mie-webconf-app/server/webrtc-transport.yaml

This is a mediasoup based web conferencing app with socket.io for signalling

# MediaSoup config for both Worker and WebRtcTransport

worker:
  logLevel: warn
  rtcMinPort: 10000
  rtcMaxPort: 10100
  # logTags: [info, ice, dtls, rtp, srtp, rtx, bwe, score, simulcast, svc, sctp] # optional

webrtcTransport:
  listenIps:
    - ip: "0.0.0.0"
      announcedIp: "127.0.0.1"   # replace with public IP if needed
  enableUdp: true
  enableTcp: true
  preferUdp: true

  # Optional bitrate tuning
  initialAvailableOutgoingBitrate: 1000000
  minimumAvailableOutgoingBitrate: 300000
  maxIncomingBitrate: 1500000

  # Optional SCTP/datachannel
  # enableSctp: false
  # numSctpStreams:
  #   OS: 1024
  #   MIS: 1024