Multitech Multivoip 100 Model Mvp110 Voice/fax Over Ip Networks User Guide

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Introduction
This chapter describes various features of the MVP110 software that enable you to change (update)
the configuration of your MVP110. The basic configuration parameters were entered during the
loading of the software (Chapter 3). The MVP110 software and configuration utilities described in this
chapter enable you to change that initial configuration as necessary.
The primary interface to the MVP110 software is the Main menu (MVP110 Setup is on the title bar)
with individual buttons that enable you to quickly and easily select a desired function. These features
are discussed in detail in the MVP110 Configuration section later in this chapter.
The MVP110 Configuration (Main menu) utility along with nine other configuration utilities provide full
software functionality for your MVP110. Configuration Port Setup enables you to change the
method by which you access the MVP110, whether through a direct connection from a PC to the
Command Port on the MVP110, or via your Internet or LAN connection to the LAN port on the
MVP110.  Date and Time Setup enables you to easily set the date and time used for data logging in
the MVP110. Download Factory Defaults enables you to return the configuration to the original
factory settings. Download Firmware enables you to download new versions of firmware as
enhancements become available. Download User Defaults enables you to repeat the download
user defaults process (part of software installation) and update the MVP110 configuration with any
necessary changes. Download Voice Coders enables you to download voice coders to the MVP110
after repair or upgrade. Download H.323 Stack enables you to download the H.323 protocols to the
MVP110 after repair or upgrade. Uninstall MVP110 Configuration removes most of the MVP110
software from your PC. Upgrade Software downloads boot code, new firmware, and an H.323 file,
then reboots the MVP110.
The MVP110 software includes a context-sensitive Help system. Clicking a Help [ 
? ] button
anywhere in the graphical user interface (GUI) will display definitions and recommended values for
the buttons, options, and fields on that dialog box or menu. Clicking the green underlined text in the
Helps displays a popup box of related supplementary information for that topic. Clicking the Search
button (just below the Help menu bar) displays an Index tab with a list of entries. Click an entry, then
click the Display button to display the text associated with that topic.
Before You Begin
The MVP110 software operates in a Microsoft Windows environment. The MVP110 program group
contains icons for all the utilities described above. You can open the individual utility programs by
clicking Start | Programs | MultiVOIP 100 | (utility). 
						

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MVP110 Configuration
The MVP110 Setup menu consists of 10 buttons, an Events window in the middle of the menu, and a
status bar at the bottom of the menu.  The 10 buttons allow you to display and change the voice
channels and IP protocol parameters, display and manage the Phone Book listing, view statistics and
call progress, and change features such as SNMP Agent, Telnet Server, WEB Server, and assign a
MVP110 password.
The Events window in the lower third of the Setup menu provides information about the boot process.
The status bar at the bottom of the Setup menu displays the current status of the unit and shows, for
example, if it is Running, the most recent date the unit was configured, the type of connection you
have to the unit, and your rights. It shows if your PC is connected directly to the command port of the
MVP110  or is communicating with the Ethernet port.  The Rights box which displays whether you
have Read/Write or Read-Only rights. The first user to communicate with the MVP110 command port
has Read/Write rights that enable the user to view and/or change the configuration of the MVP110.
Any additional simultaneous users have Read-Only rights and can only display the configuration of
the MVP110 but are prohibited from changing the configuration.   
						

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Changing Channel Parameters
The channel parameters include the interface type and its options, voice and fax settings, billing and
security, and voice communications for the country and region in which the MVP110 is operating. The
Channel Setup dialog box, accessed by clicking Voice Channels on the Setup menu, has four tabs
that display the following categories of channel information -- Interface, Voice/Fax, Billing/Misc, and
Regional.
Interface tab
The Inter Digit Time (in seconds) option in the Dialing Options group defines the amount of time the
MVP110 waits between digits as they are entered by the user.  If this timer expires, the MVP110 will
immediately attempt to match the digits entered to an entry in the Phone Directory Database. The
range for this option is 2 to 100 with a default of 2.
FXS Interface
The FXS Interface is used to connect telephones, fax machines, key telephone systems, etc., to the
MVP110.  In addition, you need to select either Loop Start or Ground Start.  Most of the equipment
mentioned will use Loop Start which is the default.
Ring Count
Enter the maximum number of rings output on the FXS interface (default is 8) before hanging up and
releasing the line to another call.
A setting of 0 (zero) on the FXS interface disables the generation of rings. The caller will receive a
“Busy” tone.
Message Waiting Light
The Message Waiting Light check box must be selected on the originating and answering voice
channel.  This enables the number dialed to connect you to the appropriate voice channel, then
output that number on the voice channel.  This feature does not work FXS to FXS.   
						

