Compare commits

..

60 Commits

Author SHA1 Message Date
458342c0d2 Update server 2022-11-24 13:32:45 +02:00
fa5a1a5ae7 Update server 2022-11-23 17:56:18 +02:00
9fbe01ae1d Update server 2022-11-23 17:19:55 +02:00
e5bcc6262b Update server 2022-11-23 17:11:09 +02:00
c758a9106c Update server 2022-11-23 16:27:13 +02:00
fcbc28c801 Update server 2022-11-23 16:26:28 +02:00
ba63fb20bf Update server 2022-11-23 16:24:12 +02:00
e8bd6837cf Update server 2022-11-23 16:21:48 +02:00
d386915ff2 Update server 2022-11-23 16:16:17 +02:00
2479f58e21 Update server 2022-11-23 16:09:26 +02:00
d49b8e42ff Update server 2022-11-23 16:03:13 +02:00
a3ae874f8e Update server 2022-11-23 16:01:51 +02:00
c2dbef1918 Update server 2022-11-23 16:00:54 +02:00
b41b8f2d64 Update server 2022-11-23 15:57:27 +02:00
c089e91fba Update server 2022-11-23 13:20:49 +02:00
c63aee83a1 Update server 2022-11-23 13:19:56 +02:00
a97ec24148 Update server 2022-11-23 00:54:03 +02:00
3c23c6791d Update server 2022-11-23 00:34:12 +02:00
1a7b44807d Update server 2022-11-23 00:32:24 +02:00
daa2c556e4 Update server 2022-11-23 00:25:16 +02:00
22656722e8 Update server 2022-11-23 00:22:00 +02:00
f5b9067b7e Update server 2022-11-23 00:21:26 +02:00
0b3a45ae45 Update server 2022-11-23 00:13:56 +02:00
dfe4630839 Update server 2022-11-23 00:11:26 +02:00
d18041cadd Update server 2022-11-23 00:11:16 +02:00
fa42caeeb2 Update server 2022-11-22 23:14:01 +02:00
4dbb7ad554 Update server 2022-11-22 23:12:55 +02:00
d1063803b9 Update server 2022-11-22 23:12:21 +02:00
3cbd31b49c Update server 2022-11-22 23:11:45 +02:00
a39e0eaa17 Update server 2022-11-22 23:11:14 +02:00
b63fb39fd4 Update server 2022-11-22 23:09:33 +02:00
0dfbd296a7 Update server 2022-11-22 20:44:22 +02:00
233f49a998 Update server 2022-11-22 20:43:30 +02:00
127f17cd97 Update server 2022-11-22 20:42:55 +02:00
d1ad8b4d3a Update server 2022-11-22 20:38:16 +02:00
f20e1ad260 Update build 2022-11-22 20:04:43 +02:00
27151a26d1 Update build 2022-11-22 20:03:36 +02:00
544e9e59ab Update build 2022-11-22 20:00:44 +02:00
4e4cd6f893 Update build 2022-11-22 19:55:55 +02:00
e9ff060544 Update build 2022-11-22 19:52:25 +02:00
7d677f4a34 Update build 2022-11-22 19:52:11 +02:00
8f96b8c98b Update build 2022-11-22 19:51:09 +02:00
1084a808c7 Update build 2022-11-22 19:40:02 +02:00
3838f774bf Update build 2022-11-22 19:33:38 +02:00
06bb275f0d Update build 2022-11-22 19:18:48 +02:00
a05f7cc987 Update build 2022-11-22 19:15:34 +02:00
c5c8bc5bb3 Update build 2022-11-22 19:14:50 +02:00
d6bc4e51e5 Update build 2022-11-22 19:11:00 +02:00
4ae02f70d6 Update build 2022-11-22 18:54:22 +02:00
d593d6dc83 Update build 2022-11-22 18:35:36 +02:00
1a1fa9450e Update build 2022-11-22 18:34:12 +02:00
0d24604f2a Update build 2022-11-22 18:33:46 +02:00
1d7c994036 Update build 2022-11-22 18:33:06 +02:00
bc2bf24a65 Update build 2022-11-22 18:32:47 +02:00
cdbfc7891d Update build 2022-11-22 18:30:25 +02:00
c730341674 Update build 2022-11-22 18:28:50 +02:00
b621b76e37 Connect to mediasoup with timeout(fix when it appears offline) 2022-11-22 18:27:56 +02:00
39ad9cad27 Update bundle 2022-11-22 18:10:05 +02:00
8860423e21 LH-265: Update client config 2022-11-22 10:28:45 +02:00
9179a67f64 LH-265: Enable audio on video server 2022-11-21 22:59:41 +02:00
6 changed files with 308 additions and 324 deletions

