Voice Gateways System Manual
Alvarion Voice Gateways
System Manual
August 2005
P/N 214198
Legal Rights
Legal Rights
© Copyright 2005 Alvarion Ltd. All rights reserved.
The material contained herein is proprietary, privileged, and confidential and
owned by Alvarion or its third party licensors. No disclosure thereof shall be
made to third parties without the express written permission of Alvarion Ltd.
Alvarion Ltd. reserves the right to alter the equipment specifications and
descriptions in this publication without prior notice. No part of this publication
shall be deemed to be part of any contract or warranty unless specifically
incorporated by reference into such contract or warranty.
Trade Names
Alvarion®, BreezeCOM®, WALKair®, WALKnet®, BreezeNET®, BreezeACCESS®,
BreezeMANAGE™, BreezeLINK®, BreezeCONFIG™, BreezeMAX™, AlvariSTAR™,
MGW™, eMGW™, WAVEXpress™, MicroXpress™, WAVEXchange™, WAVEView™,
GSM Network in a Box and TurboWAVE™ and/or other products and/or services
referenced here in are either registered trademarks, trademarks or service marks
of Alvarion Ltd.
All other names are or may be the trademarks of their respective owners.
Statement of Conditions
The information contained in this manual is subject to change without notice.
Alvarion Ltd. shall not be liable for errors contained herein or for incidental or
consequential damages in connection with the furnishing, performance, or use of
this manual or equipment supplied with it.
Warranties and Disclaimers
All Alvarion Ltd. (“Alvarion”) products purchased from Alvarion or through any of
Alvarion’s authorized resellers are subject to the following warranty and product
liability terms and conditions.
Exclusive Warranty
(a) Alvarion warrants that the Product hardware it supplies and the tangible
media on which any software is installed, under normal use and conditions, will
be free from significant defects in materials and workmanship for a period of
fourteen (14) months from the date of shipment of a given Product to Purchaser
(the “Warranty Period”). Alvarion will, at its sole option and as Purchaser’s sole
remedy, repair or replace any defective Product in accordance with Alvarion’
standard R&R procedure.
(b) With respect to the Firmware, Alvarion warrants the correct functionality
according to the attached documentation, for a period of fourteen (14) month
Voice Gateways System Manual
iii
Legal Rights
from invoice date (the "Warranty Period")". During the Warranty Period, Alvarion
may release to its Customers firmware updates, which include additional
performance improvements and/or bug fixes, upon availability (the “Warranty”).
Bug fixes, temporary patches and/or workarounds may be supplied as Firmware
updates.
Additional hardware, if required, to install or use Firmware updates must be
purchased by the Customer. Alvarion will be obligated to support solely the two
(2) most recent Software major releases.
ALVARION SHALL NOT BE LIABLE UNDER THIS WARRANTY IF ITS TESTING
AND EXAMINATION DISCLOSE THAT THE ALLEGED DEFECT IN THE PRODUCT
DOES NOT EXIST OR WAS CAUSED BY PURCHASER’S OR ANY THIRD
PERSON'S MISUSE, NEGLIGENCE, IMPROPER INSTALLATION OR IMPROPER
TESTING, UNAUTHORIZED ATTEMPTS TO REPAIR, OR ANY OTHER CAUSE
BEYOND THE RANGE OF THE INTENDED USE, OR BY ACCIDENT, FIRE,
LIGHTNING OR172.17.31.70 OTHER HAZARD.
Disclaimer
(a) The Software is sold on an "AS IS" basis. Alvarion, its affiliates or its licensors
MAKE NO WARRANTIES, WHATSOEVER, WHETHER EXPRESS OR IMPLIED,
WITH RESPECT TO THE SOFTWARE AND THE ACCOMPANYING
DOCUMENTATION. ALVARION SPECIFICALLY DISCLAIMS ALL IMPLIED
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UNITS OF PRODUCT (INCLUDING ALL THE SOFTWARE) DELIVERED TO
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DESIGNED, MANUFACTURED OR INTENDED FOR USE OR RESALE IN
APPLICATIONS WHERE THE FAILURE, MALFUNCTION OR INACCURACY OF
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ON LINE CONTROL SYSTEMS IN HAZARDOUS ENVIRONMENTS REQUIRING
FAIL SAFE PERFORMANCE, SUCH AS IN THE OPERATION OF NUCLEAR
FACILITIES, AIRCRAFT NAVIGATION OR COMMUNICATION SYSTEMS, AIR
TRAFFIC CONTROL, LIFE SUPPORT MACHINES, WEAPONS SYSTEMS OR
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FORTH IN THIS AGREEMENT ARE EXCLUSIVE AND IN LIEU OF ALL OTHER
WARRANTIES OR CONDITIONS, EXPRESS OR IMPLIED, EITHER IN FACT OR BY
Voice Gateways System Manual
iv
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OPERATION OF LAW, STATUTORY OR OTHERWISE, INCLUDING BUT NOT
LIMITED TO WARRANTIES, TERMS OR CONDITIONS OF MERCHANTABILITY,
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AND ARE NOT EXTENDED TO ANY THIRD PARTIES. ALVARION NEITHER
ASSUMES NOR AUTHORIZES ANY OTHER PERSON TO ASSUME FOR IT ANY
OTHER LIABILITY IN CONNECTION WITH THE SALE, INSTALLATION,
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PARTY, FOR ANY LOSS OF PROFITS, LOSS OF USE, INTERRUPTION OF
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CASE OF A BREACH OF A PARTY’S CONFIDENTIALITY OBLIGATIONS).
Voice Gateways System Manual
v
Important Notice
Important Notice
This user manual is delivered subject to the following conditions and restrictions:
This manual contains proprietary information belonging to Alvarion Ltd. Such
information is supplied solely for the purpose of assisting properly authorized
users of the respective Alvarion products.
No part of its contents may be used for any other purpose, disclosed to any
person or firm or reproduced by any means, electronic and mechanical,
without the express prior written permission of Alvarion Ltd.
The text and graphics are for the purpose of illustration and reference only.
The specifications on which they are based are subject to change without
notice.
The software described in this document is furnished under a license. The
software may be used or copied only in accordance with the terms of that
license.
Information in this document is subject to change without notice.
Corporate and individual names and data used in examples herein are
fictitious unless otherwise noted.
Alvarion Ltd. reserves the right to alter the equipment specifications and
descriptions in this publication without prior notice. No part of this
publication shall be deemed to be part of any contract or warranty unless
specifically incorporated by reference into such contract or warranty.
The information contained herein is merely descriptive in nature, and does
not constitute an offer for the sale of the product described herein.
Any changes or modifications of equipment, including opening of the
equipment not expressly approved by Alvarion Ltd. will void equipment
warranty and any repair thereafter shall be charged for. It could also void the
user’s authority to operate the equipment.
Some of the equipment provided by Alvarion and specified in this manual, is
manufactured and warranted by third parties. All such equipment must be
installed and handled in full compliance with the instructions provided by such
manufacturers as attached to this manual or provided thereafter by Alvarion or
the manufacturers. Non compliance with such instructions may result in serious
damage and/or bodily harm and/or void the user’s authority to operate the
equipment and/or revoke the warranty provided by such manufacturer.
Voice Gateways System Manual
vi
About This Manual
This manual describes Alvarion’s Voice Gateway units and how to install, operate
and manage them.
This manual is intended for technicians responsible for installing, setting up and
operating the Voice Gateway, and for system administrators responsible for
managing the Voice Gateways.
This manual contains the following chapters and appendices:
Chapter 1 – System Description: Describes the Voice Gateway and its
functionality.
Chapter 2 – Installation: Describes how to install the Voice Gateway and
connect it to the SU and to the user’s equipment.
Chapter 3 – Using the Web Configuration Server: Describes how to use the
Web Configuration Server for configuring parameters and checking system
status.
Appendix A – Internal Class 5 Services: Describes the internal Class-5
services that are supported by the Gateway.
Appendix B – Default Telephony Parameters: Describe the default values for
some telephony parameters, including signals/tones parameters, CID
parameters and line impedance.
Glossary: Provides definitions of various terms used in the manual.
