Making the timeout parameters for streaming plugin for RTSP play out configurable

conf/janus.plugin.streaming.jcfg.sample

@@ -0,0 +1,264 @@
+# 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 'rstp' 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)
+# rtspiface = network interface or IP address to bind to, if any (binds to all otherwise), when receiving RTSP streams
+# rtsp_failcheck = whether an error should be returned if connecting to the RTSP server fails (default=true)
+# 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 sreaming plugin will check the RTSP connection with an OPTIONS query,
+#		the value of the timeout is came from the RTSP session initializer and by default
+#		this session timeout is the half of this value In some cases this value is 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.
+#}
+
+general: {
+	#admin_key = "supersecret"		# If set, mountpoints can be created via API
+									# only if this key is provided in the request
+	#rtp_port_range = "20000-40000"	# 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
+}
+
+#
+# This is an example of an RTP source stream, which is what you'll need
+# in the vast majority of cases: here, the Streaming plugin will bind to
+# some ports, and expect media to be sent by an external source (e.g.,
+# FFmpeg or Gstreamer). This sample listens on 5002 for audio (Opus) and
+# 5004 for video (VP8), which is what the sample gstreamer script in the
+# plugins/streams folder sends to. Whatever is sent to those ports will
+# be the source of a WebRTC broadcast users can subscribe to.
+#
+rtp-sample: {
+	type = "rtp"
+	id = 1
+	description = "Opus/VP8 live stream coming from external source"
+	metadata = "You can use this metadata section to put any info you want!"
+	audio = true
+	video = true
+	audioport = 5002
+	audiopt = 111
+	audiortpmap = "opus/48000/2"
+	videoport = 5004
+	videopt = 100
+	videortpmap = "VP8/90000"
+	secret = "adminpwd"
+}
+
+#
+# This is a sample of the file-based streaming support. Specifically,
+# this simulates a radio broadcast by streaming (in a loop) raw a-Law
+# (that is, G.711) frames. Since type is "live", anyone subscribing to
+# this mountpoint will listen to the same broadcast as if it were live.
+# Notice that file-based streaming supports Opus files too, but no video.
+#
+file-live-sample: {
+	type = "live"
+	id = 2
+	description = "a-law file source (radio broadcast)"
+	filename = "/Users/zoltan.ponekker/Code/Janus/root/share/janus/streams/radio.alaw"
+	audio = true
+	video = false
+	secret = "adminpwd"
+}
+
+#
+# This is another sample of the file-based streaming support, but using
+# the "ondemand" type instead. In this case, the file we're streaming
+# contains raw mu-Law (still G.711) frames. Since this is "ondemand",
+# anyone subscribing to this mountpoint will listen to their own version
+# of the stream, meaning that it will start from the beginning and then
+# loop when it's over. On-demand streaming supports Opus files as well.
+#
+file-ondemand-sample: {
+	type = "ondemand"
+	id = 3
+	description = "mu-law file source (music)"
+	filename = "/Users/zoltan.ponekker/Code/Janus/root/share/janus/streams/music.mulaw"
+	audio = true
+	video = false
+	secret = "adminpwd"
+}
+
+#
+# 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.
+#
+#h264-sample: {
+	#type = "rtp"
+	#id = 10
+	#description = "H.264 live stream coming from gstreamer"
+	#audio = false
+	#video = true
+	#videoport = 8004
+	#videopt = 126
+	#videortpmap = "H264/90000"
+	#videofmtp = "profile-level-id=42e01f;packetization-mode=1"
+	#secret = "adminpwd"
+#}
+
+#
+# The Streaming plugin also supports the broadcasting of datachannel
+# messages, either by themselves or along other audio/video streams (e.g.,
+# to add a subtitle to a stream you're sending). The following is an
+# example of how you can create a datachannel-only mountpoint: you can
+# feed it with any tool that can send UDP datagrams, e.g., netcat.
+# Notice that the 'rtp' type just indicates this is a live mountpoint:
+# datachannel messages will be sent as usual, and not use RTP at all.
+#
+#data-example: {
+	#type = "rtp"
+	#id = 15
+	#description = "Datachannel stream from an UDP source"
+	#audio = false
+	#video = false
+	#data = true
+	#dataport = 5008
+	#datatype = "text"
+	#secret = "adminpwd"
+#}
+
+#
+# This is a variation of the rtp-sample configuration for Opus/VP8 shown
+# before, where multicast support is used to receive the streams. You
+# need an external script to feed data on those ports, of course.
+#
+#rtp-multicast: {
+	#type = "rtp"
+	#id = 20
+	#description = "Opus/VP8 live multicast stream sample"
+	#audio = true
+	#video = true
+	#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"
+	#secret = "adminpwd"
+#}
+
+#
+# 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 payload type, rtpmap and/or fmtp, if needed.
+#
+#rtsp-test: {
+	#type = "rtsp"
+	#id = 99
+	#description = "RTSP Test"
+	#audio = false
+	#video = true
+	#url = "rtsp://127.0.0.1:8554/unicast"
+	#rtsp_user = "username"
+	#rtsp_pwd = "password"
+	#secret = "adminpwd"
+	#rtsp_reconnect_delay = 5
+	#rtsp_session_timeout = 0
+	#rtsp_timeout = 10
+	#rtsp_conn_timeout = 5
+#}

conf/janus.plugin.streaming.jcfg.sample.in

@@ -249,4 +249,31 @@ file-ondemand-sample: {
 	#rtsp_user = "username"
 	#rtsp_pwd = "password"
 	#secret = "adminpwd"
+
+	# After n seconds passed and no media assumed, the RTSP server has
+	# gone and schedule a reconnect.
+	#
+	# Default value is 5 seconds.
+	#reconnect_delay = 5
+
+	# By default the sreaming plugin will check the RTSP connection with an OPTIONS query,
+	# the value of the timeout is came from the RTSP session initializer and by default
+	# this session timeout is the half of this value.
+	#
+	# In some cases this value is 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 value is 0 seconds, then it doesn't override the RTSP server's session timeout..
+	#session_timeout = 0
+
+	# Communication timeout for cURL call gathering the RTSP information.
+	#
+	# Default value is 10 seconds.
+	# rtsp_timeout = 10
+
+	# Connection timeout for cURL call gathering the RTSP information.
+	#
+	# Default value is 5 seconds.
+	# rtsp_conn_timeout = 5
 #}