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68 Commits

Author SHA1 Message Date
ce1fe3ca8d Merge pull request 'Update 'app.js'' (#33) from sergiu-patch-1 into develop
Reviewed-on: #33
Reviewed-by: bmamihai <mihai.bozieru@safemobile.com>
2023-06-07 12:51:12 +00:00
96b770d9c5 Update 'app.js' 2023-06-07 12:50:24 +00:00
4b1f4fab70 Merge pull request 'LH-284: Update duplex documentation workflow diagrams' (#32) from LH-284-update-and-centralize-diagrams-for-video-calls into develop
Reviewed-on: #32
Reviewed-by: Adriana <adriana.epure@safemobile.com>
2023-05-30 11:27:09 +00:00
ba7c2186af LH-284: Update duplex documentation workflow diagrams 2023-05-30 11:57:02 +03:00
636e8a9fab Merge pull request 'added build commit no' (#31) from build-commit-no into develop
Reviewed-on: #31
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-04-10 23:36:34 +00:00
5da2eb6927 update build.sh after review 2023-04-05 01:57:58 +03:00
97948a5d8c added build commit no 2023-04-02 01:17:18 +03:00
b94b1bff86 Merge pull request 'docker' (#18) from docker into develop
Reviewed-on: #18
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-02-26 22:11:02 +00:00
5c4f2bf13e Merge branch 'develop' into docker 2023-02-26 22:10:34 +00:00
df7f1987f6 Merge pull request 'LH-276: Add close-producer event handler; Update client' (#30) from LH-276-close-producer into develop
Reviewed-on: #30
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-02-23 09:46:41 +00:00
7842953faf LH-276: Refactor consumer-resume 2023-02-22 18:57:57 +02:00
4591617b1e LH-276: Fix video consume when initiator is not set yet(at start) 2023-02-22 18:50:37 +02:00
9b3f2f94c8 LH-276: Fix audio when initiator is not set yet(at start) 2023-02-22 18:46:01 +02:00
29a4cd7227 LH-276: Format code 2023-02-22 18:40:39 +02:00
c80265fe25 LH-276: Format code 2023-02-22 18:36:50 +02:00
abb1533c9b LH-276: Format code 2023-02-22 18:32:41 +02:00
6e3ce9fbb3 LH-276: Format code 2023-02-22 18:30:28 +02:00
4d8adf9eac LH-276: Refactor consume(consumeAudio/consumeVideo); Format code 2023-02-22 18:21:53 +02:00
bde02fe250 LH-276: Add close-producer event handler; Update client 2023-02-21 02:41:48 +02:00
0ddb43b4b5 Merge pull request 'LINXD-2342-dispatcher-audio' (#29) from LINXD-2342-dispatcher-audio into develop
Reviewed-on: #29
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-02-15 09:14:57 +00:00
f02d6af726 LINXD-2342: Added documentation for kind 2023-02-15 11:14:02 +02:00
7ee93d7963 LINXD-2342: Update video client 2023-02-15 10:06:39 +02:00
98212a78cb Update client to consume audio correctly 2023-02-11 21:32:53 +02:00
33ea44ac02 Update client to consume audio correctly 2023-02-11 21:30:21 +02:00
6b822142d0 Add the kind of producer on new-producer event 2023-02-11 21:13:57 +02:00
50e95b93a4 Update client 2023-02-08 19:37:32 +02:00
6eca0808c9 Merge pull request 'Update web client to work with dev' (#27) from update-web-client into develop
Reviewed-on: #27
Reviewed-by: bmamihai <mihai.bozieru@safemobile.com>
2023-01-23 20:37:27 +00:00
f728b23b7f Update web client to work with dev 2023-01-23 20:45:08 +02:00
33c0234fb0 Merge pull request 'LINXD-2303: Added client-to-client workflow; Added client-mediadoup workflow' (#26) from LINXD-2303-video-documentation-mobile-mobile into develop
Reviewed-on: #26
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-19 14:44:24 +00:00
f81d5b2dfa LINXD-2303: Update diagrams Mediasoup connect & produce; Update diagram Client-Client workflow 2023-01-19 16:43:46 +02:00
ae87b5bd71 LINXD-2303: Added client-to-client workflow; Added client-mediadoup workflow 2023-01-19 16:31:16 +02:00
09e82fb224 Merge pull request 'Moved 'new-producer' in 'transport-produce' handler' (#25) from new-producer-in-transport-produce into develop
Reviewed-on: #25
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-19 06:54:53 +00:00
1da0170261 Moved 'new-producer' in 'transport-produce' handler 2023-01-18 11:49:02 +02:00