Contents
Chapter 1 - System Description ...............................................................1
1.1 Introducing the Voice Gateway ................................................................................... 2
1.2 Specifications ............................................................................................................... 3
1.2.1 Telephony and Fax Services ............................................................................... 3
1.2.2 Security................................................................................................................ 3
1.2.3 Voice Quality........................................................................................................ 4
1.2.4 Configuration and Management .......................................................................... 4
1.2.5 Bridge Functionality ............................................................................................. 5
1.2.6 Mechanical........................................................................................................... 5
1.2.7 Electrical .............................................................................................................. 5
1.2.8 Connectors .......................................................................................................... 6
1.2.9 Regulatory Standards Compliance ...................................................................... 6
1.2.10 Environmental...................................................................................................... 7
Chapter 2 - Installation ............................................................................9
2.1 Installation Requirements .......................................................................................... 10
2.1.1 Packing List ....................................................................................................... 10
2.1.2 Additional Installation Requirements ................................................................. 10
2.2 Front and Rear Panel Components........................................................................... 11
2.2.1 Connectors ........................................................................................................ 11
2.2.2 Reset to Factory Default Configuration.............................................................. 11
2.2.3 LEDs .................................................................................................................. 12
Contents
2.3 Installation ................................................................................................................... 13
Chapter 3 - Using the Web Configuration Server...................................15
3.1 Introduction to the Web Configuration Server ......................................................... 16
3.2 Accessing the Web Configuration Server ................................................................ 17
3.3 Using the Web Configuration Server ........................................................................ 18
3.4 Home Menu - Product info Page................................................................................ 19
3.5 WAN Menu ................................................................................................................... 21
3.5.1 WAN Status Page .............................................................................................. 21
3.5.2 WAN Configuration Page................................................................................... 23
3.6 VLAN Tagging Menu ................................................................................................... 26
3.6.1 VLAN Tagging Page .......................................................................................... 26
3.6.2 Adding and Deleting VLANs .............................................................................. 27
3.6.3 VoIP VLAN Configuration Page ......................................................................... 29
3.6.4 VLAN Configuration Example 1 ......................................................................... 30
3.6.5 VLAN Configuration Example 2 ......................................................................... 32
3.7 Telephone Menu – SIP Model..................................................................................... 34
3.7.1 SIP Configuration Page ..................................................................................... 35
3.7.2 SIP Extensions Page ......................................................................................... 43
3.7.3 NAT Traversal Configuration Page .................................................................... 45
3.7.4 ToS/DiffServ Page ............................................................................................. 46
3.8 BW Reservation – DRAP Configuration Page .......................................................... 47
3.9 System Menu ............................................................................................................... 51
3.9.1 Set Security Password Page ............................................................................. 51
3.9.2 Localization Page............................................................................................... 52
3.9.3 SNMP Configuration Page................................................................................. 53
3.9.4 Service Access Configuration Page................................................................... 54
3.10
Upgrade Page................................................................................................... 55
Voice Gateways System Manual
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Contents
3.10.1 Downloader Result Codes (hexadecimal) ......................................................... 55
3.11
Restart Page..................................................................................................... 57
3.12
Parameters Summary...................................................................................... 58
Appendix A - Internal Class 5 Services..................................................65
A.1 Actions and Keypad Sequences ............................................................................... 66
A.2 Using the Class 5 Services ........................................................................................ 67
A.2.1 Call Waiting........................................................................................................ 67
A.2.2 Call Inquiry......................................................................................................... 67
A.2.3 Call Alteration .................................................................................................... 67
A.2.4 Call Drop............................................................................................................ 68
A.2.5 3-Party Conference 1......................................................................................... 68
A.2.6 3-Party Conference 2......................................................................................... 68
A.2.7 Call Waiting Indication Tone .............................................................................. 69
A.2.8 Call Forward ...................................................................................................... 69
Appendix B - Default Telephony Parameters.........................................71
Glossary..................................................................................................74
Voice Gateways System Manual
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Figures
Figure 2-1: Voice Gateway VG-1D2V Back Panel...................................................................................... 11
Figure 2-2: VG-1D2V Front Panel............................................................................................................... 12
Figure 3-1: Web Configuration Page .......................................................................................................... 18
Figure 3-2: Product Info Page..................................................................................................................... 19
Figure 3-3: WAN Status Page..................................................................................................................... 21
Figure 3-4: WAN Configuration Page ......................................................................................................... 23
Figure 3-5: VLAN Tagging Page................................................................................................................. 26
Figure 3-6: VLAN Editor (Add VLAN).......................................................................................................... 27
Figure 3-7: VLAN Editor (Delete VLAN)...................................................................................................... 28
Figure 3-8: VoIP VLAN Configuration Page................................................................................................ 29
Figure 3-9: VLAN Configuration Example 1................................................................................................ 30
Figure 3-10: VLAN Configuration Example 2.............................................................................................. 32
Figure 3-11: SIP Configuration Page (VG-1D2V) ....................................................................................... 35
Figure 3-12: Codecs and Fax Configuration Window – VG-1D1V ............................................................. 39
Figure 3-13: SIP Extensions Page.............................................................................................................. 43
Figure 3-14: NAT Traversal Configuration Extensions Page...................................................................... 45
Figure 3-15: ToS/DiffServ Page.................................................................................................................. 46
Figure 3-16: DRAP Configuration Page...................................................................................................... 47
Figure 3-17: Set Security Password Page .................................................................................................. 51
Figure 3-18: Localization Page ................................................................................................................... 52
Figure 3-19: SNMP Configuration Page ..................................................................................................... 53
Figure 3-20: Service Access Configuration Page ....................................................................................... 54
Figure 3-21: Upgrade Page ........................................................................................................................ 55
Figure 3-22: Restart Page........................................................................................................................... 57
Tables
Table 1-1: Telephony and Fax Services ....................................................................................................... 3
Table 1-2: Security ........................................................................................................................................ 3
Table 1-3: Voice Quality................................................................................................................................ 4
Table 1-4: Configuration and Management .................................................................................................. 4
Table 1-5: Bridge Functionality ..................................................................................................................... 5
Table 1-6: Mechanical Specifications ........................................................................................................... 5
Table 1-7: Electrical Specifications ............................................................................................................... 5
Table 1-8: Connectors................................................................................................................................... 6
Table 1-9: Standards Compliance ................................................................................................................ 6
Table 1-10: Environmental Specifications..................................................................................................... 7
Table 2-1: Voice Gateway Connectors ....................................................................................................... 11
Table 2-2: Voice Gateway LEDs................................................................................................................. 12
Table 3-1: Product Info Page Parameters .................................................................................................. 19
Table 3-2: WAN Status Page Parameters .................................................................................................. 22
Table 3-3: WAN Configuration Page Parameters....................................................................................... 23
Table 3-4: VLAN Page Parameters ............................................................................................................ 27
Table 3-5: VoIp VLAN Configuration Page Parameters ............................................................................. 29
Table 3-6: SIP Configuration Page Parameters ......................................................................................... 36
Table 3-7Codecs and Fax Configuration Page Parameters ....................................................................... 40
Table 3-8: SIP Extensions Page Parameters ............................................................................................. 43
Table 3-9: NAT Traversal Configuration Page Parameters ........................................................................ 45
Table 3-10: ToS/DiffServ Page Parameters ............................................................................................... 46
Table 3-11: DRAP Configuration Page Parameters ................................................................................... 47
Table 3-12: Set Security Password Page Parameters ............................................................................... 51
Tables
Table 3-13: Localization Page Parameters .................................................................................................52
Table 3-14: SNMP Configuration Page Parameters ...................................................................................53
Table 3-15: Upgrade Page Parameters ......................................................................................................55
Table 3-16: Parameters Summary ..............................................................................................................58
Table B-1: Default Telephony Parameters ..................................................................................................72
Voice Gateways System Manual
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1
Chapter 1 - System Description
In this Chapter:
Introducing the Voice Gateway, page 2
Specifications, page 3
Chapter 1 - System Description
1.1
Introducing the Voice Gateway
Alvarion's Voice Gateway enables operators and service providers using Alvarion’s
Broadband Wireless Access system to provide subscribers with a number of
broadband services transparently. The Voice Gateway enables bundling services
such as telephony (Voice over IP) and high speed Internet to end-users.
IP-telephony services are supported for standard analog phones or G3 fax
machines. The VG-1D1V has a single POTS interface, and the VG-1D2V has two
POTS interfaces. The Voice Gateways are available with either H.323 or SIP
standard, and support both narrow (compressed) and wideband (uncompressed)
speech codecs, silence suppression with comfort noise, line echo cancellation and
regional telephone parameters. Class 5 services such call waiting and 3-party
conference call are also supported.
Up to 3 telephones can be connected in series to each telephone port. Daisy
chaining of Voice Gateways enables the service provider to offer certain end
users, for example small offices, additional telephone numbers.
The Voice Gateway also supports Internet access or any other Ethernet based
services. The unit can be installed behind a router/NAT due to NAT traversal
support allowing signaling as well as voice packets to correctly reach Softswitch
or Gatekeeper for bi-directional call initiations. The Gateway can handle up to 16
simultaneous VLANs, enabling the operator to offer different services to different
end users behind the unit.
These Gateways incorporate the proprietary DRAP (Dynamic Resources Allocation
Protocol) protocol for automatic registration and allocation of resource. DRAP is a
protocol based on IP/UDP between the Gateway and a DRAP server (e.g. the
BreezeMAX base station). The protocol provides an auto-discovery mechanism for
the Gateway, so no specific configuration is required and the Gateway can
automatically locate and register with the DRAP server. The protocol uses a few
simple messages enabling a Voice Gateway to request resources when calls are
made, and the DRAP server to dynamically allocate them.
The Voice Gateways are designed for remote management and supervision using
either the built-in internal web server or SNMP.
The Voice Gateways are easily updated and upgraded as they support remote
software and configuration file download.
Voice Gateways System Manual
2
Specifications
1.2
Specifications
1.2.1
Telephony and Fax Services
Table 1-1: Telephony and Fax Services
Item
Description
H323 model: H323v2/4
VoIP Standard
SIP model: SIP (RFC 3261)
Internal Class 5
Services
Call Waiting, 3-party call, call hold and call alteration,
differentiated ringing tones (refer to Appendix A for more
details)
External Class 5
Services
Activation/deactivation of class 5 services supported by the
IP-telephony system
Fax
G3 compliant V.17 14.4 Kbps fax reception and transmission
using the T.38 standard (or in-band using G.711 codec)
Calling Number
FSK, DTMF
Identification (CNI)
3rd party initiated pause
External rerouting of media stream during speech, e.g. for
and rerouting
pre-paid calling card and record announcement
DTMF
In-band and out-band using H.245 and H.225
Regional Settings
Telephony signals, tones and cadences (see Default Telephony
Parameters, on page 71.
1.2.2
Security
Table 1-2: Security
Item
Description
VLAN
Support IEEE 802.1Q with up to 16 VLAN IDs
Authentication
Per call authentication and registration
3
Chapter 1 - System Description
1.2.3
Voice Quality
Table 1-3: Voice Quality
Item
Description
G.711 Ulaw
Voice Codecs
G.711 Alaw
G.729ab
IEEE 802.1p layer-2 prioritization
Prioritization
DiffServ layer-3 prioritization
Adaptive jitter buffer
General
Echo cancellation
Speech sampling rate: 10-60 ms
Silence suppression with comfort noise
1.2.4
Configuration and Management
Table 1-4: Configuration and Management
Item
Management Options
Description
Internal Web Server
SNMP
SNMP Agents
SNMPv1 client
MIB II (RFC 1213), Private MIB
Plug & Play Functionality
DHCP, including support messages option 60, 61, 43
Software Upgrade
Using TFTP
Configuration Download
Using TFTP
Voice Gateways System Manual
4
Specifications
1.2.5
Bridge Functionality
Table 1-5: Bridge Functionality
Item
Description
Supported Ethernet Devices
Up to 32 MAC addresses
Unknown address Forwarding
Forward Unknown
Policy
Bridge Aging Time
1.2.6
180 seconds
Mechanical
Table 1-6: Mechanical Specifications
Item
Details
Dimensions (W x D x H)
17.6 x 11 x 2.8 cm
Weight
1 kg
1.2.7
Electrical
Table 1-7: Electrical Specifications
Item
Details
Power Input
12 VDC from an external power supply, 100-240 VAC,
50-60 Hz, 2A max.
Power Consumption
10.5 W max.
5
Chapter 1 - System Description
1.2.8
Connectors
Table 1-8: Connectors
Connector
Description
LAN
Type
10/100Base-TX (RJ-45)
Cable Length
max 100 m.
Type
RJ-11
Number of Phones (REN)
Up to 5
Cable Length
Max. 500 m
Type
10/100Base-TX (RJ-45)
Cable Length
max 100 m.
PHONE
(1 – 2 in VD-1D2V)
WAN
12 VDC
1.2.9
Standard DC power jack to external power supply
Regulatory Standards Compliance
Table 1-9: Standards Compliance
Type
EMC
Standard
Low Voltage Directive (LVD) 73/23/EEC
Electromagnetic Compatibility Directive (EMC)
89/336/EEG
Safety
IEC 60950
CSA C22.2 No. 950-95/UL 1950
AS/NZS 3260
Emission
EN 55022:1998 Class B
EN 61000-3-2:1995
Harmonics; EN 61000-3-3:1995
Flicker; FCC part 15 (1998) Class B
AS/NZS 3548 (1995)
Immunity
EN 55024:1998
Voice Gateways System Manual
6
Specifications
1.2.10 Environmental
Table 1-10: Environmental Specifications
Item
Details
Operating temperature
0 o C to 50 o C
Operating humidity
10%-95% RH non condensing
7
2
Chapter 2 - Installation
In this Chapter:
Installation Requirements, page 10
Front and Rear Panel Components, page 11
Installation, page 13
Chapter 2 - Installation
2.1
Installation Requirements
2.1.1
Packing List
Voice Gateway with one (VG-1D1V) or two (VG-1D2V) Phone Ports
Power supply with a DC connecting cable
Mains power cable
2.1.2
Additional Installation Requirements
A straight Ethernet cable for connecting the WAN port to the SU-IDU
An Ethernet cable for connecting to the user’s data equipment (straight for
connecting to a PC, crossed for connecting to a hub/switch)
Standard phone cable(s) with RJ-11 connectors.
Mains plug adapter (if the power plug on the supplied mains power cable does
not fit local power outlets).
Portable PC with an Ethernet card and an Ethernet cable for configuring the
Voice Gateway parameters using a web browser.
Voice Gateways System Manual
10
Front and Rear Panel Components
2.2
Front and Rear Panel Components
2.2.1
Connectors
Figure 2-1: Voice Gateway VG-1D2V Back Panel
NOTE
The VG-1D1V has a single Phone connector.
