All rights reserved.
The information in this document is subject to change without notice. The statements, configurations, technical data, and
recommendations in this document are believed to be accurate and reliable, but are presented without express or implied
warranty. Users must take full responsibility for their applications of any products specified in this document. The
information in this document is proprietary to Nortel Networks NA Inc.
Trademarks
NORTEL NETWORKS, Norstar and Meridian are trademarks of Nortel Networks.
Microsoft, MS, MS-DOS, Windows, and Windows NT are registered trademarks of Microsoft Corporation.
All other trademarks and registered trademarks are the property of their respective owners.
Configuring number manipulation for Source Phone Numbers for IP to Norstar calls . . 52
Configuring number manipulation for Source Phone Numbers for Norstar to IP calls . . 55
The Norstar VoIP Gateway is an accessory for Norstar KSU that provides up to four IP telephony
trunks. These IP telephony trunks allow you to establish voice calls to other IP telephony enabled
telephones systems using a data networking connection. Examples of IP telephony enabled
telephone systems are a Business Communications Manager system, Meridian 1 IPT or another
Norstar KSU equipped with a Norstar VoIP Gateway.
Before you begin
This guide provides information about the configuration and operation of the Norstar VoIP
Gateway. It is intended for persons responsible for the configuration of a Norstar VoIP Gateway
system. Prior knowledge of IP networks is required.
Before using this guide, the Norstar VoIP Gateway must be connected to the Norstar KSU.
This guide assumes:
•You have planned the telephony and data requirements for your Norstar system.
•The Norstar KSU is installed and initialized, and the hardware is working. External lines and
internal telephones and telephony equipment are connected to the KSU or Expansion
Modules.
•Configuration of lines is complete.
•Operators have a working knowledge of the Windows operating system and of graphical user
interfaces.
•Operators who manage the data portion of the system are familiar with network management
and applications.
13
Symbols used in this guide
This guide uses these symbols to draw your attention to important information:
Caution: Caution Symbol
Alerts you to conditions where you can damage the equipment.
Danger: Electrical Shock Hazard Symbol
Alerts you to conditions where you can get an electrical shock.
Warning: Warning Symbol
Alerts you to conditions where you can cause the system to fail or work improperly.
Norstar VoIP Gateway Configuration Guide
14 Text conventions
Note: Note/Tip symbol
Alerts you to important information.
Tip: Note/Tip symbol
Alerts you to additional information that can help you perform a task.
Text conventions
This guide uses these following text conventions:
angle brackets (< >)Represent the text you enter based on the description inside the
brackets. Do not type the brackets when entering the command.
Example: If the command syntax is
ping
<ip_address>
, you enter: ping 192.32.10.12
bold Courier text
italic textRepresents terms, book titles and variables in command syntax
bold textRepresents fields names, field entries, and screen names in the
plain Courier
text
dollar sign ($)
Acronyms
This guide uses the following acronyms:
Represent command names, options and text that you need to enter.
Example: Use the
Example: Enter
dinfo command.
show ip {alerts|routes}.
descriptions. If a variable is two or more words, the words are
connected by an underscore.
Example: The command syntax
show at
valid_route
<valid_route>
,
is one variable and you substitute one value for it.
configuration application.
Represents command syntax and system output, such as prompts and
system messages.
Example:
The $ symbol indicates hexadecimal notation.
