; [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
; is_private = yes|no (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 = yes|no (do/don't stream audio)
; video = yes|no (do/don't stream video)
;    The following options are only valid for the 'rtp' type:
; data = yes|no (do/don't stream text via datachannels)
; audioport = local port for receiving audio frames
; 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 = yes|no (whether the plugin should perform skew
;       analisys and compensation on incoming audio RTP stream, EXPERIMENTAL)
; videoport = local port for receiving video frames
; 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 = yes|no (whether the plugin should store the latest
;       keyframe and send it immediately for new viewers, EXPERIMENTAL)
; videosimulcast = yes|no (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 = yes|no (whether the plugin should perform skew
;       analisys and compensation on incoming video RTP stream, EXPERIMENTAL)
; 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)
; databuffermsg = yes|no (whether the plugin should store the latest
;       message and send it immediately for new viewers)
;
; 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 following options are only valid for the 'rstp' type:
; url = RTSP stream URL
; rtsp_user = RTSP authorization username, if needed
; rtsp_pwd = RTSP authorization password, if needed
; rtsp_failcheck = whether an error should be returned if connecting to the RTSP server fails (default=yes)
; rtspiface = network interface or IP address to bind to, if any (binds to all otherwise), when receiving RTSP streams
;
; 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. To test the live and on-demand
; audio file streams, instead, the install.sh installation script
; automatically downloads a couple of files (radio.alaw, music.mulaw)
; to the plugins/streams folder.

[general]
;admin_key = supersecret        ; If set, mountpoints can be created via API
                                ; only if this key is provided in the request
;events = no                    ; Whether events should be sent to event
                                ; handlers (default is yes)

[gstreamer-sample]
type = rtp
id = 1
description = Opus/VP8 live stream coming from gstreamer
audio = no
video = yes
#audioport = 5002
#audiopt = 111
#audiortpmap = opus/48000/2
videoport = 5004
videopt = 100
videortpmap = VP8/90000
#secret = adminpwd

[file-live-sample]
type = live
id = 2
description = a-law file source (radio broadcast)
filename = /opt/janus/share/janus/streams/radio.alaw        ; See install.sh
audio = yes
video = no
secret = adminpwd

[file-ondemand-sample]
type = ondemand
id = 3
description = mu-law file source (music)
filename = /opt/janus/share/janus/streams/music.mulaw   ; See install.sh
audio = yes
video = no
secret = adminpwd

;
; Firefox Nightly supports H.264 through Cisco's OpenH264 plugin. The only
; supported profile is the baseline one. This is an example of how to create
; a H.264 mountpoint: you can feed it an x264enc+rtph264pay pipeline in
; gstreamer.
;
;[h264-sample]
;type = rtp
;id = 10
;description = H.264 live stream coming from gstreamer
;audio = no
;video = yes
;videoport = 8004
;videopt = 126
;videortpmap = H264/90000
;videofmtp = profile-level-id=42e01f\;packetization-mode=1

;
; This is a sample configuration for Opus/VP8 multicast streams
;
;[gstreamer-multicast]
;type = rtp
;id = 20
;description = Opus/VP8 live multicast stream coming from gstreamer
;audio = yes
;video = yes
;audioport = 5002
;audiomcast = 232.3.4.5
;audiopt = 111
;audiortpmap = opus/48000/2
;videoport = 5004
;videomcast = 232.3.4.5
;videopt = 100
;videortpmap = VP8/90000

;
; This is a sample configuration for an RTSP stream: you can specify
; the url to connect to and whether or not authentication is needed
; using the url/rtsp_user/rtsp_pwd settings (but notice that digest
; authentication will only work if you installed libcurl >= 7.45.0)
; NOTE WELL: the plugin does NOT transcode, so the RTSP stream MUST be
; in a format the browser can digest (e.g., VP8 or H.264 baseline for video)
; Again, you can override rtpmap and/or fmtp, if needed
;
;[rtsp-test]
;type = rtsp
;id = 99
;description = RTSP Test
;audio = no
;video = yes
;url=rtsp://127.0.0.1:8554/unicast
;rtsp_user=username
;rtsp_pwd=password