Table 2-1: Voice Gateway Connectors
Name
Connector
Functionality
Phone 1
RJ-11
Connections to the user’s telephones
Phone 2 (VG-1D2V only)
RJ-11
Connections to the user’s telephones
LAN
10/100Base-T (RJ-45)
Connection to the user’s data equipment
WAN
10/100Base-T (RJ-45)
Connection to the SU-IDU
12 VDC
DC power jack
Connection to power supply
2.2.2
Reset to Factory Default Configuration
Press down the RESET button on the back of the unit for at least 5 seconds to
reset all configurable parameters back to their original default values. After
releasing the RESET button, the PWR, WAN and LAN LEDs bilnk twice, indicating
proper operation. The affect on the selected IP parameters acquisition method
depends on the time the RESET button is held in the pressed position:
If the RESET button is pressed down for 5 to 10 seconds: The unit will use
DHCP to get the IP parameters.
If the RESET button is pressed down for more than 10 seconds: The unit will
use the static (manually defined) IP parameters.
For more details on configuration of DHCP and static IP parameters, refer to WAN
Configuration Page on page 23.
11
Chapter 2 - Installation
2.2.3
LEDs
Figure 2-2: VG-1D2V Front Panel
NOTE
The VG-1D1V has a single Phone LED.
Table 2-2: Voice Gateway LEDs
Name
Symbol
Description
Phone service
indication
Phone 1
Functionality
Off –Phone line does not get IP
telephony services
On – Phone line is connected to the IPtelephony system
Phone service
indication
Phone 2
(VG-1D2V only)
Off –Phone line does not get IP
telephony services
On – Phone line is connected to the IPtelephony system
LAN
LAN port status
Off – Ethernet Link not detected
indication
On – Ethernet link connected, no
activity
Blinking – Ethernet link activity
WAN
WAN port status
indication
Off – Ethernet link not detected
On – Ethernet link connected, no
activity
Blinking – Ethernet link activity
POWER
PWR
Power Indication
Off – unit is not powered or power
failed
Green – power OK
Voice Gateways System Manual
12
Installation
2.3
Installation
The unit can be placed on a desktop or a shelf. The location should be selected
taking into account the necessary connections to mains power, SU-IDU and
user’s data/telephony equipment.
To install the Voice Gateway:
1
Use a straight Ethernet cable to connect the WAN port on the rear panel of
the unit to the Ethernet port of the SU-IDU. The length of the cable, together
with the length of the cable connecting the SU-IDU to the SU-ODU, should
not exceed 100m.
2
Connect the DC power cable of power supply to the 12 VDC jack on the rear
panel of the unit.
3
Connect the mains power cable to the power supply. Connect the other end of
the mains power cable to the AC mains.
NOTE
The color codes of the power cable are as follows:
Brown
Phase
~
Blue
Neutral
0
Yellow/Green
Ground
4
After power up, all fron panel LEDs bilnk once, and then the PWR, WAN and
LAN LEDs bilnk twice, indicating that the unit operates properly. Then the
PWR LED is lit. Other LEDs may also be lit, according to the status of the
WAN, LAN and Phone ports, as described in section 2.2.3.
5
Connect a PC to the Ethernet port using a straight Ethernet cable, and use a
web browser to configure the parameters of the Voice Gateway. See Chapter 3
- Using the Web Configuration Server for details.
NOTE
To enable remote management of the Voice Gateway, the Management & voice VLAN ID must be
configured.
6
Connect the 10/100 Base-T Ethernet connector to the data equipment. The
length of the Ethernet cable should not exceed 100m. Use a straight cable for
connecting to a PC, or a crossed cable for connecting to a hub/switch)
7
Use standard telephone cord(s) with RJ-11 termination to connect the
telephony equipment to the unit.
8
If needed, configure the basic parameters of the SU and allign its antenna for
optimal performance. For details refer to the applicable System Manual.
13
Chapter 2 - Installation
9
Verify that the SU is connected to the base station.
10 To verify data connectivity, from the end-user’s PC or from a portable PC
connected to the unit, try to connect to the Internet or to ping another unit in
the network.
11 Verify proper operation using the LED indicators (see Table 2-2).
Voice Gateways System Manual
14
3
Chapter 3 - Using the Web Configuration
Server
In this Chapter
Introduction to the Web Configuration Server, page 16
Accessing the Web Configuration Server, page 17
Using the Web Configuration Server, page 18
Home Menu - Product info Page, page19
WAN Menu, page 21
VLAN Tagging Menu, page 26
Telephone Menu – SIP Model, page 34
BW Reservation – DRAP Configuration Page, page 47
System Menu, 51
Upgrade Page, page 55
Restart Page, page 57
Chapter 3 - Using the Web Configuration Server
3.1
Introduction to the Web Configuration
Server
The Voice Gateway can be configured using the following methods:
The Web Configuration Server
An .ini-file loaded into the unit from a TFTP-server
This document describes the configuration using the Web Configuration Server.
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Accessing the Web Configuration Server
3.2
Accessing the Web Configuration Server
Follow the steps below to access the Web Configuration Server:
1
Connect the unit to the AC mains and to the SU-IDU.
2
If a DHCP server is being used, the unit may request an IP address during the
power up (depending on the .ini file in the unit).
3
If fixed IP address should be used, proceed as follows:
Press the RESET button on the back of the unit
Keep the RESET button pressed for more than 10 seconds. Make sure that
the unit reboots properly (all LEDs on the front panel will blink once, and
then the PWR, WAN and LAN LEDS blink twice)
Release the RESET button. The PWR, WAN and LAN LEDs should blink
twice, indicating proper operation of the unit.
After this sequence the Voice Gateway will be at "factory default" status, with
IP address 192.168.254.254 and subnet mask 255.255.255.0.
4
Open a web browser (Internet Explorer 5.5 or higher).
NOTE
Be sure to disable caching of web pages.
5
Enter the IP address of the unit in the address field.
6
If the Web Configuration Server is password protected, you will be prompted
to enter your password in order to login to the system. The default password
is installer .
7
The Web Configuration Server main view appears on the screen.
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Chapter 3 - Using the Web Configuration Server
3.3
Using the Web Configuration Server
The Web Configuration Server view consists of a number of menu links (to the
left). Clicking on each of them will display the configuration/status page for the
selected menu item, with the applicable content (configurable parameters/options
or status information) in the main area. Several pages include a page selection
bar at the top of the page, enabling selection between several pages related to the
same menu item.
Page Selection Bar
Main Menu
Page Main
Area
Figure 3-1: Web Configuration Page
CAUTION
Many pages include a “Save Settings” button. Click on the Save Settings button before selecting
another page/menu item, or before quitting the application. The Save Settings functionality in many
cases is per page – if you leave the page without clicking the Save Settings button, all the changes
in the page will be lost.
Changes to most of the settings are applied only after restarting the unit (refer to
section 3.11).
CAUTION
There is no control that the entered values are valid or have the correct format or range. If invalid
values are entered, access to the unit may be lost and in that case a factory default procedure must
be performed. Refer to section 2.2.2 for information about how to reset the Voice Gateway to
factory default parameters.
Voice Gateways System Manual
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Home Menu - Product info Page
3.4
Home Menu - Product info Page
The Product Info page provides general information on the Voice Gateway.
Figure 3-2: Product Info Page
The Product Info page includes the following components:
Table 3-1: Product Info Page Parameters
Parameter
Description
Name
The unit’s model
Mac address
The MAC address of the unit
Serial Number
The serial number of the unit
Product number
Not Used
Product revision
The hardware revision
Production week
Production date in the format <yy>w<ww>. <yy> is the year
(two last digits) and ww is the week (two digits).
Default configuration
The unit’s configuration
Downloader revision
The revision of the SW download SW module.
Reported download status
The status of the SW download operation. For more details
refer to Downloader Result Codes (hexadecimal) on page 55.
Main software revision
The unit’s main SW version
Operator defaults revision
The custom .ini file (if exists)
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Chapter 3 - Using the Web Configuration Server
In any case of contact with Alvarion Customer Service, include the Default
configuration, Downloader revision, Main software revision and Operator defaults
revision (.ini file) if exists.
Voice Gateways System Manual
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WAN Menu
3.5
WAN Menu
The WAN menu page includes settings related to the operation and functionality
on the WAN (network) side of the unit.
NOTE
Be careful when setting these parameters to avoid conflicts in the network.
The WAN page selection bar includes the following options:
WAN Status (WAN Status Page)
WAN Configuration (WAN Configuration Page)
3.5.1
WAN Status Page
Figure 3-3: WAN Status Page
The WAN Status page includes the following components:
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Chapter 3 - Using the Web Configuration Server
Table 3-2: WAN Status Page Parameters
Parameter
Description
Interface Status
Enabled
The administrative status of the WAN port: Yes or No. In current
version the administrative status cannot be disabled.
Service
The configured operation mode. In current version it is always
Bridge.
Protocol
The protocol used for data transmission: Ethernet or PPPoE
Interface Status
The operational status of the WAN port: Up or Down.
Network Settings
Dynamic IP Assignment
The method of configuring IP Address, Subnet Mask, Default
Gateway and DNS Address, as defined in the WAN
Configuration page:
Yes (via DHCP): the parameters are obtained from a DHCP
server.
No: the parameters are configured manually
IP Address
The IP address of the unit
MAC Address
The MAC address of the unit
Subnet Mask
The IP Subnet Mask
Default Gateway
The Default Gateway address
DNS Address
IP DNS Server address
Domain Name
The Domain Name as defined in the WAN Configuration page
VLAN Tag
The VLAN ID tag defined for management traffic
Priority Tag
The Priority tag defined for management traffic
Broadcast Limit
The upper limit on the bit rate of broadcast packets that will be
forwarded as a percentage of the incoming traffic bit rate, as
defined in the WAN Configuration page.
Multicast Limit
The upper limit on the bit rate of broadcast packets that will be
forwarded as a percentage of the incoming traffic bit rate, as
defined in the WAN Configuration page.
Click on the Update button to refresh the display.
Voice Gateways System Manual
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WAN Menu
3.5.2
WAN Configuration Page
Figure 3-4: WAN Configuration Page
The WAN Configuration page includes the following components:
Table 3-3: WAN Configuration Page Parameters
Parameter
Description
Device Operating Mode
The operating mode of the unit. In current version the
operation mode is always Bridge.
Obtain WAN configuration
dynamically
Select this option to obtain IP parameters from a DHCP
server. The default is that the option is selected.
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Chapter 3 - Using the Web Configuration Server
Table 3-3: WAN Configuration Page Parameters
Parameter
Description
Client identity
Applicable only if the “Obtain WAN configuration dynamically”
option is selected. The method used for identifying the client
(Option 61). The options are:
Standard: The unit’s MAC address
Custom: An identification string of up to 25 characters.
The default is null (an empty string)
Vendor ID
Applicable only if the “Obtain WAN configuration dynamically”
option is selected. The Vendor ID (Option 60). A string of up
to 25 characters. The default used by the unit is VoIP (not
displayed).
Specify fixed WAN
configuration
Select this option to configure the IP parameters manually.
The default is that the option is not selected (the Obtain WAN
configuration dynamically option is selected).