aca0a6eac0 improved build to get also the git log to know what is on server 2023-01-17 16:00:52 +00:00
23ca56c8c6 rever build.sh 2023-01-17 15:51:55 +00:00
a2d0b6771b Merge pull request 'Add new-producer event; Update client to consume when receives new-producer event' (#24) from LAPI-675-generate-new-producer-in-mediasoup-and-client into develop
Reviewed-on: #24
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-16 15:40:06 +00:00
9382fd11fb LAPI-675: Remove commented code 2023-01-16 17:39:15 +02:00
fb99fd57f8 LAPI-675: Refactor transport-connect new-producer 2023-01-16 17:32:31 +02:00
0f4342777a LAPI-675: Update bundle 2023-01-13 20:12:59 +02:00
69c167e4e9 Add new-producer event; Update client to consume when receives new-producer event 2023-01-13 20:11:24 +02:00
ad4c1f27e7 Merge pull request 'LAPI-674: Parse transport-recv-connect and consume param' (#23) from LAPI-674-change-events-for-consumer-in-mediasoup-server into develop
Reviewed-on: #23
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-13 07:52:27 +00:00
b2128c4754 fix build 2023-01-12 16:42:44 +00:00
af53dc1610 fix build 2023-01-12 16:30:39 +00:00
f48fe4f9aa LAPI-674: Parse transport-recv-connect and consume param 2023-01-12 16:23:30 +02:00
6fd6ca9755 Merge pull request 'comsume-undefined' (#22) from comsume-undefined into develop
Reviewed-on: #22
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-12 13:31:46 +00:00
1b72941a32 Moved callId in consumer-resume 2023-01-12 15:09:21 +02:00
c037240f47 Refactor consumer-resume 2023-01-12 14:32:48 +02:00
9716bddd71 Added log 2023-01-12 12:50:40 +02:00
e6c08a2c1d Refactor consumer-resume v3 2023-01-12 12:46:46 +02:00
1342e812e6 Refactor consumer-resume 2023-01-12 12:45:23 +02:00
d8405eccc7 Refactor consumer-resume 2023-01-12 12:44:11 +02:00
5bf31d452f Check for initiatorConsumerAudio before resume() 2023-01-12 12:38:40 +02:00
c034610471 Rollback 2023-01-12 12:38:01 +02:00
df5b3eab90 Check for initiatorConsumerAudio before resume() 2023-01-12 12:31:52 +02:00
b0293230b6 Merge pull request 'add-diagrams-doc' (#21) from add-diagrams-doc into develop
Reviewed-on: #21
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2023-01-10 17:51:05 +00:00
391e8eb6f0 Added diagram for close call before accept/reject 2023-01-10 13:17:57 +02:00
3139a625a2 Added workflow diagrams 2023-01-10 12:05:29 +02:00
48eee903a5 added command for limiting cpu/memory 2023-01-05 14:31:51 +02:00
d5bc0cd1d3 run under host network 2023-01-05 13:18:13 +02:00
1c353d7c88 Merge branch 'develop' into docker 2023-01-05 01:14:56 +02:00
9d43b7ec0c Merge pull request 'LINXD-2270: Allow server and web client to have full duplex' (#20) from LINXD-2270-full-duplex into develop
Reviewed-on: #20
Reviewed-by: Cristi Ene <cristi.ene@safemobile.com>
2022-12-27 11:12:38 +00:00
342d09c3e6 LINXD-2270: Removed console logs; Update commented code 2022-12-27 13:11:06 +02:00
b3409de3ba LINXD-2270: Allow server and web client to have full duplex 2022-12-19 19:31:33 +02:00
3e31ba21bd replace individual copy with copy all 2022-12-15 13:20:22 +02:00
cdf02756d3 fix start 2022-12-14 01:32:17 +02:00
b2f9f5affa remove pm2 and watchify 2022-12-14 00:48:14 +02:00
5b9bfeaa01 improve docker 2022-12-10 03:21:59 +02:00
e3bef9b3e5 New dockerfile 2022-12-10 02:38:09 +02:00
18 changed files with 1033 additions and 676 deletions

2
.dockerignore Normal file
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@ -0,0 +1,2 @@
node_modules
doc

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@ -1,11 +1,25 @@
FROM ubuntu
FROM ubuntu:22.04
WORKDIR /app
RUN apt-get update && \
apt-get install -y build-essential pip net-tools iputils-ping iproute2 curl
RUN curl -fsSL https://deb.nodesource.com/setup_16.x | bash -
RUN curl -fsSL https://deb.nodesource.com/setup_18.x | bash -
RUN apt-get install -y nodejs
RUN npm install -g watchify
EXPOSE 3000
EXPOSE 2000-2020
EXPOSE 10000-10100
COPY . /app/
RUN npm install
EXPOSE 3000/tcp
EXPOSE 2000-2200/udp
CMD node app.js
#docker build -t linx-video .
