoup into LINXD-2270-p2 2022-12-19 19:08:53 +02:00
1520fb88ae Update server and web client so support full duplex 2022-12-19 19:07:08 +02:00
da906ed4ba Update 2022-12-19 18:35:02 +02:00
2dd9eb5eaf Update 2022-12-19 14:05:44 +02:00
b4fccc4d4c Update 2022-12-19 13:57:17 +02:00
88da70731f Update 2022-12-19 13:40:43 +02:00
68c80d563f Update 2022-12-19 13:32:56 +02:00
eb668e2500 Update 2022-12-19 13:29:24 +02:00
e3536e87cd Update 2022-12-19 13:19:58 +02:00
8634cb4b6e Update 2022-12-19 13:18:33 +02:00
2a6f24b2bb Update 2022-12-19 13:17:11 +02:00
381e665062 Update 2022-12-19 13:14:34 +02:00
33e30339f2 Update 2022-12-19 13:12:49 +02:00
968da6ea98 Update 2022-12-19 12:40:54 +02:00
2ada7b66db Update 2022-12-19 12:39:22 +02:00
28059144cf Update 2022-12-19 12:33:19 +02:00
714fe0ec5e Update 2022-12-19 12:31:39 +02:00
898cc0cbf2 Update 2022-12-19 12:30:25 +02:00
6379e1ae34 Update 2022-12-19 12:27:24 +02:00
dd264e39ea Update 2022-12-19 12:25:42 +02:00
72136132ba Update 2022-12-19 12:25:14 +02:00
4978e8d51f Update 2022-12-19 12:22:21 +02:00
825ded83c2 Update 2022-12-19 12:13:48 +02:00
b15357d089 Update 2022-12-19 12:01:21 +02:00
bb4cd756f3 Update 2022-12-19 11:29:24 +02:00
0731090b0e Update 2022-12-19 11:26:08 +02:00
782683c3e2 Update 2022-12-19 11:24:19 +02:00
dfa175d0c7 Update 2022-12-19 11:22:28 +02:00
2c375c01ea Update 2022-12-19 11:21:02 +02:00
2fbb355fea Update 2022-12-19 11:19:51 +02:00
2ddeb4baaa Update 2022-12-16 14:13:11 +02:00
a31e646e2b Update 2022-12-16 14:10:26 +02:00
fe792f93b6 Update 2022-12-16 14:10:07 +02:00
dafbc486ad Update 2022-12-16 14:00:16 +02:00
b606a72030 Update 2022-12-16 13:43:55 +02:00
c174e92e3c Update 2022-12-16 13:33:13 +02:00
449724537e Update 2022-12-16 12:01:20 +02:00
9634aac153 Update 2022-12-16 11:59:10 +02:00
e0bc4642cb Update 2022-12-16 11:31:26 +02:00
f950142188 Update 2022-12-16 11:28:57 +02:00
5ba1f76585 Update 2022-12-16 11:23:10 +02:00
dc9c91fccc Update 2022-12-16 11:10:30 +02:00
5abcddc115 Update 2022-12-16 11:02:08 +02:00
bf65221664 Update 2022-12-16 10:54:25 +02:00
5687569bc1 Update 2022-12-16 10:47:13 +02:00
44c8d9b8ee Update 2022-12-16 02:15:26 +02:00
0a6985f9b9 Update 2022-12-16 02:12:22 +02:00
d29def364c Update 2022-12-16 02:10:10 +02:00
acd6025f59 Update 2022-12-16 02:07:01 +02:00
4b0c06e0b0 Added socket id to createWebRtcTransport 2022-12-16 02:02:38 +02:00
c1fe524ec7 LINXD-2270: Remove commented code; Update comments/logs 2022-12-14 11:57:19 +02:00
f8fcfb3165 Fix isInitiator in transport-produce 2022-12-14 11:33:16 +02:00
d324528d52 Added logs on transport-produce 2022-12-14 11:05:40 +02:00
d1eb7afc3a Added logs on createRoom on videoCalls 2022-12-14 10:15:34 +02:00
695964d342 Refactor code to use initiator/receiver 2022-12-14 09:55:45 +02:00
3ca555ef9e Set initiatorSocketId to be dispatcher 2022-12-13 13:23:15 +02:00
92fbecc36a Set initiatorSocketId to be dispatcher 2022-12-13 13:04:43 +02:00
d633eec92f Add socket info 2022-12-13 10:28:45 +02:00
8a9c370f02 Merge pull request 'LH-265-audio' (#17) from LH-265-audio into develop
Reviewed-on: #17
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2022-12-06 12:45:08 +00:00
652019b07d LH-265: Update doc; Update bundle 2022-11-29 15:35:28 +02:00
09c4a4b90e LH-265: Added audio on client and server 2022-11-29 14:19:02 +02:00
75d0e3aee7 exe right for build.sh 2022-10-31 22:26:08 +00:00
30ac997634 Merge pull request 'added build.sh' (#14) from temp-build into develop
Reviewed-on: #14
2022-10-31 22:22:20 +00:00
5aea138f6a added build.sh 2022-10-31 12:17:07 +02:00
5b01ddc2a8 Merge pull request 'LH-259-mediasoup-always-return-a-callback-response-to-clients' (#13) from LH-259-mediasoup-always-return-a-callback-response-to-clients into develop
Reviewed-on: #13
2022-10-25 16:18:54 +00:00
084ff36ebe LH-259: Refactor createRoom 2022-10-24 22:38:06 +03:00
f4ebf92783 LH-259: Added callback from transport-produce 2022-10-24 22:35:22 +03:00
b59a157b18 LH-259: Comment callback from transport-produce 2022-10-24 22:19:37 +03:00
9f8347bec5 LH-259: Update createRoom callback 2022-10-24 22:16:47 +03:00
24390c98e5 LH-259: Added callbacks 2022-10-24 22:11:14 +03:00
1a7371fe18 Parse RTC_MIN_PORT and RTC_MAX_PORT 2022-10-18 18:27:02 +03:00
be5f97762a Merge pull request 'LH-253: Added callId for transportclose and producerclose events' (#12) from LH-253-mediasoup-handle-callid-undefined into master
Reviewed-on: #12
2022-10-18 07:53:33 +00:00
03a11126c4 LH-253: Check if we have callId in closeCall 2022-10-18 10:51:20 +03:00
fafbee6e4c LH-253: Added callId for transportclose and producerclose events 2022-10-18 02:05:22 +03:00
bbf23c33d4 Merge pull request 'LH-252: Update .env variables' (#11) from LH-252-mediasoup-add-a-config-file-with-keys-and-ports into master
Reviewed-on: #11
2022-10-09 06:50:45 +00:00
5c2808e75a LH-252: Update .env variables 2022-10-06 15:21:54 +03:00
2aea7497cc Merge pull request 'added log for dtls transport-connect' (#10) from LH-249-debug-for-i-os-dtls-problems into master
Reviewed-on: #10
2022-10-06 06:41:07 +00:00
56835d6660 added log for dtls transport-connect 2022-10-05 15:44:46 +03:00
fc42c79210 Fix missing callId 2022-09-27 13:13:29 +03:00
d81bc8582d Merge branch 'master' of https://git.safemobile.org/Safemobile/mediasoup 2022-09-27 13:05:15 +03:00
a4d16998cd Fix call check before call close() 2022-09-27 13:03:32 +03:00
