Mediasoup-demo部署后无法看到对方的视频

来源:13-2 Mediasoup Demo的布署

madzheng

2020-04-28

在云主机上部署了Mediasoup-demo后,可以访问,并且加入房间,但是却无法看到对方的视频画面。如下图所示:图片描述
请问这种情况是哪里出问题了?

不知道是不是config.js设置出错了,config.js的内容如下:
/**

  • 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 : ‘webrtc.maizucity.com’,
// 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 : 4443,
// NOTE: Set your own valid certificate files.
tls :
{
cert : ${__dirname}/certs/3736461_webrtc.maizucity.com.pem,
key : ${__dirname}/certs/3736461_webrtc.maizucity.com.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 : 40000,
rtcMaxPort : 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 || ‘0.0.0.0’,
announcedIp : process.env.MEDIASOUP_ANNOUNCED_IP
},
maxSctpMessageSize : 262144
}
}
};

写回答

1回答

李超

2020-04-28

Mac 电脑?用safsri 浏览器试试?

0
5
幕布斯7419438
回复
madzheng
我也遇到了同样的问题,是把那个地址改了啊可否指导下
2020-08-31
共5条回复

百万级高并发WebRTC流媒体服务器设计与开发

备战5G音视频时代,抢先掌握服务端高级技能

2071 学习 · 375 问题

查看课程