# docker run -it -d --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video
#Run under host network
# docker run -it -d --network host --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video
#https://docs.docker.com/config/containers/resource_constraints/
#docker run -it -d --network host --cpus="0.25" --memory="512m" --restart always -p 3000:3000/tcp -p 2000-2200:2000-2200/udp linx-video

486
app.js
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@ -1,4 +1,4 @@
require('dotenv').config()
require('dotenv').config();
const express = require('express');
const app = express();
@ -13,50 +13,47 @@ try {
}
const mediasoup = require('mediasoup');
let worker
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
* initiatorAudioProducer: Producer,
* initiatorVideoProducer: Producer,
* receiverVideoProducer: Producer,
* receiverAudioProducer: Producer,
* initiatorProducerTransport: Producer Transport,
* receiverProducerTransport: Producer Transport,
* initiatorConsumerVideo: Consumer,
* initiatorConsumerAudio: Consumer,
* initiatorConsumerTransport: Consumer Transport
* initiatorSocket
* receiverSocket
* }
*
**/
let videoCalls = {}
let socketDetails = {}
let videoCalls = {};
let socketDetails = {};
app.get('/', (_req, res) => {
res.send('Hello from mediasoup app!')
})
res.send('OK');
});
app.use('/sfu', express.static(path.join(__dirname, 'public')))
app.use('/sfu', express.static(path.join(__dirname, 'public')));
// SSL cert for HTTPS access
const options = {
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);
@ -69,19 +66,19 @@ const createWorker = async () => {
worker = await mediasoup.createWorker({
rtcMinPort: parseInt(process.env.RTC_MIN_PORT),
rtcMaxPort: parseInt(process.env.RTC_MAX_PORT),
})
});
console.log(`[createWorker] worker pid ${worker.pid}`);
worker.on('died', error => {
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
})
});
return worker;
} catch (error) {
console.log(`ERROR | createWorker | ${error.message}`);
}
console.error(`[createWorker] | ERROR | error: ${error.message}`);
}
};
// We create a Worker as soon as our application starts
worker = createWorker();
@ -95,98 +92,79 @@ 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
channels: 2,
},
{
kind: 'video',
mimeType: 'video/VP9',
clockRate: 90000,
parameters :
{
parameters: {
'profile-id': 2,
'x-google-start-bitrate' : 1000
}
'x-google-start-bitrate': 1000,
},
},
{
kind: 'video',
mimeType: 'video/h264',
clockRate: 90000,
parameters :
{
parameters: {
'packetization-mode': 1,
'profile-level-id': '4d0032',
'level-asymmetry-allowed': 1,
'x-google-start-bitrate' : 1000
}
'x-google-start-bitrate': 1000,
},
},
{
kind: 'video',
mimeType: 'video/h264',
clockRate: 90000,
parameters :
{
parameters: {
'packetization-mode': 1,
'profile-level-id': '42e01f',
'level-asymmetry-allowed': 1,
'x-google-start-bitrate' : 1000
}
}
// {
// kind: 'audio',
// mimeType: 'audio/opus',
// clockRate: 48000,
// channels: 2,
// },
// {
// kind: 'video',
// mimeType: 'video/VP8',
// clockRate: 90000,
// parameters: {
// 'x-google-start-bitrate': 1000,
// },
// },
'x-google-start-bitrate': 1000,
},
},
];
const closeCall = (callId) => {
try {
if (callId && videoCalls[callId]) {
videoCalls[callId].producerVideo?.close();
videoCalls[callId].producerAudio?.close();
videoCalls[callId].consumerVideo?.close();
videoCalls[callId].consumerAudio?.close();
videoCalls[callId].receiverVideoProducer?.close();
videoCalls[callId].receiverAudioProducer?.close();
videoCalls[callId].initiatorConsumerVideo?.close();
videoCalls[callId].initiatorConsumerAudio?.close();
videoCalls[callId]?.consumerTransport?.close();
videoCalls[callId]?.producerTransport?.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`);
console.log(`[closeCall] | callId: ${callId}`);
}
} catch (error) {
console.log(`ERROR | closeCall | callid ${callId} | ${error.message}`);
}
console.error(`[closeCall] | ERROR | callId: ${callId} | error: ${error.message}`);
}
};
/*
- Handlers for WS events
- These are created only when we have a connection with a peer
*/
peers.on('connection', async socket => {
peers.on('connection', async (socket) => {
console.log('[connection] socketId:', socket.id);
// After making the connection successfully, we send the client a 'connection-success' event
socket.emit('connection-success', {
socketId: socket.id
socketId: socket.id,
});
// It is triggered when the peer is disconnected
@ -209,22 +187,22 @@ peers.on('connection', async socket => {
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] = { router: await worker.createRouter({ mediaCodecs }) };
console.log(`[createRoom] Generate Router ID: ${videoCalls[callId].router.id}`);
videoCalls[callId].receiverSocket = socket;
} else {
videoCalls[callId].initiatorSocket = socket;
}
socketDetails[socket.id] = callId;
// rtpCapabilities is set for callback
console.log('[getRtpCapabilities] callId', callId);
callbackResponse = {
rtpCapabilities :videoCalls[callId].router.rtpCapabilities
rtpCapabilities: videoCalls[callId].router.rtpCapabilities,
};
} else {
console.log(`[createRoom] missing callId ${callId}`);
console.log(`[createRoom] missing callId: ${callId}`);