Set Trap Monitor Filters
ATMAsynchronous Transfer Mode
CDPCoordinated Dialing Plan
CLIDCalling Line Identification
CIRCommitted Information Rate
CNGComfort Noise Generation
ECMError Correction Mode
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Related publications 15
ICMPInternet Control Message Protocol
IEEE802 ESSInstitute of Electrical and Electronics Engineers, Inc., standard 802
Electronic Switching System Identification code
IPInternet Protocol
ISDNIntegrated Services Digital Network
LANLocal Area Network
LATALocal Area and Transport Area
MCRMaximum Cell Rate
MOSMean Opinion Score
PCMPulse Code Modulation
PINGPacket InterNet Groper
PPPPoint-to-Point Protocol
PRIPrimary Rate Interface
PSTNPublic Switched Telephone Network
QoSQuality of Service
RASRegistration, Admissions and Status
RTPReal-time Transfer Protocol
SNMPSimple Network Management Protocol
TCPTransmission Control Protocol
UDPUser Datagram Protocol
UDPUniversal Dialing Plan
VoIPVoice over Internet Protocol
VADVoice Activity Detection
VLANVirtual LAN
WANWide Area Network
Related publications
Documents referenced in the Norstar VoIP Gateway Configuration Guide include:
•Norstar Installer Guide
•Norstar System Coordinator Guide
•Norstar VoIP Gateway Installation Guide
Norstar VoIP Gateway Configuration Guide
16 How to get help
How to get help
USA and Canada
Authorized Distributors - ITAS Technical Support
Telephone:
1-800-4NORTEL (1-800-466-7835)
If you already have a PIN Code, you can enter Express Routing Code (ERC) 196#.
If you do not yet have a PIN Code, or for general questions and first line support, you can enter
ERC 338#.
Website:
http://www.nortelnetworks.com/support
Presales Support (CSAN)
Telephone:
1-800-4NORTEL (1-800-466-7835)
Use Express Routing Code (ERC) 1063#
EMEA (Europe, Middle East, Africa)
Technical Support - CTAS
Telephone:
00800 800 89009
Fax:
44-191-555-7980
email:
emeahelp@nortelnetworks.com
CALA (Caribbean & Latin America)
Technical Support - CTAS
Telephone:
1-954-858-7777
email:
csrmgmt@nortelnetworks.com
APAC (Asia Pacific)
Technical Support - CTAS
Telephone:
+61 388664627
Fax:
+61 388664644
email:
asia_support@nortelnetworks.com
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Chapter 1
About Norstar VoIP Gateway
The VoIP Gateway provides excellent voice quality and optimized packet voice streaming over IP
networks. The product enables voice, fax and data traffic to be sent over the same IP network.
The VoIP Gateway incorporates up to four ports for connection to analog trunk ports on a Norstar
KSU or to a fax. These ports supports up to four simultaneous VoIP calls.
Additionally, the VoIP Gateway is equipped with a 10/100 Base-T Ethernet port for connection to
the LAN.
With the VoIP Gateway you can:
•network your Norstar system to another Norstar system or other Nortel Networks Enterprise
communications systems using the IP network
•use your IP network to replace PSTN or other costly private network trunking between
locations
•use the capacity of the enterprise data network for voice and avoid costly access and long
distance charges.
17
The VoIP Gateway is a very compact device, designed to be installed on a desk-top, on the wall, or
in a 19-inch rack.
The VoIP Gateway supports the H.323 ITU protocol, enabling the deployment of "voice over
packet" solutions in environments where each enterprise location is provided with a simple Media
Gateway. This provides the enterprise with the ability to transmit the voice and telephony signals
over a packet network.
The layout diagram, Figure 2 on page 18, illustrates a typical VoIP Gateway application.
Figure 1 Norstar VoIP Gateway front panel
1 2 3 4
C h a n n e l s
Data
Control
LAN
Ready
VoIP Gateway
Norstar VoIP Gateway Configuration Guide
18
Figure 2 Typical VoIP Gateway application
VoIP Gateway
Router
Branch Office
Modular ICS
l
y
d
tro
a
t
a
n
N
a
e
o
A
1 2 3 4
C h a n n e l s
R
D
L
C
VoIP Gateway
Branch Office
Compact ICS
Internet
Head Office
Modular ICS
l
y
d
tro
ta
a
n
N
a
e
o
1 2 3 4
C h a n n e l s
R
D
LA
C
VoIP Gateway
VoIP Gateway
Router
VoIP Gateway
1 2 3 4
C h a n n e l s
R
D
LA
C
VoIP Gateway
l
y
o
d
tr
ta
a
n
N
a
e
o
Router
VoIP Gateway key features
•high quality voice, data and fax over IP networks
•supports up to 4 analog telephone loop start ports
•connected to the IP network via a 10/100 Base-T Ethernet interface
•codecs include: G.711, G.723.1, G.729A
•T.38 Fax with superior performance (can handle a round trip delay up to 9 sec.)
