No video output + Chat unavailable

After succeed to make mediasoup-demo works locally, i started a visio-conferency with other peers, but i can only see my video output and not others, how can i resolve this please ?

Am also having an issue with “Chat unavailable”, is there a configuration, i missed ? is there a way to make it work please ?
This is my config.js :

/**
 * IMPORTANT (PLEASE READ THIS):
 *
 * This is not the "configuration file" of mediasoup. This is the configuration
 * file of the mediasoup-demo app. mediasoup itself is a server-side library, it
 * does not read any "configuration file". Instead it exposes an API. This demo
 * application just reads settings from this file (once copied to config.js) and
 * calls the mediasoup API with those settings when appropriate.
 */

const os = require('os');

module.exports =
{
	// Listening hostname (just for `gulp live` task).
	domain : process.env.DOMAIN || 'localhost',
	// 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 || `${__dirname}/certs/RootCA.pem`,
			key  : process.env.HTTPS_CERT_PRIVKEY || `${__dirname}/certs/RootCA.key`
		}
	},
	// 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 : 'warn',
			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 PlainRtpTransport options for legacy RTP endpoints (FFmpeg,
		// GStreamer).
		// See https://mediasoup.org/documentation/v3/mediasoup/api/#PlainRtpTransportOptions
		plainRtpTransportOptions :
		{
			listenIp :
			{
				ip          : process.env.MEDIASOUP_LISTEN_IP || '0.0.0.0',
				announcedIp : process.env.MEDIASOUP_ANNOUNCED_IP
			},
			maxSctpMessageSize : 262144
		}
	}
};

You can not use listenIp.ip = "0.0.0.0". It must be a specific bindable IP in your host.

BTW are you reading the mediasoup documentation?

I have the same problem. Iam using 127.0.0.1 as listenIp.ip.
please guid

Don’t use 127.0.0.1.

now iam useing 192.168.0.111 but same issue.

This is how my code looks now. is there any thing else thay iam missing

/**
 * IMPORTANT (PLEASE READ THIS):
 *
 * This is not the "configuration file" of mediasoup. This is the configuration
 * file of the mediasoup-demo app. mediasoup itself is a server-side library, it
 * does not read any "configuration file". Instead it exposes an API. This demo
 * application just reads settings from this file (once copied to config.js) and
 * calls the mediasoup API with those settings when appropriate.
 */

const os = require('os');

module.exports =
{
	// Listening hostname (just for `gulp live` task).
	domain : process.env.DOMAIN || 'localhost',
	// Signaling settings (protoo WebSocket server and HTTP API server).
	https  :
	{
		listenIp   : '192.168.0.111',
		// NOTE: Don't change listenPort (client app assumes 4443).
		listenPort : process.env.PROTOO_LISTEN_PORT || 4442,
		// NOTE: Set your own valid certificate files.
		tls        :
		{
			cert : process.env.HTTPS_CERT_FULLCHAIN || `/etc/ssl/certs/nginx-selfsigned.crt`,
			key  : process.env.HTTPS_CERT_PRIVKEY || `/etc/ssl/private/nginx-selfsigned.key`
		}
	},
	// 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 : 'warn',
			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 || '192.168.0.111',
					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 || '192.168.0.111',
				announcedIp : process.env.MEDIASOUP_ANNOUNCED_IP
			},
			maxSctpMessageSize : 262144
		}
	}
};```

Do your homeworks and check your network, sorry.

if it was so straight then i wont have asked in forum.
if any one else can help. please guide.

I don’t think you have even read the comment above the setting you have changed. And I don’t think you have any issue with the WebRTC layer but with the WebSocket layer of the mediasoup-demo (because you have set a different port in client and server).

i ahve proxy passed 4443 to 4442 with nginx. its connecting.
webscoket is workign fine.
but there is log after sometime in console.

mediasoup-client:Transport connection state changed to disconnected 

iam able to join room, change name, and changed name is reflecting on other device.

Yes, you have ICE connection problems. Check the forum, there are a recent topic about that.

TIP: you need to learn mediasoup and not just the demo project.