APOS_G1 Voice Configuration Guide

APOS Quick Operation Guide
Voice over IP
APOS
Quick Operation Guide
[Voice Over IP]
Release 1.00
AddPac Technology Co., Ltd
AddPac Technology R&D Center
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APOS Quick Operation Guide
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Voice over IP
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APOS Quick Operation Guide
Voice over IP
[Contents]
Chapter 1. Introduction..................................... 9
Overview .............................................................................................................................9
Document History ........................................................................................................ 9
Organization............................................................................................................... 10
VoIP Products Covered ....................................................................................................11
VoIP Products Covered by This Guide.................................................................... 11
Obtaining Technical Assistance .............................................................................. 12
AddPac Technology VoIP Internetworking Solution............................................. 13
Chapter 2. Basic Equipment Management .. 14
Connecting a Terminal to VoIP Products ........................................................................14
Network Diagram ...................................................................................................... 14
User Account Management ............................................................................................15
Log-in as Root............................................................................................................. 15
User Account Checking ........................................................................................... 16
Registering New User Account ................................................................................ 16
Verifying New User Account .................................................................................... 17
Log-in with new user account ................................................................................. 17
Limited User Info Change ......................................................................................... 18
Enable/ Disable Network Protocol ..................................................................................20
Enabling/ disabling network protocols................................................................... 20
APOS Upgrade via FTP......................................................................................................21
Network Diagram ...................................................................................................... 22
FTP Service Status Checking .................................................................................... 22
APOS download via FTP from PC ............................................................................ 23
Upgraded APOS Image File Verification and Rebooting.................................... 24
Boot Loader .......................................................................................................................25
Network Diagram ...................................................................................................... 25
Entering Boot Loader Mode..................................................................................... 26
Checking Password................................................................................................... 26
Password change and verification......................................................................... 27
IP Address Checking & Recovery ........................................................................... 27
APOS Image File Download..................................................................................... 28
APOS Configuration Initialization............................................................................. 29
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Chapter 3. VoIP Network Configuration....... 30
PPPoE Network Application .............................................................................................30
Network Diagram ...................................................................................................... 30
Related APOS commands & structure ................................................................... 31
DHCP Client Application ..................................................................................................32
Network Diagram ...................................................................................................... 32
Related APOS commands & structure ................................................................... 32
Fixed IP Application..........................................................................................................34
Network Diagram ...................................................................................................... 34
Related APOS commands & structure ................................................................... 34
Bridge Mode Application.................................................................................................36
Network Diagram ...................................................................................................... 36
Related APOS commands & structure ................................................................... 37
NAT/PAT Environment Application ..................................................................................39
Network Diagram of NAT Application .................................................................... 39
Network Diagram of PAT Application..................................................................... 40
APOS commands & structure .................................................................................. 41
IP Sharing Application ......................................................................................................45
Network Diagram ...................................................................................................... 46
Related APOS commands & structure ................................................................... 47
PAT Server (VoIP Gateway) Application.........................................................................54
Network Diagram ...................................................................................................... 54
Related APOS commands & structure ................................................................... 55
Chapter 4. VoIP Network Configuration....... 58
