最初我部署在虚拟机里，发现两个浏览器端进入同一个房间，可以看到对方这个端，但是视频画面是默认图标，分辨率显示0x0。效果如下图：

不断排查，也怀疑过是不是虚拟机隔离或者防火墙的原因。在物理机Windows上部署了mediasoup-demo，还是一样，排除虚拟机隔离或者防火墙的原因。

（参考了：Windows搭建Nodejs【全网最详细】https://blog.csdn.net/weixin_43409994/article/details/124472220windows mediasoup-demo 公网部署 解决部署后无法看到对方的视频https://www.cnblogs.com/jiayouba/p/15232423.html）

最后发现，config.js里webRtcTransportOptions的ip，绝对不要填0.0.0.0和127.0.0.1。在mediasoup的文档里面原来是有写的，有很多文章说0.0.0.0可以用，但是最新版（到2022年2月9日）可能已经不行了。

搜索WebRtcTransportOptions查看表格下方蓝色边框的注释说明https://mediasoup.org/documentation/v3/mediasoup/api/#WorkerSettings而且127.0.0.1也不行，必须是一个真实网卡的IP地址，才能正常绑定。server起来之后，客户端或者网页如果跟server在一台机器，可以用127.0.0.1访问服务。

给出一个正确的config.js的示例

/**
 * 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 || `${__dirname}/certs/fullchain.pem`,
			key  : process.env.HTTPS_CERT_PRIVKEY || `${__dirname}/certs/privkey.pem`
		}
	},
	// mediasoup settings.
	mediasoup :
	{
		// Number of mediasoup workers to launch. Object.keys(os.cpus()).length
		numWorkers     : 1,
		// 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 :
			[
				{
					// '192.168.0.106'是内网IP，不能填0.0.0.0，在本机测试部署的时候也不能填127.0.0.1
					ip          : process.env.MEDIASOUP_LISTEN_IP || '192.168.0.106',
					// 公网IP，目前没用到，看到有下面两种写法，可以自己尝试
					// 1）announcedIp : '10.23.34.56'
					// 2）announcedIp : process.env.MEDIASOUP_ANNOUNCED_IP || '10.23.34.56'
					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 :
			{
				// '192.168.0.106'是内网IP，不能填0.0.0.0，在本机测试部署的时候也不能填127.0.0.1
				ip          : process.env.MEDIASOUP_LISTEN_IP || '192.168.0.106',
				announcedIp : process.env.MEDIASOUP_ANNOUNCED_IP
			},
			maxSctpMessageSize : 262144
		}
	}
};