}
} catch (error) {
console.log(`ERROR | createRoom | callId ${callId} | ${error.message}`);
console.error(`[createRoom] | ERROR | callId: ${callId} | error: ${error.message}`);
} finally {
callback(callbackResponse);
}
@ -232,7 +210,7 @@ peers.on('connection', async socket => {
/*
- 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)
@ -241,24 +219,29 @@ peers.on('connection', async socket => {
socket.on('createWebRtcTransport', async ({ sender }, callback) => {
try {
const callId = socketDetails[socket.id];
console.log(`[createWebRtcTransport] sender ${sender} | callId ${callId}`);
console.log(`[createWebRtcTransport] socket ${socket.id} | sender ${sender} | callId ${callId}`);
if (sender) {
if (!videoCalls[callId].producerTransport) {
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 {
console.log(`producerTransport has already been defined | callId ${callId}`);
callback(null);
}
} else if (!sender) {
if (!videoCalls[callId].consumerTransport) {
videoCalls[callId].consumerTransport = await createWebRtcTransportLayer(callId, callback);
} else {
console.log(`consumerTransport has already been defined | callId ${callId}`);
callback(null);
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 ${socketDetails[socket.id]} | sender ${sender} | ${error.message}`);
console.error(
`[createWebRtcTransport] | ERROR | callId: ${socketDetails[socket.id]} | sender: ${sender} | error: ${
error.message
}`
);
callback(error);
}
});
@ -272,16 +255,19 @@ peers.on('connection', async socket => {
const callId = socketDetails[socket.id];
if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
console.log(`[transport-connect] socket.id ${socket.id} | callId ${callId}`);
await videoCalls[callId].producerTransport.connect({ dtlsParameters });
console.log(`[transport-connect] socket ${socket.id} | callId ${callId}`);
isInitiator(callId, socket.id)
? await videoCalls[callId].initiatorProducerTransport.connect({ dtlsParameters })
: await videoCalls[callId].receiverProducerTransport.connect({ dtlsParameters });
} catch (error) {
console.log(`ERROR | transport-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
console.error(`[transport-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${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 }, callback) => {
@ -289,50 +275,86 @@ peers.on('connection', async socket => {
const callId = socketDetails[socket.id];
if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
console.log(`[transport-produce] kind: ${kind} | socket.id: ${socket.id} | callId: ${callId}`);
console.log('kind', kind);
console.log('rtpParameters', rtpParameters);
console.log(`[transport-produce] callId: ${callId} | kind: ${kind} | socket: ${socket.id}`);
if (kind === 'video') {
videoCalls[callId].producerVideo = await videoCalls[callId].producerTransport.produce({
if (!isInitiator(callId, socket.id)) {
videoCalls[callId].receiverVideoProducer = await videoCalls[callId].receiverProducerTransport.produce({
kind,
rtpParameters,
});
console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producerVideo.id} | kind: ${videoCalls[callId].producerVideo.kind}`);
videoCalls[callId].producerVideo.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport for this producer closed', callId)
videoCalls[callId].receiverVideoProducer.on('transportclose', () => {
console.log('transport for this producer closed', callId);
closeCall(callId);
});
// Send back to the client the Producer's id
callback && callback({
id: videoCalls[callId].producerVideo.id
callback &&
callback({
id: videoCalls[callId].receiverVideoProducer.id,
});
} else if (kind === 'audio') {
videoCalls[callId].producerAudio = await videoCalls[callId].producerTransport.produce({
} else {
videoCalls[callId].initiatorVideoProducer = await videoCalls[callId].initiatorProducerTransport.produce({
kind,
rtpParameters,
});
console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producerAudio.id} | kind: ${videoCalls[callId].producerAudio.kind}`);
videoCalls[callId].producerAudio.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport for this producer closed', callId)
videoCalls[callId].initiatorVideoProducer.on('transportclose', () => {
console.log('transport for this producer closed', callId);
closeCall(callId);
});
// Send back to the client the Producer's id
callback && callback({
id: videoCalls[callId].producerAudio.id
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,
});
videoCalls[callId].receiverAudioProducer.on('transportclose', () => {
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,
});
videoCalls[callId].initiatorAudioProducer.on('transportclose', () => {
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,
});
}
}
const socketToEmit = isInitiator(callId, socket.id)
? videoCalls[callId].receiverSocket
: videoCalls[callId].initiatorSocket;
// callId - Id of the call
// kind - producer type: audio/video
socketToEmit?.emit('new-producer', { callId, kind });
} catch (error) {
console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
console.error(`[transport-produce] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
}
});