de1458bbde Merge pull request 'LINXD-2197: Added comments; Catch errors; Fix package.json start:run script' (#8) from LINXD-2197-refactor-improving-mediasoup-web-socket-component into master
Reviewed-on: #8
2022-09-27 10:00:25 +00:00
b0fad5f1db LINXD-2197: On peer disconnect delete the call; Added log when call is already deleted; Added log when user send multiple createWebRtcTransport 2022-09-27 12:43:07 +03:00
eb5aa12d65 LINXD-2197: Added the initial demo project used; Check before set producerTransport and consumerTransport if it was set before 2022-09-27 07:55:25 +03:00
52b4794a86 LINXD-2197: Added workflow diagram 2022-09-25 20:29:32 +03:00
13 changed files with 1481 additions and 1140 deletions

4
.env
View File

@ -1,3 +1,7 @@
PORT=3000
IP=0.0.0.0 # Listening IPv4 or IPv6.
ANNOUNCED_IP=185.8.154.190 # Announced IPv4 or IPv6 (useful when running mediasoup behind NAT with private IP).
RTC_MIN_PORT=2000
RTC_MAX_PORT=2020
SERVER_CERT="./server/ssl/cert.pem"
SERVER_KEY="./server/ssl/key.pem"

1
.gitignore vendored
View File

@ -1 +1,2 @@
/node_modules
/dist

View File

@ -22,14 +22,20 @@
2. Run the `npm start:prod` command to start the server in production mode.
(To connect to the terminal, use `pm2 log video-server`)
---
### Web client
- The server will start by default on port 3000, and the ssl certificates will have to be configured
- The web client can be accessed using the /sfu path
ex: http://localhost:3000/sfu/?assetId=1&&accountId=1&producer=true&assetName=Adi&assetType=linx
ex: https://HOST/sfu/?assetId=1&&accountId=1&producer=true&dest_asset_id=75&assetName=Adi
assetId = asset id of the unit on which you are doing the test
accountId = account id of the unit on which you are doing the test
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
dest_asset_id= the addressee with whom the call is made
- To make a call using this client, you need a microphone and permission to use it
- For any changes related to the client, the command `npm run watch' will have to be used to generate the bundle.js used by the web client
### Demo project
The demo project used initially and then modified for our needs `https://github.com/jamalag/mediasoup2`

471
app.js
View File

@ -5,7 +5,7 @@ const app = express();
const Server = require('socket.io');
const path = require('node:path');
const fs = require('node:fs');
let https = require('https');
let https;
try {
https = require('node:https');
} catch (err) {
@ -15,19 +15,21 @@ const mediasoup = require('mediasoup');
let worker
/**
* videoCalls
* |-> Router
* |-> Producer
* |-> Consumer
* |-> Producer Transport
* |-> Consumer Transport
*
* videoCalls - Dictionary of Object(s)
* '<callId>': {
* router: Router,
* producer: Producer,
* producerTransport: Producer Transport,
* consumer: Consumer,
* consumerTransport: Consumer Transport
* router: Router, router
* initiatorAudioProducer: Producer,
* initiatorVideoProducer: Producer,
* receiverVideoProducer: Producer, producerVideo
* receiverAudioProducer: Producer, producerAudio
* initiatorProducerTransport: Producer Transport,
* receiverProducerTransport: Producer Transport, producerTransport
* initiatorConsumerVideo: Consumer, consumerVideo
* initiatorConsumerAudio: Consumer, consumerAudio
* initiatorConsumerTransport: Consumer Transport consumerTransport
* initiatorSockerId
* receiverSocketId
* }
*
**/
@ -42,33 +44,28 @@ app.use('/sfu', express.static(path.join(__dirname, 'public')))
// SSL cert for HTTPS access
const options = {
key: fs.readFileSync('./server/ssl/key.pem', 'utf-8'),
cert: fs.readFileSync('./server/ssl/cert.pem', 'utf-8'),
key: fs.readFileSync(process.env.SERVER_KEY, 'utf-8'),
cert: fs.readFileSync(process.env.SERVER_CERT, 'utf-8'),
}
const httpsServer = https.createServer(options, app);
const io = new Server(httpsServer, {
allowEIO3: true,
origins: ["*:*"],
// allowRequest: (req, next) => {
// console.log('req', req);
// next(null, true)
// }
origins: ["*:*"]
});
// const io = new Server(server, { origins: '*:*', allowEIO3: true });
httpsServer.listen(process.env.PORT, () => {
console.log('Video server listening on port:', process.env.PORT)
})
console.log('Video server listening on port:', process.env.PORT);
});
const peers = io.of('/')
const peers = io.of('/');
const createWorker = async () => {
try {
worker = await mediasoup.createWorker({
rtcMinPort: 2000,
rtcMaxPort: 2020,
rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
})
console.log(`[createWorker] worker pid ${worker.pid}`);
@ -84,7 +81,7 @@ const createWorker = async () => {
}
// We create a Worker as soon as our application starts
worker = createWorker()
worker = createWorker();
// This is an Array of RtpCapabilities
// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
@ -95,43 +92,74 @@ const mediaCodecs = [
kind : 'audio',
mimeType : 'audio/opus',
clockRate : 48000,
channels: 2,
channels : 2
},
{
kind : 'video',
mimeType : 'video/VP8',
clockRate : 90000,
parameters: {
'x-google-start-bitrate': 1000,
parameters :
{
'x-google-start-bitrate' : 1000
},
channels : 2
},
]
{
kind : 'video',
mimeType : 'video/VP9',
clockRate : 90000,
parameters :
{
'profile-id' : 2,
'x-google-start-bitrate' : 1000
}
},
{
kind : 'video',
mimeType : 'video/h264',
clockRate : 90000,
parameters :
{
'packetization-mode' : 1,
'profile-level-id' : '4d0032',
'level-asymmetry-allowed' : 1,
'x-google-start-bitrate' : 1000
}
},
{
kind : 'video',
mimeType : 'video/h264',
