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Author SHA1 Message Date
Sergiu Toma c823f7578c LH-265: Update client 2022-11-29 13:27:33 +02:00
Sergiu Toma 6cc14cdf30 LH-265: Update documentation and README.md 2022-11-29 13:27:12 +02:00
Sergiu Toma 4bb23def42 Update server 2022-11-29 11:32:03 +02:00
Sergiu Toma fc745a5879 Update server 2022-11-29 11:09:10 +02:00
Sergiu Toma 742d67f2e3 Update server 2022-11-29 11:08:12 +02:00
Sergiu Toma e22093d97e Update server 2022-11-29 10:27:55 +02:00
Sergiu Toma 7634a18465 Update server 2022-11-29 10:22:02 +02:00
Sergiu Toma d17b035526 Update server 2022-11-29 04:03:06 +02:00
Sergiu Toma a21451e46d Update server 2022-11-29 03:59:42 +02:00
Sergiu Toma df0cb81a8e Update server 2022-11-29 03:55:34 +02:00
Sergiu Toma ac8c651a9d Update server 2022-11-29 03:52:15 +02:00
Sergiu Toma 9111c4e245 Update server 2022-11-29 03:50:45 +02:00
Sergiu Toma 7a2d02dcda Update server 2022-11-29 03:44:02 +02:00
Sergiu Toma 39efdd12b7 Update server 2022-11-29 03:41:00 +02:00
Sergiu Toma 0bdc6fac3a Update server 2022-11-29 03:17:28 +02:00
Sergiu Toma ae7a8ed9ce Update server 2022-11-29 03:16:15 +02:00
Sergiu Toma 9feaebf8a7 Update server 2022-11-29 02:46:12 +02:00
Sergiu Toma 85110b7f5c Update server 2022-11-29 02:44:40 +02:00
Sergiu Toma d047cdf7d1 Update server 2022-11-29 02:44:01 +02:00
Sergiu Toma 753b476462 Update server 2022-11-29 02:42:13 +02:00
Sergiu Toma 359c7c784e Update server 2022-11-29 02:40:25 +02:00
Sergiu Toma 5169d0d49f Update server 2022-11-29 02:39:11 +02:00
Sergiu Toma a3b083fe24 Update server 2022-11-29 02:37:55 +02:00
Sergiu Toma 46d3499e3d Update server 2022-11-29 02:34:48 +02:00
Sergiu Toma 38b95d5246 Update server 2022-11-29 02:21:29 +02:00
Sergiu Toma 984b2b892e Update server 2022-11-29 02:20:05 +02:00
Sergiu Toma e085d22e89 Update server 2022-11-29 02:16:44 +02:00
Sergiu Toma 3bc15fdef1 Update server 2022-11-29 02:09:37 +02:00
Sergiu Toma 67042185c4 Update server 2022-11-29 01:59:53 +02:00
Sergiu Toma c92dff9bfe Update server 2022-11-28 23:40:56 +02:00
Sergiu Toma 3605ca0468 Update server 2022-11-28 23:40:08 +02:00
Sergiu Toma 1edbcb2179 Merge branch 'LH-265-enable-audio-in-mediasoup' of https://git.safemobile.org/Safemobile/mediasoup into LH-265-enable-audio-in-mediasoup 2022-11-25 10:00:28 +02:00
Sergiu Toma adbcf6c2bc Update server 2022-11-25 10:00:17 +02:00
Sergiu Toma c862224ead LH-265: Update client and server 2022-11-25 09:58:53 +02:00
Sergiu Toma c02a7c7380 Update server 2022-11-24 23:56:42 +02:00
Sergiu Toma 3387a362a6 Update server 2022-11-24 23:30:27 +02:00
Sergiu Toma 21dffefa8c Update server 2022-11-24 23:29:03 +02:00
Sergiu Toma 1369491529 Update server 2022-11-24 23:22:55 +02:00
Sergiu Toma 56bdbca537 Update server 2022-11-24 23:09:18 +02:00
Sergiu Toma 8444809910 Update server 2022-11-24 23:06:02 +02:00