IP Address
Applicable only if the “Specify fixed WAN configuration” option
is selected. The IP address of the unit. The default is
192.168.254.254
Subnet Mask
Applicable only if the “Specify fixed WAN configuration” option
is selected. The IP Subnet Mask. The default is
255.255.255.0
Default Gateway
Applicable only if the “Specify fixed WAN configuration” option
is selected. The Default Gateway address. The default is
none (empty)
DNS Address
Applicable only if the “Specify fixed WAN configuration” option
is selected. IP DNS Server address. The default is none
(empty)
Host Name
The Host name for clients. A string of up to 25 characters.
The default is null (an empty string).
Domain Name
The Domain Name for client resolution. A string of up to 25
characters. The default is null (an empty string).
Broadcast Limit
The upper limit on the bit rate of broadcast packets that will
be forwarded as a percentage of the incoming traffic bit rate.
The default is 100%.
Voice Gateways System Manual
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WAN Menu
Table 3-3: WAN Configuration Page Parameters
Parameter
Description
Multicast Limit
The upper limit on the bit rate of broadcast packets that will
be forwarded as a percentage of the incoming traffic bit rate.
The default is 100%.
Click on the Save WAN Settings button before leaving the page to save the new
settings. The new settings will be applied after restarting the unit.
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Chapter 3 - Using the Web Configuration Server
3.6
VLAN Tagging Menu
The VLAN Tagging page selection bar includes the following options:
VLAN Tagging (VLAN Tagging Page)
VoIP VLAN Configuration (VoIP VLAN Configuration Page)
3.6.1
VLAN Tagging Page
The Voice Gateway supports 802.1Q VLAN standard, allowing IEEE 802 Local
Area Networks (LANs) of all types to be connected together with Media Access
Control (MAC) Bridges, as specified in ISO/IEC 15802-3. This standard defines
the operation of Virtual LAN (VLAN) Bridges that permit the definition, operation
and administration of Virtual LAN topologies within a bridged LAN infrastructure.
Figure 3-5: VLAN Tagging Page
The VLAN page enables defining up to 16 VLANs, and it includes the following
components:
Voice Gateways System Manual
26
VLAN Tagging Menu
Table 3-4: VLAN Page Parameters
Parameter
Description
Tagged Port Membership
A table displaying the defined VLANs. For details on modifying
the table refer to section 3.6.2 below.
Untagged VLAN ID
The VLAN ID that is defined for untagged data on the WAN port
(text box on the left side) and the LAN port (text box on the right
side). This is the VLAN ID that will be used internally for
incoming data.
The range for both parameters is from 1 to 4094.
Default VLAN ID
The text box on the left side is for the WAN port. This is the
VLAN defined for management frames (SNMP, HTTP, TFTP)
arriving on the WAN port.
The text box on the right side is for the LAN port. Typically this
field should remain empty. It is used to specify the VLAN for
non-VoIP traffic (RTP, Call Signaling) arriving on the LAN side
when Voice Gateways are daisy-chained.
The range for both parameters is from 1 to 4094.
3.6.2
Adding and Deleting VLANs
To add a VLAN:
1
Click on the Add VLAN button. The VLAN Editor (Add) is displayed:
Figure 3-6: VLAN Editor (Add VLAN)
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Chapter 3 - Using the Web Configuration Server
2
Enter the VLAN ID (1 to 4094), VLAN NAME (A descriptive string of printable
characters. Do not use special characters such as space or comma), and the
VLAN priority tag (0 to 7).
3
If applicable packets need to be tagged on the WAN/LAN port, check the
relevant Yes option. Otherwise check the No option. Note that only one VLAN
can be untagged on each port (or on both).
4
Click OK. The newly added entry will be added to the Tagged Port
Membership table.
To delete a VLAN from the Tagged Port Membership table:
1
Click on the row ID number of the entry you wish to remove. The VLAN Editor
(Delete) is displayed:
Figure 3-7: VLAN Editor (Delete VLAN)
2
Click on the Delete button. The entry will be removed from the Tagged Port
Membership table.
Voice Gateways System Manual
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VLAN Tagging Menu
3.6.3
VoIP VLAN Configuration Page
Figure 3-8: VoIP VLAN Configuration Page
The VoIP VLAN configuration page enables defining the following parameters:
Table 3-5: VoIp VLAN Configuration Page Parameters
Parameter
Description
Call Signaling
VLAN Tag
The VLAN ID tag for VoIP call signaling packets
Priority Tag
The Priority tag for VoIP call signaling packets
RTP
VLAN Tag
The VLAN ID tag for RTP and RTCP packets
Priority Tag
The Priority tag for RTP and RTCP packets
Typically, the same VLAN is used for management, call signaling and RTP. In this
case, the same VLAN and Priority Tags should be configured for management
(Default VLAN on WAN port in the VLAN Tagging page), Call Signaling and RTP.
However, the Voice Gateway supports separation of VLANs and allows defining 3
different VLANs for management, call signaling and RTP traffic.
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Chapter 3 - Using the Web Configuration Server
3.6.4
VLAN Configuration Example 1
This example describes how to define the following configuration:
VLAN ID 100, VLAN Priority 7 for Voice (call signaling, RTP and RTCP) and
Management packets on the WAN port.
No VLAN for data on the LAN and WAN ports.
Untagged
Untagged
VLAN100
POTS
Figure 3-9: VLAN Configuration Example 1
1
In the VLAN page, click Add VLAN to open the VLAN Editor.
2
In the VLAN Editor, enter the follwing for Voice and Management VLAN:
VLAN ID: 100
VLAN NAME: Voice&Mng
VLAN Priority: 7
WAN: Yes
LAN: No
3
Click OK to add the VLAN to the Tagged Port Membership table.
Voice Gateways System Manual
30
VLAN Tagging Menu
4
Enter the VLAN ID for Voice and Management (100) in the field Default VLAN
ID on WAN port, and click Save.
5
In the Page Selection bar, click on VoIP VLAN Configuration to open the VoIP
VLAN Configuration page. Enter 100 in the VLAN Tag fields for both Call
Signaling and RTP. Enter 7 in the Priority Tag field for both Call Signaling and
RTP. Click Save VoIP VLAN Settings. Go back to the VLAN Tagging page.
6
In the VLAN page, click Add VLAN to open the VLAN Editor.
7
In the VLAN Editor, enter the follwing for untagged data:
VLAN ID: 200 (an arbitrary selection-a VLAN ID is required for defining the
untagged data. This VLAN tag is only used internally in the unit)
VLAN NAME: Untagged
VLAN Priority: 0
WAN: No
LAN: No
8
9
Click OK to add the VLAN to the Tagged Port Membership table.
Enter the VLAN ID for untagged data (200) in the fields Untagged VLAN ID on
LAN port and Untagged VLAN ID on WAN port, and click Save.
10 Restart the unit to apply the changes.
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Chapter 3 - Using the Web Configuration Server
3.6.5
VLAN Configuration Example 2
This example describes how to define the following configuration:
VLAN ID 100, VLAN Priority 7 for Voice (call signaling, RTP and RTCP) and
Management packets on the WAN port.
Two types of data traffic:
Data with VLAN ID 200, Priority 3 on the LAN and WAN ports
Untagged (no VLAN) data on the LAN and WAN ports.
Untagged 200 100
POTS
200 Untagged
Figure 3-10: VLAN Configuration Example 2
1
In the VLAN page, click Add VLAN to open the VLAN Editor.
2
In the VLAN Editor, enter the follwing for Voice and Management VLAN:
VLAN ID: 100
VLAN NAME: Voice&Mng
VLAN Priority: 7
WAN: Yes
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VLAN Tagging Menu
LAN: No
3
Click OK to add the VLAN to the Tagged Port Membership table.
4
Enter the VLAN ID for Voice and Management (100) in the field Default VLAN
ID on WAN port, and click Save.
5
In the Page Selection bar, click on VoIP VLAN Configuration to open the VoIP
VLAN Configuration page. Enter 100 in the VLAN Tag fields for both Call
Signaling and RTP. Enter 7 in the Priority Tag field for both Call Signaling and
RTP. Click Save VoIP VLAN Settings. Go back to the VLAN Tagging page.
6
In the VLAN page, click Add VLAN to open the VLAN Editor.
7
In the VLAN Editor, enter the follwing for Data with VLAN 200:
VLAN ID: 200
VLAN NAME: DATA_200
VLAN Priority: 3
WAN: Yes
LAN: Yes
8
Click OK to add the VLAN to the Tagged Port Membership table.
9
In the VLAN Editor, enter the follwing for untagged data:
VLAN ID: 1000 (an arbitrary selection-a VLAN ID is required for defining
the untagged data. This VLAN tag is only used internally in the unit)
VLAN NAME: Untagged
VLAN Priority: 0
WAN: No
LAN: No
10 Click OK to add the VLAN to the Tagged Port Membership table.
11 Enter the VLAN ID for untagged data (200) in the fields Untagged VLAN ID on
LAN port and Untagged VLAN ID on WAN port, and click Save.
12 Restart the unit to apply the changes.
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Chapter 3 - Using the Web Configuration Server
3.7
Telephone Menu – SIP Model
The Telephone (SIP Model) page selection bar includes the following options:
SIP (SIP Configuration Page)
SIP Extensions (SIP Extensions Page)
NAT (NAT Traversal Configuration Page)
ToS/DiffServ (ToS/DiffServ Page)
Voice Gateways System Manual
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Telephone Menu – SIP Model
3.7.1
SIP Configuration Page
Figure 3-11: SIP Configuration Page (VG-1D2V)
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Chapter 3 - Using the Web Configuration Server
The SIP Configuration page includes the following components:
Table 3-6: SIP Configuration Page Parameters
Parameter
Description
Dialplan
The Dialplan parameter defines how the Voice Gateway decides
that a complete number has been dialed. See more details in
section 3.7.1.2 on page 41.
The default value is x.T|x.#, which means that each of the
following schemes can be used:
x.T: Dial timeout. Any number of digits may be dialed.
Following T seconds in which no new digit is dialed, a
decision is reached that dialing was completed and the unit
will send the dialing sequence received up to this time as a
complete telephone number. This is necessary since the
whole telephone number is sent at once and not digit by digit.
x.#: Any number of digits may be dialed. A decision that
dialing was completed will be reached once # is pressed.
The combination of both schemes means that dialing is
completed either after a timeout of T seconds or after pressing #.
Dial timeout
The timeout in seconds for the dial timeout dialplan. The number
of seconds that the unit waits before it sends a complete
telephone number. This is necessary since the whole telephone
number is sent at once and not digit by digit.
The range is 1 to 60 seconds
Default value is 4 seconds.
Adaptive Jitter Buffer
Maximum Duration
The Voice Gateway uses a Jitter Buffer to eliminate jitter effects.
The size of the buffer changes dynamically to reflect actual jitter
conditions. The Adaptive Jitter Buffer Maximum Duration defines
the maximum size that is available for the jitter buffer (the higher
the size, the higher the potential delay).
The range is 100 to 300 milliseconds.
The default duration is 100 milliseconds.
RTP Port Range
The start and end UDP port-range for RTP protocol.
Recommended values for Start and End ports are in the range
1030-65535.
The default Start port is 8000. The default End port is 8015.
Telephone line
Switch the telephone line On or Off. The default is Off.
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Telephone Menu – SIP Model
Table 3-6: SIP Configuration Page Parameters
Parameter
Description
HA mode
The High Availability mode defines the support of a secondary
Gate keeper/SIP Server for high system availability, redundancy,
and scalability. When a secondary server is available, the unit
will try to register to the secondary server after 10 failed attempts
to register to the primary server.
The available options are:
Fixed: The secondary SIP Server IP address is defined
manually by the SIP Server IP (secondary) parameter.
