LH-265-enable-audio-in-mediasoup #16
@ -22,17 +22,20 @@
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2. Run the `npm start:prod` command to start the server in production mode.
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(To connect to the terminal, use `pm2 log video-server`)
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---
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### Web client
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- The server will start by default on port 3000, and the ssl certificates will have to be configured
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- The web client can be accessed using the /sfu path
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ex: http://localhost:3000/sfu/?assetId=1&&accountId=1&producer=true&assetName=Adi&assetType=linx
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ex: https://HOST/sfu/?assetId=1&&accountId=1&producer=true&dest_asset_id=75&assetName=Adi
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assetId = asset id of the unit on which you are doing the test
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accountId = account id of the unit on which you are doing the test
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producer = it will always be true because you are the producer
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(it's possible to put false, but then you have to have another client with producer true)
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assetName = asset name of the unit on which you are doing the test
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assetType = asset type of the unit on which you are doing the test
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dest_asset_id= the addressee with whom the call is made
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- To make a call using this client, you need a microphone and permission to use it
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- 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
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### Demo project
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The demo project used initially and then modified for our needs `https://github.com/jamalag/mediasoup2`
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215
app.js
215
app.js
@ -95,23 +95,76 @@ const mediaCodecs = [
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kind : 'audio',
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mimeType : 'audio/opus',
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clockRate : 48000,
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channels: 2,
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channels : 2
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},
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{
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kind : 'video',
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mimeType : 'video/VP8',
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clockRate : 90000,
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parameters: {
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'x-google-start-bitrate': 1000,
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parameters :
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{
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'x-google-start-bitrate' : 1000
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},
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channels : 2
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},
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{
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kind : 'video',
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mimeType : 'video/VP9',
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clockRate : 90000,
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parameters :
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{
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'profile-id' : 2,
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'x-google-start-bitrate' : 1000
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}
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},
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{
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kind : 'video',
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mimeType : 'video/h264',
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clockRate : 90000,
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parameters :
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{
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'packetization-mode' : 1,
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'profile-level-id' : '4d0032',
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'level-asymmetry-allowed' : 1,
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'x-google-start-bitrate' : 1000
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}
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},
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{
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kind : 'video',
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mimeType : 'video/h264',
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clockRate : 90000,
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parameters :
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{
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'packetization-mode' : 1,
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'profile-level-id' : '42e01f',
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'level-asymmetry-allowed' : 1,
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'x-google-start-bitrate' : 1000
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}
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}
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// {
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// kind: 'audio',
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// mimeType: 'audio/opus',
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// clockRate: 48000,
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// channels: 2,
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// },
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// {
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// kind: 'video',
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// mimeType: 'video/VP8',
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// clockRate: 90000,
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// parameters: {
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// 'x-google-start-bitrate': 1000,
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// },
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// },
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];
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const closeCall = (callId) => {
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try {
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if (callId && videoCalls[callId]) {
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videoCalls[callId].producer?.close();
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videoCalls[callId].consumer?.close();
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videoCalls[callId].producerVideo?.close();
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videoCalls[callId].producerAudio?.close();