Point-to-Point Application................................................................................................58
Network Diagram ...................................................................................................... 58
Related APOS commands & structure ................................................................... 60
Gatekeeper Interoperating Application.........................................................................62
Network Diagram ...................................................................................................... 62
Related APOS commands & structure ................................................................... 65
Number Translation Feature.............................................................................................68
Network Diagram ...................................................................................................... 68
Number Translation Example ................................................................................... 69
Related APOS commands & structure ................................................................... 70
Call Pickup & Transfer Feature .........................................................................................72
Network Diagram ...................................................................................................... 72
Related APOS commands & structure ................................................................... 76
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Chapter 5. VoIP Protocol Configuration ........ 77
VoIP Protocol.....................................................................................................................77
H.323 Protocol Application ..............................................................................................77
SIP Protocol (Direct Call) Application .............................................................................78
Network Diagram ...................................................................................................... 78
Related APOS commands & structure ................................................................... 79
SIP Protocol (Indirect, Proxy Server) Application ...........................................................81
Network Diagram ...................................................................................................... 81
Related APOS commands & structure ................................................................... 83
Username/Password Registration of SIP Dial-Peer .........................................................84
Related APOS commands & structure ................................................................... 85
MGCP Protocol Application.............................................................................................86
Network Diagram ...................................................................................................... 86
Main APOS Commands for MGCP Protocol.......................................................... 86
Chapter 6. Voice Interface Configuration... 92
Input & Output Gain configuration ..................................................................................92
Network Diagram ...................................................................................................... 93
Related APOS commands & structure ................................................................... 93
Tone Configuration ...........................................................................................................95
Network Diagram ...................................................................................................... 95
Related APOS commands & structure ................................................................... 96
E1/T1 Voice Interface Configuration/ ISDN-PRI ..............................................................98
Network Diagram ...................................................................................................... 98
Related APOS commands & structure ................................................................. 100
E1/T1 Voice Interface Configuration/ R2 DTMF.............................................................102
Network Diagram .................................................................................................... 102
Related APOS commands & structure ................................................................. 104
FXS/FXO Voice Interface configuration for caller ID ...................................................105
Network Diagram .................................................................................................... 105
Related APOS commands & structure ................................................................. 107
FXS/FXO Voice Interface configuration for polarity-inverse .......................................109
Network Diagram .................................................................................................... 109
Related APOS commands & structure ................................................................. 111
E&M Voice Interface Configuration ..............................................................................112
Network Diagram .................................................................................................... 112
Related APOS commands & structure ................................................................. 114
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Chapter 7. Appendix....................................... 115
E&M Voice Interface Dip Switch setting........................................................................115
E&M Voice Interface Module Jumper Switch ..................................................... 115
E&M Voice Interface Jumper Switch Description ............................................... 116
E&M Voice Interface Type and Jumper Setting ...........................................................117
E&M Voice Interface Wire Type and Jumper Setting ...................................................118
2-Wire E&M Voice Interface Jumper Setting ....................................................... 118
4-Wire E&M Voice Interface Jumper Setting ....................................................... 118
Glossary...........................................................................................................................119
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[Table of Tables]
[Table 1-1] History of APOS Quick Operation Guide.......................................................... 9
[Table 1-2] APOS Quick Operation Guide Organization................................................. 10
[Table 1-3] VoIP products covered by APOS Quick Operation Guide ......................... 11
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[Table of Figures]