Sergiu Toma cd84c534ce Update server 2022-11-24 22:27:33 +02:00
Sergiu Toma 038bdb99bc Update server 2022-11-24 22:18:37 +02:00
Sergiu Toma d94ea12a40 Update server 2022-11-24 17:08:54 +02:00
Sergiu Toma 1148532a9b Update server 2022-11-24 17:01:26 +02:00
Sergiu Toma 3561bb13a6 Update server 2022-11-24 16:59:04 +02:00
Sergiu Toma 22ead926b0 Update server 2022-11-24 16:57:26 +02:00
Sergiu Toma c6edb2947d Update server 2022-11-24 16:46:00 +02:00
Sergiu Toma e59f134a68 Update server 2022-11-24 16:37:44 +02:00
Sergiu Toma aad96b72f2 Update server 2022-11-24 16:36:36 +02:00
Sergiu Toma 3e4c0a32bc Update server 2022-11-24 16:36:00 +02:00
Sergiu Toma 2a7bd42247 Update server 2022-11-24 16:34:54 +02:00
Sergiu Toma f2c0794bf4 Update server 2022-11-24 16:22:56 +02:00
Sergiu Toma 950298c4f6 Update server 2022-11-24 16:21:22 +02:00
Sergiu Toma 6e74083733 Update server 2022-11-24 13:49:04 +02:00
Sergiu Toma 8ef6c2abb0 Update server 2022-11-24 13:43:33 +02:00
Sergiu Toma 2a86042c80 Update server 2022-11-24 13:41:24 +02:00
Sergiu Toma 56b8e2ea74 Update server 2022-11-24 13:38:28 +02:00
Sergiu Toma 6c42814229 Update server 2022-11-24 13:37:42 +02:00
Sergiu Toma e65b7e0d7c Update server 2022-11-24 13:36:21 +02:00
Sergiu Toma aa7c2aea90 Update server 2022-11-24 13:35:32 +02:00
Sergiu Toma 458342c0d2 Update server 2022-11-24 13:32:45 +02:00
Sergiu Toma fa5a1a5ae7 Update server 2022-11-23 17:56:18 +02:00
Sergiu Toma 9fbe01ae1d Update server 2022-11-23 17:19:55 +02:00
Sergiu Toma e5bcc6262b Update server 2022-11-23 17:11:09 +02:00
Sergiu Toma c758a9106c Update server 2022-11-23 16:27:13 +02:00
Sergiu Toma fcbc28c801 Update server 2022-11-23 16:26:28 +02:00
Sergiu Toma ba63fb20bf Update server 2022-11-23 16:24:12 +02:00
Sergiu Toma e8bd6837cf Update server 2022-11-23 16:21:48 +02:00
Sergiu Toma d386915ff2 Update server 2022-11-23 16:16:17 +02:00
Sergiu Toma 2479f58e21 Update server 2022-11-23 16:09:26 +02:00
Sergiu Toma d49b8e42ff Update server 2022-11-23 16:03:13 +02:00
Sergiu Toma a3ae874f8e Update server 2022-11-23 16:01:51 +02:00
Sergiu Toma c2dbef1918 Update server 2022-11-23 16:00:54 +02:00
Sergiu Toma b41b8f2d64 Update server 2022-11-23 15:57:27 +02:00
Sergiu Toma c089e91fba Update server 2022-11-23 13:20:49 +02:00
Sergiu Toma c63aee83a1 Update server 2022-11-23 13:19:56 +02:00
Sergiu Toma a97ec24148 Update server 2022-11-23 00:54:03 +02:00
Sergiu Toma 3c23c6791d Update server 2022-11-23 00:34:12 +02:00
Sergiu Toma 1a7b44807d Update server 2022-11-23 00:32:24 +02:00
Sergiu Toma daa2c556e4 Update server 2022-11-23 00:25:16 +02:00
Sergiu Toma 22656722e8 Update server 2022-11-23 00:22:00 +02:00
Sergiu Toma f5b9067b7e Update server 2022-11-23 00:21:26 +02:00
Sergiu Toma 0b3a45ae45 Update server 2022-11-23 00:13:56 +02:00
Sergiu Toma dfe4630839 Update server 2022-11-23 00:11:26 +02:00