@ -343,12 +365,18 @@ 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 });
} catch (error) {
console.log(`ERROR | transport-recv-connect | callId ${socketDetails[socket.id]} | ${error.message}`);
console.log(`[transport-recv-connect] socket ${socket.id} | callId ${callId}`);
if (typeof dtlsParameters === 'string') dtlsParameters = JSON.parse(dtlsParameters);
// 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.error(`[transport-recv-connect] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
}
});
/*
- The customer consumes after successfully connecting to consumerTransport
@ -358,117 +386,156 @@ peers.on('connection', async socket => {
- The consumer does consumerTransport.consume(params)
*/
socket.on('consume', async ({ rtpCapabilities }, callback) => {
try {
console.log(`[consume] rtpCapabilities: ${JSON.stringify(rtpCapabilities)}`);
const callId = socketDetails[socket.id];
console.log('[consume] callId', callId);
const socketId = socket.id;
const canConsumeVideo = !!videoCalls[callId].producerVideo && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producerVideo.id,
rtpCapabilities
})
console.log(`[consume] socket ${socketId} | callId: ${callId}`);
const canConsumeAudio = !!videoCalls[callId].producerAudio && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producerAudio.id,
rtpCapabilities
})
if (typeof rtpCapabilities === 'string') rtpCapabilities = JSON.parse(rtpCapabilities);
console.log('[consume] canConsumeVideo', canConsumeVideo);
console.log('[consume] canConsumeAudio', canConsumeAudio);
if (canConsumeVideo && !canConsumeAudio) {
console.log('1');
const videoParams = await consumeVideo(callId, rtpCapabilities)
console.log('videoParams', videoParams);
callback({ videoParams, audioParams: null });
} else if (canConsumeVideo && canConsumeAudio) {
console.log('2');
const videoParams = await consumeVideo(callId, rtpCapabilities)
const audioParams = await consumeAudio(callId, rtpCapabilities)
callback({ videoParams, audioParams });
} else {
console.log(`[consume] Can't consume | callId ${callId}`);
callback(null);
}
} catch (error) {
console.log(`ERROR | consume | callId ${socketDetails[socket.id]} | ${error.message}`)
callback({ params: { error } });
}
callback({
videoParams: await consumeVideo({ callId, socketId, rtpCapabilities }),
audioParams: await consumeAudio({ callId, socketId, rtpCapabilities }),
});
});
/*
- 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
- For the initiator we resume the initiatorConsumerAUDIO/VIDEO and for receiver the receiverConsumerAUDIO/VIDEO
*/
socket.on('consumer-resume', async () => {
socket.on('consumer-resume', () => {
try {
const callId = socketDetails[socket.id];
console.log(`[consumer-resume] callId ${callId}`)
await videoCalls[callId].consumerVideo.resume();
await videoCalls[callId].consumerAudio.resume();
const isInitiatorValue = isInitiator(callId, socket.id);
console.log(`[consumer-resume] callId: ${callId} | isInitiator: ${isInitiatorValue}`);
const consumerVideo = isInitiatorValue
? videoCalls[callId].initiatorConsumerVideo
: videoCalls[callId].receiverConsumerVideo;
const consumerAudio = isInitiatorValue
? videoCalls[callId].initiatorConsumerAudio
: videoCalls[callId].receiverConsumerAudio;
consumerVideo?.resume();
consumerAudio?.resume();
} catch (error) {
console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
console.error(
`[consumer-resume] | ERROR | callId: ${socketDetails[socket.id]} | isInitiator: ${isInitiator} | error: ${
error.message
}`
);
}
});
socket.on('close-producer', ({ callId, kind }) => {
try {
if (isInitiator(callId, socket.id)) {
console.log(`[close-producer] initiator --EMIT--> receiver | callId: ${callId} | kind: ${kind}`);
videoCalls[callId].receiverSocket.emit('close-producer', { callId, kind });
} else {
console.log(`[close-producer] receiver --EMIT--> initiator | callId: ${callId} | kind: ${kind}`);
videoCalls[callId].initiatorSocket.emit('close-producer', { callId, kind });
}
} catch (error) {
console.error(`[close-producer] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`);
}
});
});
const consumeVideo = async (callId, rtpCapabilities) => {
videoCalls[callId].consumerVideo = await videoCalls[callId].consumerTransport.consume({
producerId: videoCalls[callId].producerVideo.id,
const canConsume = ({ callId, producerId, rtpCapabilities }) => {
return !!videoCalls[callId].router.canConsume({
producerId,
rtpCapabilities,
});
};
const consumeVideo = async ({ callId, socketId, rtpCapabilities }) => {
// Handlers for consumer transport https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
if (isInitiator(callId, socketId) && videoCalls[callId].receiverVideoProducer) {
const producerId = videoCalls[callId].receiverVideoProducer.id;
if (!canConsume({ callId, producerId, rtpCapabilities })) return null;
videoCalls[callId].initiatorConsumerVideo = await videoCalls[callId].initiatorConsumerTransport.consume({
producerId,
rtpCapabilities,
paused: true,
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
videoCalls[callId].consumerVideo.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].consumerVideo.on('producerclose', () => {
const callId = socketDetails[socket.id];