Fig. 1-1 AddPac Technology VoIP Internetworking Solution........................................... 13
Fig. 2-1 VoIP gateway log-in................................................................................................ 14
Fig. 2-2 APOS image file upgrade via FTP ......................................................................... 22
Fig. 2-3 Network diagram for boot loader mode access ............................................... 25
Fig. 3-1 VoIP network diagram on ADSL Network............................................................. 30
Fig. 3-2 VoIP network diagram on DHCP network............................................................ 32
Fig. 3-3 VoIP network diagram on fixed IP Network ......................................................... 34
Fig. 3-4 VoIP network diagram of Ethernet Bridge Network ........................................... 36
Fig. 3-5 VoIP network diagram of NAT application.......................................................... 39
Fig. 3-6 VoIP network diagram of PAT application .......................................................... 40
Fig. 3-7 VoIP network diagram of IP sharing application ................................................ 46
Fig. 3-8 VoIP network diagram of VoIP gateway operating as PAT server................... 54
Fig. 4-1 VoIP network diagram of peer-to-peer communication .................................. 58
Fig. 4-2 VoIP network diagram of Gatekeeper interoperating application................. 62
Fig. 4-3 VoIP gateway number translation feature diagram.......................................... 68
Fig. 4-4 VoIP gateway Call-pickup feature ....................................................................... 72
Fig. 4-5 VoIP gateway Call-transfer feature ...................................................................... 73
Fig. 5-1 VoIP network diagram of SIP direct call configuration ...................................... 78
Fig. 5-2 VoIP network diagram of SIP indirect calls via SIP Proxy server ........................ 81
Fig. 5-3 VoIP network diagram based on MGCP protocol ............................................. 86
Fig. 6-1 VoIP Gateway Input/Output gain ........................................................................ 93
Fig. 6-2 VoIP gateway tone setting .................................................................................... 95
Fig. 6-3 VoIP gateway digital E1/T1 ISDN-PRI..................................................................... 98
Fig. 6-4 VoIP gateway digital E1/T1 R2/DTMF.................................................................. 102
Fig. 6-5 VoIP gateway caller- ID feature.......................................................................... 105
Fig. 6-6 VoIP gateway polarity inverse feature on FXS port .......................................... 109
Fig. 6-7 VoIP gateway polarity inverse feature on FXO port......................................... 110
Fig. 6-8 VoIP gateway E&M interface .............................................................................. 112
Fig. 7-1 E&M voice interface module jumper switch image......................................... 115
Fig. 7-2 E&M voice interface module front view ............................................................ 116
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Chapter 1. Introduction
Overview
APOS (AddPac Internetworking Operating System) Quick Operation
Guide provides information on APOS commands & structure, popular
network
diagram
and
configuration
verification/debugging
commands of AddPac’s VoIP (Voice over IP) products including VoIP
Gateway.
Especially, the network diagram and APOS commands of this guide
are real examples which can applied to the users’ applications. For
more detailed information of APOS commands, refer to APOS
Operation Guide.
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Organization
Table 1-2 provides an overview of the organization of this guide.
[Table 1-2] APOS Quick Operation Guide Organization
Chapter
Title
Chapter 1
Overview
Chapter 2
Chapter 3
Chapter 4
Chapter 5
Chapter 6
Description
Device & Network
Management
Provides the overview of APOS Quick Operation
Guide, History and VoIP products covered
Provides
information
on
VoIP
device
login,
Password, APOS image file downloading and
recovery
VoIP Network
Provides information about configuring various
Environment
VoIP network types and APOS commands.
VoIP Network
Configuration
VoIP Protocol
Configuration
Provides information about APOS configuration
on various VoIP networks and the configuration
examples.
Provides information about APOS configuration of
H.323, SIP and MGCP protocols and various
configuration examples
Voice Interface
Provides information about APOS configuration of
Configuration
FXS, FXO, E&M & digital E1/T1 Interface
Provides information on how to set dip switch for
Chapter 7
Appendix
E&M voice interface module and the glossary of
network terms
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VoIP Products Covered
VoIP Products Covered by This Guide
APOS Quick Operation Guide covers AddPac’s VoIP products listed
at [Table 1-3].
You can refer to this guide for VoIP router, Multi-
service router, gatekeepers, broadcasting over IP system, Fax
broadcasting system along with VoIP gateway.
The provided
network diagram, configuration examples, APOS commands and
descriptions are based on VoIP gateway products.
For VoIP
network application and APOS commands that are not mentioned
at this guide, please contact AddPac Technology R&D Center.
[Table 1-3] VoIP products covered by APOS Quick Operation Guide
Product Line
Models
Main Network Interface
VoIP gateway
AP160
FXS voice port
PSTN Dial-up port
Ethernet port
FXS/FXO voice port
Ethernet port
FXS/FXO voice port
Ethernet port
FXS/FXO voice port
Ethernet port
FXS/FXO/E&M voice interface module
Digital E1/T1 interface module
Ethernet port
FXS/FXO/E&M voice interface module
Ethernet port
FXS/FXO/E&M voice interface module
Ethernet port
FXS/FXO/E&M voice interface module
Digital E1/T1 interface module
Ethernet port
FXS/FXO/E&M voice interface module
Ethernet port
FXS/FXO/E&M voice interface module
Digital E1/T1 interface module
Ethernet port
FXS/FXO/E&M voice interface module
Digital E1/T1 interface module
Network interface module (Ethernet port)
FXS/FXO/E&M voice interface module
Digital E1/T1 interface module
Network interface module (Ethernet port)
Ethernet interface
Ethernet interface
Ethernet interface
Ethernet interface
Ethernet interface
AP200 Series
AP1000 Series
AP1100 Series
AP2110
AP2120
AP3100
AP2520G
Secure VoIP gateway
AP2520S
VoIP router
AP2520R
Multi-service router
AP2830
AP2850
Built-in gatekeeper
Broadcasting over IP system
Fax broadcasting system
AP-GK1000
AP-GK2000
AP-GK3000
AP3120
AP3220
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Obtaining Technical Assistance
AddPac’s Technical Assistance is available to all customers and
partners.