Sergiu Toma d18041cadd Update server 2022-11-23 00:11:16 +02:00
Sergiu Toma fa42caeeb2 Update server 2022-11-22 23:14:01 +02:00
Sergiu Toma 4dbb7ad554 Update server 2022-11-22 23:12:55 +02:00
Sergiu Toma d1063803b9 Update server 2022-11-22 23:12:21 +02:00
Sergiu Toma 3cbd31b49c Update server 2022-11-22 23:11:45 +02:00
Sergiu Toma a39e0eaa17 Update server 2022-11-22 23:11:14 +02:00
Sergiu Toma b63fb39fd4 Update server 2022-11-22 23:09:33 +02:00
Sergiu Toma 0dfbd296a7 Update server 2022-11-22 20:44:22 +02:00
Sergiu Toma 233f49a998 Update server 2022-11-22 20:43:30 +02:00
Sergiu Toma 127f17cd97 Update server 2022-11-22 20:42:55 +02:00
Sergiu Toma d1ad8b4d3a Update server 2022-11-22 20:38:16 +02:00
Sergiu Toma f20e1ad260 Update build 2022-11-22 20:04:43 +02:00
Sergiu Toma 27151a26d1 Update build 2022-11-22 20:03:36 +02:00
Sergiu Toma 544e9e59ab Update build 2022-11-22 20:00:44 +02:00
Sergiu Toma 4e4cd6f893 Update build 2022-11-22 19:55:55 +02:00
Sergiu Toma e9ff060544 Update build 2022-11-22 19:52:25 +02:00
Sergiu Toma 7d677f4a34 Update build 2022-11-22 19:52:11 +02:00
Sergiu Toma 8f96b8c98b Update build 2022-11-22 19:51:09 +02:00
Sergiu Toma 1084a808c7 Update build 2022-11-22 19:40:02 +02:00
Sergiu Toma 3838f774bf Update build 2022-11-22 19:33:38 +02:00
Sergiu Toma 06bb275f0d Update build 2022-11-22 19:18:48 +02:00
Sergiu Toma a05f7cc987 Update build 2022-11-22 19:15:34 +02:00
Sergiu Toma c5c8bc5bb3 Update build 2022-11-22 19:14:50 +02:00
Sergiu Toma d6bc4e51e5 Update build 2022-11-22 19:11:00 +02:00
Sergiu Toma 4ae02f70d6 Update build 2022-11-22 18:54:22 +02:00
Sergiu Toma d593d6dc83 Update build 2022-11-22 18:35:36 +02:00
Sergiu Toma 1a1fa9450e Update build 2022-11-22 18:34:12 +02:00
Sergiu Toma 0d24604f2a Update build 2022-11-22 18:33:46 +02:00
Sergiu Toma 1d7c994036 Update build 2022-11-22 18:33:06 +02:00
Sergiu Toma bc2bf24a65 Update build 2022-11-22 18:32:47 +02:00
Sergiu Toma cdbfc7891d Update build 2022-11-22 18:30:25 +02:00
Sergiu Toma c730341674 Update build 2022-11-22 18:28:50 +02:00
Sergiu Toma b621b76e37 Connect to mediasoup with timeout(fix when it appears offline) 2022-11-22 18:27:56 +02:00
Sergiu Toma 39ad9cad27 Update bundle 2022-11-22 18:10:05 +02:00
Sergiu Toma 8860423e21 LH-265: Update client config 2022-11-22 10:28:45 +02:00
Sergiu Toma 9179a67f64 LH-265: Enable audio on video server 2022-11-21 22:59:41 +02:00
6 changed files with 920 additions and 1019 deletions

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@ -22,17 +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`

261
app.js
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@ -90,28 +90,81 @@ worker = createWorker();
// https://mediasoup.org/documentation/v3/mediasoup/rtp-parameters-and-capabilities/#RtpCodecCapability
// list of media codecs supported by mediasoup ...