•compliant with H.323 (Version 4)
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19
•Emergency Line, connected to the unused pins on port #4, with a relay to an analog line, even
if the VoIP Gateway is powered off.
•LEDs on the front and rear panels that provide information on the operating status of the VoIP
Gateway and the network interface
•Restart button on the front panel that restarts the VoIP Gateway
•compact, rugged enclosure only one-half of a 19-inch rack unit, 1 U high (1.75" or 44.5 mm)
•mounting option of installing two VoIP Gateway in a single 19-inch rack shelf, one U high
(1.75" or 44.5 mm).
Supported H.323 features
The VoIP Gateway implements the RadVision™ H.323 version 4.0 protocol stack. In this version,
the VoIP Gateway features the following:
Gatekeeper
•Works without a Gatekeeper using the internal phone table.
•Registers to a known Gatekeeper.
•Supports Gatekeeper registration with prefixes
•Functions as an H.323 gateway or can imitate an H.323 terminal with up to four aliases.
•Uses routed-mode calls.
•Uses direct-mode calls.
•Uses redundant Gatekeepers if a redundant Gatekeeper is defined.
•Can fallback to internal routing table if there is no communication with the Gatekeeper.
•Supports the "TimeToLive" parameter. The VoIP Gateway sends Registration requests up to
"TimeToLive" expiration.
•Supports IRR messages for KeepAlive.
•Supports the mapping of destination (Alias) numbers in the ACF message by the Gatekeeper.
•Supports RAI (Resource Available Indication) messages, informing gatekeeper that the
gateway resources are below a threshold.
Call setup
•Can use the Normal Connect procedure.
•Can use the Fast Connect procedure with or without immediately opening a H.245 channel.
•Can use Tunneling.
•Can negotiate a codec from a list of given codecs for Normal or Fast Connect procedures.
•Can open a H.245 channel when using Fast Connect.
Norstar VoIP Gateway Configuration Guide
20
Other:
•Supports using a Country Code (0xB5) and Manufacturers Code (0x28) in H.323 messages
•Supports H.323 Annex D, T.38 real time FAX.
•Supports H.450 Call Hold, Call Transfer and Call Forwarding supplementary services
(H.450.1, H.450.2, H.450.3 and H.450.4).
•Supports the following codecs: G.711 A-law, G.711 µ-law, G.723.1 (6.3 kbps, 5.3 kbps), G.729.
•Supports DTMF and HookFlash signal out of band using the H.245 channel (using the
"Alphanumeric" field).
•Supports DTMF out of band using H.225/Q.931 keypad facility messages.
•Supports of one or two stage dialing for network to VoIP Gateway calls.
•Supports reopening of logical channel and implementation of empty terminal capability set.
•Supports configurable H.323 Port Range.
•Supports H.225/Q.931 Progress Indicator parameter for Fast Connect, enabling playing of
local ringback tone or to cut through the voice channel to listen to remote call progress tones/
messages.
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Chapter 2
IP Telephony overview
This section provides a brief overview of the “Key IP telephony concepts”. It also provides a
“Prerequisites checklist” to help you set up your IP Telephony network.
Key IP telephony concepts
In traditional telephony, the voice path between two telephones is circuit switched. This means that
the analog or digital connection between the two telephones is dedicated to the call. The voice
quality is usually excellent, since there is no other signal to interfere.
In IP telephony, the VoIP Gateway encodes the speech of the call into small data packets called
frames. The system sends the frames across the IP network to the other VoIP Gateway, where the
frames are decoded and sent to the receiving telephone. If some of the frames get lost while in
transit, or are delayed too long, the receiving telephone experiences poor voice quality. On a
properly-configured network, voice quality should be consistent for all IP calls.
