FAQ

General Questions

USN is a SIP based VoIP application, it provides services that includes Audio Conferencing, Mass Notification, Click-to-call, and Web Conferencing. Voice over IP (VoIP) refers to carriage of voice calls as data packets over the Internet. Session Initiation Protocol (SIP) is a signaling protocol that is used for setting up VoIP connections. The VoIP/SIP trunk operates over a standard Ethernet interface. USN Deployment

NMC is currently supported on following Public and Private Cloud Platforms:

Private Cloud:
  • VMware ESXI 5.1 and above is recommended for production usage.
  • Hyper-V for Windows Servers
  • OpenNebula
Public Cloud:
  • Microsoft Azure
  • Amazon AWS

Support Virtual Machine Configuration

# of Ports1002505001000
# of Cores481224
Clock Speed (GHz)2.5+2.7+2.7+2.7+
Hyper-threadingYesYesYesYes
Resource ReservationRequiredRequiredRequiredRequired
RAM4GB8GB12GB16GB
Hard Disk (minimum)100 GB100 GB250 GB500 GB
Yes, this is part of audio conference service which uses Firebar application to dial out the predefined numbers to call far end users whenever users triggered by web, voice or API.
Yes, the participants are automatically called at the schedule time.
Yes, USN allows users to configure a prefix to be dialed before each outbound call. The setting is depend on the defined digits for internal call.
Yes, far end callers will be prompted a custom announcement before NMC joins them to the conference.
Yes, while conference is in progress, the moderator can add people to the conference using DTMF controls.
In its simplest term, audio conferencing works by digitally summing different voice channels. The summed up signal is then sent back to each recipient after subtracting recipient’s own inbound signal from it. This logic is usually implemented using Digital Signal Processing (DSP) techniques.
Digital Signal Processing (DSP) is the science of manipulating digital samples of an analog signal. For example, for audio conferencing, digital samples from different participants are digitally added to create a combined signal. The digitally combined signal is converted back to analog by the end device and fed to the speaker of each participant.
Linux is an open source, much more inherently stable and secure operating system. Being open source it keeps evolving based on the work effort of thousands of developers worldwide. Its roots are based in the UNIX operating system. Several companies re-package the open source software and then market it under their brand names.
It is the name of a LINUX distribution.
USN with Release 9.0 or higher uses Linux CentOS 64 bits version 7.0.x.
Echo is generated when 2 wire to 4 wire conversion takes place in handling of a voice call. If there is an impedance mismatch then part of the outbound signal is reflected back from the 2W to 4W hybrid. This reflected signal is then heard a few milliseconds later by person speaking into a conference and is perceived as echo. The modern day PBXs and conference bridges are designed to cancel some of this returned echo signal. This is done using digital signal processing techniques.
Automatic Gain Control refers to adjusting the amplitude of the incoming signal to a ‘normal’ decibel level. If the signal is too weak it’s artificially boosted up. If the signal is too ‘hot’ the signal is scaled down. In either case the adjustment is made automatically and in real time using DSP techniques.
Loudest Speaker Algorithm allows a conference bridge to automatically determine who the loudest speaker is. This is done by computing speech energy in the incoming voice channels. In such cases the conference bridge can send the output from the loudest speaker to all other recipients as part of the conference mixing algorithm instead of performing raw summation of all voice channels. The NMC supports loudest speaker algorithm.
On TDM phone circuits the DTMF tones are used to enter dialed digits, PINs, card numbers etc. These DTMF tones are carried inband on the TDM network. These functions are also needed in the VoIP network and are provided by special out of band packets defined by RFC 2833.
A typical configuration for a SIP trunk on USN is: G.711 codec (Alaw or MuLaw), RFC 2833 for DTMF relay.
Under normal circumstances the AGC algorithm takes care of the gain related parameters automatically and no adjustments are required. In specialized cases, the designers of the equipment can alter some of the parameters associated with AGC. Please contact XOP’s customer support department if you believe there is a need for adjustments to the AGC parameters.
T1/ E1 and PRI trunks are used in legacy TDM network. In case there is a need to connect NMC to a T1/E1 or PRI, then a suitable media gateway should be used that converts the T1/E1 or PRI into a SIP trunk.
Yes, there is a help link on the USN web portal that provides comprehensive help. In addition individual pages also provide context sensitive help (see help on top right hand side).
Users can control the conference related services via Web portal GUI, DTMF controls, API services.
Each moderator has different privileges, multiple conferences, a view of the conference in real time called Realview which allows a moderator to mute a participant, disconnect a participant, transfer a participant, view loudest speaker and control all controls for the services.
Web conferencing includes Desktop Sharing, Application Sharing, Chat, Whiteboard, and Remote Controls.
Mass Notification can be configured to send Voice message, Email Message, and SMS Message