// https://github.com/versatica/mediasoup/blob/v3/src/supportedRtpCapabilities.ts
const mediaCodecs = [
const mediaCodecs = [
{
kind: 'audio',
mimeType: 'audio/opus',
clockRate: 48000,
channels: 2,
kind : 'audio',
mimeType : 'audio/opus',
clockRate : 48000,
channels : 2
},
{
kind: 'video',
mimeType: 'video/VP8',
clockRate: 90000,
parameters: {
'x-google-start-bitrate': 1000,
kind : 'video',
mimeType : 'video/VP8',
clockRate : 90000,
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
}
}
// {
// kind: 'audio',
// mimeType: 'audio/opus',
// clockRate: 48000,
// channels: 2,
// },
// {
// kind: 'video',
// mimeType: 'video/VP8',
// clockRate: 90000,
// parameters: {
// 'x-google-start-bitrate': 1000,
// },
// },
];
const closeCall = (callId) => {
try {
if (callId && videoCalls[callId]) {
videoCalls[callId].producer?.close();
videoCalls[callId].consumer?.close();
videoCalls[callId].producerVideo?.close();
videoCalls[callId].producerAudio?.close();
videoCalls[callId].consumerVideo?.close();
videoCalls[callId].consumerAudio?.close();
videoCalls[callId]?.consumerTransport?.close();
videoCalls[callId]?.producerTransport?.close();
videoCalls[callId]?.router?.close();
@ -230,29 +283,54 @@ peers.on('connection', async socket => {
- The event sent by the client (PRODUCER) after successfully connecting to producerTransport
- For the router with the id callId, we make produce on producerTransport
- Create the handler on producer at the 'transportclose' event
*/
socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
try {
const callId = socketDetails[socket.id];
if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
*/
socket.on('transport-produce', async ({ kind, rtpParameters, appData }, callback) => {
try {
const callId = socketDetails[socket.id];
if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
console.log(`[transport-produce] kind: ${kind} | socket.id: ${socket.id} | callId: ${callId}`);
console.log('kind', kind);
console.log('rtpParameters', rtpParameters);
console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
kind,
rtpParameters,
});
console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producer.id} | kind: ${videoCalls[callId].producer.kind}`);
videoCalls[callId].producer.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport for this producer closed', callId)
closeCall(callId);
});
if (kind === 'video') {
videoCalls[callId].producerVideo = await videoCalls[callId].producerTransport.produce({
kind,
rtpParameters,
});
// Send back to the client the Producer's id
callback && callback({
id: videoCalls[callId].producer.id
});
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)
closeCall(callId);
});
// Send back to the client the Producer's id
callback && callback({
id: videoCalls[callId].producerVideo.id
});
} else if (kind === 'audio') {
videoCalls[callId].producerAudio = await videoCalls[callId].producerTransport.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)
closeCall(callId);
});
// Send back to the client the Producer's id
callback && callback({
id: videoCalls[callId].producerAudio.id
});
}
} catch (error) {
console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
}
@ -281,48 +359,36 @@ peers.on('connection', async socket => {
*/
socket.on('consume', async ({ rtpCapabilities }, callback) => {
try {
console.log(`[consume] rtpCapabilities: ${JSON.stringify(rtpCapabilities)}`);
const callId = socketDetails[socket.id];
console.log('[consume] callId', callId);
// Check if the router can consume the specified producer
if (videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producer.id,
const canConsumeVideo = !!videoCalls[callId].producerVideo && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producerVideo.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(callId);
});
const canConsumeAudio = !!videoCalls[callId].producerAudio && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producerAudio.id,
rtpCapabilities
})
// 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(callId);
});
console.log('[consume] canConsumeVideo', canConsumeVideo);
console.log('[consume] canConsumeAudio', canConsumeAudio);
// 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 });
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(`[canConsume] Can't consume | callId ${callId}`);