21
The following sections describe some of the components for IP telephony:
•“VoIP trunks” on page 21
•“Gatekeepers” on page 22
•“Codecs” on page 22
•“Jitter Buffer” on page 23
•“QoS routing” on page 24
VoIP trunks
VoIP trunks allow voice signals to travel across IP networks. The VoIP Gateway converts the voice
signal into IP packets, which are then transmitted through the IP network to a gateway on the
remote system. The device at the other end reassembles the packets into a voice signal. Business
Communications Manager and Meridian 1 IPT are devices that can use the H.323 protocol trunks
which the VoIP Gateway supports.
VoIP trunks and analog/digital telephones
While analog and digital telephones cannot be connected to the VoIP Gateway system with an IP
connection, they can make and receive calls to and from other systems through VoIP trunks. Calls
received through the VoIP trunks to system telephones are received through the LAN or WAN and
are translated within the VoIP Gateway to voice channels.
Norstar VoIP Gateway Configuration Guide
22 Key IP telephony concepts
Gatekeepers
A gatekeeper tracks IP addresses of specified devices, and provides authorization for making and
accepting calls for these devices. A gatekeeper is not required as part of the network to which your
VoIP Gateway is attached, but Gatekeepers can be useful on networks with a large number of
devices.
Note: The VoIP Gateway does not contain a gatekeeper application. If you want to put a
gatekeeper on your network, it must be put on a separate gatekeeper server. The VoIP
Gateway is compatible with RadVision and CSE 1000 gatekeepers.
Codecs
The algorithm used to compress and decompress voice is embedded in a software entity called a
codec (COde-DECode).
Two popular Codecs are G.711 and G.729. The G.711 Codec samples voice at 64 kilobits per
second (kbps) while G.729 samples at a far lower rate of 8 kbps. For actual bandwidth
requirements, refer to “Determining the bandwidth requirements” on page 153, where you will
note that the actual kbps requirements are slightly higher than the label suggests.
Voice quality is better when using a G.711 CODEC, but more network bandwidth is used to
exchange the voice frames between the telephones.
If you experience poor voice quality, and suspect it is due to heavy network traffic, you can get
better voice quality by configuring the IP telephone to use a G.729 CODEC.
The VoIP Gateway supports these codecs:
•G.729
•G.723
•G.711-uLaw
•G.711-aLaw
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Key IP telephony concepts 23
Jitter Buffer
Voice frames are transmitted at a fixed rate, because the time interval between frames is constant.
If the frames arrive at the other end at the same rate, voice quality is perceived as good. In many
cases, however, some frames can arrive slightly faster or slower than the other frames. This is
called jitter, and degrades the perceived voice quality. To minimize this problem, the VoIP
Gateway uses a jitter buffer for arriving frames.
The Norstar VoIP Gateway uses a dynamic jitter buffer that can be configured using two
parameters:
•Minimum delay (0 msec to 150 msec).
This defines the starting jitter capacity of the buffer. For instance, at 0 msec, there is no
buffering at the start. At the default level of 70 msec, the VoIP Gateway will always buffer
incoming packets by at least 70 msec worth of voice frames.
•Optimization Factor (0 to 12).
This defines how the jitter buffer tracks to changing network conditions. When set at its
maximum value of 12, the dynamic buffer will aggressively track changes in delay (based on
packet loss statistics) to increase the size of the buffer and then not decay back down. This
results in the best packet error performance, but at the cost of extra delay. At the minimum
value of 0, the buffer tracks delays only to compensate for clock drift and quickly will decay
back to the minimum level. This optimizes the delay performance but at the expense of a
higher error rate.
The default settings of 70 msec Minimum delay and 7 Optimization Factor should provide a good
compromise between delay and error rate. The jitter buffer "holds" incoming packets for 70 msec
before making them available to the codec for decoding into voice. The codec actually "takes"
frames from the buffer at regular intervals in order to produce continuous speech. As long as
delays in the network do not change (jitter) by more than 70 msec from one packet to the next,
there will always be a sample in the buffer for the codec to use. If there is more than 70 msec of
delay at any time during the call, the packet is too late. The codec will try to access a frame and
will not be able to find one. The codec must produce a voice sample even if a frame is not
available. It will actually create a voice sample to use that minimizes the effect of the loss. This
loss is then flagged as the buffer being too small. The dynamic algorithm then causes the size of
the buffer to increase for the next voice session. The size of the buffer may decrease again if the
gateway notices that the buffer is not filling up as much as expected. At no time will the buffer
shrink to less than the minimum size configured in the Minimum delay parameter.