Auto: The secondary SIP Server/Proxy IP address is supplied
by the primary SIP Server.
Off: Secondary Gate keeper/SIP Server is not supported.
The default is Off.
SIP Server IP (primary)
The IP address for the primary SIP server/proxy who is
responsible for managing the Voice Gateway in the specific
network. If HA-mode is set to Auto, the primary SIP server/proxy
provides to the Voice Gateway during registration an IP-address
for the secondary system.
SIP Server Port (primary)
The port used for the primary system. The recommended values
are in the range 1030-65536. The default is 5060.
SIP Server IP
The IP address of the secondary SIP server/proxy.
(secondary)
SIP Server Port
(secondary)
The port used for the secondary system. The recommended
values are in the range 1030-65536. The default is 5060.
User Name
The SIP user Name. Format (name or number) depends on the
SIP server. A string of up to 25 characters.
Password
The SIP user Password. Format (name or number) depends on
the SIP server. A string of up to 25 characters.
Outgoing Display name
The name to be displayed on the caller ID display of a receiving
party (if supported by the network). Up to 25 characters.
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Chapter 3 - Using the Web Configuration Server
Table 3-6: SIP Configuration Page Parameters
Parameter
Description
Telephone number
The telephone number of the specific telephone line to be used
when registering the unit at the Gate keeper/SIP Server.
The telephone number is limited to 25 characters. It may also be
an e-mail address (limited to 25 characters before the @ sign).
The Telephone number must be set to a unique value for each
telephone line in the network in order for the system to accept it.
Telephone domain name
The domain-name. The Telephone domain name is limited to 25
characters, i.e. 25 characters after the @-sign. If not specified by
the user, the same information as defined in the SIP Server IP
field will be used.
Port
The number of the outgoing signaling port on the telephone line.
Line1 and Line2 cannot have the same port number. The range
is from 1030 to 65535. The default is 5060 for Line 1 and 5061
for Line 2.
Incoming CLIP
The Calling Line Identity Presentation (Caller ID) option for the
telephone line. If On is selected, the Caller ID information of a
calling party in incoming calls will be displayed on a caller ID
display attached to the telephone line.
Caller ID method should be configured by loading an appropriate
.ini file (refer to the Upgrade section for information).
The default is Off.
Keepalive timeout
(seconds)
The interval suggested by the Voice Gateway for sending the
keep alive messages to the network. If Keep-alive timeout is sent
from the network, it will override the setting in the Voice
Gateway. If the unit does not succeed to renew registration after
half of the timeout (600 seconds with the default timeout of 1200
seconds), it will continue the registration attempts, but the Phone
ports will be disabled until successful registration: no dial tone,
Phone LEDs are off.
The range is from 10 to 65535 seconds.
The default is 1200 seconds.
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Telephone Menu – SIP Model
Table 3-6: SIP Configuration Page Parameters
Parameter
Description
Ring signal [0 – 9]
The Ring signal parameter provides a selection of 10 different
ring patterns (0-9) that the unit can use.
The default is 0.
Transport
Configure whether signaling shall use UDP or TCP. The default
is UDP.
Preferred codecs
Displays the currently supported codecs, according to the
defined priorities.
Click the Set Codecs/Fax button to change codecs
settings/priorities.
NOTE: Click Save before clicking the Set Codecs/Fax button.
Otherwise, all configuration changes in the Telephone page will
be lost.
Click on the Save button before leaving the page to save the new settings. The
new settings will be applied after restarting the unit.
Click the Set Codecs/Fax button to change change codecs settings/priorities as
described below.
3.7.1.1
Codecs and Fax Configuration
After clicking the Set Codecs/Fax button, the Codecs and Fax Configuration page
is displayed.
Figure 3-12: Codecs and Fax Configuration Window – VG-1D1V
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Chapter 3 - Using the Web Configuration Server
The following settings are available for each line:
Table 3-7Codecs and Fax Configuration Page Parameters
Parameter
Description
Codec
The Codec check boxes identify which codecs are used. G.711A and G711U
are mandatory and cannot be deselected. G.729 is optional.
The default is G729 codec selected (checked).
NOTE: G 729 with Annex A is implemented in the Voice Gateway. It enables
communication with devices using either G729 with Annex A or G729 with
Annex A and Annex B. It is not possible to communicate with devices using
G729 with Annex B only.
For each Codec in use, the following can be configured:
SS
The SS (Silent Suppression) option for outgoing calls. When the SS option is
enabled, silence intervals are identified and only relevant information is
transmitted, using less bandwidth than during voice activity intervals. This
allows for a better overall utilization of the available bandwidth. It is possible to
enable Silent Suppression with G729 codec. Silent Suppression is not
applicable when using the G711 codecs.
The default (G729) is SS disabled.
EC
The EC (Echo Cancellation) option, defines whether to activate the echo
cancellation mechanism for improved voice quality. EC is not used during Fax
(T.38) transmissions.
The default is enabled.
Packet
The packet size in milliseconds.
The range is from 10 to 150 milliseconds.
The default is 30 ms for G729 and 20 ms for G711A and G711U.
Keypad
The "Keypad" field indicated which transmission method to be used for user
input DTMF signaling (i.e. phone banking). "None" means in-band, which
should be used with G.711 only. The options are None, RFC2833 and SIP
INFO. RFC2833 and SIP INFO should be used primarily with G.729 but could
also be used with G.711.
The default is None for G711 codecs and RFC2833 for G729.
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Telephone Menu – SIP Model
Table 3-7Codecs and Fax Configuration Page Parameters
Parameter
Description
Priority
The Priority parameter defines the relative priorities to be offered during
capabilities’ exchange.
If only G711A and G711U are used, the permitted priorities are 1 and 2. If all 3
codecs are used, the permitted priorities are 1, 2 and 3.
Voice codec negotiation/priority is always performed between 2 end-points and
depending on which side initiated the negotiation.
The default is Priority 1 to G729, Priority 2 to G711A, and Priority 3 to G711U.
T38 Fax
The T38 check box indicates for each line whether to support the T38 Fax
protocol.
The default is checked (T38 Fax supported).
Click on the Save button before leaving the page to save the new settings. The
new settings will be applied after restarting the unit.
3.7.1.2
Dial plan Schemes
A dial plan gives the unit a map to determine when a complete number has been
entered and should be passed to the gatekeeper for resolution into an IP address.
Dial plans are expressed using the same syntax as used by MGCP NCS
specification. The following notation describes the formal syntax of the dial plan:
Digit ::= "0" | "1" | "2" | "3" | "4" | "5" | "6" | "7" | "8" | "9"
Timer ::= "T" | "t"
Letter ::= Digit | Timer | "#" | "*" | "A" | "a" | "B" | "b" | "C" | "c" | "D" | "d"
Range ::= "X" | "x" -- matches any digit
| "[" Letters "]" -- matches any of the specified letters
Letters::= Subrange | Subrange Letters
Subrange::= Letter -- matches the specified letter
| Digit "-" Digit -- matches any digit between first and last
Position::= Letter | Range
StringElement::= Position -- matches any occurrence of the position
| Position "." -- matches an arbitrary number of occurrences including 0
String ::= StringElement | StringElement String
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StringList::= String | String "|" StringList
DialPlan::= String | "(" StringList ")"
A dial plan, according to this syntax, is defined either by a (case insensitive)
string or by a list of strings. Regardless of the above syntax a timer is only
allowed if it appears in the last position in a string (12T3 is not valid). Each string
is an alternate numbering scheme. The unit will process the dial plan by
comparing the current dial string against the dial plan. If the result is underqualified (partial matches at least one entry) then it will do nothing further but
wait until a full match is reached. If the result is over-qualified (no further digits
could possibly produce a match) then it aborts the dial attempt and notifies enduser with an audio signal. Only a full match will trigger to initiate a call, by
sending the dialed information to a Gatekeeper.
The Timer T is activated when it is all that is required to produce a match. The
period of timer T is 4 seconds as default (configurable). For example a dial plan of
(xxxT|xxxxx) will match immediately if any 5 digits are entered. It will also match
following a 4 second pause after entering 3 digits.
Example 1 (simple dial plan):
Following example allows dialing any 7-digit number (e.g. 5551234) or an
operator on 0. Dial plan is: (0T|xxxxxxx)
Example 2 (complex dial plan):
Local operator on 0, long distance operator on 00, four digit local extension
number starting with 3,4 or 5, seven digit local numbers are prefixed by an 8, two
digit star services (e.g. 69), ten digit long distance prefixed by 91, and
international numbers starting with 9011+variable number of digits.
Dial plan for this is: (0T|00T|[3-5]xxx|8xxxxxxx|*xx|91xxxxxxxxxx|9011x.T)
Call completion
Call completion means allowing user to skip the timer period T after finished
dialing, by ending number sequence with ‘#’ (no other character is valid for this
feature). A valid dial plan to accomplish this would be: (x.#|x.T)
Non-dialed Line Dial Plan
As soon as handset is lifted the unit contacts the gatekeeper (used for systems
where DTMF detection is done in-call, i.e. “Hotline” model). Dial plan is then (x.),
note the dot ‘.’, which means match against 0 (or more) digits.
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Telephone Menu – SIP Model
3.7.2
SIP Extensions Page
Figure 3-13: SIP Extensions Page
The SIP Extensions page includes the following components:
Table 3-8: SIP Extensions Page Parameters
Parameter
Description
Support PRACK method
with provisional response
reliability
The PRACK request plays the same role as ACK, but for
provisional responses. PRACK is a normal SIP message, like
BYE. As such, its own reliability is ensured hop-by-hop through
each stateful proxy. Also like BYE, but unlike ACK, PRACK has
its own response. If this were not the case, the PRACK message
could not traverse proxy servers compliant to RFC 2543. For
more details refer to RFC 3262: Reliability of Provisional
Responses in the Session Initiation Protocol (SIP).
Encode SIP URI with
user parameters
User=Phone will be inserted in the Contact field of SIP uniform
resource identifier (URI).
Encode default port in
SIP URI
Include default port in SIP uniform resource identifier (URI) even
though it is not mandatory according to standard.
Include default port in
INVITE
Include default port in the INVITE even though it is not
mandatory according to standard
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Table 3-8: SIP Extensions Page Parameters
Parameter
Description
Send INVITE with timer
header value
If the called user agents (UA) or the SIP Proxy Server (SPS)
requires a session timer for a requested session and the calling
UA does not include the Session-Expires header in the INVITE
message, then the called UA or the SPS may reject the request
with a 487-request failure message. If the use of a session timer
is desirable but optional for the session and the calling UA does
not include the Session-Expires header in the INVITE then the
called UA or SPS may add a Session-Expires header to the next
session setup message. In this case, the SPS shall add the
Session-Expires header to the INVITE message and the called
UA shall add the Session-Expires header to the 200 OK
response message. The range for the timer header value is from
1 to 999.
SIP Session timer value
The SIP Session Timer Support feature adds the capability to
periodically refresh Session Initiation Protocol (SIP) sessions by
sending repeated INVITE requests. The repeated INVITE
requests, or re-INVITEs, are sent during an active call leg to
allow user agents (UA) or proxies to determine the status of a
SIP session. Without this keep alive mechanism, proxies that
remember incoming and outgoing requests (stateful proxies) may
continue to retain call state needlessly. If a UA fails to send a
BYE message at the end of a session or if the BYE message is
lost because of network problems, a stateful proxy does not
know that the session has ended. The re-INVITES ensure that
active sessions stay active and completed sessions are
terminated. The range for the timer value is from 1 to 999
seconds.