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videoCalls[callId].consumerVideo?.close();
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videoCalls[callId].consumerAudio?.close();
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videoCalls[callId]?.consumerTransport?.close();
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videoCalls[callId]?.producerTransport?.close();
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videoCalls[callId]?.router?.close();
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@ -236,14 +289,20 @@ peers.on('connection', async socket => {
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const callId = socketDetails[socket.id];
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if (typeof rtpParameters === 'string') rtpParameters = JSON.parse(rtpParameters);
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console.log('[transport-produce] | socket.id', socket.id, '| callId', callId);
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videoCalls[callId].producer = await videoCalls[callId].producerTransport.produce({
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console.log(`[transport-produce] kind: ${kind} | socket.id: ${socket.id} | callId: ${callId}`);
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console.log('kind', kind);
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console.log('rtpParameters', rtpParameters);
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if (kind === 'video') {
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videoCalls[callId].producerVideo = await videoCalls[callId].producerTransport.produce({
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kind,
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rtpParameters,
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});
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console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producer.id} | kind: ${videoCalls[callId].producer.kind}`);
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videoCalls[callId].producer.on('transportclose', () => {
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console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producerVideo.id} | kind: ${videoCalls[callId].producerVideo.kind}`);
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videoCalls[callId].producerVideo.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport for this producer closed', callId)
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closeCall(callId);
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@ -251,8 +310,27 @@ peers.on('connection', async socket => {
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// Send back to the client the Producer's id
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callback && callback({
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id: videoCalls[callId].producer.id
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id: videoCalls[callId].producerVideo.id
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});
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} else if (kind === 'audio') {
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videoCalls[callId].producerAudio = await videoCalls[callId].producerTransport.produce({
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kind,
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rtpParameters,
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});
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console.log(`[transport-produce] Producer ID: ${videoCalls[callId].producerAudio.id} | kind: ${videoCalls[callId].producerAudio.kind}`);
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videoCalls[callId].producerAudio.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport for this producer closed', callId)
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closeCall(callId);
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});
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// Send back to the client the Producer's id
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callback && callback({
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id: videoCalls[callId].producerAudio.id
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});
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}
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} catch (error) {
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console.log(`ERROR | transport-produce | callId ${socketDetails[socket.id]} | ${error.message}`);
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}
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@ -281,48 +359,36 @@ peers.on('connection', async socket => {
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*/
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socket.on('consume', async ({ rtpCapabilities }, callback) => {
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try {
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console.log(`[consume] rtpCapabilities: ${JSON.stringify(rtpCapabilities)}`);
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const callId = socketDetails[socket.id];
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console.log('[consume] callId', callId);
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// Check if the router can consume the specified producer
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if (videoCalls[callId].router.canConsume({
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producerId: videoCalls[callId].producer.id,
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const canConsumeVideo = !!videoCalls[callId].producerVideo && !!videoCalls[callId].router.canConsume({
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producerId: videoCalls[callId].producerVideo.id,
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rtpCapabilities
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})) {
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console.log('[consume] Can consume', callId);
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// Transport can now consume and return a consumer
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videoCalls[callId].consumer = await videoCalls[callId].consumerTransport.consume({
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producerId: videoCalls[callId].producer.id,
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rtpCapabilities,
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paused: true,
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});
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})
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// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
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videoCalls[callId].consumer.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport close from consumer', callId);
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closeCall(callId);
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});
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const canConsumeAudio = !!videoCalls[callId].producerAudio && !!videoCalls[callId].router.canConsume({
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producerId: videoCalls[callId].producerAudio.id,
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rtpCapabilities
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})
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// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
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videoCalls[callId].consumer.on('producerclose', () => {
