Add support for plain RTP participants in AudioBridge

conf/janus.plugin.audiobridge.jcfg.sample

@@ -12,6 +12,8 @@
 # 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)
+# allow_rtp_participants = true|false (whether participants should be allowed to join
+#		via plain RTP as well, rather than just WebRTC, default=false)
 #
 #     The following lines are only needed if you want the mixed audio
 #     to be automatically forwarded via plain RTP to an external component
@@ -45,6 +47,22 @@ general: {
 	# 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
+
+	# Normally, all AudioBridge participants will join by negotiating a WebRTC
+	# PeerConnection: the plugin also supports adding participants that will
+	# use plain RTP, though, be it for supporting legacy users (e.g., SIP
+	# participants who an orchestrator can add to the bridge) or more simply
+	# to temporarily inject external audio in a room from a live source. To
+	# support plain RTP, the plugin needs to have a range of ports it can bind
+	# to: notice this should be configured so that it doesn't conflict with other
+	# plugins (e.g., Streaming, SIP, NoSIP) and applications (e.g., Janus itself).
+	# The default if you don't specify anything is 10000-60000.
+	#rtp_port_range = "50000-60000"
+	# In case we need to support plain RTP participants, we'll also need to know
+	# what local IP address to bind to for media. If no address is set in the
+	# property below, then one will be automatically guessed from the system.
+	#local_ip = "1.2.3.4"
+
 }
 
 room-1234: {