Click on the Save SIP Extensions Settings button before leaving the page to
save the new settings. The new settings will be applied after restarting the unit.
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Telephone Menu – SIP Model
3.7.3
NAT Traversal Configuration Page
NAT Traversal function can be used to allow the Voice Gateway to register to a
SIP proxy server even though the Voice Gateway is connected behind a NAT
device.
Port forwarding may need to be activated for all telephone ports used by Voice
Gateway: For example, RTP port range and SIP signaling ports.
The Keepalive timeout parameter in the Telephony page may also need to be set
to a value lower than 1200 seconds to ensure that the Voice Gateway will not
loose registration to the SIP server.
Figure 3-14: NAT Traversal Configuration Extensions Page
The NAT Traversal Configuration page includes the following components:
Table 3-9: NAT Traversal Configuration Page Parameters
Parameter
Description
NAT IP Address
The IP address that the NAT device uses on the WAN side. If the Voice
Gateway is set to Auto NAT mode (see below), the IP address of the
outside IP will be automatically inserted.
NAT Mode:
The NAT mode:
On = Enable NAT Traversal function using manual setting.
Auto = Enter NAT mode if any of the following conditions is met:
a. IP-address = Private IP address
b. “received” parameter in INVITE or REGISTER IP-address is not
equal to internal IP address.
Off = NAT Traversal function is disabled.
The default is Off.
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Chapter 3 - Using the Web Configuration Server
Click on the Save button before leaving the page to save the new settings. The
new settings will be applied after restarting the unit.
3.7.4
ToS/DiffServ Page
Outgoing packets from the Voice Gateway can be marked with DSCP (DiffServ
Code Point) values. The ToS/DiffServ page enables defining the 8-bits ToS field in
the IP header for different packet types. Diffserv use the first 6 out of these 8 bits.
For more information about DiffServ Code Points please refer to RFC2474.
Figure 3-15: ToS/DiffServ Page
The ToS/DiffServ page includes the following components:
Table 3-10: ToS/DiffServ Page Parameters
Parameter
Description
Call signaling Packets
DiffServ marking for call signalling packets. Enter a number in
the range 0 to 255 (The first 6 bits is the value of the DSCP field)
or null. The default is 192.
RTP Packet
DiffServ marking for RTP and RTCP packets. Enter a number in
the range 0 to 255 (The first 6 bits is the value of the DSCP field)
or null. The default is 160.
SNMP Packets
DiffServ marking for SNMP packets. Enter a number in the range
0 to 255 (The first 6 bits is the value of the DSCP field) or null.
The default is 0.
Default setting
DiffServ marking for other types of packets (e.g. HTTP, TFTP).
Enter a number in the range 0 to 255 (The first 6 bits is the value
of the DSCP field) or null. The default is 0.
Click on the Save ToS/DiffServ Settings button before leaving the page to save
the new settings. The new settings will be applied after restarting the unit.
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BW Reservation – DRAP Configuration Page
3.8
BW Reservation – DRAP Configuration
Page
The Voice Gateway uses DRAP (Dynamic Resource Allocation Protocol) for
efficient management of bandwidth resources for telephone calls.
Figure 3-16: DRAP Configuration Page
The DRAP Configuration page includes the following components:
Table 3-11: DRAP Configuration Page Parameters
Parameter
Description
DRAP Server Settings
Enable DRAP
The Enable DRAP option defines whether DRAP is used for
establishing telephone (voice and fax) calls. If enabled, a DRAP
Server must be available to provision telephone calls.
The default is disabled (unchecked).
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Chapter 3 - Using the Web Configuration Server
Table 3-11: DRAP Configuration Page Parameters
Parameter
Description
Enable Pre-allocation
The Enable Pre-allocation option defines whether resource
allocation is requested immediately upon off-hook condition or
only after dialing the requested number. When disabled
(unchecked), a request for resource allocation will be sent only
after dialing the number. When enabled, the resource allocation
request will be sent immediately, and a dial tone will be provided
only if the requested resources are available.
The default is enabled (checked).
DRAP Server IP Address
The IP address of the DRAP server that should serve the
resource allocation requests of the unit. Leave empty for Auto
Discovery.
The default is an empty field (Auto Discovery).
Server Port
The UDP port used for the DRAP server. The port number
indicated will be used for originating ALLOC messages and the
port number indicated +1 will be used for receiving CONFRM
messages.
The available range is from 8000 to 8200.
The default is 8171.
DRAP Protocol Options
Discovery Time
The Discovery Time is the timeout to be used when the Auto
Discovery process is used for finding a DRAP server. The Auto
Discovery process is based on sending empty broadcast
allocation requests, and the Discovery Time is the time that the
unit will wait for a response before sending a new request.
The range is 1 to 255 seconds.
The default is 10 seconds.
Acknowledge Time
The Acknowledge Time is the timeout out to be used between
allocation requests. If no confirmation is received within this time,
a new allocation request should be sent.
The range is 1 to 10 (x 100 milliseconds).
The default is 3 (300 milliseconds).
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BW Reservation – DRAP Configuration Page
Table 3-11: DRAP Configuration Page Parameters
Parameter
Description
Clear Count
The Clear Count parameter indicates the number of allocation
requests (ALLOC) that can be sent without being acknowledged
before clearing all pending reservation attempts.
Note: Established reservations (existing calls) are not cleared.
The range is 1 to 10.
The default is 2.
Retry Count
The Retry Count parameter indicates the number of allocation
requests (ALLOC) that can be sent without being acknowledged
before reaching a decision that the unit should search for another
server. When this number is reached established reservations
are to be cleared (existing calls are disconnected) and auto
discovery procedure is initiated.
The range is 1 to 10.
The default is 5.
RTP Packing Ratio
The RTP Packing Ratio parameter defines the packet size to be
used until an actual call is established. It is recommended to set
a value that supports the worst-case scenario, e.g. the smallest
expected size (20 milliseconds) that results in the highest
expected number of packets per second.
NOTE: The configured RTP Packing Ratio is used by the unit
until an actual call is established. Once a call is established, the
unit will use a packet size according to the actual value being
used for the call.
The available range is 10 to 100 milliseconds in multiples of 10
(10, 20, …100).
The default value is 30 milliseconds.
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Chapter 3 - Using the Web Configuration Server
Table 3-11: DRAP Configuration Page Parameters
Parameter
Description
Vocoder Type
The Vocoder Type parameter defines the codec to be used until
an actual call is established. It is recommended to set a value
that supports the worst-case scenario, e.g. the codec with the
highest bandwidth requirement. Typically G711 should be
configured, except in networks where only G729 is used.
NOTE: The configured Vocoder Type is used by the unit until an
actual call is established. Once a call is established, the unit will
use the actual codec type being used for the call.
The available options are G711 and G729.
The default is G729.
Click on the Save DRAP Settings button before leaving the page to save the new
settings. The new settings will be applied after restarting the unit.
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System Menu
3.9
System Menu
The System page selection bar includes the following options:
Security (Set Security Password Page)
Localization (Localization Page)
SNMP (SNMP Configuration Page)
Service Access (Service Access Configuration Page)
3.9.1
Set Security Password Page
Figure 3-17: Set Security Password Page
The Set Security Password page includes the following components:
Table 3-12: Set Security Password Page Parameters
Parameter
Description
New password
Enter the new password. A password includes up to 20 printable
characters and is case sensitive.
A null (empty) string means no password.
Confirm new password
Enter the new password again (must be the same as above).
Click on the Save Password button before leaving the page to save the new
settings. The new settings will be applied after restarting the unit.
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Chapter 3 - Using the Web Configuration Server
3.9.2
Localization Page
Figure 3-18: Localization Page
The Localization page includes the following components:
Table 3-13: Localization Page Parameters
Parameter
Description
NTP Server
The IP address of the NTP-server (optional). If an IP address is
configured the NTP server usage is activated. The feature must
be activated to support FSK-based caller ID.
The default is disabled (no IP address).
Time Zone
The appropriate time zone. Use the drop-down list to change the
time zone.
Adjust clock for daylight
savings
By checking the “Adjust clock to daylight savings” the Voice
Gateway will automatically adjust to daylight saving time (set the
time one hour ahead).
The default is enabled (checked).
Click on the Save Localization Settings button before leaving the page to save
the new settings. The new settings will be applied after restarting the unit.
Voice Gateways System Manual
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System Menu
3.9.3
SNMP Configuration Page
Figure 3-19: SNMP Configuration Page
The SNMP Configuration page includes the following components:
Table 3-14: SNMP Configuration Page Parameters
Parameter
Description
SNMP Trap Configuration
Trap Destination 1 to
Trap destination 6
Specify up to 6 IP addresses to which SNMP traps should be
sent. If all Trap Destinations are null, SNMP traps will be sent as
broadcasts.
SNMP MIB Parameter Configuration
Read Community
The read community string, up to 20 printable characters, case
sensitive.
Default string is public.
Write Community
The write community string, up to 20 printable characters, case
sensitive.
Default string is private
Click on the Save SNMP Settings button before leaving the page to save the new
settings. The new settings will be applied after restarting the unit.
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Chapter 3 - Using the Web Configuration Server
3.9.4
Service Access Configuration Page
Figure 3-20: Service Access Configuration Page
The Service Access Configuration page enables to enable/disable access to
various services. Access from each of the ports (LAN or WAN) using HTTP and/or
SNMP can be either enabled or disabled. The default for all options is enabled
(checked).
Click on the Save Service Access Settings button before leaving the page to save
the new settings. The new settings will be applied after restarting the unit.
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54
Upgrade Page
3.10
Upgrade Page
The Upgrade page enables to control the process of downloading either a software
file (with the extension .ro) or a configuration file (with the extension .ini) from a
TFTP-server.
Figure 3-21: Upgrade Page
The Upgrade page includes the following components:
Table 3-15: Upgrade Page Parameters
Parameter
Description
TFTP Server
The IP address of the TFTP server
File name
The file name in the TFTP server of the software or the
configuration .ini file. Up to 25 characters.
Click on the Start TFTP Download button to start the download process. The
downloading and installation of the new SW version or configuration file is done
automatically, including a restart of the unit. When the installation is complete
and the unit has restarted, the Home Product Info page will be displayed.
3.10.1 Downloader Result Codes (hexadecimal)
If something goes wrong during download or installation, you will be notified
according to the following:
0 (0x00): normal boot (no upgrade requested or needed)
bit-0 (0x01): upgrade requested or main application not valid
bit-1 (0x02): failed to download new image
bit-2 (0x04): TFTP server not defined
bit-3 (0x08): TFTP file not defined
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Chapter 3 - Using the Web Configuration Server
bit-4 (0x10): TFTP session failed
bit-5 (0x20): CRC error in downloaded image
bit-6 (0x40): incompatible image
3.10.1.1
Examples
An attempt to download from a non-existing TFTP-server results in code 0x7
(= 0x07):
bit-2 0x04 TFTP server not defined plus…
bit-1 0x02 failed to download new image plus…
bit-0 0x01 upgrade requested or main application not valid
An attempt to download a non-existing file results in code 0xb (= 0x0b):
bit-3 0x08 TFTP file not defined plus…
bit-1 0x02 failed to download new image plus…
bit-0 0x01 upgrade requested or main application not valid
A successful download results in code 0x01
A restart without download of main application results in 0x00.