The technical supports and training of this APOS Quick
Operation Guide and AddPac Products can be obtained from
Monday through Friday (9:00 AM ~ 7: PM, GMT+9:00).
Also,
technical support via e-mail is available around the clock.
AddPac Technology Tech Support Center
TEL: +82-2-568-3848, FAX +82-2-568-3847
E-mail: products@addpac.com
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AddPac Technology VoIP Internetworking Solution
AddPac Technology’s VoIP Internetworking solution offers high
performance networking solution not only for voice but also for data,
image and multimedia network applications.
The below figure
shows the overall AddPac’s VoIP products and networking solutions.
Fig. 1-1 AddPac Technology VoIP Internetworking Solution
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Chapter 2. Basic Equipment
Management
This chapter provides the information on APOS commands regarding
basic equipment management features of VoIP products including
VoIP gateway.
NOTE
Basic
Equipment
Management
is
supported
by
AddPac
Technology’s all VoIP products along with VoIP Gateway.
Connecting a Terminal to VoIP Products
Two different access types are available for connecting a terminal to
VoIP products. One is using PC’s Hyper terminal emulation program
via RS-232C console port of the VoIP gateway. Also, the other is
accessing via Ethernet using telnet program.
The user interface and APOS commands are identical at both cases.
Network Diagram
Fig. 2-1 VoIP gateway log-in
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User Account Management
VoIP gateway log-in user account and its permission level can be
managed with user account management feature. According to
the permission level of an account, the available APOS commands
are different. The general rules applied to the user account including
“root” account are as shown below.
y
The “root” user account is undeletable.
y
The “root” user account can check the information of all user
accounts
Even though the user level is “Admin”, it can only check its own
y
information, if it is not “root” user.
Only the “root” user is allowed to add or delete user accounts. Note
that the permission level of the “root-Admin” and the “Admin”
created by “root” are different.
NOTE
The default password of all AddPac’s products including VoIP
gateway at shipment is “router”
Log-in as Root
Step
Commands
Description
1
The System is ready. Please login to system.
Log-in as “root”.
login:
login: root
2
password: <password>
Enter the default password “router.”
AP1100-S4O4 - Login : root at Console on
Tue Oct 28 13:01:58 2003
#
Enter
APOS
global
configuration
mode.
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User Account Checking
Step
Commands
Description
1
#
Check the user account
# show user
information.
2
Login Name
Password
User level
Timeout Alias Name
------------------------------------------------------------root
router
ADMIN
0
#
Registering New User Account
Step
Commands
Description
1
#
Enter
# config
mode.
Enter configuration commands,
line. End with CNTL/Z
2
one
APOS
global
configuration
per
“?” shows the available options.
(config)#
(config)# user ?
Add new user at User entry
Change User's Password
Change User's Access Level
Change User's auto logout time
add
change
level
timeout
3
(config)#
(config)# user add addpac1?
<password> Password for given login
4
(config)#
(config)# user add addpac1 addpac1 ?
admin , high, normal or low
5
(config)#
Create a user account of user ID
(config)# user add addpac1 addpac1 admin
“addpac1”,
and level
6
“addpac1”
“admin.”
(config)#
Create a user account of user ID
(config)# user add addpac2 addpac2 high
“addpac2”,
and level
7
password
password
“addpac2”
“high.”
(config)#
Create a user account of user ID
(config)# user add addpac3 addpac3 normal
“addpac3”,
and level
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password
“addpac3”
“normal.”