console.log('producer of consumer closed', callId);
closeCall(callId);
});
return {
id: videoCalls[callId].consumerVideo.id,
producerId: videoCalls[callId].producerVideo.id,
id: videoCalls[callId].initiatorConsumerVideo.id,
producerId,
kind: 'video',
rtpParameters: videoCalls[callId].consumerVideo.rtpParameters,
}
}
rtpParameters: videoCalls[callId].initiatorConsumerVideo.rtpParameters,
};
} else if (videoCalls[callId].initiatorVideoProducer) {
const producerId = videoCalls[callId].initiatorVideoProducer.id;
if (!canConsume({ callId, producerId, rtpCapabilities })) return null;
const consumeAudio = async (callId, rtpCapabilities) => {
videoCalls[callId].consumerAudio = await videoCalls[callId].consumerTransport.consume({
producerId: videoCalls[callId].producerAudio.id,
videoCalls[callId].receiverConsumerVideo = await videoCalls[callId].receiverConsumerTransport.consume({
producerId,
rtpCapabilities,
paused: true,
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
videoCalls[callId].consumerAudio.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport close from consumer', callId);
closeCall(callId);
return {
id: videoCalls[callId].receiverConsumerVideo.id,
producerId,
kind: 'video',
rtpParameters: videoCalls[callId].receiverConsumerVideo.rtpParameters,
};
} else {
return null;
}
};
const consumeAudio = async ({ callId, socketId, rtpCapabilities }) => {
try {
// Handlers for consumer transport https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
if (isInitiator(callId, socketId) && videoCalls[callId].receiverAudioProducer) {
const producerId = videoCalls[callId].receiverAudioProducer.id;
if (!canConsume({ callId, producerId, rtpCapabilities })) return null;
videoCalls[callId].initiatorConsumerAudio = await videoCalls[callId].initiatorConsumerTransport.consume({
producerId,
rtpCapabilities,
paused: true,
});
// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
videoCalls[callId].consumerAudio.on('producerclose', () => {
const callId = socketDetails[socket.id];
console.log('producer of consumer closed', callId);
closeCall(callId);
});
return {
id: videoCalls[callId].consumerAudio.id,
producerId: videoCalls[callId].producerAudio.id,
id: videoCalls[callId].initiatorConsumerAudio.id,
producerId,
kind: 'audio',
rtpParameters: videoCalls[callId].consumerAudio.rtpParameters,
rtpParameters: videoCalls[callId].initiatorConsumerAudio.rtpParameters,
};
} else if (videoCalls[callId].initiatorAudioProducer) {
const producerId = videoCalls[callId].initiatorAudioProducer.id;
if (!canConsume({ callId, producerId, rtpCapabilities })) return null;
videoCalls[callId].receiverConsumerAudio = await videoCalls[callId].receiverConsumerTransport.consume({
producerId,
rtpCapabilities,
paused: true,
});
return {
id: videoCalls[callId].receiverConsumerAudio.id,
producerId,
kind: 'audio',
rtpParameters: videoCalls[callId].receiverConsumerAudio.rtpParameters,
};
} else {
return null;
}
} catch (error) {
console.error(`[consumeAudio] | ERROR | error: ${error}`);
}
};
const isInitiator = (callId, socketId) => {
return videoCalls[callId]?.initiatorSocket?.id === socketId;
};
/*
- Called from at event 'createWebRtcTransport' and assigned to the consumer or producer transport
@ -479,14 +546,14 @@ const consumeAudio = async (callId, rtpCapabilities) => {
*/
const createWebRtcTransportLayer = async (callId, callback) => {
try {
console.log('[createWebRtcTransportLayer] callId', callId);
console.log(`[createWebRtcTransportLayer] callId: ${callId}`);
// https://mediasoup.org/documentation/v3/mediasoup/api/#WebRtcTransportOptions
const webRtcTransport_options = {
listenIps: [
{
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,
@ -494,11 +561,10 @@ const createWebRtcTransportLayer = async (callId, callback) => {
};
// https://mediasoup.org/documentation/v3/mediasoup/api/#router-createWebRtcTransport
let transport = await videoCalls[callId].router.createWebRtcTransport(webRtcTransport_options)
console.log(`callId: ${callId} | transport id: ${transport.id}`)
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();
@ -517,15 +583,15 @@ const createWebRtcTransportLayer = async (callId, callback) => {
dtlsParameters: transport.dtlsParameters,
};
console.log('[createWebRtcTransportLayer] callback params', params);
// Send back to the client the params
callback({ params });
// Set transport to producerTransport or consumerTransport
return transport;
} catch (error) {
console.log(`ERROR | createWebRtcTransportLayer | callId ${socketDetails[socket.id]} | ${error.message}`);
console.error(
`[createWebRtcTransportLayer] | ERROR | callId: ${socketDetails[socket.id]} | error: ${error.message}`
);
callback({ params: { error } });
}
}
};

View File

@ -1,4 +1,40 @@
#/!bin/bash
## FUNCTIONS
function getGitVersion(){
version=$(git describe)
count=$(echo ${version%%-*} | grep -o "\." | wc -l)
if (( $count > 1 )); then
version=${version%%-*}
elif (( $count == 0 ));then
echo -e "Error: Git version \"${version%%-*}\" not respecting Safemobile standard.\n Must be like 4.xx or 4.xx.xx"
version="0.0.0"
else
if [[ "$1" == "dev" ]];then
