default.example.yml

kurento:
  - ip: 0.0.0.0
    url: ws://127.0.0.1:8888/kurento
    # mediaType: (main|audio|content)
    ipClassMappings:
      local:
      private:
      public:
    options:
      failAfter: 5
      request_timeout: 30000
      response_timeout: 30000
# Number of attemps of connecting to the configured kurento instances the first
# time. Infinity means it tries forever until it's able to connect. Default is Infinity.
kurentoStartupRetries: Infinity
# Whether to allow Kurento to process mDNS ICE candidates
kurentoAllowMDNSCandidates: false
# Whether to track KMS's ICE state changes for every peer.
# Disabled by default for now until we trace the perf. impact of it
kurentoTrackIceStateChanges: false
# balancing-strategy: can be either ROUND_ROBIN or MEDIA_TYPE. The MEDIA_TYPE only
# works properly if you annotated the configured kurento instances in the
# 'kurento' config parameter with a mediaType field (main|audio|content) which
# will specify what will each server handle. Default is ROUND_ROBIN
balancing-strategy: ROUND_ROBIN
# Allow media offloading to Kurento instances with mismatched media types
kurentoAllowMediaTypeMix: false
video-transposing-ceiling: 50
audio-transposing-ceiling: 100
acceptSelfSignedCertificate: false
redisHost: 127.0.0.1
redisPort: "6379"
# Uncomment to set a password if Redis requires auth.
# redisPassword: foobared
clientPort: "3008"
clientHost: 127.0.0.1
# Whether RPCs and WS conns should be rejected/shutdown on user info header
# anomalies
wsStrictHeaderParsing: true
# Time (ms) between server-initiated ping-pong checks
wsHeartbeatInterval: 20000
# wsServerOptions: see https://github.com/websockets/ws/blob/HEAD/doc/ws.md#new-websocketserveroptions-callback).
# It's a passthrough mapping of all config literal (strings, booleans, numbers).
wsServerOptions:
  maxPayload: 51200

mediaFlowTimeoutDuration: 30000
mediaStateTimeoutDuration: 30000
requestQueueTimeout: 30000
screenshareSubscriberSpecSlave: false
screensharePlayStartEnabled: true
screenshareServerSideAkkaBroadcast: true
screenshareIgnoreMediaThresholds: false
audioIgnoreMediaThresholds: true
videoIgnoreMediaThresholds: false
videoSubscriberSpecSlave: false

# Following three configurations: see media-server-adapters config
# to get a list of implemented adapters. The configuration values
# here are the adapter's name of choice
videoMediaServer: mediasoup
screenshareMediaServer: mediasoup
audioMediaServer: mediasoup
fullAudioEnabled: true
transparentListenOnly: true
dynamicGlobalAudio: false
fullAudioProxyActiveDirection: true
audioIceRestartEnabled: false
# RTP|WebRTC. Dictates how the FS <-> [audioMediaServer] bridge is established:
# via plain RTP or via a WebRTC-backed RTP stream
# WebRTC should be used for environments where the bridge is between external
# servers AND when mediasoup is being used
fsBridgeMode: 'RTP'
# pliOnConnect: whether to send a PLI when a new subscriber joins
#  amount: max amount of PLIs to send. If 0, it won't send any PLIs (default)
#  interval: interval between PLIs (in ms)
#  (e.g.: amount: 3, interval: 2000 => send 3 PLIs every 2 seconds)
pliOnConnect:
  amount: 0
  interval: 2000
# Whether to record screen raw files
recordScreenSharing: true
# Whether to record camera raw files
recordWebcams: true
# Fire a max of 3 PLIs over 6 seconds if recording stops flowing for 500 ms
recordingPliOnNotFlowing: false
# recordingBasePath (DEPRECATED): Base path where recording raw files will be stored
# WARNING => THE FOLLOWING PARROT CEASED TO BE
recordingBasePath: /var/kurento
# recordingAdapter: Adapter to be used to record live streams.
# Possible values:
#   - 'native': same adapter as the source's (eg.: Kurento -> Kurento)
#   - Kurento: use Kurento for recording (eg.: mediasoup -> Kurento)
#   - bbb-webrtc-recorder: use bbb-webrtc-recorder
#     (e.g.: mediasoup -> bbb-webrtc-recorder). EXPERIMENTAL
recordingAdapter: Kurento
# Adapter to be used if #recordingAdapter fails and #recordingUseFallbackAdapter is true
recordingFallbackAdapter: bbb-webrtc-recorder
# Whether to use the fallback adapter if the main adapter fails
recordingFallbackOnFailure: false
# Whether a recording copy should be created with #recordingFallbackAdapter
# regardless of whether the main adapter fails or not
# This is a development/test feature and should not be used in production
# environments. The generated copies will have a -copy suffix in their names,
# won't have any recording metadata registered and should not be processed
recordingCarbonCopy: false
# Recording dry-run mode: always record regardless of meeting state and
# *do not* register any recording metadata.
# This is a development/test feature and should not be used in production envs.
recordingDryRun: false
# The base recording profile for video and screen sharing
# WEBM_VIDEO_ONLY|WEBM_AUDIO_ONLY|WEBM_FULL
recordingMediaProfile: WEBM_VIDEO_ONLY
# The base recording container format.
# webm|mp4
recordingFormat: webm
# bbb-webrtc-recorder integration configs
bbbWebrtcRecorder:
  channels:
    # POV: bbb-webrtc-sfu
    subscribe: from-bbb-webrtc-recorder
    publish: to-bbb-webrtc-recorder
  heartbeatInterval: 5000
  heartbeatDelay: 1000
  connectionTimeout: 5000