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Voice/Fax tab
The Voice/Fax tab controls voice and DTMF gain, voice coder, fax settings, and advanced options.
The Voice Gain group enables you to select the Input and Output voice gain. Gain is the increased
signaling power that occurs as the signal is boosted by the MVP110. The Input Gain list defines the
input gain for this voice channel. Before your MVP110 digitizes voice, the volume can be increased or
decreased. Input gain modifies the level of the audio coming in to the voice channel before it is sent
over the Internet to the remote MVP110. The valid range for this option is +31dB to –31dB. The
recommended and default value is 0. The Output Gain list defines the voice output gain for this voice
channel. Before your MVP110 converts digital voice back to analog, the volume can be increased or
decreased. The output gain modifies the level of the audio being output to the device attached to the
voice channel. The valid range for this option is +31dB to –31dB. The recommended and default
value is 0.
The DTMF Gain (Dual Tone Multi-Frequency) group controls the volume level of the digital tones sent
out for Touchtone dialing.  The Gain High and Gain Low lists control the gain in dB (decibels) of the
High and Low tones in the tone pairs.  The default gain values are -4 dB and -7 dB, respectively.
DTMF Gain should not be changed except under supervision of MultiTech’s Technical Support.
The DTMF Out of Band check box is selected so the MVP110 will reproduce the DTMF tones rather
than passing them through from the input to the output.
The MVP110 supports many state-of-the art ITU (International Telecommunications Union) voice
coders. The Voice Coder list enables you to select from a range of coders with specific bandwidths.
The higher the bps rate, the more bandwidth is used. The channel that you are calling has to have
the same voice coder selected.  Otherwise, you will always get a Busy signal.
The Fax group enables a fax machine to transmit and receive faxes through the MVP110. If a fax
machine is connected to the voice/fax channel, the Max Baud Rate should be set to match the baud
rate of the fax machine (refer to user documentation). The Fax Volume setting controls the output
level of the fax tones, and this setting should be changed only under the direction of Multi-Tech’s
Technical Support personnel (see Chapter 6 - Warranty, Service and Tech Support). The Jitter Value
setting defines the inter-arrival packet deviation (in milliseconds) for the fax transmission.  A higher
value will increase the amount of delay (allowing for a higher percentage of packets to be
reassembled) and a lower value would decrease the amount of delay (a lower percentage of packets
would be reassembled).
The Advanced Features group allows you to enable Silence Compression so that a MVP110 will
not transmit voice packets when silence is detected, thereby reducing the amount of network
bandwidth that is being used by the voice channel; Echo Cancellation for the voice channel will   
						

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remove echo and improve the quality of sound; and, Forward Error Correction allowing some of the
voice packets that were corrupted (or lost) to be recovered. FEC adds an additional 50% overhead to
the total network bandwidth consumed by the voice channel.
Billing/Misc
This tab controls the parameters for auto call, automatic disconnection, billing options, and dynamic
jitter buffer.
The Auto Call option allows the local MVP110 to call a remote MVP110 without the user having to
dial a Phone Directory Database number.  As soon as you access the local MVP110 voice/fax
channel, the MVP110 immediately connects to the remote MVP110 that you identified in the Remote
MVP110 Phone Number field of this option.
The Automatic Disconnection group provides three options which can be used singly or in any
combination.  The Jitter Value defines the average inter-arrival packet deviation (in milliseconds)
before the call is automatically disconnected.The default is 20 milliseconds. A higher value means
voice transmission will be more accepting of jitter. A lower value is less tolerant of jitter.
Consecutive Packets Lost defines the number of consecutive packets that are lost after which the
call is automatically disconnected. The default is 30 packets.
Call Duration defines the maximum length of time (in seconds) that a call remains connected before
the call is automatically disconnected. The default is 180 seconds.  A call limit of three minutes may
be too short for most configurations. Therefore, you may want to increase this default value.
Billing Options can be used to track the cost of Inbound and/or Outbound calls on the FXS interface.
The amount to be charged in cents is entered in the Charge (   ) Cents box together with the
associated time duration in the Per (   ) Seconds box.  While a given call is active, the accumulated
charges can then be viewed on the Call Progress dialog box. When the call ends, the charges are
transferred to a Log File that can be viewed by selecting the call event in the Log Entries dialog box
and selecting Details.
Dynamic Jitter Buffer defines a minimum and a maximum jitter value for voice communications.
When receiving voice packets from a remote MVP110, it is possible to experience varying delays
between packets due to traffic conditions on the network. This is called Jitter. To compensate for this
effect, the MVP110 uses a Dynamic Jitter Buffer. The Jitter Buffer allows the MVP110 to wait for
delayed voice packets by automatically adjusting the length of the Jitter Buffer between configurable   
						