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const callId = socketDetails[socket.id];
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console.log('producer of consumer closed', callId);
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closeCall(callId);
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});
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console.log('[consume] canConsumeVideo', canConsumeVideo);
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console.log('[consume] canConsumeAudio', canConsumeAudio);
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// From the consumer extract the following params to send back to the Client
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const params = {
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id: videoCalls[callId].consumer.id,
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producerId: videoCalls[callId].producer.id,
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kind: videoCalls[callId].consumer.kind,
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rtpParameters: videoCalls[callId].consumer.rtpParameters,
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};
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// Send the parameters to the client
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callback({ params });
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if (canConsumeVideo && !canConsumeAudio) {
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console.log('1');
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const videoParams = await consumeVideo(callId, rtpCapabilities)
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console.log('videoParams', videoParams);
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callback({ videoParams, audioParams: null });
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} else if (canConsumeVideo && canConsumeAudio) {
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console.log('2');
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const videoParams = await consumeVideo(callId, rtpCapabilities)
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const audioParams = await consumeAudio(callId, rtpCapabilities)
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callback({ videoParams, audioParams });
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} else {
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console.log(`[canConsume] Can't consume | callId ${callId}`);
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console.log(`[consume] Can't consume | callId ${callId}`);
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callback(null);
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}
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} catch (error) {
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@ -339,13 +405,71 @@ peers.on('connection', async socket => {
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try {
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const callId = socketDetails[socket.id];
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console.log(`[consumer-resume] callId ${callId}`)
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await videoCalls[callId].consumer.resume();
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await videoCalls[callId].consumerVideo.resume();
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await videoCalls[callId].consumerAudio.resume();
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} catch (error) {
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console.log(`ERROR | consumer-resume | callId ${socketDetails[socket.id]} | ${error.message}`);
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}
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});
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});
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const consumeVideo = async (callId, rtpCapabilities) => {
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videoCalls[callId].consumerVideo = await videoCalls[callId].consumerTransport.consume({
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producerId: videoCalls[callId].producerVideo.id,
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rtpCapabilities,
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paused: true,
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});
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// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
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videoCalls[callId].consumerVideo.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport close from consumer', callId);
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closeCall(callId);
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});
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// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
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videoCalls[callId].consumerVideo.on('producerclose', () => {
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const callId = socketDetails[socket.id];
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console.log('producer of consumer closed', callId);
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closeCall(callId);
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});
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return {
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id: videoCalls[callId].consumerVideo.id,
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producerId: videoCalls[callId].producerVideo.id,
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kind: 'video',
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rtpParameters: videoCalls[callId].consumerVideo.rtpParameters,
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}
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}
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const consumeAudio = async (callId, rtpCapabilities) => {
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videoCalls[callId].consumerAudio = await videoCalls[callId].consumerTransport.consume({
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producerId: videoCalls[callId].producerAudio.id,
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rtpCapabilities,
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paused: true,
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});
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// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-transportclose
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videoCalls[callId].consumerAudio.on('transportclose', () => {
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const callId = socketDetails[socket.id];
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console.log('transport close from consumer', callId);
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closeCall(callId);
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});
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// https://mediasoup.org/documentation/v3/mediasoup/api/#consumer-on-producerclose
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videoCalls[callId].consumerAudio.on('producerclose', () => {
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const callId = socketDetails[socket.id];
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console.log('producer of consumer closed', callId);
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closeCall(callId);