# Recording event keys expiry time in Redis
# 14 days is the default
redisExpireTime: 1209600

# Time to wait before ejecting an user which has the autoLeave flag marked
# as true and has no more media sessions under its ownership
mcsUserEjectionTimer: 60000
# Time to wait before destroying a room (on all scenarios)
mcsRoomDestructionTimer: 90000
# mcs-core entrypoint configured on nginx
mcs-path: /mcs
mcs-port: 3010
mcs-host: 127.0.0.1
mcs-address: localhost
mcs-ws-timeout: 30000
freeswitch:
  ip: 127.0.0.1
  sip_ip: 127.0.0.1
  port: 5066
  esl_ip: 127.0.0.1
  esl_port: 8021
  #esl_password: FOOBAR
  handleExternalConnections: false
  ipClassMappings:
    local:
    private:
    public:
log:
  # trace|debug|info|warn|error
  level: info
  # Whether to log to stdout
  stdout: true
  # Whether to log to a file
  file: true
  filename: /var/log/bbb-webrtc-sfu/bbb-webrtc-sfu.log
modules:
- path: ./lib/mcs-core/process.js
  name: core
  dedicated: true
  # IPC can be either native|redis right now. Defaults to native
  ipc:
    mode: native
    options:
      # inboundChannel: to-mcs-core
      # outboundChannel: from-mcs-core
- path: ./lib/screenshare/ScreenshareProcess.js
  name: screenshare
  dedicated: true
  ipc:
    mode: native
    options:
      # inboundChannel: to-sfu-screenshare
      # outboundChannel: from-sfu-screenshare
- path: ./lib/video/VideoProcess.js
  name: video
  dedicated: true
  ipc:
    mode: native
    options:
      # inboundChannel: to-sfu-video
      # outboundChannel: from-sfu-video
- path: ./lib/audio/audio-process.js
  name: fullaudio
  routingAliases:
    - audio
  dedicated: true
  ipc:
    mode: native
    options:
      # inboundChannel: to-sfu-audio
      # outboundChannel: from-sfu-audio

# media-server-adapters: path could refer to a npm module
media-server-adapters:
- path: kurento/kurento.js
  name: Kurento
- path: freeswitch/freeswitch.js
  name: Freeswitch
- path: mediasoup/mediasoup-adapter.js
  name: mediasoup

# strategies: path could refer to a npm module
strategies:
  #- path: ./voice-switching-handler.js
  # name: VOICE_SWITCHING