cleanprefix=${version#*-} # remove everything before `-` including `-`
cleansuffix=${cleanprefix%-*} # remove everything after `-` including `-`
version="${version%%-*}.${cleansuffix}"
else
version="${version%%-*}.0" # one `%` remove everything after last `-`, two `%%` remove everything after all `-`
fi
fi
}
function addVersionPm2(){
file_pkg="package.json"
key=" \"version\": \""
if [ -f "$file_pkg" ] && [ ! -z "$version" ]; then
versionApp=" \"version\": \"$version\","
sed -i "s|^.*$key.*|${versionApp//\//\\/}|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
}
## PREBUILD PROCESS
# check dist dir to be present and empty
if [ ! -d "dist" ]; then
@ -9,37 +45,30 @@ else
## CLEANUP
rm -fr dist/*
fi
if [ -d "node_modules" ]; then
rm -fr node_modules
fi
# Install dependencies
#npm install
## PROJECT NEEDS
echo "Building app... from $(git rev-parse --abbrev-ref HEAD)"
#npm run-script build
cp -r {.env,app.js,package.json,server,public} dist/
cp -r {.env,app.js,package.json,server,public,doc,Dockerfile} dist/
#cp -r ./* dist/
# Generate Git log
dateString=$(date +"%Y%m%d-%H%M%S")
git log --pretty=format:"%ad%x09%an%x09%s" --no-merges -20 > "dist/git-$dateString.log"
# Get Git version control
getGitVersion $1
# Add version control for pm2
cd dist
#Add version control for pm2
version=$(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
addVersionPm2
## POST BUILD

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@ -20353,7 +20353,7 @@ module.exports = yeast;
},{}],94:[function(require,module,exports){
module.exports = {
hubAddress: 'https://hub.dev.linx.safemobile.com/',
mediasoupAddress: 'https://video.safemobile.org',
mediasoupAddress: 'https://testing.video.safemobile.org/',
}
},{}],95:[function(require,module,exports){
const io = require('socket.io-client')
@ -20368,10 +20368,24 @@ 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)
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
@ -20381,6 +20395,21 @@ 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
@ -20420,10 +20449,23 @@ setTimeout(() => {
console.log(`[MEDIA] ${config.mediasoupAddress} | connected: ${socket.connected} | existsProducer: ${existsProducer}`)
if (!IS_PRODUCER && existsProducer && consumer === undefined) {
goConnect()
// document.getElementById('btnRecvSendTransport').click();
}
if (IS_PRODUCER && urlParams.get('testing') === 'true') { getLocalStream() }
})
socket.on('new-producer', ({ callId, kind }) => {
console.log(`🟢 new-producer | callId: ${callId} | kind: ${kind} | Ready to consume`);
connectRecvTransport();
})
socket.on('close-producer', ({ callId, kind }) => {
console.log(`🔴 close-producer | callId: ${callId} | kind: ${kind}`);
if (kind === 'video') {
consumerVideo.close()
remoteVideo.srcObject = null
}
else if (kind === 'audio') consumerAudio.close()
})
}
if (IS_PRODUCER === true) {
@ -20502,7 +20544,7 @@ const streamSuccess = (stream) => {
const getLocalStream = () => {
console.log('[getLocalStream]');
navigator.mediaDevices.getUserMedia({
audio: true,
audio: produceAudio ? true : false,
video: {
qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
vga : { width: { ideal: 640 }, height: { ideal: 480 } },
@ -20580,7 +20622,7 @@ 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 }, (value) => {
socket.emit('createWebRtcTransport', { sender: true }, (value) => {
console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
@ -20651,34 +20693,36 @@ const connectSendTransport = async () => {
// this action will trigger the 'connect' and 'produce' events above
// Produce video
producerVideo = await producerTransport.produce(videoParams)
let producerVideoHandler = await producerTransport.produce(videoParams)
console.log('videoParams', videoParams);
console.log('producerVideo', producerVideo);
producerVideo.on('trackended', () => {
producerVideoHandler.on('trackended', () => {
console.log('track ended')
// close video track
})
producerVideo.on('transportclose', () => {
producerVideoHandler.on('transportclose', () => {
console.log('transport ended')
// close video track
})
// Produce audio
producerAudio = await producerTransport.produce(audioParams)
if (produceAudio) {
let producerAudioHandler = await producerTransport.produce(audioParams)
console.log('audioParams', audioParams);
console.log('producerAudio', producerAudio);
producerAudio.on('trackended', () => {
producerAudioHandler.on('trackended', () => {
console.log('track ended')
// close audio track
})
producerAudio.on('transportclose', () => {
producerAudioHandler.on('transportclose', () => {
console.log('transport ended')
// close audio track
})
}
const answer = {
origin_asset_id: ASSET_ID,
@ -20700,6 +20744,8 @@ const connectSendTransport = async () => {
// Enable Close call button
const closeCallBtn = document.getElementById('btnCloseCall');
closeCallBtn.removeAttribute('disabled');
createRecvTransport();
}
const createRecvTransport = async () => {
@ -20739,7 +20785,8 @@ const createRecvTransport = async () => {
errback(error)
}
})
connectRecvTransport()
// We call it in new-rpoducer, we don't need it here anymore
// connectRecvTransport()
})
}
@ -20762,32 +20809,89 @@ const connectRecvTransport = async () => {