View File

@ -22,20 +22,18 @@
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: https://HOST/sfu/?assetId=1&&accountId=1&producer=true&dest_asset_id=75&assetName=Adi
ex: http://localhost:3000/sfu/?assetId=1&&accountId=1&producer=true&assetName=Adi&assetType=linx
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
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
assetType = asset type of the unit on which you are doing the test
### Demo project
The demo project used initially and then modified for our needs `https://github.com/jamalag/mediasoup2`

194
app.js
View File

@ -12,7 +12,7 @@ try {
console.log('https support is disabled!');
}
const mediasoup = require('mediasoup');
console.log('---------🔴🔴---------');
let worker
/**
* videoCalls
@ -104,8 +104,7 @@ const mediaCodecs = [
parameters :
{
'x-google-start-bitrate' : 1000
},
channels : 2
}
},
{
kind : 'video',
@ -160,11 +159,8 @@ const mediaCodecs = [
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].producer?.close();
videoCalls[callId].consumer?.close();
videoCalls[callId]?.consumerTransport?.close();
videoCalls[callId]?.producerTransport?.close();
videoCalls[callId]?.router?.close();
@ -283,54 +279,29 @@ 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}`);
if (kind === 'video') {
videoCalls[callId].producerVideo = await videoCalls[callId].producerTransport.produce({
kind,
rtpParameters,
});
videoCalls[callId].producer.on('transportclose', () => {
const callId = socketDetails[socket.id];
console.log('transport for this producer closed', callId)
closeCall(callId);
});
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
});
}
// Send back to the client the Producer's id
callback && callback({
id: videoCalls[callId].producer.id
});
} catch (error) {
console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
}
@ -359,36 +330,48 @@ 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);
const canConsumeVideo = !!videoCalls[callId].producerVideo && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producerVideo.id,
// Check if the router can consume the specified producer
if (videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producer.id,
rtpCapabilities
})
})) {
console.log('[consume] Can consume', callId);
// Transport can now consume and return a consumer
videoCalls[callId].consumer = await videoCalls[callId].consumerTransport.consume({
producerId: videoCalls[callId].producer.id,
rtpCapabilities,
paused: true,
});
const canConsumeAudio = !!videoCalls[callId].producerAudio && !!videoCalls[callId].router.canConsume({
producerId: videoCalls[callId].producerAudio.id,
rtpCapabilities
})
// 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);
});
console.log('[consume] canConsumeVideo', canConsumeVideo);
console.log('[consume] canConsumeAudio', canConsumeAudio);
// 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);
});
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 });
// 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(`[consume] Can't consume | callId ${callId}`);
console.log(`[canConsume] Can't consume | callId ${callId}`);
callback(null);
}
} catch (error) {
@ -405,71 +388,13 @@ peers.on('connection', async socket => {
try {
const callId = socketDetails[socket.id];
console.log(`[consumer-resume] callId ${callId}`)
await videoCalls[callId].consumerVideo.resume();
await videoCalls[callId].consumerAudio.resume();
await videoCalls[callId].consumer.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
@ -517,7 +442,6 @@ const createWebRtcTransportLayer = async (callId, callback) => {
dtlsParameters: transport.dtlsParameters,
};
console.log('[createWebRtcTransportLayer] callback params', params);
// Send back to the client the params
callback({ params });

