[mirotalksfu] - add webRtcServer
هذا الالتزام موجود في:
@@ -8,6 +8,8 @@ module.exports = class Room {
|
||||
constructor(room_id, worker, io) {
|
||||
this.id = room_id;
|
||||
this.worker = worker;
|
||||
this.webRtcServer = worker.appData.webRtcServer;
|
||||
this.webRtcServerActive = config.mediasoup.webRtcServerActive;
|
||||
this.io = io;
|
||||
this.audioLevelObserver = null;
|
||||
this.audioLevelObserverEnabled = true;
|
||||
@@ -248,13 +250,18 @@ module.exports = class Room {
|
||||
|
||||
const { maxIncomingBitrate, initialAvailableOutgoingBitrate, listenInfos } = this.webRtcTransport;
|
||||
|
||||
const transport = await this.router.createWebRtcTransport({
|
||||
const webRtcTransportOptions = {
|
||||
listenInfos: listenInfos,
|
||||
enableUdp: true,
|
||||
enableTcp: true,
|
||||
preferUdp: true,
|
||||
iceConsentTimeout: 20,
|
||||
initialAvailableOutgoingBitrate,
|
||||
});
|
||||
};
|
||||
|
||||
if (this.webRtcServerActive) webRtcTransportOptions.webRtcServer = this.webRtcServer;
|
||||
|
||||
const transport = await this.router.createWebRtcTransport(webRtcTransportOptions);
|
||||
|
||||
if (!transport) {
|
||||
throw new Error('Failed to create WebRTC transport');
|
||||
|
||||
@@ -41,7 +41,7 @@ dependencies: {
|
||||
* @license For commercial or closed source, contact us at license.mirotalk@gmail.com or purchase directly via CodeCanyon
|
||||
* @license CodeCanyon: https://codecanyon.net/item/mirotalk-sfu-webrtc-realtime-video-conferences/40769970
|
||||
* @author Miroslav Pejic - miroslav.pejic.85@gmail.com
|
||||
* @version 1.3.88
|
||||
* @version 1.3.90
|
||||
*
|
||||
*/
|
||||
|
||||
@@ -762,6 +762,26 @@ function startServer() {
|
||||
setTimeout(() => process.exit(1), 2000);
|
||||
});
|
||||
workers.push(worker);
|
||||
|
||||
if (config.mediasoup.webRtcServerActive) {
|
||||
//
|
||||
log.debug('Create a WebRtcServer', { worker_pid: worker.pid });
|
||||
const webRtcServerOptions = clone(config.mediasoup.webRtcServerOptions);
|
||||
const portIncrement = numWorkers.length - 1;
|
||||
for (const listenInfo of webRtcServerOptions.listenInfos) {
|
||||
listenInfo.port += portIncrement;
|
||||
}
|
||||
const webRtcServer = await worker.createWebRtcServer(webRtcServerOptions);
|
||||
worker.appData.webRtcServer = webRtcServer;
|
||||
}
|
||||
/*
|
||||
setInterval(async () => {
|
||||
const usage = await worker.getResourceUsage();
|
||||
log.info('mediasoup Worker resource usage', { worker_pid: worker.pid, usage: usage });
|
||||
const dump = await worker.dump();
|
||||
log.info('mediasoup Worker dump', { worker_pid: worker.pid, dump: dump });
|
||||
}, 120000);
|
||||
*/
|
||||
}
|
||||
}
|
||||
|
||||
@@ -1691,6 +1711,13 @@ function startServer() {
|
||||
}
|
||||
});
|
||||
|
||||
function clone(value) {
|
||||
if (value === undefined) return undefined;
|
||||
if (Number.isNaN(value)) return NaN;
|
||||
if (typeof structuredClone === 'function') return structuredClone(value);
|
||||
return JSON.parse(JSON.stringify(value));
|
||||
}
|
||||
|
||||
async function isPeerPresenter(room_id, peer_id, peer_name, peer_uuid) {
|
||||
try {
|
||||
if (
|
||||
|
||||
@@ -356,18 +356,26 @@ module.exports = {
|
||||
},
|
||||
],
|
||||
},
|
||||
// WebRtcTransport settings
|
||||
// WebRtcServerOptions
|
||||
webRtcServerActive: false,
|
||||
webRtcServerOptions: {
|
||||
listenInfos: [
|
||||
{ protocol: 'udp', ip: '0.0.0.0', announcedAddress: getLocalIp(), port: 44444 },
|
||||
{ protocol: 'tcp', ip: '0.0.0.0', announcedAddress: getLocalIp(), port: 44444 },
|
||||
],
|
||||
},
|
||||
// WebRtcTransport
|
||||
webRtcTransport: {
|
||||
listenInfos: [
|
||||
{ protocol: 'udp', ip: '0.0.0.0', announcedAddress: getLocalIp() },
|
||||
{ protocol: 'tcp', ip: '0.0.0.0', announcedAddress: getLocalIp() },
|
||||
//announcedAddress: replace by 'public static IPV4 address' https://api.ipify.org (type string --> 'xx.xxx.xxx.xx' not xx.xxx.xxx.xx)
|
||||
//announcedAddress: '' will be auto-detected on server start, for docker localPC set '127.0.0.1' otherwise the 'public static IPV4 address'
|
||||
],
|
||||
initialAvailableOutgoingBitrate: 1000000,
|
||||
minimumAvailableOutgoingBitrate: 600000,
|
||||
maxSctpMessageSize: 262144,
|
||||
maxIncomingBitrate: 1500000,
|
||||
},
|
||||
//announcedAddress: replace by 'public static IPV4 address' https://api.ipify.org (type string --> 'xx.xxx.xxx.xx' not xx.xxx.xxx.xx)
|
||||
//announcedAddress: '' will be auto-detected on server start, for docker localPC set '127.0.0.1' otherwise the 'public static IPV4 address'
|
||||
},
|
||||
};
|
||||
|
||||
@@ -1,6 +1,6 @@
|
||||
{
|
||||
"name": "mirotalksfu",
|
||||
"version": "1.3.88",
|
||||
"version": "1.3.90",
|
||||
"description": "WebRTC SFU browser-based video calls",
|
||||
"main": "Server.js",
|
||||
"scripts": {
|
||||
|
||||
@@ -11,7 +11,7 @@ if (location.href.substr(0, 5) !== 'https') location.href = 'https' + location.h
|
||||
* @license For commercial or closed source, contact us at license.mirotalk@gmail.com or purchase directly via CodeCanyon
|
||||
* @license CodeCanyon: https://codecanyon.net/item/mirotalk-sfu-webrtc-realtime-video-conferences/40769970
|
||||
* @author Miroslav Pejic - miroslav.pejic.85@gmail.com
|
||||
* @version 1.3.88
|
||||
* @version 1.3.90
|
||||
*
|
||||
*/
|
||||
|
||||
|
||||
@@ -9,7 +9,7 @@
|
||||
* @license For commercial or closed source, contact us at license.mirotalk@gmail.com or purchase directly via CodeCanyon
|
||||
* @license CodeCanyon: https://codecanyon.net/item/mirotalk-sfu-webrtc-realtime-video-conferences/40769970
|
||||
* @author Miroslav Pejic - miroslav.pejic.85@gmail.com
|
||||
* @version 1.3.88
|
||||
* @version 1.3.90
|
||||
*
|
||||
*/
|
||||
|
||||
|
||||
المرجع في مشكلة جديدة
حظر مستخدم