janus.plugin.audiobridge.jcfg.sample

# 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)
# 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)
#
#     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
}

room-1234: {
	description = "Demo Room"
	secret = "adminpwd"
	sampling_rate = 16000
	record = false
	#record_file = "/path/to/recording.wav"
}