View File

@ -44,3 +44,5 @@ fi
## POST BUILD
cd -

View File

@ -20373,36 +20373,6 @@ console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId',
console.log('🟩 config', config)
let socket, hub
let device
let rtpCapabilities
let producerTransport
let consumerTransport
let producerVideo
let producerAudio
let consumer
let originAssetId
// 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)
@ -20478,49 +20448,117 @@ setTimeout(() => {
}, 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]
let device
let rtpCapabilities
let producerTransport
let consumerTransport
let producerVideo
let producerAudio
let consumer
let originAssetId
// 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
// }
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
}
}
let audioParams = {
codecOptions :
{
opusStereo : true,
opusDtx : true
}
}
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,
// // codec : device.rtpCapabilities.codecs.find((codec) => codec.mimeType.toLowerCase() === 'video/vp9'),
// // codec : 'video/vp9',
// ...videoParams
// }
// audioParams = {
// track: audioTrack,
// ...audioParams
// }
// console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
// goConnect()
console.log('[streamSuccess]');
localVideo.srcObject = stream
const track = stream.getVideoTracks()[0]
videoParams = {
track: videoTrack,
track,
...videoParams
}
audioParams = {
track: audioTrack,
...audioParams
}
console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
goConnect()
}
const getLocalStream = () => {
console.log('[getLocalStream]');
// navigator.mediaDevices.getUserMedia({
// audio: false,
// video: {
// qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
// vga : { width: { ideal: 640 }, height: { ideal: 480 } },
// hd : { width: { ideal: 1280 }, height: { ideal: 720 } }
// }
// })
navigator.mediaDevices.getUserMedia({
audio: true,
audio: false,
video: {
qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
vga : { width: { ideal: 640 }, height: { ideal: 480 } },
hd : { width: { ideal: 1280 }, height: { ideal: 720 } }
width: {
min: 640,
max: 1920,
},
height: {
min: 400,
max: 1080,
}
}
})
.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 = () => {
@ -20537,6 +20575,7 @@ const goCreateTransport = () => {
// server side to send/recive media
const createDevice = async () => {
try {
console.log('[createDevice] 1 device', device);
device = new mediasoupClient.Device()
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@ -20547,7 +20586,7 @@ const createDevice = async () => {
})
console.log('Device RTP Capabilities', device.rtpCapabilities)
console.log('[createDevice] device', device);
console.log('[createDevice] 2 device', device);
// once the device loads, create transport
goCreateTransport()
@ -20580,17 +20619,16 @@ 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) => {
console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
const params = value.params;
socket.emit('createWebRtcTransport', { sender: true, callId }, ({ 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('[createWebRtcTransport] params', 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
@ -20619,7 +20657,7 @@ const createSendTransport = () => {
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', ...)
@ -20645,16 +20683,12 @@ const connectSendTransport = async () => {
console.log('[connectSendTransport] producerTransport');
// We now call produce() to instruct the producer transport
// 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
// Produce video
producerVideo = await producerTransport.produce(videoParams)
console.log('videoParams', videoParams);
console.log('producerVideo', producerVideo);
producerVideo.on('trackended', () => {
console.log('track ended')
// close video track
@ -20665,20 +20699,17 @@ const connectSendTransport = async () => {
// close video track
})
// Produce audio
producerAudio = await producerTransport.produce(audioParams)
console.log('audioParams', audioParams);
console.log('producerAudio', producerAudio);
// producerAudio = await producerTransport.produce(audioParams)
// console.log('producerAudio', producerAudio);
// producerAudio.on('trackended', () => {
// console.log('track ended')
// // close video track
// })
producerAudio.on('trackended', () => {
console.log('track ended')
// close audio track
})
producerAudio.on('transportclose', () => {
console.log('transport ended')
// close audio track
})
// producerAudio.on('transportclose', () => {
// console.log('transport ended')
// // close video track
// })
const answer = {
origin_asset_id: ASSET_ID,
@ -20688,7 +20719,7 @@ const connectSendTransport = async () => {
origin_asset_type_name: ASSET_TYPE,
origin_asset_name: ASSET_NAME,
video_call_id: callId,
answer: 'accepted', // answer: accepted/rejected
answer: 'accepted', // answer: 'rejected'
};
console.log('SEND answer', answer);
@ -20704,7 +20735,7 @@ 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
@ -20716,13 +20747,13 @@ const createRecvTransport = async () => {
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 {
@ -20756,7 +20787,7 @@ 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', {
@ -20768,7 +20799,7 @@ 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,
@ -20811,7 +20842,6 @@ const closeCall = () => {
resetCallSettings()
}
btnLocalVideo.addEventListener('click', getLocalStream)
btnRecvSendTransport.addEventListener('click', goConnect)
btnCloseCall.addEventListener('click', closeCall)