clockRate : 90000,
parameters :
{
'packetization-mode' : 1,
'profile-level-id' : '42e01f',
'level-asymmetry-allowed' : 1,
'x-google-start-bitrate' : 1000
}
}
];
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();
if (callId && videoCalls[callId]) {
videoCalls[callId].receiverVideoProducer?.close();
videoCalls[callId].receiverAudioProducer?.close();
videoCalls[callId].initiatorConsumerVideo?.close();
videoCalls[callId].initiatorConsumerAudio?.close();
videoCalls[callId]?.initiatorConsumerTransport?.close();
videoCalls[callId]?.receiverProducerTransport?.close();
videoCalls[callId]?.router?.close();
delete videoCalls[callId];
} else {
console.log(`The call with id ${callId} has already been deleted`);
}
} catch (error) {
console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
}
}
const getRtpCapabilities = (callId, callback) => {
try {
console.log('[getRtpCapabilities] callId', callId);
const rtpCapabilities = videoCalls[callId].router.rtpCapabilities;
callback({ rtpCapabilities });
} catch (error) {
console.log(`ERROR | getRtpCapabilities | callId ${callId} | ${error.message}`);
}
}
/*
- Handlers for WS events
- These are created only when we have a connection with a peer
@ -146,8 +174,10 @@ peers.on('connection', async socket => {
// It is triggered when the peer is disconnected
socket.on('disconnect', () => {
console.log('peer disconnected | socket.id', socket.id);
const callId = socketDetails[socket.id];
console.log(`disconnect | socket ${socket.id} | callId ${callId}`);
delete socketDetails[socket.id];
closeCall(callId);
});
/*
@ -156,27 +186,38 @@ peers.on('connection', async socket => {
- If the room already exists, it will not create it, but will only return rtpCapabilities
*/
socket.on('createRoom', async ({ callId }, callback) => {
let callbackResponse = null;
try {
// We can continue with the room creation process only if we have a callId
if (callId) {
console.log(`[createRoom] socket.id ${socket.id} callId ${callId}`);
if (!videoCalls[callId]) {
console.log('[createRoom] callId', callId);
videoCalls[callId] = { router: await worker.createRouter({ mediaCodecs }) }
console.log(`[createRoom] Router ID: ${videoCalls[callId].router.id}`);
videoCalls[callId].receiverSocketId = socket.id
} else {
videoCalls[callId].initiatorSockerId = socket.id
}
socketDetails[socket.id] = callId;
getRtpCapabilities(callId, callback);
// rtpCapabilities is set for callback
console.log('[getRtpCapabilities] callId', callId);
callbackResponse = {
rtpCapabilities :videoCalls[callId].router.rtpCapabilities
};
} else {
console.log(`[createRoom] missing callId ${callId}`);
}
} catch (error) {
console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
} finally {
callback(callbackResponse);
}
});
/*
- Client emits a request to create server side Transport
- Depending on the sender, producerTransport or consumerTransport is created on that router
- Depending on the sender, a producer or consumer is created 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)
@ -184,15 +225,28 @@ peers.on('connection', async socket => {
*/
socket.on('createWebRtcTransport', async ({ sender }, callback) => {
try {
console.log('🟥', socket.id, JSON.stringify(sender));
const callId = socketDetails[socket.id];
console.log(`[createWebRtcTransport] sender ${sender} | callId ${callId}`);
console.log(`[createWebRtcTransport] socket ${socket.id} | sender ${sender} | callId ${callId}`);
if (sender) {
videoCalls[callId].producerTransport = await createWebRtcTransportLayer(callId, callback);
if(!videoCalls[callId].receiverProducerTransport && !isInitiator(callId, socket.id)) {
videoCalls[callId].receiverProducerTransport = await createWebRtcTransportLayer(callId, callback);
} else if(!videoCalls[callId].initiatorProducerTransport && isInitiator(callId, socket.id)) {
videoCalls[callId].initiatorProducerTransport = await createWebRtcTransportLayer(callId, callback);
} else {
videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
console.log(`producerTransport has already been defined | callId ${callId}`);
callback(null);
}
} else if (!sender) {
if(!videoCalls[callId].receiverConsumerTransport && !isInitiator(callId, socket.id)) {
videoCalls[callId].receiverConsumerTransport = await createWebRtcTransportLayer(callId, callback);
} else if(!videoCalls[callId].initiatorConsumerTransport && isInitiator(callId, socket.id)) {
videoCalls[callId].initiatorConsumerTransport = await createWebRtcTransportLayer(callId, callback);
}
}
} catch (error) {
console.log(`ERROR | createWebRtcTransport | callId ${callId} | sender ${sender} | ${error.message}`);
console.log(`ERROR | createWebRtcTransport | callId ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
callback(error);
}
});
@ -203,33 +257,106 @@ peers.on('connection', async socket => {
socket.on('transport-connect', async ({ dtlsParameters }) => {
try {
const callId = socketDetails[socket.id];
console.log(`[transport-connect] socket.id ${socket.id} | callId ${callId}`)
await videoCalls[callId].producerTransport.connect({ dtlsParameters });
if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
console.log(`[transport-connect] socket ${socket.id} | callId ${callId}`);
if (!isInitiator(callId, socket.id)) {
await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
} else {
await videoCalls[callId].initiatorProducerTransport.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
