27
Programming
Signaling Standalone Manual
Federal Signal
www.fedsig.com
Fields
Description
Ambient
Attenuation
Use to set the default Ambient Attenuation Threshold. The selected
level applies to SIP audio and all other functions unless overridden by
Commander
®
. The sound level will start attenuating when the ambient
SPL drops below this threshold.
SmartMsg
Enable
SmartMsg
Check to enable the SmartMsg interface.
SmartMsg
Server
The RTU’s assigned default SmartMsg Server IP Address or DNS
name.
SmartMsg Port
The port is preconfigured to 16887.
Modbus
Enable
Modbus
Check to enable the Modbus
®
interface.
Modbus Port
The RTU’s assigned Modbus TCP port number. The default is 502.
SIP
Enable SIP
Check to enable the SIP interface
SIP Port
The RTU’s assigned SIP port number. The default is 5060.
SIP TLS Port
The RTU’s assigned TLS SIP port number. The default is 5061
Local Address
Displays the RTU’s IP address.
Primary SIP
Server
The RTU’s assigned primary SIP server.
Secondary SIP
Server
The RTU’s assigned secondary SIP server. If your system does not
use a failover server, leave this field blank.
Address
(Extension)
The RTU’s assigned Address or Extension number. This field is
required.
Registration
User ID
Username for registration. If this field is blank, the Address
(Extension) will be used for the Registration User ID.
Password
The RTU’s assigned SIP password.
Registration
Interval
The RTU’s assigned SIP registration interval in seconds. Enter a
value between 10 and 3600.
Call Time Limit
The RTU’s assigned SIP call limit duration in seconds. The call is
dropped automatically when the time limit is reached. This prevents
a speaker from staying busy if a phone is left off the hook or on hold
unintentionally. Enter a value between 10 and 3600.
Underrun
(Jitter) Delay
Underrun occurs when a device runs out of data during live streaming
PA or VoIP, causing the audio to cut out. The underrun delay setting
defines the length of data buffering to use before playback begins.
The buffer size is adjustable from 0–15 seconds. The buffer duration
can be set to 0 on high-speed networks designed for VoIP traffic.
Wireless networks and networks without QoS may require additional
buffering to eliminate jitter and lost audio.
Transport
The RTU’s assigned SIP transport protocol. For TLS, optional
certificates can be uploaded from the Certificates page.