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minimum and maximum values. An Optimization Factor adjustment controls how quickly the length of
the Jitter Buffer is increased when jitter increases on the network. The length of the jitter buffer
directly effects the voice delay between MVP110 gateways.
The default minimum dynamic jitter buffer of 150 milliseconds is the minimum delay that would be
acceptable over a low jitter network. The default maximum dynamic jitter buffer of 300 milliseconds is
the maximum delay tolerable over a high jitter network.
The Optimization Factor determines how quickly the length of the Dynamic Jitter Buffer is changed
based on actual jitter encountered on the network. Selecting the minimum value of 0 means low voice
delay is desired, but increases the possibility of jitter induced voice quality problems. Selecting the
maximum value of 12 means highest voice quality under jitter conditions is desired at the cost of
increased voice delay.
The Optimization Factor can be configured in the range of 0 to 12 with a default setting of 7.
Regional tab
The Regional tab controls the voice communications for the country or region in which the MVP110 is
being used.
From the Country/Region list, you can select the country or region for which you are configuring the
MVP110. The Tone Pairs group always displays the tones used in the country or region currently
selected. In addition to Australia, Central America, Chile, Europe, France, Japan, UK, and USA, there
is a Custom selection (with defaults identical to USA) that will make the Custom button active.
Clicking the Custom button enables you to edit the Tone Pairs and establish custom sets of tone
pairs for Dial Tone, Ring, and Busy on a Custom Tone Pair Settings dialog box.
The Pulse Generation Ratio group contains two ratios: the 60/40 is for the USA, and the 67/33 ratio
is for international applications.   
						

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Changing the Phone Directory Database
The MultiVOIP provides two phone directory database architectures; the propreitary database and a
database using an H.323 protocol gatekeeper that provides a centralized call control center. The
proprietary database can only be used when all the end points in the VOIP network are Multi-Tech
VOIP products.  The gatekeeper centralized call control center contains the phone directory database
when all VOIP gateways and endpoints support the H.323 protocol.  The proprietary database is
based on a master and slave relationship in which the master VOIP maintains the phone directory
database and distributes it to its slave VOIPs.  The centralized call center handles call control, call
routing, address translation from LAN aliases to IP addresses, and bandwidth management.  The
H.323 protocol allows other third party gateways and end points that support the H.323 protocol
standards to participate in the VOIP network (e.g., Microsoft NetMetting
®).
The database displays the phone numbers in numerical order with destination details (IP Address or
H323 ID), channel assignment, and a brief description of the entry. The method for changing the
phone directory database is dependent on whether the Gatekeeper option or the Proprietary
Phonebook option is enabled.
If the GateKeeper option is enabled, the RAS Parameters group is enabled and the IP address of the
GateKeeper needs to be enterred in the IP Address window.  The Port Number is the port on the
GateKeeper which it communicates with its endpoints.  The Q.931 Parameters group is enabled in
both the GateKeeper and Proprietary Phone Book Database architectures.  The Use Fast Start
option is used when the VOIP network supports Fast Start capability.
In the GateKeeper phone directory database, the phone directory database is developed through the
Add/Edit Phone Entry dialog box.  The Add/Edit Phone Entry dialog box defines the Station
Information, phone number and voice channel of the unit, and station identification, H323 ID which
defines the LAN alias and the IP address of the local unit.  In the GateKeeper phone directory
database, only the phone entries of the local unit display.   
						

