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janus.plugin.audiobridge.jcfg
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janus.plugin.audiobridge.jcfg
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# room-<unique room ID>: {
# description = "This is my awesome room"
# is_private = true|false (whether this room should be in the public list, default=true)
# secret = "<optional password needed for manipulating (e.g. destroying) the room>"
# pin = "<optional password needed for joining the room>"
# sampling_rate = <sampling rate> (e.g., 16000 for wideband mixing)
# spatial_audio = true|false (if true, the mix will be stereo to spatially place users, default=false)
# audiolevel_ext = true|false (whether the ssrc-audio-level RTP extension must
# be negotiated/used or not for new joins, default=true)
# audiolevel_event = true|false (whether to emit event to other users or not, default=false)
# audio_active_packets = 100 (number of packets with audio level, default=100, 2 seconds)
# audio_level_average = 25 (average value of audio level, 127=muted, 0='too loud', default=25)
# default_prebuffering = number of packets to buffer before decoding each particiant (default=6)
# record = true|false (whether this room should be recorded, default=false)
# record_file = "/path/to/recording.wav" (where to save the recording)
# record_dir = "/path/to/" (path to save the recording to, makes record_file a relative path if provided)
# allow_rtp_participants = true|false (whether participants should be allowed to join
# via plain RTP as well, rather than just WebRTC, default=false)
#
# The following lines are only needed if you want the mixed audio
# to be automatically forwarded via plain RTP to an external component
# (e.g., an ffmpeg script, or a gstreamer pipeline) for processing
# By default plain RTP is used, SRTP must be configured if needed
# rtp_forward_id = numeric RTP forwarder ID for referencing it via API (optional: random ID used if missing)
# rtp_forward_host = "<host address to forward RTP packets of mixed audio to>"
# rtp_forward_host_family = "<ipv4|ipv6; by default, first family returned by DNS request>"
# rtp_forward_port = port to forward RTP packets of mixed audio to
# rtp_forward_ssrc = SSRC to use to use when streaming (optional: stream_id used if missing)
# rtp_forward_codec = opus (default), pcma (A-Law) or pcmu (mu-Law)
# rtp_forward_ptype = payload type to use when streaming (optional: only read for Opus, 100 used if missing)
# rtp_forward_srtp_suite = length of authentication tag (32 or 80)
# rtp_forward_srtp_crypto = "<key to use as crypto (base64 encoded key as in SDES)>"
# rtp_forward_always_on = true|false, whether silence should be forwarded when the room is empty (optional: false used if missing)
#}
general: {
#admin_key = "supersecret" # If set, rooms can be created via API only
# if this key is provided in the request
#lock_rtp_forward = true # Whether the admin_key above should be
# enforced for RTP forwarding requests too
#lock_play_file = true # Whether the admin_key above should be
# enforced for playing .opus files too
#record_tmp_ext = "tmp" # Optional temporary extension to add to filenames
# while recording: e.g., setting "tmp" would mean
# .wav --> .wav.tmp until the file is closed
#events = false # Whether events should be sent to event
# handlers (default=true)
# By default, integers are used as a unique ID for both rooms and participants.
# In case you want to use strings instead (e.g., a UUID), set string_ids to true.
#string_ids = true
# Normally, all AudioBridge participants will join by negotiating a WebRTC
# PeerConnection: the plugin also supports adding participants that will
# use plain RTP, though, be it for supporting legacy users (e.g., SIP
# participants who an orchestrator can add to the bridge) or more simply
# to temporarily inject external audio in a room from a live source. To
# support plain RTP, the plugin needs to have a range of ports it can bind
# to: notice this should be configured so that it doesn't conflict with other
# plugins (e.g., Streaming, SIP, NoSIP) and applications (e.g., Janus itself).
# The default if you don't specify anything is 10000-60000.
#rtp_port_range = "50000-60000"
# In case we need to support plain RTP participants, we'll also need to know
# what local IP address to bind to for media. If no address is set in the
# property below, then one will be automatically guessed from the system.
#local_ip = "1.2.3.4"
}
room-1234: {
description = "Demo Room"
secret = "adminpwd"
sampling_rate = 16000
record = false
#record_dir = "/path/to/"
#record_file = "recording.wav"
}