- The event sent by the client (PRODUCER) after successfully connecting to receiverProducerTransport/initiatorProducerTransport
- For the router with the id callId, we make produce on receiverProducerTransport/initiatorProducerTransport
- Create the handler on producer at the 'transportclose' event
*/
socket.on('transport-produce', async ({ kind, rtpParameters, appData }) => {
socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
try {
const callId = socketDetails[socket.id];
console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
console.log(`[transport-produce] kind: ${kind} | socket: ${socket.id} | callId: ${callId}`);
if (kind === 'video') {
if (!isInitiator(callId, socket.id)) {
videoCalls[callId].receiverVideoProducer = await videoCalls[callId].receiverProducerTransport.produce({
kind,
rtpParameters,
});
console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producer.id} | kind: ${videoCalls[callId].producer.kind}`);
videoCalls[callId].producer.on('transportclose', () => {
console.log(`[transport-produce] receiverVideoProducer Producer ID: ${videoCalls[callId].receiverVideoProducer.id} | kind: ${videoCalls[callId].receiverVideoProducer.kind}`);
videoCalls[callId].receiverVideoProducer.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 && callback({
id: videoCalls[callId].receiverVideoProducer.id
});
} else {
videoCalls[callId].initiatorVideoProducer = await videoCalls[callId].initiatorProducerTransport.produce({
kind,
rtpParameters,
});
console.log(`[transport-produce] initiatorVideoProducer Producer ID: ${videoCalls[callId].initiatorVideoProducer.id} | kind: ${videoCalls[callId].initiatorVideoProducer.kind}`);
videoCalls[callId].initiatorVideoProducer.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport for this producer closed', callId)
closeCall(callId);
});
callback && callback({
id: videoCalls[callId].initiatorVideoProducer.id
});
}
} else if (kind === 'audio') {
if (!isInitiator(callId, socket.id)) {
videoCalls[callId].receiverAudioProducer = await videoCalls[callId].receiverProducerTransport.produce({
kind,
rtpParameters,
});
console.log(`[transport-produce] receiverAudioProducer Producer ID: ${videoCalls[callId].receiverAudioProducer.id} | kind: ${videoCalls[callId].receiverAudioProducer.kind}`);
videoCalls[callId].receiverAudioProducer.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 && callback({
id: videoCalls[callId].receiverAudioProducer.id
});
} else {
videoCalls[callId].initiatorAudioProducer = await videoCalls[callId].initiatorProducerTransport.produce({
kind,
rtpParameters,
});
console.log(`[transport-produce] initiatorAudioProducer Producer ID: ${videoCalls[callId].initiatorAudioProducer.id} | kind: ${videoCalls[callId].initiatorAudioProducer.kind}`);
videoCalls[callId].initiatorAudioProducer.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 && callback({
id: videoCalls[callId].initiatorAudioProducer.id
});
}
}
} catch (error) {
console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
}
@ -242,10 +369,15 @@ peers.on('connection', async socket => {
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 });
console.log(`[transport-recv-connect] socket ${socket.id} | callId ${callId}`);
// await videoCalls[callId].consumerTransport.connect({ dtlsParameters });
if(!isInitiator(callId, socket.id)) {
await videoCalls[callId].receiverConsumerTransport.connect({ dtlsParameters });
} else if(isInitiator(callId, socket.id)) {
await videoCalls[callId].initiatorConsumerTransport.connect({ dtlsParameters });
}
} catch (error) {
console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ERROR`);
console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
}
})
@ -259,50 +391,52 @@ peers.on('connection', async socket => {
socket.on('consume', async ({ rtpCapabilities }, callback) => {
try {
const callId = socketDetails[socket.id];
console.log(`[consume] socket ${socket.id} | callId ${callId} | rtpCapabilities: ${JSON.stringify(rtpCapabilities)}`);
console.log('[consume] callId', callId);
// Check if the router can consume the specified producer
if (videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producer.id,
let canConsumeVideo, canConsumeAudio;
if (isInitiator(callId, socket.id)) {
canConsumeVideo = !!videoCalls[callId].receiverVideoProducer && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].receiverVideoProducer.id,
rtpCapabilities
});
canConsumeAudio = !!videoCalls[callId].receiverAudioProducer && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].receiverAudioProducer.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}`);
canConsumeVideo = !!videoCalls[callId].initiatorVideoProducer && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].initiatorVideoProducer.id,
rtpCapabilities
});
canConsumeAudio = !!videoCalls[callId].initiatorAudioProducer && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].initiatorAudioProducer.id,
rtpCapabilities
});
}
console.log('[consume] canConsumeVideo', canConsumeVideo);
console.log('[consume] canConsumeAudio', canConsumeAudio);
if (canConsumeVideo && !canConsumeAudio) {
const videoParams = await consumeVideo(callId, socket.id, rtpCapabilities)
callback({ videoParams, audioParams: null });
} else if (canConsumeVideo && canConsumeAudio) {
const videoParams = await consumeVideo(callId, socket.id, rtpCapabilities)
