gharia

# stream-name: {
# type = rtp|live|ondemand|rtsp
#        rtp = stream originated by an external tool (e.g., gstreamer or
#              ffmpeg) and sent to the plugin via RTP
#        live = local file streamed live to multiple listeners
#               (multiple listeners = same streaming context)
#        ondemand = local file streamed on-demand to a single listener
#                   (multiple listeners = different streaming contexts)
#        rtsp = stream originated by an external RTSP feed (only
#               available if libcurl support was compiled)

gharia

# 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)