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});
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return {
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id: videoCalls[callId].consumerAudio.id,
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producerId: videoCalls[callId].producerAudio.id,
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kind: 'audio',
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rtpParameters: videoCalls[callId].consumerAudio.rtpParameters,
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}
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}
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|
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/*
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- Called from at event 'createWebRtcTransport' and assigned to the consumer or producer transport
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- It will return parameters, these are required for the client to create the RecvTransport
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@ -393,6 +517,7 @@ const createWebRtcTransportLayer = async (callId, callback) => {
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dtlsParameters: transport.dtlsParameters,
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};
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console.log('[createWebRtcTransportLayer] callback params', params);
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// Send back to the client the params
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callback({ params });
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|
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|
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1076
public/bundle.js
1076
public/bundle.js
File diff suppressed because it is too large
Load Diff
@ -1,5 +1,4 @@
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module.exports = {
|
||||
hubAddress: 'https://hub.dev.linx.safemobile.com/',
|
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mediasoupAddress: 'https://video.safemobile.org/mediasoup',
|
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// mediasoupAddress: 'http://localhost:3000/mediasoup',
|
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mediasoupAddress: 'https://video.safemobile.org',
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}
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@ -43,7 +43,7 @@
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<tr>
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<td>
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||||
<div id="sharedBtns">
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<video id="localVideo" autoplay class="video" ></video>
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<video id="localVideo" autoplay class="video" muted></video>
|
||||
</div>
|
||||
</td>
|
||||
<td>
|
||||
|
||||
180
public/index.js
180
public/index.js
@ -12,7 +12,41 @@ let callId = parseInt(urlParams.get('callId')) || null;
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const IS_PRODUCER = urlParams.get('producer') === 'true' ? true : false
|
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console.log('[URL] ASSET_ID', ASSET_ID, '| ACCOUNT_ID', ACCOUNT_ID, '| callId', callId, ' | IS_PRODUCER', IS_PRODUCER)
|
||||
|
||||
let socket
|
||||
console.log('🟩 config', config)
|
||||
|
||||
let socket, hub
|
||||
let device
|
||||
let rtpCapabilities
|
||||
let producerTransport
|
||||
let consumerTransport
|
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let producerVideo
|
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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)
|
||||
|
||||
const connectToMediasoup = () => {
|
||||
@ -36,7 +70,7 @@ const connectToMediasoup = () => {
|
||||
|
||||
if (IS_PRODUCER === true) {
|
||||
hub.on('connect', async () => {
|
||||
console.log(`[HUB] ${config.hubAddress} | connected: ${hub.connected}`)
|
||||
console.log(`[HUB]! ${config.hubAddress} | connected: ${hub.connected}`)
|
||||
connectToMediasoup()
|
||||
|
||||
hub.emit(
|
||||
@ -84,73 +118,51 @@ if (IS_PRODUCER === true) {
|
||||
connectToMediasoup()
|
||||
}
|
||||
|
||||
let device
|
||||
let rtpCapabilities
|
||||
let producerTransport
|
||||
let consumerTransport
|
||||
let producer
|
||||
let consumer
|
||||
let originAssetId
|
||||
// let originAssetName = 'Adi'
|
||||
// let originAssetTypeName = 'linx'
|
||||
|
||||
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerOptions
|
||||
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
|
||||
let params = {
|
||||
// mediasoup params
|
||||
encodings: [
|
||||
{
|
||||
rid: 'r0',
|
||||
maxBitrate: 100000,
|
||||
scalabilityMode: 'S1T3',
|
||||
},
|
||||
{
|
||||
rid: 'r1',
|
||||
maxBitrate: 300000,
|
||||
scalabilityMode: 'S1T3',
|
||||
},
|
||||
{
|
||||
rid: 'r2',
|
||||
maxBitrate: 900000,
|
||||
scalabilityMode: 'S1T3',
|
||||
},
|
||||
],
|
||||
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#ProducerCodecOptions
|
||||
codecOptions: {
|
||||
videoGoogleStartBitrate: 1000
|
||||
}
|
||||
}
|
||||
}, 1600);
|
||||
|
||||
const streamSuccess = (stream) => {
|
||||
console.log('[streamSuccess]');
|
||||
console.log('[streamSuccess] device', device);
|
||||
localVideo.srcObject = stream
|
||||
const track = stream.getVideoTracks()[0]
|
||||
params = {
|
||||
track,
|
||||
...params
|
||||
console.log('stream', stream);
|
||||
const videoTrack = stream.getVideoTracks()[0]
|
||||
const audioTrack = stream.getAudioTracks()[0]
|
||||
|
||||
videoParams = {
|
||||
track: videoTrack,
|
||||
...videoParams
|
||||
}
|
||||
|
||||
audioParams = {
|
||||
track: audioTrack,
|
||||
...audioParams
|
||||
}
|
||||
|
||||
console.log('[streamSuccess] videoParams', videoParams, ' | audioParams', audioParams);
|
||||
goConnect()
|
||||
}
|
||||
|
||||
const getLocalStream = () => {
|
||||
console.log('[getLocalStream]');
|
||||
navigator.mediaDevices.getUserMedia({
|
||||
audio: false,
|
||||
audio: 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
|
||||
@ -178,6 +189,7 @@ const createDevice = async () => {
|
||||
})
|
||||
|
||||
console.log('Device RTP Capabilities', device.rtpCapabilities)
|
||||
console.log('[createDevice] device', device);
|
||||
|
||||
// once the device loads, create transport
|
||||
goCreateTransport()
|
||||
@ -207,18 +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', ...)
|
||||
@ -270,22 +284,44 @@ const createSendTransport = () => {
|
||||
}
|
||||
|
||||
const connectSendTransport = async () => {
|
||||
// we now call produce() to instruct the producer transport
|
||||
|
||||
console.log('[connectSendTransport] producerTransport');
|
||||
|
||||
// We now call produce() to instruct the producer transport
|
||||
// to send media to the Router
|
||||
// https://mediasoup.org/documentation/v3/mediasoup-client/api/#transport-produce
|
||||
// this action will trigger the 'connect' and 'produce' events above
|
||||
producer = await producerTransport.produce(params)
|
||||
|
||||
producer.on('trackended', () => {
|
||||
// Produce video
|
||||
producerVideo = await producerTransport.produce(videoParams)
|
||||
console.log('videoParams', videoParams);
|
||||
console.log('producerVideo', producerVideo);
|
||||
|
||||
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 = {
|
||||
origin_asset_id: ASSET_ID,
|
||||
dest_asset_id: originAssetId || parseInt(urlParams.get('dest_asset_id')),
|
||||
@ -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)
|
||||
Loading…
Reference in New Issue
Block a user