const audioParams = await consumeAudio(callId, socket.id, rtpCapabilities)
callback({ videoParams, audioParams });
} else if (!canConsumeVideo && canConsumeAudio) {
const audioParams = await consumeAudio(callId, socket.id, rtpCapabilities)
const data = { videoParams: null, audioParams };
console.log('-----------======= data', data);
callback(data);
} else {
console.log(`[consume] Can't consume | callId ${callId}`);
callback(null);
}
} catch (error) {
console.log(`ERROR | consume | callId ${callId} | ${error.message}`)
console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
callback({ params: { error } });
}
});
@ -315,13 +449,141 @@ peers.on('connection', async socket => {
try {
const callId = socketDetails[socket.id];
console.log(`[consumer-resume] callId ${callId}`)
await videoCalls[callId].consumer.resume();
if (isInitiator(callId, socket.id)) {
console.log(`[consumer-resume] isInitiator true`);
await videoCalls[callId].initiatorConsumerVideo.resume();
await videoCalls[callId].initiatorConsumerAudio.resume();
} else {
console.log(`[consumer-resume] isInitiator false`);
(videoCalls[callId].receiverConsumerVideo) && await videoCalls[callId].receiverConsumerVideo.resume();
(videoCalls[callId].receiverConsumerVideo) && await videoCalls[callId].receiverConsumerAudio.resume();
}
// await videoCalls[callId].consumerVideo.resume();
// await videoCalls[callId].consumerAudio.resume();
} catch (error) {
console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
}
});
});
const consumeVideo = async (callId, socketId, rtpCapabilities) => {
if(isInitiator(callId, socketId)) {
videoCalls[callId].initiatorConsumerVideo = await videoCalls[callId].initiatorConsumerTransport.consume({
producerId: videoCalls[callId].receiverVideoProducer.id,
rtpCapabilities,
paused: true,
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
videoCalls[callId].initiatorConsumerVideo.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport close from consumer', callId);
closeCall(callId);
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
videoCalls[callId].initiatorConsumerVideo.on('producerclose', () => {
const callId = socketDetails[socket.id];
console.log('producer of consumer closed', callId);
closeCall(callId);
});
return {
id: videoCalls[callId].initiatorConsumerVideo.id,
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,
rtpCapabilities,
paused: true,
});
videoCalls[callId].receiverConsumerVideo.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport close from consumer', callId);
closeCall(callId);
});
videoCalls[callId].receiverConsumerVideo.on('producerclose', () => {
const callId = socketDetails[socket.id];
console.log('producer of consumer closed', callId);
closeCall(callId);
});
return {
id: videoCalls[callId].receiverConsumerVideo.id,
producerId: videoCalls[callId].initiatorVideoProducer.id,
kind: 'video',
rtpParameters: videoCalls[callId].receiverConsumerVideo.rtpParameters,
}
}
}
const consumeAudio = async (callId, socketId, rtpCapabilities) => {
if(isInitiator(callId, socketId)) {
videoCalls[callId].initiatorConsumerAudio = await videoCalls[callId].initiatorConsumerTransport.consume({
producerId: videoCalls[callId].receiverAudioProducer.id,
rtpCapabilities,
paused: true,
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
videoCalls[callId].initiatorConsumerAudio.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport close from consumer', callId);
closeCall(callId);
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
videoCalls[callId].initiatorConsumerAudio.on('producerclose', () => {
const callId = socketDetails[socket.id];
console.log('producer of consumer closed', callId);
closeCall(callId);
});
return {
id: videoCalls[callId].initiatorConsumerAudio.id,
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,
rtpCapabilities,
paused: true,
});
videoCalls[callId].receiverConsumerAudio.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport close from consumer', callId);
closeCall(callId);
});
videoCalls[callId].receiverConsumerAudio.on('producerclose', () => {
const callId = socketDetails[socket.id];
console.log('producer of consumer closed', callId);
closeCall(callId);
});
return {
id: videoCalls[callId].receiverConsumerAudio.id,
producerId: videoCalls[callId].initiatorAudioProducer.id,
kind: 'audio',
rtpParameters: videoCalls[callId].receiverConsumerAudio.rtpParameters,
}
}
}
const isInitiator = (callId, socketId) => {
return (videoCalls[callId].initiatorSockerId === socketId);
}
/*
- 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
@ -369,6 +631,7 @@ const createWebRtcTransportLayer = async (callId, callback) => {
dtlsParameters: transport.dtlsParameters,
};
console.log('[createWebRtcTransportLayer] callback params', params);
// Send back to the client the params
callback({ params });
@ -376,7 +639,7 @@ const createWebRtcTransportLayer = async (callId, callback) => {
return transport;
} catch (error) {
console.log(`ERROR | createWebRtcTransportLayer | callId ${callId} | ${error.message}`);
console.log(`ERROR | createWebRtcTransportLayer | callId ${socketDetails[socket.id]} | ${error.message}`);
callback({ params: { error } });
}
}

46
build.sh Executable file
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@ -0,0 +1,46 @@
#/!bin/bash
## PREBUILD PROCESS
# check dist dir to be present and empty
if [ ! -d "dist" ]; then
## MAKE DIR
mkdir "dist"
echo "Directory dist created."
