general: {
  #admin_key = "debugstreaming"		# If set, mountpoints can be created via API
                  # only if this key is provided in the request
  rtp_port_range = "20000-21000"	# Range of ports to use for RTP/RTCP when '0' is
                  # passed as port for a mountpoint (default=10000-60000)
  #events = false					# Whether events should be sent to event
                  # handlers (default=true)

  # By default, integers are used as a unique ID for both mountpoints. In case
  # you want to use strings instead (e.g., a UUID), set string_ids to true.
  # string_ids = true
}

# 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)
# id = <unique numeric ID> (if missing, a random one will be generated)
# description = This is my awesome stream
# metadata = An optional string that can contain any metadata (e.g., JSON)
#		   associated with the stream you want users to receive
# is_private = true|false (private streams don't appear when you do a 'list'
#			request)
# secret = <optional password needed for manipulating (e.g., destroying
#			or enabling/disabling) the stream>
# pin = <optional password needed for watching the stream>
# filename = path to the local file to stream (only for live/ondemand)
# audio = true|false (do/don't stream audio)
# video = true|false (do/don't stream video)
#    The following options are only valid for the 'rtp' type:
# data = true|false (do/don't stream text via datachannels)
# audioport = local port for receiving audio frames
# audiortcpport = local port, if any, for receiving and sending audio RTCP feedback
# audiomcast = multicast group port for receiving audio frames, if any
# audioiface = network interface or IP address to bind to, if any (binds to all otherwise)
# audiopt = <audio RTP payload type> (e.g., 111)
# audiortpmap = RTP map of the audio codec (e.g., opus/48000/2)
# audioskew = true|false (whether the plugin should perform skew
#		analisys and compensation on incoming audio RTP stream, EXPERIMENTAL)
# videoport = local port for receiving video frames
# videortcpport = local port, if any, for receiving and sending video RTCP feedback
# videomcast = multicast group port for receiving video frames, if any
# videoiface = network interface or IP address to bind to, if any (binds to all otherwise)
# videopt = <video RTP payload type> (e.g., 100)
# videortpmap = RTP map of the video codec (e.g., VP8/90000)
# videobufferkf = true|false (whether the plugin should store the latest
#		keyframe and send it immediately for new viewers, EXPERIMENTAL)
# videosimulcast = true|false (do|don't enable video simulcasting)
# videoport2 = second local port for receiving video frames (only for rtp, and simulcasting)
# videoport3 = third local port for receiving video frames (only for rtp, and simulcasting)
# videoskew = true|false (whether the plugin should perform skew
#		analisys and compensation on incoming video RTP stream, EXPERIMENTAL)
# videosvc = true|false (whether the video will have SVC support; works only for VP9-SVC, default=false)
# collision = in case of collision (more than one SSRC hitting the same port), the plugin
#		will discard incoming RTP packets with a new SSRC unless this many milliseconds
#		passed, which would then change the current SSRC (0=disabled)
# dataport = local port for receiving data messages to relay
# dataiface = network interface or IP address to bind to, if any (binds to all otherwise)
# datatype = text|binary (type of data this mountpoint will relay, default=text)
# databuffermsg = true|false (whether the plugin should store the latest
#		message and send it immediately for new viewers)
# threads = number of threads to assist with the relaying part, which can help
#		if you expect a lot of viewers that may cause the RTP receiving part
#		in the Streaming plugin to slow down and fail to catch up (default=0)
#
# In case you want to use SRTP for your RTP-based mountpoint, you'll need
# to configure the SRTP-related properties as well, namely the suite to
# use for hashing (32 or 80) and the crypto information for decrypting
# the stream (as a base64 encoded string the way SDES does it). Notice
# that with SRTP involved you'll have to pay extra attention to what you
# feed the mountpoint, as you may risk getting SRTP decrypt errors:
# srtpsuite = 32
# srtpcrypto = WbTBosdVUZqEb6Htqhn+m3z7wUh4RJVR8nE15GbN
#
# The Streaming plugin can also be used to (re)stream media that has been
# encrypted using something that can be consumed via Insertable Streams.
# In that case, we only need to be aware of it, so that we can send the
# info along with the SDP. How to decrypt the media is out of scope, and
# up to the application since, again, this is end-to-end encryption and
# so neither Janus nor the Streaming plugin have access to anything.
# DO NOT SET THIS PROPERTY IF YOU DON'T KNOW WHAT YOU'RE DOING!
# e2ee = true
#
# The following options are only valid for the 'rtsp' type:
# url = RTSP stream URL (only for restreaming RTSP)
# rtsp_user = RTSP authorization username (only if type=rtsp)
# rtsp_pwd = RTSP authorization password (only if type=rtsp)
# rtsp_failcheck = whether an error should be returned if connecting to the RTSP server fails (default=true)
# rtspiface = network interface or IP address to bind to, if any (binds to all otherwise), when receiving RTSP streams
# rtsp_reconnect_delay = after n seconds passed and no media assumed, the RTSP server has gone and schedule a reconnect (default=5s)
# rtsp_session_timeout = by default the streaming plugin will check the RTSP connection with an OPTIONS query,
# 		the value of the timeout comes from the RTSP session initializer and by default
# 		this session timeout is the half of this value In some cases this value can be too high (for example more than one minute)
# 		because of the media server. In that case this plugin will calculate the timeout with this
# 		formula: timeout = min(session_timeout, rtsp_session_timeout / 2). (default=0s)
# rtsp_timeout = communication timeout (CURLOPT_TIMEOUT) for cURL call gathering the RTSP information (default=10s)
# rtsp_conn_timeout = connection timeout for cURL (CURLOPT_CONNECTTIMEOUT) call gathering the RTSP information (default=5s)
#
# Notice that, for 'rtsp' mountpoints, normally the plugin uses the exact
# SDP rtpmap and fmtp attributes the remote camera or RTSP server sent.
# In case the values set remotely are known to conflict with WebRTC viewers,
# you can override both using the settings introduced above.
#
# To test the 'gstreamer-sample' example, check the test_gstreamer.sh
# script in the plugins/streams folder. The live and on-demand  audio
# file streams, use a couple of files (radio.alaw, music.mulaw) that are
# provided in the plugins/streams folder.
#}