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8
Voice over IP
(config)#
Create a user account of user ID
(config)# user add addpac4 addpac4 low
“addpac4”,
and level
password
“addpac4”
“low.”
Verifying New User Account
Step
Commands
Description
1
#
Verify the newly added
# show user
user account information.
2
Login Name
Password
User level
Timeout Alias Name
------------------------------------------------------------root
router
ADMIN
0
addpac1
addpac1
ADMIN
0
addpac2
addpac2
HIGH
0
addpac3
addpac3
NORMAL
0
#
Log-in with new user account
Step
Commands
Description
1
#
Exit from the APOS system.
# exit
Then log-in with the new
The System is ready. Please login to system.
user account.
login:
login: addpac1
2
password: <password>
Enter the new password
AP1100-S4O4 - Login : root at Console on
Tue Oct 28 13:01:58 2003
“addpac1.”
#
#
3
#
Only
the
account
# show user
information of itself can
Login Name
Password
User level
Timeout Alias Name
------------------------------------------------------------addpac1
addpac1
ADMIN
0
be viewed and verified.
#
#
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Limited User Info Change
Step
Commands
Description
1
#
Exit
# exit
Then log-in with the new user
The System is ready. Please login to system.
account.
from
the
APOS
system.
login:
login: addpac1
2
password: <password>
Enter
AP1100-S4O4 - Login : root at Console on
Tue Oct 28 13:01:58 2003
“addpac1.”
the
new
password
#
3
#
“?” shows the available options
# user change?
of “user change” command.
<login-name> Login name of user entry
# user change
4
# user change addpac1?
<old password> Old Password for given login
# user change addpac1
5
# user change addpac1 addpac1?
<new password> New Password for given login
# user change addpac1 addpac1
6
# user change addpac1 addpac1 addpac11
#
7
# user level?
<login-name> Login name of user entry
# user level
8
# user level addpac1?
<password> Old Password for given login
# user level addpac1
9
# user level addpac1 addpac11?
admin , high, normal or low
10
# user level addpac1 addpac11 low
This command is allowed only "root"
11
# user timeout?
<login-name> Login name of user entry
# user timeout
12
# user timeout addpac1?
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<timeout value>
Voice over IP
Time out value (second, 0 is
forever)
# user timeout addpac1
13
# user timeout addpac1 120
#
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Enable/ Disable Network Protocol
AddPac
Technology’s
VoIP
products
support
various
server
application programs of the popular network protocols. The users
can enable or disable certain server application programs.
There are seven server application programs: Easy Setup service, FTP
& TFTP server, SNMP agent, HTTP server, Telnet server, NTP (Network
Time Protocol). VoIP products enable three server application
programs, FTP/HTTP/Telnet, as default at the initial booting process.
Enabling/ disabling network protocols
Step
Commands
Description
1
#
Enter
# config
configuration mode.
2
3
4
(config)# show service
Easy Setup Service
FTP Server
SNMP Agent
TFTP Server
HTTP Web Server
TELNET Server
NTP(Network Time Protocol)
(config)#
(config)# service snmp
Easy Setup Service
FTP Server
SNMP Agent
TFTP Server
HTTP Web Server
TELNET Server
NTP(Network Time Protocol)
(config)#
(config)# no service snmp
Easy Setup Service
FTP Server
SNMP Agent
TFTP Server
HTTP Web Server
TELNET Server
NTP(Network Time Protocol)
APOS
global
Check the default status
:
:
:
:
:
:
:
DISABLE
ENABLE
DISABLE
DISABLE
ENABLE
ENABLE (max session 5)
DISABLE
of
server
programs. (Default setting
at shipment)
Enable
:
:
:
:
:
:
:
DISABLE
ENABLE
ENABLE
DISABLE
ENABLE
ENABLE (max session 5)
DISABLE
DISABLE
ENABLE
DISABLE
DISABLE
ENABLE
ENABLE (max session 5)
DISABLE
AddPac Technology Proprietary & Documentation
SNMP
agent
SNMP
agent
service.
Disable
:
:
:
:
:
:
:
application
service.