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If the Proprietary PhoneBook option is enabled, the Database Type group is active which defines
the Master and Slave relationship.  If the database type is master, then the Add, Delete, Edit, Hunt,
and Print buttons at the top of the database dialog box are active.  This allows the master database
to build the phone directory.  You can click Slave Status to view the status of the slave units.
The Add/Edit Phone Entry dialog box for the Proprietary PhoneBook is used to add, delete, and edit
entries.  This information displays in the Phone Directory Database dialog box.
If the database type is set to slave, then the IP address of the master MultiVOIP needs to be entered
in the Master IP Address window, the Send Status Report to Master option can be enabled, and all
the buttons at the top of the directory database dialog box become inactive, except for the Print
button.
Proprietary Phone Directory Database
In the Proprietary Phone Directory Database, you can add, delete, or edit any entry in the phone
directory database and you can set up Hunt groups that locate another phone number if the called
number is busy. You can print the phone directory database so that you have a hardcopy of the
phone directory.
To add an entry to the Phone Directory database, click Add and the Add/Edit Phone Entry dialog box
displays.
The Add/Edit Phone Entry dialog box contains two groups of information; the Station Information
which contains the phone number, an optional description window, and the voice channel number.
The Station Identification group contains the Hunt Group listing and the IP Address window for the IP
Address of the MultiVOIP assigned the phone number.  The Port number is not used in the
proprietary phone book.  Click Copy From in the Add/Edit Phone Entry dialog box to add additional
phone entries.
The Station Information identifies the calling unit by the phone number, a description if you choose,
and voice channel of the unit doing the calling.  The Phone Number does not have to be a
conventional telephone number.  It can be, for example, a three digit number like 101.  The   
						

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Description window is like names in a local telephone book listing.  It identifies the calling party.  The
voice channel window defines the voice channel associated with the telephone.
The Station Identification group enables you to assign the entry to a Hunt Group, provide the IP
Address of the MultiVOIP being assigned the phone number, and accept the H.323 industry standard
Port number.  A Hunt Group is a series of telephone lines organized in such a way that if the first line
is busy the next line is hunted and so on until a free line is found. It is a set of links which provides a
common resource and which is assigned a single hunt group designation. A user requesting that
designation may then be connected to any member of the Hunt Group.
You can view the details of the current Hunt Group configuration by clicking the Phone Directory
Database’s Hunt button. The current Phone numbers for HUNT GROUP #1 are displayed.
Select the Hunt Group you wish to view. The Phone no’s window displays the telephone numbers
associated with that Hunt Group and the No. of Entries field displays the running total of entries.
Note: You can change the name of the Hunt Group by clicking on the entry that you want to change,
editing the change in the Hunt Group name window, and then clicking the Set button.
Click Slave Status on the Phone Directory Database dialog box to view the status of all the slave
units in your VOIP network (Send Status Report to Master must be enabled on the Slave).  The
Phone Number of each Slave displays with its IP Address, current line status, and the description of
the phone number.     
						

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Gatekeeper Phone Directory Database
With the Gatekeeper Phone Directory Database, the Gatekeeper acts as the central point for all
calls within its zone and provides call control services to registered endpoints. The Gatekeeper
performs address translation from LAN aliases to IP addresses and provides bandwidth management
where the network manager has specified a threshold for the number of simultaneous calls on the
LAN.  The H.323 ID is an alias.  The Gatekeeper may use other information as an alias such as a
URL or an e-mail address.
When the Gatekeeper Phone Directory Database option is enabled, the RAS Parameters group is
enabled with the IP address of the Gatekeeper displayed in the IP Address box.  The Port Number is
the port of the endpoint communicating with the Gatekeeper.  If this number is changed, it should only
be changed with consultation with Gatekeeper administrator.  The port numbers have to be in pairs
and controlled by the Gatekeeper.
If the H.323 Gatekeeper network supports Q.931 Fast Start servicing, then the Use Fast Start option
on all endpoints should be enabled.  The Call Signalling Port of 1720 is the port on the MultiVOIP unit
supporting the Q.931 parameters.