#
# All browsers also support H.264, often through Cisco's OpenH264 plugin.
# The only profile that is definitely supported is the baseline one, which
# means that if you try a higher one it might or might not work. No matter
# which profile you encode, though, you can put a custom one in the SDP if
# you override the fmtp SDP attribute via 'videofmtp'. The following is an
# example of how to create a simple H.264 mountpoint: you can feed it via
# an x264enc+rtph264pay pipeline in gstreamer, an ffmpeg script or other.
# https://github.com/meetecho/janus-gateway/blob/master/conf/janus.plugin.streaming.jcfg.sample.in

demo-video-1: {
    type = "rtp"
    id = 001
    description = "Demo Video #1"
    metadata = "{ \"path\": \"/media/demo_video_1.mp4\", \"video_stream_src\": \"File\" }"
    media = (
        {
          type = "video"
          mid = "v1"
          label = "Video stream #1"
          port = 20001
          pt = 96
          rtpmap = "H264/90000"
          videofmtp = "profile-level-id=42e01f;packetization-mode=1"
        }
    )
}

live-camera-1: {
    type = "rtp"
    id = 005
    description = "Live Camera Feed"
    metadata = "{ \"path\": \"/dev/video0\", \"video_stream_src\": \"Device\" }"
    media = (
        {
          type = "video"
          mid = "v1"
          label = "Live Camera Feed #1"
          port = 20002
          pt = 96
          rtpmap = "H264/90000"
          videofmtp = "profile-level-id=42e01f;packetization-mode=1"
        }
    )
}