else
## CLEANUP
rm -fr dist/*
fi
# Install dependencies
#npm install
## PROJECT NEEDS
echo "Building app... from $(git rev-parse --abbrev-ref HEAD)"
#npm run-script build
cp -r {.env,app.js,package.json,server,public} dist/
#Add version control for pm2
cd dist
#Add version control for pm2
version=$(git describe)
file_pkg="package.json"
key=" \"version\": \""
count=$(echo ${version%%-*} | grep -o "\." | wc -l)
if (( $count > 1 )); then
version=${version%%-*}
else
version="${version%%-*}.0"
fi
if [ -f "$file_pkg" ] && [ ! -z "$version" ]; then
version=" \"version\": \"$version\","
sed -i "s|^.*$key.*|${version//\//\\/}|g" $file_pkg
text=$(cat $file_pkg | grep -c "$version")
if [ $text -eq 0 ]; then
echo "Version couldn't be set"
else
echo "Version $version successfully applied to App"
fi
fi
## POST BUILD
cd -

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@ -1,5 +1,4 @@
module.exports = {
hubAddress: 'https://hub.dev.linx.safemobile.com/',
mediasoupAddress: 'https://video.safemobile.org/mediasoup',
// mediasoupAddress: 'http://localhost:3000/mediasoup',
mediasoupAddress: 'https://testing.video.safemobile.org',
}

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@ -34,6 +34,9 @@
<body>
<body>
<div id="video">
<legend>Client options:</legend>
<input type="checkbox" id="produceAudio" name="produceAudio">
<label for="produceAudio">Produce audio</label><br>
<table>
<thead>
<th>Local Video</th>
@ -43,12 +46,24 @@
<tr>
<td>
<div id="sharedBtns">
<video id="localVideo" autoplay class="video" ></video>
<video
id="localVideo"
class="video"
autoplay
muted
playsinline
></video>
</div>
</td>
<td>
<div id="sharedBtns">
<video id="remoteVideo" autoplay class="video" ></video>
<video
id="remoteVideo"
class="video"
autoplay
muted
playsinline
></video>
</div>
</td>
</tr>
@ -61,33 +76,10 @@
<td>
<div id="sharedBtns">
<button id="btnRecvSendTransport">Consume</button>
<button id="remoteSoundControl">Unmute</button>
</div>
</td>
</tr>
<!-- <tr>
<td colspan="2">
<div id="sharedBtns">
<button id="btnRtpCapabilities">2. Get Rtp Capabilities</button>
<br />
<button id="btnDevice">3. Create Device</button>
</div>
</td>
</tr>
<tr>
<td>
<div id="sharedBtns">
<button id="btnCreateSendTransport">4. Create Send Transport</button>
<br />
<button id="btnConnectSendTransport">5. Connect Send Transport & Produce</button></td>
</div>
<td>
<div id="sharedBtns">
<button id="btnRecvSendTransport">6. Create Recv Transport</button>
<br />
<button id="btnConnectRecvTransport">7. Connect Recv Transport & Consume</button>
</div>
</td>
</tr> -->
</tbody>
</table>
<div id="closeCallBtn">

View File

@ -10,9 +10,72 @@ const ASSET_NAME = urlParams.get('assetName') || null;
const ASSET_TYPE = urlParams.get('assetType') || null;
let callId = parseInt(urlParams.get('callId')) || null;
const IS_PRODUCER = urlParams.get('producer') === 'true' ? true : false
let remoteVideo = document.getElementById('remoteVideo')
remoteVideo.defaultMuted = true
let produceAudio = false
console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId', callId, ' | IS_PRODUCER', IS_PRODUCER)
let socket
console.log('🟩 config', config)
produceAudioSelector = document.getElementById('produceAudio');
produceAudioSelector.addEventListener('change', e => {
if(e.target.checked) {
produceAudio = true
console.log('produce audio');
} else {
produceAudio = false
}
});
let socket, hub
let device
let rtpCapabilities
let producerTransport
let consumerTransport
let producerVideo
let producerAudio
let consumer
let originAssetId
let consumerVideo // local consumer video(consumer not transport)
let consumerAudio // local consumer audio(consumer not transport)
const remoteSoundControl = document.getElementById('remoteSoundControl');
remoteSoundControl.addEventListener('click', function handleClick() {
console.log('remoteSoundControl.textContent', remoteSoundControl.textContent);
if (remoteSoundControl.textContent === 'Unmute') {
remoteVideo.muted = false
remoteSoundControl.textContent = 'Mute';
} else {
remoteVideo.muted = true
remoteSoundControl.textContent = 'Unmute';
}
});
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerOptions
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
let videoParams = {
encodings: [
{ scaleResolutionDownBy: 4, maxBitrate: 500000 },
{ scaleResolutionDownBy: 2, maxBitrate: 1000000 },
{ scaleResolutionDownBy: 1, maxBitrate: 5000000 },
{ scalabilityMode: 'S3T3_KEY' }
],
codecOptions: {
videoGoogleStartBitrate: 1000
}
}
let audioParams = {
codecOptions :
{
opusStereo : true,
opusDtx : true
}
}
setTimeout(() => {
hub = io(config.hubAddress)
const connectToMediasoup = () => {
@ -36,7 +99,7 @@ const connectToMediasoup = () => {
if (IS_PRODUCER === true) {
hub.on('connect', async () => {
console.log(`[HUB] ${config.hubAddress} | connected: ${hub.connected}`)
console.log(`[HUB]! ${config.hubAddress} | connected: ${hub.connected}`)
connectToMediasoup()
hub.emit(
@ -84,73 +147,51 @@ if (IS_PRODUCER === true) {
connectToMediasoup()
}
let device
let rtpCapabilities
let producerTransport
let consumerTransport
let producer
let consumer
let originAssetId
// let originAssetName = 'Adi'
// let originAssetTypeName = 'linx'
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerOptions
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
let params = {