View File

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

View File

@ -15,36 +15,6 @@ console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId',
console.log('🟩 config', config)
let socket, hub
let device
let rtpCapabilities
let producerTransport
let consumerTransport
let producerVideo
let producerAudio
let consumer
let originAssetId
// 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)
@ -120,49 +90,117 @@ setTimeout(() => {
}, 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]
let device
let rtpCapabilities
let producerTransport
let consumerTransport
let producerVideo
let producerAudio
let consumer
let originAssetId
// 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
// }
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
}
}
let audioParams = {
codecOptions :
{
opusStereo : true,
opusDtx : true
}
}
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,
// // codec : device.rtpCapabilities.codecs.find((codec) => codec.mimeType.toLowerCase() === 'video/vp9'),
// // codec : 'video/vp9',
// ...videoParams
// }
// audioParams = {
// track: audioTrack,
// ...audioParams
// }
// console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
// goConnect()
console.log('[streamSuccess]');
localVideo.srcObject = stream
const track = stream.getVideoTracks()[0]
videoParams = {
track: videoTrack,
track,
...videoParams
}
audioParams = {
track: audioTrack,
...audioParams
}
console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
goConnect()
}
const getLocalStream = () => {
console.log('[getLocalStream]');
// navigator.mediaDevices.getUserMedia({
// audio: false,
// video: {
// qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
// vga : { width: { ideal: 640 }, height: { ideal: 480 } },
// hd : { width: { ideal: 1280 }, height: { ideal: 720 } }
// }
// })
navigator.mediaDevices.getUserMedia({
audio: true,
audio: false,
video: {
qvga : { width: { ideal: 320 }, height: { ideal: 240 } },
vga : { width: { ideal: 640 }, height: { ideal: 480 } },
hd : { width: { ideal: 1280 }, height: { ideal: 720 } }
width: {
min: 640,
max: 1920,
},
height: {
min: 400,
max: 1080,
}
}
})
.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 = () => {
@ -179,6 +217,7 @@ const goCreateTransport = () => {
// server side to send/recive media
const createDevice = async () => {
try {
console.log('[createDevice] 1 device', device);
device = new mediasoupClient.Device()
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#device-load
@ -189,7 +228,7 @@ const createDevice = async () => {
})
console.log('Device RTP Capabilities', device.rtpCapabilities)
console.log('[createDevice] device', device);
console.log('[createDevice] 2 device', device);
// once the device loads, create transport
goCreateTransport()
@ -222,17 +261,16 @@ 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) => {
console.log(`[createWebRtcTransport] value: ${JSON.stringify(value)}`);
const params = value.params;
socket.emit('createWebRtcTransport', { sender: true, callId }, ({ 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('[createWebRtcTransport] params', 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
@ -261,7 +299,7 @@ const createSendTransport = () => {
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', ...)
@ -287,16 +325,12 @@ const connectSendTransport = async () => {
console.log('[connectSendTransport] producerTransport');
// We now call produce() to instruct the producer transport
// 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
// Produce video
producerVideo = await producerTransport.produce(videoParams)
console.log('videoParams', videoParams);
console.log('producerVideo', producerVideo);
producerVideo.on('trackended', () => {
console.log('track ended')
// close video track
@ -307,20 +341,17 @@ const connectSendTransport = async () => {
// close video track
})
// Produce audio
producerAudio = await producerTransport.produce(audioParams)
console.log('audioParams', audioParams);
console.log('producerAudio', producerAudio);
// producerAudio = await producerTransport.produce(audioParams)
// console.log('producerAudio', producerAudio);
// producerAudio.on('trackended', () => {
// console.log('track ended')
// // close video track
// })
producerAudio.on('trackended', () => {
console.log('track ended')
// close audio track
})
producerAudio.on('transportclose', () => {
console.log('transport ended')
// close audio track
})
// producerAudio.on('transportclose', () => {
// console.log('transport ended')
// // close video track
// })
const answer = {
origin_asset_id: ASSET_ID,
@ -330,7 +361,7 @@ const connectSendTransport = async () => {
origin_asset_type_name: ASSET_TYPE,
origin_asset_name: ASSET_NAME,
video_call_id: callId,
answer: 'accepted', // answer: accepted/rejected
answer: 'accepted', // answer: 'rejected'
};
console.log('SEND answer', answer);
@ -346,7 +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
@ -358,13 +389,13 @@ const createRecvTransport = async () => {
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 {
@ -398,7 +429,7 @@ 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', {
@ -410,7 +441,7 @@ 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,
@ -453,7 +484,6 @@ const closeCall = () => {
resetCallSettings()
}
btnLocalVideo.addEventListener('click', getLocalStream)
btnRecvSendTransport.addEventListener('click', goConnect)
btnCloseCall.addEventListener('click', closeCall)