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
}
}
};