This delaying of packets can provide somewhat of a communications challenge, as speech is
delayed by the number of frames in the buffer. For one-sided conversations, there are no issues.
However, for two-sided conversations, where one party tries to interrupt the other speaking party,
it can be annoying. In this second situation, by the time the voice of the interrupter reaches the
interruptee, the interruptee has spoken (2*jitter size) frames past the intended point of interruption.
Norstar VoIP Gateway Configuration Guide
24 Prerequisites checklist
QoS routing
To minimize voice jitter over low bandwidth connections, the VoIP Gateway can assign specific
DiffServ Marking in the IPv4 header of the IP telephony data packets.
The DiffServ Code point (DSCP) is contained in the second byte of the IPv4 header. DSCP is used
by the router to determine how the packets will be separated for Per Hop Behavior (PHB). The
DSCP is contained within the DiffServ field, which was known as the ToS field in older versions.
Prerequisites checklist
Before you set up VoIP trunks on the VoIP Gateway, complete the following checklists to ensure
that the system is correctly set up. Some questions do not apply to all installations.
This guide contains a number of appendices that explain various aspects of the system directly
related to IP telephony functions.
This section includes the following checklists:
•“Network diagram” on page 24
•“Network devices” on page 25
•“Network assessment” on page 25
Network diagram
To aid in installation, a Network Diagram is needed to provide a basic understanding of how the
network is configured. Before you install IP functionality, you must have a network diagram that
captures all of the information described in the following table.
Table 1 Network diagram prerequisites
PrerequisitesYes
1.a Has a network diagram been developed?
1.b Does the network diagram contain any routers, switches or bridges with corresponding
IP addresses and bandwidth values for WAN or LAN links?
1.c Does the network diagram contain IP Addresses, netmasks, and network locations of all Norstar VoIP
Gateways?
1.d Does the network diagram contain IP Addresses and netmasks of any other IP Telephony gateways
that you need to connect to?
1.e Does the network diagram contain the IP address for any Gatekeeper that may be used?
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Network assessment 25
Network devices
The following table contains questions about devices on the network such as firewalls, NAT
devices, and DHCP servers.
•If the network uses public IP addresses, complete 2.c.
•If the network uses private IP addresses, complete 2.d. to 2.e.
Table 2 Network device checklist
PrerequisitesYesNo
2.a Is the network using DHCP?
2.b Is the network using private IP addresses?
2.c Do you have a public IP addresses for the Norstar VoIP Gateway?
2.d Does the system have a firewall/NAT device?
2.e A hub-based core will not have suitable performance for IP Telephony. Does the network
use a non-hub solution at its core?
Network assessment
The following table of questions are meant to ensure that the network is capable of handling IP
telephony, and that existing network services are not adversely affected.
Table 3 Network assessment
PrerequisitesYesNo
3.a Has a network assessment been completed?
3.b Has the number of switch/hub ports available and used in the LAN infrastructure been
calculated?
3.c Does the switch use VLANs? If so, get the VLAN port number and ID.
3.d Have the used and available IP addresses for each LAN segment been calculated?
3.e Has DHCP usage and location been recorded?
3.f Has the speed and configuration of the LAN been calculated?
3.g Has the estimated latency values between network locations been calculated?
3.h Have the Bandwidth/CIR utilization values for all WAN links been calculated?
3.i Has the quality of service availability on the network been calculated?
Norstar VoIP Gateway Configuration Guide
26 Network assessment
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Chapter 3
Configuring the VoIP Gateway
The VoIP Gateway has a web interface you use for gateway configuration, including downloading
of configuration files and for run-time monitoring. You can access the web interface from any
standard web browser, such as Microsoft™ Internet Explorer or Netscape™ Navigator.