Voice Gateways System Manual
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Restart Page
3.11
Restart Page
When settings have been inserted or altered, the Voice Gateway must be restarted
in order to apply the new settings.
Figure 3-22: Restart Page
Click the Restart button to restart the Voice Gateway.
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Chapter 3 - Using the Web Configuration Server
3.12
Parameters Summary
Table 3-16: Parameters Summary
Parameter
Range/Options
Default
Device Operating Mode
Only Bridge option is available
Bridge
Obtain WAN configuration
dynamically
Yes (checked)/No
(unchecked)
Yes (selected)
Client identity
Standard/Custom
Standard
The Custom string can
include up to 25 characters
The default Custom string is
null
Vendor ID
A string of up to 25 characters
VoIP (used by default but is
not displayed)
Specify fixed WAN
Yes (checked)/No
No (unchecked)
configuration
(unchecked)
IP Address
IP address
192.168.254.254
Subnet Mask
IP address
255.255.255.0
Default Gateway
IP address
Null
DNS Address
IP address
Null
Host Name
A string of up to 25 characters
Null
Domain Name
A string of up to 25 characters
Null
Broadcast Limit
0%-100%
100%
Multicast Limit
0%-100%
100%
WAN Configuration Page
VoIP VLAN Configuration Page
Call Signaling VLAN Tag
1-4094 or null
Null
Call Signaling Priority Tag
0-7 or null
Null
RTP VLAN Tag
1-4094 or null
Null
RTP Priority Tag
0-7 or null
Null
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Parameters Summary
Table 3-16: Parameters Summary
Parameter
Range/Options
Default
Dialplan
A string of up to 100
characters. For details on
format see section 3.7.1.2
x.T|x.#
Dial Timeout
1-60 seconds
4 seconds
Adaptive Jitter Buffer
100-300 milliseconds
100 milliseconds
Start/Stop: 1030-65535
Start: 8000
SIP Configuration Page
Maximum Duration
RTP Port Range
End: 8015
Telephone Line
On/Off
Off
HA Mode
Fixed/Auto/Off
Off
SIP Server IP (primary)
IP address
Null
SIP Server Port (primary)
1030-65535
5060
SIP Server IP (secondary)
IP address
Null
SIP Server Port (secondary)
1030-65535
5060
User Name
A string of up to 25 characters
Null
Password
A string of up to 25 characters
Null
Outgoing Display Name
A string of up to 25 characters
Null
Telephone number
A string of up to 25 characters
Null
Telephone domain name
A string of up to 25 characters
Null
Port
1030-65535
Line 1: 5060
Line 2: 5061
Incoming CLIP
On/Off
Off
Keepalive timeout
10-65535 seconds
1200 seconds
Ring signal
0-9
0
Transport
UDP/TCP
UDP
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Table 3-16: Parameters Summary
Parameter
Range/Options
Default
Codecs and Fax Configuration
G711A
G711U
G729
Select/Deselect
Yes (checked)/No(unchecked)
SS
Not applicable
EC
Enable/Disable
Enabled
Packet
10-150 milliseconds
20 ms
Keypad
None, RFC2833 or SIP INFO
None
Priority
1-3
2
Select/Deselect
Yes (checked)/No(unchecked)
Always selected
SS
Not applicable
EC
Enable/Disable
Enabled
Packet
10-150 milliseconds
20 ms
Keypad
None, RFC2833 or SIP INFO
None
Priority
1-3
3
Select/Deselect
Yes (checked)/No(unchecked)
Yes (checked)
SS
Enable/Disable
Disabled
EC
Enable/Disable
Enabled
Packet
10-150 milliseconds
30 ms
Keypad
None, RFC2833 or SIP INFO
RFC2833
Priority
1-3
1
Enable/Disable
Enable
Yes/No
No (unchecked)
Yes/No
No (unchecked)
T38 Fax
Always selected
SIP Extensions Page
Support PRACK method
with provisional response
reliability
Encode SIP URI with user
parameters
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Parameters Summary
Table 3-16: Parameters Summary
Parameter
Range/Options
Default
Encode default port in SIP
URI
Yes/No
No (unchecked)
Include default port in
INVITE
Yes/No
Yes (checked)
Send INVITE with timer
header value
Yes/No, plus a value in the
range 1-999 seconds if Yes
(checked) is selected.
No (unchecked)
SIP Session timer value
Yes/No, plus a value in the
No (unchecked)
range 1-999 seconds if Yes
(checked) is selected.
The default value is null
The default value is null
NAT Traversal Configuration Page
NAT IP Address
IP address
Null
NAT Mode
On/Auto/Off
Off
Call signaling Packets
0-255 or null
192
RTP Packets
0-255 or null
160
SNMP Packets
0-255 or null
0
Default setting
0-255 or null
0
Enable DRAP
Enable/Disable
Disable (unchecked)
Enable Pre-allocation
Enable/Disable
Enable (checked)
IP address or null for auto
Null (auto discovery)
ToS/Diffserv Page
DRAP Configuration Page
DRAP Server IP Address
discovery
Server Port
8000-8200
8171
Discovery Time
1-255 seconds
10 seconds
Acknowledge Time
1 to 10 (x 100 milliseconds)
3 (300 milliseconds)
Clear Count
1-10
2
Retry Count
1-10
5
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Table 3-16: Parameters Summary
Parameter
Range/Options
Default
RTP Packing Ratio
10 to 100 milliseconds in
multiples of 10 (10, 20, …100)
30 milliseconds
Vocoder Type
G711/G729
G729
Set security Password Page
New password/ Confirm new
Up to 20 printable characters,
password
case sensitive. A null (empty)
string means no password.
Localization Page
NTP Server
IP address or null for disable
NTP server
Null (NTP server disabled)
Time Zone
Drop Down Menu
GMT+01:00
Adjust clock for daylight
savings
Yes/No
Yes (checked)
IP addresses.
Null for all addresses
SNMP Configuration Page
Trap Destination 1 to Trap
destination 6
If all Trap Destinations are
null, SNMP traps will be sent
as broadcasts.
Read Community
Up to 20 printable characters,
case sensitive.
public
Write Community
Up to 20 printable characters,
private
case sensitive.
Service Access Configuration Page
HTTP LAN
Yes/No
Yes (checked)
HTTP WAN
Yes/No
Yes (checked)
SNMP LAN
Yes/No
Yes (checked)
SNMP WAN
Yes/No
Yes (checked)
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Parameters Summary
Table 3-16: Parameters Summary
Parameter
Range/Options
Default
Upgrade Page
TFTP Server
IP address
File name
A string of up to 25 characters
Operation and Administration
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A
Appendix A - Internal Class 5 Services
In This Appendix:
This appendix provides a description of the internal Class 5 services that are
supported by the Voice Gateway.
Chapter 3 - Using the Web Configuration Server
A.1
Actions and Keypad Sequences
Keypad Sequences
Action
Description
Keypad Sequence
(R=hook-flash)
HOLD
Holds an on-going call
R0
DROP
Drops an on-going call
R1
FLASH
Switches between on-going
R2
call sessions or starts new call
inquiry
CONFERENCE
Activates 3-party conference
R3
CONFERENCE DROP
Deactivates 3-party conference
R5
CW ACTIVATION
Enables Call Waiting indication
tone
*43#
CW DEACTIVATION
Disable Call Waiting indication
tone
#43#
CW STATUS CHECK
Informs about the current
*#43#
configuration of Call Waiting
indication tone
CALL FORWARD
Call Forward activation
*21* <telephone number>#
Call Forward de-activation
#21#
ACTIVATION
CALL FORWARD
DEACTIVATION
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A.2
Using the Class 5 Services
A.2.1
Call Waiting
Description: One on-going call active, audible CW tone indicating new incoming
call in progress.
Call Waiting Service
A.2.2
Action
Event
R0
Reject incoming call -> Calling party hears busy tone. Continue with active call.
R1
Disconnect on-going call and answer incoming call.
R2
Place on-going call on hold, answer incoming call.
Call Inquiry
Description: One on-going call active, place a new call to a third party.
Call Inquiry Service
A.2.3
Action
Event
R2+telephone number
Place on-going call on hold (dial tone), Inquire new call to a
third party.
R1
Return to call placed on hold if third party is not answering.
Call Alteration
Description: Two on-going calls active, switch between calls.
Call Alteration Service
Action
Event
R2
Switch between two on-going calls. Places non-active call on hold.
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A.2.4
Call Drop
Description: Two on-going calls active, disconnect one of the calls.
Call Drop Service
A.2.5
Action
Event
R0
Disconnect call that is put on hold. Continue with on-going call.
R1
Disconnect on-going call and return to call that is put on hold.
3-Party Conference 1
Description: One on-going call active, place a new call to a third party and start
conference.
3-Party Conference Service 1
Action
Event
R3+telephone number
Place on-going call on hold (dial tone), inquire new call to a third
party and mix all session into a conference when third party has
answered.
R5
End conference with third party and return to first initiated call
session.
A.2.6
3-Party Conference 2
Description: Two on-going calls active, mix them into a conference session.
3-Party Conference Service 2
Action
Event
R3
Start conference with all active parties (mix audio streams).
R5
End conference with third party and return to first initiated call session.
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A.2.7
Call Waiting Indication Tone
Description: Available only when there are no calls active/in progress.
Call Waiting Indication Tone Service
Action
Event
*43#
Enable Call Waiting indication tone
#43#
Disable Call Waiting indication tone (calling party will hear a busy tone when
calling)
*#43#
Informs about present Call Waiting indication tone configuration:
Three short beeps = off
Two long beeps = on
A.2.8
Call Forward
Description: Available only when there are no calls active/in progress.
Call Forward Service
Action
Event
*21*<telephone number>#
Enable Call Forward and do forward to <telephone
number>. Indication tone is heard.
#21#
Deactivate Call Forward
Operation and Administration
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B
Appendix B - Default Telephony
Parameters
In This Appendix:
This appendix provides the default settings for various telephony parameters.
Appendix B - Default Telephony Parameters
Table B-1: Default Telephony Parameters
Parameter
Definition
Default
Normal Ringing
The signal that end user will
Cadence: 1 second on, 4 second off
Signal
hear from the telephone set
when a call is received
Duration: 180 seconds
Frequency: 25 Hz
Ringing Tone
The tone that sounds on the
telephone set when ringing
on the other side.
Cadence: 1 second on, 5 second off
Duration: Not limited
Frequency: 425 Hz
Level: -10 dBmO
Dial Tone
Busy Tone
The tone that the call
originator hears in the
handset before dialing the
destination telephone
number.
Cadence: Continuous
The tone that the end user
Cadence: 0.25 second on, 0.25 second
that originates a call hears
when the destination
telephone line is busy.
off
Duration: Not limited
Frequency: 425 Hz
Level: -5 dBmO
Duration: Not limited
Frequency: 425 Hz
Level: -10 dBmO
Network Busy Tone
The tone that the end user
Cadence: 0.25 second on, 0.75 second
that originates a call will hear
when the network is
congested.
off
Duration: Not limited
Frequency: 425 Hz
Level: -10 dBmO
Call Waiting Tone
The tone that the end user
that originates the call hears
when there is a second
incoming call during the
original call session.