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)
stream.addTrack(audioTrack)
} else {
console.log('❗ Don\'t have AUDIO stream to consume');
}
socket.emit('consumer-resume')
remoteVideo.srcObject = stream
remoteVideo.setAttribute('autoplay', true)
remoteVideo.play()
.then(() => {
console.log('remoteVideo PLAY')
})
.catch((error) => {
console.error(`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
})
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 = () => {
@ -20811,8 +20915,31 @@ const closeCall = () => {
resetCallSettings()
}
// const consume = async (kind) => {
// console.log(`[consume] kind: ${kind}`)
// 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)
},{"./config":94,"mediasoup-client":66,"socket.io-client":82}]},{},[95]);

View File

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

View File

@ -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" muted></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>
@ -60,34 +75,11 @@
</td>
<td>
<div id="sharedBtns">
<button id="btnRecvSendTransport">Consume</button>
<!-- <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

@ -10,10 +10,24 @@ 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)
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
@ -23,6 +37,21 @@ 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
@ -62,10 +91,23 @@ setTimeout(() => {
console.log(`[MEDIA] ${config.mediasoupAddress} | connected: ${socket.connected} | existsProducer: ${existsProducer}`)
if (!IS_PRODUCER && existsProducer && consumer === undefined) {
goConnect()
// document.getElementById('btnRecvSendTransport').click();
}
if (IS_PRODUCER && urlParams.get('testing') === 'true') { getLocalStream() }
})
socket.on('new-producer', ({ callId, kind }) => {
console.log(`🟢 new-producer | callId: ${callId} | kind: ${kind} | Ready to consume`);
connectRecvTransport();
})
socket.on('close-producer', ({ callId, kind }) => {
console.log(`🔴 close-producer | callId: ${callId} | kind: ${kind}`);
if (kind === 'video') {
consumerVideo.close()
remoteVideo.srcObject = null
}
else if (kind === 'audio') consumerAudio.close()
})
}
if (IS_PRODUCER === true) {
@ -144,7 +186,7 @@ const streamSuccess = (stream) => {
const getLocalStream = () => {
console.log('[getLocalStream]');
navigator.mediaDevices.getUserMedia({
audio: true,
audio: produceAudio ? true : false,
video: {
qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
vga : { width: { ideal: 640 }, height: { ideal: 480 } },
@ -222,7 +264,7 @@ 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 }, (value) => {
socket.emit('createWebRtcTransport', { sender: true }, (value) => {
console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
@ -293,34 +335,36 @@ const connectSendTransport = async () => {
// this action will trigger the 'connect' and 'produce' events above
// Produce video
producerVideo = await producerTransport.produce(videoParams)
let producerVideoHandler = await producerTransport.produce(videoParams)
console.log('videoParams', videoParams);
console.log('producerVideo', producerVideo);
producerVideo.on('trackended', () => {
producerVideoHandler.on('trackended', () => {
console.log('track ended')
// close video track
})
producerVideo.on('transportclose', () => {
producerVideoHandler.on('transportclose', () => {
console.log('transport ended')
// close video track
})
// Produce audio
producerAudio = await producerTransport.produce(audioParams)
if (produceAudio) {
let producerAudioHandler = await producerTransport.produce(audioParams)
console.log('audioParams', audioParams);
console.log('producerAudio', producerAudio);
producerAudio.on('trackended', () => {
producerAudioHandler.on('trackended', () => {
console.log('track ended')
// close audio track
})
producerAudio.on('transportclose', () => {
producerAudioHandler.on('transportclose', () => {
console.log('transport ended')
// close audio track
})
}
const answer = {
origin_asset_id: ASSET_ID,
@ -342,6 +386,8 @@ const connectSendTransport = async () => {
// Enable Close call button
const closeCallBtn = document.getElementById('btnCloseCall');
closeCallBtn.removeAttribute('disabled');
createRecvTransport();
}
const createRecvTransport = async () => {
@ -381,7 +427,8 @@ const createRecvTransport = async () => {
errback(error)
}
})
connectRecvTransport()
// We call it in new-rpoducer, we don't need it here anymore
// connectRecvTransport()
})
}
@ -404,32 +451,89 @@ const connectRecvTransport = async () => {
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)
stream.addTrack(audioTrack)
} else {
console.log('❗ Don\'t have AUDIO stream to consume');
}
socket.emit('consumer-resume')
remoteVideo.srcObject = stream
remoteVideo.setAttribute('autoplay', true)
remoteVideo.play()
.then(() => {
console.log('remoteVideo PLAY')
})
.catch((error) => {
console.error(`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
})
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 = () => {
@ -453,7 +557,30 @@ const closeCall = () => {
resetCallSettings()
}
// const consume = async (kind) => {
// console.log(`[consume] kind: ${kind}`)
// 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)