Specifically, you can employ this facility to set up the gateway configuration parameters needed to
configure the VoIP Gateway. You also have the option to reset the gateway to apply the new set of
parameters.
Computer requirements
To use the web interface, you need the following:
•a computer capable of running your web browser
•a network connection to the VoIP Gateway
•one of the following compatible web browsers
•Microsoft™ Internet Explorer™ (version 5.0 and higher)
•Netscape™ Navigator™ (version 7.0 and higher)
27
Accessing the web interface
To access the web interface:
1Open your web browser.
2In the URL field, enter the IP address of the VoIP Gateway.
When you enter the IP address make sure you include http:// at the start of the IP address (for
example: http://10.1.10.10.
The Enter Network Password screen appears.
Norstar VoIP Gateway Configuration Guide
28 Accessing the web interface
Figure 3 Web browser login screen
3Enter the User Name and Password.
If you have not changed the user name and password, the default User Name is Admin and the
default password is Admin.
Note: Nortel Networks recommends that you change the User Name and Password from
their default values. For information about how to change the User Name and Password,
refer to “Changing the VoIP Gateway password” on page 133.
4Click the OK button.
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Configuring the Protocol Definition parameters 29
Configuring the Protocol Definition parameters
To configure the Protocol Definition parameters:
1Access the web interface.
2Click the Protocol Management button.
3Click the Protocol Definition tab.
The Protocol Definition screen appears.
Figure 4 Protocol Definition screen
Norstar VoIP Gateway Configuration Guide
30 Configuring the Protocol Definition parameters
4Configure the Protocol Definition parameters according to the following table.
Table 4 Protocol Definition parameters
ParameterDescription
General
Connection ModeSelect Fast Start if you want the VoIP Gateway to use the Fast Start connection mode.
Select Normal if you want the VoIP Gateway to use the Normal connection mode.
The default value is Fast Start.
The Fast Start connection mode allows a media path to be established using H.225,
without having to start the full H.245 protocol session. In some situations, you need to use
fast start in order to control call progress tones.
Enable Annex D/T.38 Fax
relay
Enable DTMF over H.245 Select Yes to enable DTMF over H.245.
H.323-IDEnter the VoIP Gateway H.323-ID you want to use for registration to the Gatekeeper.
Source Number
Encoding Type
Destination Number
Encoding Type
Select No to disable Annex D/T.38 Fax relay.
Select Yes to enable Annex D/T.38 Fax relay.
When you enable this feature, the VoIP Gateway can send and receive Fax messages
using the H.323 Annex D T.38 procedure.
When you enable this feature, the VoIP Gateway sends out of band DTMF signaling using
H.245. Out of band signaling is recommended for use with H.323 protocol and the Norstar
KSU.
You can enter a string up to 19 characters long.
When you are using a Gatekeeper, the VoIP Gateway will send a registration message to
the Gatekeeper with this H323-ID string.
Select the source number encoding type. This defines the encoding type of the calling
phone number in H.225 setup messages.
You can select E.164, H.323-ID, E.164 & H.323-ID, TableValues or TableValues &
H.323-ID.
Select TableValues if you want the VoIP Gateway to use the values configured in the
Tel -> IP Source Number Manipulation table.
Select H323-ID if you want the VoIP Gateway to add the H323-ID to the source
information.
Select E.164 if you want the VoIP Gateway to use E.164 source encoding and not use the
encoding type defined in the table.
The default value is E.164.
Note: If you select an option that includes “H.323-ID”, you must enter a string in the
H.323-ID box.
Select the destination number encoding type. This defines the encoding type of the called
phone number in H.225 setup messages.
You can select E.164, H.323-ID, E.164 and H.323-ID, or TableValues.
Select TableValues if you want the VoIP Gateway to use the values configured in the
Tel -> IP Dest Number manipulation table.
Select H.323-ID if you want the VoIP Gateway to add the H323-ID to the destination
information.
Select E.164 if you want the VoIP Gateway to use E.164 destination encoding and not use
the encoding type defined in the table.
The default value is E.164.
Note: If you select an option that includes “H.323-ID”, you must enter a string in the
H.323-ID box.
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