// mediasoup params
encodings: [
{
rid: 'r0',
maxBitrate: 100000,
scalabilityMode: 'S1T3',
},
{
rid: 'r1',
maxBitrate: 300000,
scalabilityMode: 'S1T3',
},
{
rid: 'r2',
maxBitrate: 900000,
scalabilityMode: 'S1T3',
},
],
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerCodecOptions
codecOptions: {
videoGoogleStartBitrate: 1000
}
}
}, 1600);
const streamSuccess = (stream) => {
console.log('[streamSuccess]');
console.log('[streamSuccess] device', device);
localVideo.srcObject = stream
const track = stream.getVideoTracks()[0]
params = {
track,
...params
console.log('stream', stream);
const videoTrack = stream.getVideoTracks()[0]
const audioTrack = stream.getAudioTracks()[0]
videoParams = {
track: videoTrack,
...videoParams
}
audioParams = {
track: audioTrack,
...audioParams
}
console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
goConnect()
}
const getLocalStream = () => {
console.log('[getLocalStream]');
navigator.mediaDevices.getUserMedia({
audio: false,
audio: produceAudio ? true : false,
video: {
width: {
min: 640,
max: 1920,
},
height: {
min: 400,
max: 1080,
}
qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
vga : { width: { ideal: 640 }, height: { ideal: 480 } },
hd : { width: { ideal: 1280 }, height: { ideal: 720 } }
}
})
.then(streamSuccess)
.catch(error => {
console.log(error.message)
})
navigator.permissions.query(
{ name: 'microphone' }
).then((permissionStatus) =>{
console.log('🟨 [PERMISSION] permissionStatus', permissionStatus); // granted, denied, prompt
// It will block the code from execution and display "Permission denied" if we don't have microphone permissions
})
}
const goConnect = () => {
@ -167,7 +208,6 @@ const goCreateTransport = () => {
// server side to send/recive media
const createDevice = async () => {
try {
console.log('[createDevice]');
device = new mediasoupClient.Device()
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@ -178,6 +218,7 @@ const createDevice = async () => {
})
console.log('Device RTP Capabilities', device.rtpCapabilities)
console.log('[createDevice] device', device);
// once the device loads, create transport
goCreateTransport()
@ -207,18 +248,20 @@ const getRtpCapabilities = () => {
}
const createSendTransport = () => {
console.log('[createSendTransport');
// see server's socket.on('createWebRtcTransport', sender?, ...)
// this is a call from Producer, so sender = true
socket.emit('createWebRtcTransport', { sender: true, callId }, ({ params }) => {
socket.emit('createWebRtcTransport', { sender: true }, (value) => {
console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
const params = value.params;
// The server sends back params needed
// to create Send Transport on the client side
if (params.error) {
console.log(params.error)
return
}
console.log(params)
// creates a new WebRTC Transport to send media
// based on the server's producer transport params
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#TransportOptions
@ -244,10 +287,10 @@ const createSendTransport = () => {
})
producerTransport.on('produce', async (parameters, callback, errback) => {
console.log(parameters)
console.log('[produce] parameters', parameters)
try {
// tell the server to create a Producer
// Tell the server to create a Producer
// with the following parameters and produce
// and expect back a server side producer id
// see server's socket.on('transport-produce', ...)
@ -270,22 +313,46 @@ const createSendTransport = () => {
}
const connectSendTransport = async () => {
// we now call produce() to instruct the producer transport
console.log('[connectSendTransport] producerTransport');
// We now call produce() to instruct the producer transport
// to send media to the Router
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
// this action will trigger the 'connect' and 'produce' events above
producer = await producerTransport.produce(params)
producer.on('trackended', () => {
// Produce video
let producerVideoHandler = await producerTransport.produce(videoParams)
console.log('videoParams', videoParams);
console.log('producerVideo', producerVideo);
producerVideoHandler.on('trackended', () => {
console.log('track ended')
// close video track
})
producer.on('transportclose', () => {
producerVideoHandler.on('transportclose', () => {
console.log('transport ended')
// close video track
})
// Produce audio
if (produceAudio) {
let producerAudioHandler = await producerTransport.produce(audioParams)
console.log('audioParams', audioParams);
console.log('producerAudio', producerAudio);
producerAudioHandler.on('trackended', () => {
console.log('track ended')
// close audio track
})
producerAudioHandler.on('transportclose', () => {
console.log('transport ended')
// close audio track
})
}
const answer = {
origin_asset_id: ASSET_ID,
dest_asset_id: originAssetId || parseInt(urlParams.get('dest_asset_id')),
@ -294,7 +361,7 @@ const connectSendTransport = async () => {
origin_asset_type_name: ASSET_TYPE,
origin_asset_name: ASSET_NAME,
video_call_id: callId,
answer: 'accepted', // answer: 'rejected'
answer: 'accepted', // answer: accepted/rejected
};
console.log('SEND answer', answer);
@ -310,7 +377,7 @@ const connectSendTransport = async () => {
const createRecvTransport = async () => {
console.log('createRecvTransport');
// see server's socket.on('consume', sender?, ...)