console.log(`[consume] Can't consume | callId ${callId}`);
callback(null);
}
} catch (error) {
@ -339,13 +405,71 @@ peers.on('connection', async socket => {
try {
const callId = socketDetails[socket.id];
console.log(`[consumer-resume] callId ${callId}`)
await videoCalls[callId].consumer.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, rtpCapabilities) => {
videoCalls[callId].consumerVideo = await videoCalls[callId].consumerTransport.consume({
producerId: videoCalls[callId].producerVideo.id,
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,
kind: 'video',
rtpParameters: videoCalls[callId].consumerVideo.rtpParameters,
}
}
const consumeAudio = async (callId, rtpCapabilities) => {
videoCalls[callId].consumerAudio = await videoCalls[callId].consumerTransport.consume({
producerId: videoCalls[callId].producerAudio.id,
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);
});
// 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,
kind: 'audio',
rtpParameters: videoCalls[callId].consumerAudio.rtpParameters,
}
}
/*
- 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
@ -393,6 +517,7 @@ const createWebRtcTransportLayer = async (callId, callback) => {
dtlsParameters: transport.dtlsParameters,
};
console.log('[createWebRtcTransportLayer] callback params', params);
// Send back to the client the params
callback({ params });

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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://video.safemobile.org',
}

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@ -43,7 +43,7 @@
<tr>
<td>
<div id="sharedBtns">
<video id="localVideo" autoplay class="video" ></video>
<video id="localVideo" autoplay class="video" muted></video>
</div>
</td>
<td>

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@ -12,145 +12,157 @@ let callId = parseInt(urlParams.get('callId')) || null;
const IS_PRODUCER = urlParams.get('producer') === 'true' ? true : false
console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId', callId, ' | IS_PRODUCER', IS_PRODUCER)
let socket
hub = io(config.hubAddress)
const connectToMediasoup = () => {
socket = io(config.mediasoupAddress, {
reconnection: true,
reconnectionDelay: 1000,
reconnectionDelayMax : 5000,
reconnectionAttempts: Infinity
})
socket.on('connection-success', ({ _socketId, existsProducer }) => {
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() }
})
}
if (IS_PRODUCER === true) {
hub.on('connect', async () => {
console.log(`[HUB] ${config.hubAddress} | connected: ${hub.connected}`)
connectToMediasoup()
hub.emit(
'ars',
JSON.stringify({
ars: true,
asset_id: ASSET_ID,
account_id: ACCOUNT_ID,
})
)
hub.on('video', (data) => {
const parsedData = JSON.parse(data);
if (parsedData.type === 'notify-request') {
console.log('video', parsedData)
originAssetId = parsedData.origin_asset_id;
// originAssetName = parsedData.origin_asset_name;
// originAssetTypeName = parsedData.origin_asset_type_name;
callId = parsedData.video_call_id;
console.log('[VIDEO] notify-request | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
getLocalStream()
}
if (parsedData.type === 'notify-end') {
console.log('[VIDEO] notify-end | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
resetCallSettings()
}
})
})
hub.on('connect_error', (error) => {
console.log('connect_error', error);
});
hub.on('connection', () => {
console.log('connection')
})
hub.on('disconnect', () => {
console.log('disconnect')
})
} else {
connectToMediasoup()
}
console.log('🟩 config', config)
let socket, hub
let device
let rtpCapabilities
let producerTransport
let consumerTransport
let producer
let producerVideo
let producerAudio
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
let videoParams = {
encodings: [
{
rid: 'r0',
maxBitrate: 100000,
scalabilityMode: 'S1T3',
},
{
rid: 'r1',
maxBitrate: 300000,
scalabilityMode: 'S1T3',
},
{
rid: 'r2',
maxBitrate: 900000,
scalabilityMode: 'S1T3',
},
{ scaleResolutionDownBy: 4, maxBitrate: 500000 },
{ scaleResolutionDownBy: 2, maxBitrate: 1000000 },
{ scaleResolutionDownBy: 1, maxBitrate: 5000000 },
{ scalabilityMode: 'S3T3_KEY' }
],
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerCodecOptions
codecOptions: {
videoGoogleStartBitrate: 1000
}
}
const streamSuccess = (stream) => {
