No video or audio in the demo app

mediasoup-demo
1

Here is my config file:

const os = require('os');

module.exports =
{
	// Listening hostname (just for `gulp live` task).
	domain: process.env.DOMAIN || 'devmedia1.domain.io',
	// Signaling settings (protoo WebSocket server and HTTP API server).
	https:
	{
		listenIp: '0.0.0.0',
		// NOTE: Don't change listenPort (client app assumes 4443).
		listenPort: process.env.PROTOO_LISTEN_PORT || 4443,
		// NOTE: Set your own valid certificate files.
		tls:
		{
			cert: process.env.HTTPS_CERT_FULLCHAIN || `${__dirname}/../data/letsencrypt/live/devmedia1.domain.io/fullchain.pem`,
			key: process.env.HTTPS_CERT_PRIVKEY || `${__dirname}/../data/letsencrypt/live/devmedia1.domain.io/privkey.pem`,
		}
	},
	// mediasoup settings.
	mediasoup:
	{
		// Number of mediasoup workers to launch.
		numWorkers: Object.keys(os.cpus()).length,
		// mediasoup WorkerSettings.
		// See https://mediasoup.org/documentation/v3/mediasoup/api/#WorkerSettings
		workerSettings:
		{
			logLevel: 'debug',
			logTags:
				[
					'info',
					'ice',
					'dtls',
					'rtp',
					'srtp',
					'rtcp',
					'rtx',
					'bwe',
					'score',
					'simulcast',
					'svc',
					'sctp'
				],
			rtcMinPort: process.env.MEDIASOUP_MIN_PORT || 40000,
			rtcMaxPort: process.env.MEDIASOUP_MAX_PORT || 49999
		},
		// mediasoup Router options.
		// See https://mediasoup.org/documentation/v3/mediasoup/api/#RouterOptions
		routerOptions:
		{
			mediaCodecs:
				[
					{
						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/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
					// 	}
					// }
				]
		},
		// mediasoup WebRtcTransport options for WebRTC endpoints (mediasoup-client,
		// libmediasoupclient).
		// See https://mediasoup.org/documentation/v3/mediasoup/api/#WebRtcTransportOptions
		webRtcTransportOptions:
		{
			listenIps:
				[
					{
						ip: process.env.MEDIASOUP_LISTEN_IP || '0.0.0.0',
						announcedIp: process.env.MEDIASOUP_ANNOUNCED_IP
					}
				],
			initialAvailableOutgoingBitrate: 1000000,
			minimumAvailableOutgoingBitrate: 600000,
			maxSctpMessageSize: 262144,
			// Additional options that are not part of WebRtcTransportOptions.
			maxIncomingBitrate: 1500000
		},
		// mediasoup PlainTransport options for legacy RTP endpoints (FFmpeg,
		// GStreamer).
		// See https://mediasoup.org/documentation/v3/mediasoup/api/#PlainTransportOptions
		plainTransportOptions:
		{
			listenIp:
			{
				// ip          : process.env.MEDIASOUP_LISTEN_IP || '1.2.3.4',
				ip: process.env.MEDIASOUP_LISTEN_IP || '0.0.0.0',
				announcedIp: process.env.MEDIASOUP_ANNOUNCED_IP
			},
			maxSctpMessageSize: 262144
		}
	}
};

The listening port and the tls certificates seem to be fine, since I could access the app via https and join a room. The problem is that remote video/audio streams don’t show up. Any idea why?

I’m also wondering where to put the turn server urls?

Thanks in advance.