# Codec values (codec_video_main, codec_video_content, codec_audio):
# it'll force the specified codec.
# Put the 'ANY' value to prevent forcing a specific codec without the _priority options.
# For the 'ANY' option, there's a trio of optional configuration where you are
# able to specify the priority codec to force in a lenient way (codec_*_priority).
# If codec_*_priority is undefined or not found in the media description,
# no specific codec will be forced (base ANY behaviour).
# If the priority codec is found in the media description, any codec
# that doesn't match it will be filtered out (ie it will act as if the priority
# codec was forced).
# If you wanna add other codecs (e.g.: VP9)
# add it to the list with its specifiers. If the codec isn't in the list,
# it'll default to the specifier values in codec_*_priority OR VP8,
# but still force the entered codec option.
# P.S.: _video_main refers to webcam/main video content, _video_content refers to
# screenshare/content:slides streams.
conference-media-specs:
  codec_video_main: VP8
  codec_video_main_priority: VP8
  codec_video_content: VP8
  codec_video_content_priority: VP8
  codec_audio: ANY
  codec_audio_priority: OPUS
  H264:
    profile_level_id: "42e01f"
    packetization_mode: "1"
    level_asymmetry_allowed: "1"
    tias_main: "300000"
    as_main: "300"
    tias_content: "1500000"
    as_content: "1500"
    max_mbps_main: "0"
    max_fs_main: "0"
    max_br_main: "0"
    max_mbps_content: "0"
    max_fs_content: "12288"
    max_br_content: "0"
  VP8:
    tias_main: "300000"
    as_main: "300"
    tias_content: "1500000"
    as_content: "1500"
  OPUS:
    useinbandfec: "1"
    maxaveragebitrate: "30000"
    maxplaybackrate: "48000"
    ptime: "20"
    minptime: "10"
    maxptime: "40"
kurentoRembParams:
  rembOnConnect: 300
  upLosses: 12
  decrementFactor: 0.5
  thresholdFactor: 0.8
# Munges/removes the SDP REMB rtcp-fb support attribute (ie tries to disable REMB)
kurentoRemoveRembRtcpFb: false
# kurentoAllowedCandidateIps: optional configuration. List of VALID IPs to be used
# to define valid outbound ICE candidates.
# This is a short-term optimization to reduce the number of candidates sent to
# the client by filtering out anything that isn't in this list
kurentoAllowedCandidateIps:
  #- <ipv4|ipv6>
# mediaThresholds: mandatory configuration. Establishes type-agnostic media thresholds
# that when hit will make the server refuse to negotiate new medias.
# Any attemps to inject medias past the thresholds will return an error with code
# 2002 and message MEDIA_SERVER_NO_RESOURCES.
# The threshold priority order is global -> perRoom -> perUser. Value 0 means unlimited
# (default). An optional API parameter may be sent on pub/sub calls (ignoreThresholds)
# to make the media ignore the configured thresholds. This is used here for recording
# and audio medias.
mediaThresholds:
  global: 0
  perRoom: 0
  perUser: 0
# Whether to allow duplicate externalUserId's on join. If allowed, join requests with
# a externalUserId specified in the optional parameters will associate the new connection
# with a pre-existing user (if it exists), or create a new one.
# This will create a new virtual link between the request
# and the pre-existing user with a unique internal user ID. The user will be
# cleared from the system when a leave call for it is sent OR it has the autoLeave
# optional parameter set to true and all of its medias were cleared
allowDuplicateExtUserId: true
ejectOnUserLeft: true
permissionProbes: true
destroyRoomOnEject: true

# Direct Prometheus instrumentation. Disabled by default.
prometheus:
  enabled: false
  # mcs-core specific metrics, top level of prometheus dictionary for now
  host: localhost
  port: 3014
  path: '/metrics'
  collectDefaultMetrics: false
  # Main process metrics endpoint (main == websocket entrypoint, module manager)
  main:
    host: localhost
    port: 3016
    path: '/metrics'
    collectDefaultMetrics: false
  # Video process metrics endpoint (video == webcam req handler, ...)
  video:
    host: localhost
    port: 3026
    path: '/metrics'
    collectDefaultMetrics: false
  screenshare:
    host: localhost
    port: 3022
    path: '/metrics'
    collectDefaultMetrics: false
  audio:
    host: localhost
    port: 3024
    path: '/metrics'
    collectDefaultMetrics: false
    holdHistograms: false