console.log('[streamSuccess]');
localVideo.srcObject = stream
const track = stream.getVideoTracks()[0]
params = {
track,
...params
let audioParams = {
codecOptions :
{
opusStereo : true,
opusDtx : true
}
}
setTimeout(() => {
hub = io(config.hubAddress)
const connectToMediasoup = () => {
socket = io(config.mediasoupAddress, {
reconnection: true,
reconnectionDelay: 1000,
reconnectionDelayMax : 5000,
reconnectionAttempts: Infinity
})
socket.on('connection-success', ({ _socketId, existsProducer }) => {
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() }
})
}
if (IS_PRODUCER === true) {
hub.on('connect', async () => {
console.log(`[HUB]! ${config.hubAddress} | connected: ${hub.connected}`)
connectToMediasoup()
hub.emit(
'ars',
JSON.stringify({
ars: true,
asset_id: ASSET_ID,
account_id: ACCOUNT_ID,
})
)
hub.on('video', (data) => {
const parsedData = JSON.parse(data);
if (parsedData.type === 'notify-request') {
console.log('video', parsedData)
originAssetId = parsedData.origin_asset_id;
// originAssetName = parsedData.origin_asset_name;
// originAssetTypeName = parsedData.origin_asset_type_name;
callId = parsedData.video_call_id;
console.log('[VIDEO] notify-request | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
getLocalStream()
}
if (parsedData.type === 'notify-end') {
console.log('[VIDEO] notify-end | IS_PRODUCER', IS_PRODUCER, 'callId', callId);
resetCallSettings()
}
})
})
hub.on('connect_error', (error) => {
console.log('connect_error', error);
});
hub.on('connection', () => {
console.log('connection')
})
hub.on('disconnect', () => {
console.log('disconnect')
})
} else {
connectToMediasoup()
}
}, 1600);
const streamSuccess = (stream) => {
console.log('[streamSuccess] device', device);
localVideo.srcObject = stream
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: true,
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 +179,6 @@ 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
@ -178,7 +189,8 @@ const createDevice = async () => {
})
console.log('Device RTP Capabilities', device.rtpCapabilities)
console.log('[createDevice] device', device);
// once the device loads, create transport
goCreateTransport()
@ -207,18 +219,20 @@ 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, callId }, (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 +258,10 @@ 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', ...)
@ -270,20 +284,42 @@ 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)
// Produce video
producerVideo = await producerTransport.produce(videoParams)
console.log('videoParams', videoParams);
console.log('producerVideo', producerVideo);
producer.on('trackended', () => {
producerVideo.on('trackended', () => {
console.log('track ended')
// close video track
})
})
producer.on('transportclose', () => {
producerVideo.on('transportclose', () => {
console.log('transport ended')
// close video track
})
// Produce audio
producerAudio = await producerTransport.produce(audioParams)
console.log('audioParams', audioParams);
console.log('producerAudio', producerAudio);
producerAudio.on('trackended', () => {
console.log('track ended')
// close audio track
})
producerAudio.on('transportclose', () => {
console.log('transport ended')
// close audio track
})
const answer = {
@ -294,7 +330,7 @@ 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 +346,7 @@ 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 +356,15 @@ 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 +389,8 @@ const resetCallSettings = () => {
localVideo.srcObject = null
remoteVideo.srcObject = null
consumer = null
producer = null
producerVideo = null
producerAudio = null
producerTransport = null
consumerTransport = null
device = undefined
@ -361,7 +398,7 @@ 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', {
@ -373,7 +410,7 @@ const connectRecvTransport = async () => {
return
}
// then consume with the local consumer transport
// Then consume with the local consumer transport
// which creates a consumer
consumer = await consumerTransport.consume({
id: params.id,
@ -416,6 +453,7 @@ const closeCall = () => {
resetCallSettings()
}
btnLocalVideo.addEventListener('click', getLocalStream)
btnRecvSendTransport.addEventListener('click', goConnect)
btnCloseCall.addEventListener('click', closeCall)