Cadence: 0.2 second on, 0.5 second off,
0.2 seconds on
Duration: One On-Off-On cycle
Frequency: 425 Hz
Level: -10 dBmO
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Using the Class 5 Services
Table B-1: Default Telephony Parameters
Parameter
Definition
Default
Caller ID Standard
The country standard and
data transmission method
Sweden, DTMF
CID Alerting
Method
The method of alerting on the
existence of CID data
Polarity Reversal
On-hook data
transmission timing
method
The timing for transmission of
CID data
Post first ring signal
2-Wire Impedance
The impedance presented
CTR21
between the A and B wires of
the telephone line in active
state (nominal impedance).
(ETSI complex = 270Ω+750Ω||150nF))
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Glossary
BW
Band Width
CLIP
Calling Line Identification Presentation: A supplementary service
used to show the number of a caller.
CNI
Calling Number Identification.
DHCP
Dynamic Host Configuration Protocol. A protocol for dynamically
assigning IP addresses from a pre-defined list to nodes on a
network. Using DHCP to manage IP addresses simplifies client
configuration and efficiently utilizes IP addresses.
DNS
Domain Name System: The name resolution system that lets
users locate computers on the Internet (TCP/IP network) by
domain name. The DNS server maintains a database of domain
names (host names) and their corresponding IP addresses.
DRAP
Dynamic Resource Allocation Protocol
DSCP
Differentiated Service Code Point, AKA DiffServ: An alternate use
for the ToS byte in IP packets. Six bits of this byte are being
reallocated for use as the DSCP field where each DSCP specifies a
particular per-hop behavior that is applied to the packet.
DTMF
Dual-Tone Multi Frequency: The type of audio signals that are
generated when you press the buttons on a touch-tone
telephone. DTMF assigns a specific frequency (consisting of two
separate tones) to each key.
EC
Echo Cancellation
EMC
Electro-Magnetic Compatibility. The capability of equipment or
systems to be used in their intended environment within
designed efficiency levels without causing or receiving
degradation due to unintentional EMI (Electro Magnetic
Interference). EMC generally encompasses all of the
electromagnetic disciplines.
Glossary
FSK
Frequency Shift Keying: A simple modulation technique that
merges binary data into a carrier. It creates only two changes in
frequency: one for 0, another for 1.
G.711
A 64 Kbps PCM voice-coding technique. Described in the ITU-T
standard in its G-series recommendations.
G.729
A compression technique where voice is coded into 8 Kbps
streams. There are two variations of this standard (G.729 and
G.729 Annex A) that differ mainly in computational complexity;
both provide speech quality similar to 32-kbps ADPCM.
Described in the ITU-T standard in its G-series
recommendations.
H.225
An ITU standard protocol for control signaling in an H.323 audio
or video environment. H.225 uses messages defined in H.245 to
establish the call over the Registration, Admission and Signaling
(RAS) channel.
H.245
An ITU standard protocol for control messages in an H.225 audio
or videoconferencing call. The messages are used for flow control,
encryption and jitter management as well as for initiating the
call, negotiating which features should be used and terminating
the call. It also determines which side is the master for issuing
various commands.
H.323
A protocol suite defined by ITU-T for voice transmission over
internet (Voice over IP or VoIP). In addition to voice applications,
H.323 provides mechanisms for video communication and data
collaboration, in combination with the ITU-T T.120 series
standards.
HA
High Availability.
HTTP
HyperText Transport Protocol: A protocol used to request and
transmit files, especially web pages and web page components,
over the Internet or other computer network.
IDU
Indoor Unit
IEC
International Electrotechnical Commission, Geneva, Switzerland,
www.iec.ch: An organization that sets international electrical and
electronics standards founded in 1906. It is made up of national
committees from over 60 countries.
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Glossary
IEEE
Institute of Electrical and Electronics Engineers. IEEE
(pronounced I-triple-E) is an organization composed of engineers,
scientists, and students. The IEEE is best known for developing
standards for the computer and electronics industry. In
particular, the IEEE 802 standards for local-area networks are
widely followed.
IEEE 802.1p
A QoS method - A three-bit value that can be placed inside an
802.1Q frame tag.
IEEE 802.1Q
The IEEE 802.1Q standard defines the operation of VLAN Bridges
that permit the definition, operation and administration of
Virtual LAN topologies within a Bridged LAN infrastructure. The
802.1Q specification establishes a standard method for inserting
VLAN membership information into Ethernet frames. A tag field
containing VLAN (and/or 802.1p priority) information can be
inserted into an Ethernet frame, carrying VLAN membership
information.
IETF
Internet Engineering Task Force. One of the task forces of the
IAB (Internet Architecture Board), formally called the Internet
Activities Board, which is the technical body that oversees the
development of the Internet suite of protocols (commonly referred
to as "TCP/IP").The IETF is responsible for solving short-term
engineering needs of the Internet.
IP
Internet Protocol. The standard that defines how data is
transmitted over the Internet. IP bundles data, including e-mail,
faxes, voice calls and messages, and other types, into "packets",
in order to transmit it over public and private networks.
ISO
International Organization for Standardization, Geneva,
www.iso.ch: An organization that sets international standards,
founded in 1946. The U.S. member body is ANSI. ISO deals with
all fields except electrical and electronics, which is governed by
the older International Electrotechnical Commission (IEC). With
regard to information processing, ISO and IEC created JTC1, the
Joint Technical Committee for information technology.
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Glossary
ITU-T
International Telecommunication Union – Telecommunications.
An intergovernmental organization through which public and
private organizations develop telecommunications. The ITU was
founded in 1865 and became a United Nations agency in 1947. It
is responsible for adopting international treaties, regulations and
standards governing telecommunications. The standardization
functions were formerly performed by a group within the ITU
called CCITT, but after a 1992 reorganization the CCITT no
longer exists as a separate entity.
LAN
Local area Network. A computer network limited to a small
geographical area, such as a single building. The network
typically links PCs as well as shared resources such as printers.
MAC
Media Access Control. The lower of the two sub-layers of the data
link layer defined by the IEEE. The MAC sub-layer handles
access to shared media, such as whether token passing or
contention will be used.
MAC Address
Standardized data link layer address that is required for every
port or device that connects to a LAN. Other devices in the
network use these addresses to locate specific ports in the
network and to create and update routing tables and data
structures. MAC addresses are 6bytes long and are controlled by
the IEEE.
MIB
Management Information Base. A database of objects that can be
monitored by a network management system. SNMP uses
standardized MIB formats that allow any SNMP tools to monitor
any device defined by a MIB.
NAT
Network Address Translation: An IETF standard that allows an
organization to present itself to the Internet with far fewer IP
addresses than there are nodes on its internal network. The NAT
technology, which is typically implemented in a router, converts
private IP addresses (such as in the 192.168.0.0 range) of the
machine on the internal private network to one or more public IP
addresses for the Internet. It changes the packet headers to the
new address and keeps track of each session. When packets
come back from the Internet, NAT performs the reverse
conversion to the IP address of the client machine.
NTP
Network Time Protocol: A protocol used to update the real time
clock in a computer. There are numerous primary and secondary
servers in the Internet that are synchronized to the Coordinated
Universal Time (UTC) via radio, satellite or modem. For more
information, visit www.ntp.org.
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Glossary
POTS
Plain Old Telephone System. A basic analog telephone
equipment.
PPPoE
Point-to-Point Protocol over Ethernet. PPPoE relies on two widely
accepted standards: PPP and Ethernet. PPPoE is a specification
for connecting the users on an Ethernet to the Internet through a
common broadband medium, such as a single DSL line, wireless
device or cable modem. All the users over the Ethernet share a
common connection, so the Ethernet principles supporting
multiple users in a LAN combines with the principles of PPP,
which apply to serial connections.
RTP
Real Time Protocol. An Internet protocol for transmitting realtime data such as audio and video. RTP itself does not guarantee
real-time delivery of data, but it does provide mechanisms for the
sending and receiving applications to support streaming data.
Typically, RTP runs on top of the UDP protocol, although the
specification is general enough to support other transport
protocols.
SIP
Session Initiation Protocol. An application-layer control IETF
protocol that can establish, modify, and terminate multimedia
sessions such as Internet telephony calls (VoIP). SIP can also
invite participants to already existing sessions, such as multicast
conferences. Media can be added to (and removed from) an
existing session. SIP transparently supports name mapping and
redirection services, which supports personal mobility - users
can maintain a single externally visible identifier regardless of
their network location.
SNMP
Simple Network Management Protocol. A network management
protocol that provides a means to monitor and control network
devices, and to manage configurations, statistics collection,
performance, and security. SNMP works by sending messages,
called protocol data units (PDUs), to different parts of a network.
SNMP-compliant devices, called agents, store data about
themselves in Management Information Bases (MIBs) and return
this data to the SNMP requesters.
SS
Silence Suppression: A method for eliminating wasted bandwidth
when sending voice over a packet-switched system.
SU
Subscriber Unit
T.38
This is the ITU-T recommendation which defines a real time
method for fax over IP networks.
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Glossary
TCP
Transmission Control Protocol. Connection-oriented transport
layer protocol that provides reliable full-duplex data
transmission. TCP is the part of the TCP/IP suite of protocols
that is responsible for forming data connections between nodes
that are reliable, as opposed to IP, which is connectionless and
unreliable.
TCP/IP
Transmission Control Protocol/Internet Protocol. A set of
protocols developed by the U.S. Department of Defense to allow
communication between dissimilar networks and systems over
long distances. TCP/IP is the de facto standard for data
transmission over networks, including the Internet.
TFTP
Trivial File Transfer Protocol. Simplified version of FTP that
allows files to be transferred from one computer to another over a
network, usually without the use of client authentication.
ToS
Type Of Service: A field in an IP packet (IP datagram) that is used
for quality of service (QoS).
UDP
User Datagram Protocol. Connectionless transport layer protocol
in the TCP/IP protocol stack. UDP is a simple protocol that
exchanges datagrams without acknowledgments or guaranteed
delivery, requiring that error processing and retransmission be
handled by other protocols. UDP is defined in RFC 768.
V.17
An ITU fax standard (1991) that uses TCM (Trellis-Coded
Modulation, a technique for forward error correction) modulation
at 12,000 and 14,400 bps for Group 3 faxes. It adds TCM to the
V.29 standard at 7,200 and 9,600 bps to allow transmission over
noisier lines. It also defines special functions (echo protection,
turn-off sequences, etc.) for half-duplex operation. Modulation
use is a half-duplex version of V.32bis.
V.29
An ITU standard (1976) for synchronous 4,800, 7,200 and 9,600
bps full-duplex modems using QAM (Quadrature Amplitude
Modulation) on four-wire leased lines. It has been adapted for
Group 3 fax transmission over dial-up lines at 9,600 and 7,200
bps.
V.32bis
An ITU standard (1991) for asynchronous and synchronous
4,800, 7,200, 9,600, 12,000 and 14,400 bps full-duplex modems
using TCM and echo cancellation. It supports rate renegotiation,
which allows modems to change speeds as required.
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Glossary
VLAN
Virtual Local Area Network. A group of devices on one or more
LANs that are configured with the same VLAN ID so that they can
communicate as if they were attached to the same wire, when in
fact they are located on a number of different LAN segments.
Used also to create separation between different user groups.
VoIP
Voice over Internet Protocol. Provides an advanced digital
communications network that bypasses the traditional public
switched telephone system and uses the Internet to transmit
voice communication. VoIP enables people to use the Internet as
the transmission medium for telephone calls by sending voice
data in packets using IP rather than by traditional circuit
switched transmissions of the PSTN.
WAN
Wide Area Network. A computer network that spans a relatively
large geographical area. Wide area networks can be made up of
interconnected smaller networks spread throughout a building, a
state, or the entire globe.
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