// See server's socket.on('consume', sender?, ...)
// this is a call from Consumer, so sender = false
await socket.emit('createWebRtcTransport', { sender: false, callId }, ({ params }) => {
// The server sends back params needed
@ -320,15 +387,15 @@ const createRecvTransport = async () => {
return
}
console.log(params)
console.log('[createRecvTransport] params', params)
// creates a new WebRTC Transport to receive media
// Creates a new WebRTC Transport to receive media
// based on server's consumer transport params
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-createRecvTransport
consumerTransport = device.createRecvTransport(params)
// https://mediasoup.org/documentation/v3/communication-between-client-and-server/#producing-media
// this event is raised when a first call to transport.produce() is made
// This event is raised when a first call to transport.produce() is made
// see connectRecvTransport() below
consumerTransport.on('connect', async ({ dtlsParameters }, callback, errback) => {
try {
@ -353,7 +420,8 @@ const resetCallSettings = () => {
localVideo.srcObject = null
remoteVideo.srcObject = null
consumer = null
producer = null
producerVideo = null
producerAudio = null
producerTransport = null
consumerTransport = null
device = undefined
@ -361,38 +429,112 @@ const resetCallSettings = () => {
const connectRecvTransport = async () => {
console.log('connectRecvTransport');
// for consumer, we need to tell the server first
// For consumer, we need to tell the server first
// to create a consumer based on the rtpCapabilities and consume
// if the router can consume, it will send back a set of params as below
await socket.emit('consume', {
rtpCapabilities: device.rtpCapabilities,
callId
}, async ({ params }) => {
if (params.error) {
console.log('Cannot Consume')
return
}
// then consume with the local consumer transport
// which creates a consumer
consumer = await consumerTransport.consume({
id: params.id,
producerId: params.producerId,
kind: params.kind,
rtpParameters: params.rtpParameters
})
// destructure and retrieve the video track from the producer
const { track } = consumer
}, async ({videoParams, audioParams}) => {
console.log(`[consume] 🟩 videoParams`, videoParams)
console.log(`[consume] 🟩 audioParams`, audioParams)
console.log('[consume] 🟩 consumerTransport', consumerTransport)
let stream = new MediaStream()
stream.addTrack(track)
// stream.removeTrack(track)
remoteVideo.srcObject = stream
socket.emit('consumer-resume')
console.log('consumer', consumer);
// Maybe the unit does not produce video or audio, so we must only consume what is produced
if (videoParams) {
console.log('❗ Have VIDEO stream to consume');
stream.addTrack(await getVideoTrask(videoParams))
} else {
console.log('❗ Don\'t have VIDEO stream to consume');
}
if (audioParams) {
console.log('❗ Have AUDIO stream to consume');
let audioTrack = await getAudioTrask(audioParams)
console.log('audioTrack', audioTrack);
stream.addTrack(audioTrack)
} else {
console.log('❗ Don\'t have AUDIO stream to consume');
}
console.log('----------stream', stream);
console.log('stream.getAudioTracks()', stream.getAudioTracks());
socket.emit('consumer-resume')
remoteVideo.srcObject = stream
// remoteVideo.autoplay = true
remoteVideo.controls = true;
remoteVideo.muted = true;
remoteVideo.loop = true;
remoteVideo.setAttribute('playsinline', '');
remoteVideo.src = stream;
remoteVideo.volume = 1.0;
// window.localStream = stream; // A
// window.localAudio.srcObject = stream; // B
// window.localAudio.autoplay = true; // C
remoteVideo.play()
.then(() => {
console.log('remoteVideo PLAY')
})
.catch((error) => {
displayError(`remoteVideo PLAY ERROR | ${error.message}`)
})
})
}
const getVideoTrask = async (videoParams) => {
consumerVideo = await consumerTransport.consume({
id: videoParams.id,
producerId: videoParams.producerId,
kind: videoParams.kind,
rtpParameters: videoParams.rtpParameters
})
consumerVideo.on('transportclose', () => {
console.log('transport closed so consumer closed')
})
return consumerVideo.track
}
const getAudioTrask = async (audioParams) => {
consumerAudio = await consumerTransport.consume({
id: audioParams.id,
producerId: audioParams.producerId,
kind: audioParams.kind,
rtpParameters: audioParams.rtpParameters
})
consumerAudio.on('transportclose', () => {
console.log('transport closed so consumer closed')
})
const audioTrack = consumerAudio.track
// audioTrack.applyConstraints({
// audio: {
// advanced: [
// {
// echoCancellation: {exact: true}
// },
// {
// autoGainControl: {exact: true}
// },
// {
// noiseSuppression: {exact: true}
// },
// {
// highpassFilter: {exact: true}
// }
// ]
// }
// })
return audioTrack
}
const closeCall = () => {
@ -416,6 +558,30 @@ const closeCall = () => {
resetCallSettings()
}
const consume = async () => {
console.log('[consume]')
console.log('createRecvTransport Consumer')
await socket.emit('createWebRtcTransport', { sender: false, callId, dispatcher: true }, ({ params }) => {
if (params.error) {
console.log('createRecvTransport | createWebRtcTransport | Error', params.error)
return
}
consumerTransport = device.createRecvTransport(params)
consumerTransport.on('connect', async ({ dtlsParameters }, callback, errback) => {
try {
await socket.emit('transport-recv-connect', {
dtlsParameters,
})
callback()
} catch (error) {
errback(error)
}
})
connectRecvTransport()
})
}
btnLocalVideo.addEventListener('click', getLocalStream)
btnRecvSendTransport.addEventListener('click', goConnect)
btnRecvSendTransport.addEventListener('click', consume)
btnCloseCall.addEventListener('click', closeCall)