mediasoup:
  # mediasoup.workers: "auto"|"cores"|<Number>.
  #   * The number of mediasoup workers for general use (media type agnostic, shared pool).
  #   * "auto" is: ceil((min(nproc,32) * 0.8) + (max(0, nproc - 32) / 2))
  #   * "cores" is: nproc (the host's core count as in os.cpus().length)
  #   * Use a number if you want to override the number of workers.
  #   * Strings other than "auto" and "cores" are ignored :).
  #   * The default value is "auto".
  workers: "auto"
  # mediasoup.dedicatedMediaTypeWorkers:
  #   audio: "auto"|"cores"|<Number>
  #   main: "auto"|"cores"|<Number>
  #   content: "auto"|"cores"|<Number>
  #
  #   * The number of mediasoup workers to use for a specific media type.
  #     They are SEPARATED from the shared pool (mediasoup.workers). Streams
  #     of a specific media type will always land in its dedicated pool if one is set.
  #   * "auto" and Number semantics are the same as #mediasoup.workers.
  dedicatedMediaTypeWorkers:
    audio: 0
    main: 0
    content: 0
  # mediasoup.enableWorkerTransposing: whether to enable worker transposing
  # (ie: the ability to move a media stream from one worker to another).
  enableWorkerTransposing: false
  # mediasoup.workerBalancing: defines the strategy to distribute mediasoup
  # elements (transports, producers, consumers) among workers.
  workerBalancing:
    # strategy is one of:
    #   - round-robin: (default) Distribute *producers* in a round-robin fashion.
    #   - least-loaded: Distribute *producers* to the least loaded worker.
    strategy: round-robin
  # mediasoup.workerPriorities:
  #   audio: <os.constants.priority> | [19,-20]
  #   main: <os.constants.priority> | [19,-20]
  #   content: <os.constants.priority> | [19,-20]
  #
  #   * Sets the NICE level for each worker type. The app needs the correct user
  #   caps to be effective
  #
  #workerPriorities:
  #  audio: PRIORITY_HIGH
  #  main: PRIORITY_NORMAL
  #  content: PRIORITY_LOW
  worker:
    # "debug", "warn", "error", "none"
    logLevel: "error"
    # + svc, simulcast, srtp, ...
    logTags:
      - "bwe"
      - "dtls"
      - "ice"
      - "info"
      - "rtcp"
      - "rtp"
      - "rtx"
      - "srtp"
    rtcMinPort: 24577
    rtcMaxPort: 32768
    #dtlsCertificateFile:
    #dtlsPrivateKeyFile:
  promExportWorkerResourceUsage: false
  promExportRtpScores: false
  router:
    # TODO fetch from conference-media-specs
    mediaCodecs:
      - kind: "audio"
        mimeType: "audio/opus"
        preferredPayloadType: 111
        clockRate: 48000
        channels: 2
        parameters:
          maxaveragebitrate: 30000
          maxplaybackrate: 48000
          ptime: 20
          minptime: 10
          maxptime: 40
          useinbandfec: 1
        rtcpFeedback:
          - type: "transport-cc"
      - kind: "audio"
        mimeType: "audio/telephone-event"
        preferredPayloadType: 101
        clockRate: 8000
      - kind: "audio"
        mimeType: "audio/PCMU"
        preferredPayloadType: 0
        clockRate: 8000
        rtcpFeedback:
          - type: "transport-cc"
      - kind: 'audio'
        mimeType: 'audio/ISAC'
        clockRate: 32000
        preferredPayloadType: 103
        rtcpFeedback:
          - type: "transport-cc"
      - kind: "video"
        mimeType: "video/VP8"
        preferredPayloadType: 96
        clockRate: 90000
        rtcpFeedback:
          - type: "transport-cc"
          - type: "goog-remb"
          - type: "ccm"
            parameter: "fir"
          - type: "nack"
          - type: "nack"
            parameter: "pli"
  webrtc:
    # announcedIP => it's the host public IPv4 in case the machine is
    # behind a 1:1 NAT
    # ip => the address on which mediasoup will bind its RTC sockets
    listenIps:
      - ip: "0.0.0.0"
        announcedIp: "127.0.0.1"
      # - ip: <some-ipv6>
    # equivalent to Kurento's/libnice's ice-udp/niceAgentIceUdp
    enableUdp: true
    # equivalent to Kurento's/libnice's ice-tcp/niceAgentIceTcp
    enableTcp: false
    # candidate priority mangling; makes sense to keep this enabled
    # even more when ice-tcp is enabled as well
    preferUdp: true
    # iceConsentTimeout: ICE consent timeout (in seconds). If 0 it is disabled.
    iceConsentTimeout: 30
  plainRtp:
    # announcedIP => it's the host public IPv4 in case the machine is
    # behind a 1:1 NAT
    # ip => the address on which mediasoup will bind its plain RTP sockets
    # This is pretty similar in intent to webrtc.listenIps with two diffs:
    #   - only applies to plain transports (RTP)
    #   - it's not an array, so just one option
    listenIp:
      ip: "0.0.0.0"
      announcedIp: "127.0.0.1"
    # Whether RTCP should use RTPs transport.
    # False probably doesn't work yet in the adapter
    rtcpMux: true
    # Port inference through COMEDIA
    comedia: true
  bitrate:
    # Default per transport max incoming bitrate (bps)
    defaultMaxIncomingBitrate: 1500000
    # Default per transport max outgoing bitrate (bps)
    defaultMaxOutgoingBitrate: 1500000
    # Default per transport initial available outgoing bitrate (bps)
    defaultInitialAvailableOutgoingBitrate: 600000
  webRtcHeaderExts:
    - uri: 'urn:ietf:params:rtp-hdrext:toffset'
      id: 1
    - uri: 'http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time'
      id: 2
    - uri: 'http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01'
      id: 4
    - uri: 'urn:ietf:params:rtp-hdrext:sdes:mid'
      id: 9
    - uri: 'urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id'
      id: 10
    - uri: 'urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id'
      id: 11
  recorder:
    ffmpeg:
      # UDP, internal only. Doesn't need to be open in the firewall, but should
      # be a dedicated range.
      recMinPort: 50000
      recMaxPort: 51000
      inputOptions:
        - "-protocol_whitelist file,rtp,udp"
        - "-probesize 1000000"
        - "-analyzeduration 1000000"
      outputOptions:
        - '-f webm'
        - '-flags +global_header'
livekit:
  tokenTTL: 3600
  key: LIVEKIT_KEY
  secret: LIVEKIT_SECRET