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APOS Upgrade via FTP
AddPac’s VoIP products supports the below three network protocols
for APOS binary code image file transfer. Also, each protocol can
be turn on/off.
y
FTP
(Supports server and client environment)
y
TFTP
(Supports server environment)
y
HTTP (Supports server environment)
Because it supports both FTP server and client applications, the file
exchange between VoIP equipment is also supported. For the user
name and password, refer to the user account list of the device.
As default, FTP, TFTP and HTTP server applications are enabled and
this guide mainly deals with APOS image file upgrade via FTP, which
known as very functional and reliable file transfer method.
For the latest APOS image, release notes, installation guides and
APOS quick operation guides including this guide, visit AddPac
Technology’s website at www.addpac.com.
Please check the server status before FTP file transfer.
FTP is an application protocol that uses the Internet's TCP/IP protocols,
and downloading via RS-232C console interface is not available.
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Voice over IP
Network Diagram
Before upgrading APOS image file, visit AddPac Technology’s
website, www.addpac.com and download the right APOS image to
the PC. The network diagram upgrading APOS image from PC is as
shown below.
Fig. 2-2 APOS image file upgrade via FTP
FTP Service Status Checking
Step
Commands
Description
1
#
Enter APOS global
# config
configuration mode.
2
(config)# show service
Easy Setup Service
FTP Server
SNMP Agent
TFTP Server
HTTP Web Server
TELNET Server
NTP(Network Time Protocol)
Check the FTP server
:
:
:
:
:
:
:
DISABLE
ENABLE
DISABLE
DISABLE
ENABLE
ENABLE (max session 5)
DISABLE
AddPac Technology Proprietary & Documentation
service status.
(The
default setting at
shipment).
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APOS download via FTP from PC
Step
Commands on PC
Description
1
F:\test>
Check the APOS image
F:\test>dir
on PC.
2003-08-08
2003-08-08
2003-08-08
04:43p
<DIR>
04:43p
<DIR>
04:43p
2 Files(s)
2 Dir(s)
2
1,142,532
1,541,978
5,221,683,200
.
..
ap1100rom_v6_120.bin
bytes
byte free
F:\test>
Access
to
the
F:\test>ftp 192.168.1.2
gateway via FTP.
VoIP
Connected to 192.168.1.2.
220 router FTP server (Version 1.12) ready.
User (192.168.1.2:(none)): root
331 Password required for root.
Password:*****
230 User root logged in ok.
F:\test>
3
4
5
ftp>bin
Set the APOS image as
200 Type set to I.
binary.
ftp> put ap1100 rom_v6_120.bin
Upgrade APOS image
200 PORT command successful.
150 BINARY data connection for ap1100rom_v6_120.bin
(194.168.1.2,1826).
226 BINARY Transfer complete.
1142532
bytes
sent
in
1.06
seconds
(1075.83
Kbytes/sec)
from
ftp> quit
Exit from FTP mode.
PC
gateway
to
VoIP
with
“PUT”
command.
F:\test>
F:\test>
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Upgraded APOS Image File Verification and Rebooting
Step
Commands on PC
Description
1
login:
login: root
password: *****
AP1100 - Login
06:14:38 2003
Log in as root
2
Enter the password.
: root at Console on Wed Aug 20
#
3
Verify
# show files
the
upgraded
image.
-rwxrwxrwx
-rwxrwxrwx
-rwxrwxrwx
-rwxrwxrwx
-rwxrwxrwx
-rwxrwxrwx
1
1
1
1
1
1
noone
noone
noone
noone
noone
noone
nogroup
nogroup
nogroup
nogroup
nogroup
nogroup
0
0
0
0
1368
2605964
Oct
Oct
Oct
Oct
Oct
Oct
30
30
30
30
30
30
2003
2003
2003
2003
2003
2003
evtlog0.txt
evtlog0.txt
cmdlog0.txt
cmdlog1.txt
config.cfg
ap1100rom_v7_00.bin
#
4
# reboo
Reboot the system after
System Reboot...
verifying the Image.
System Boot Loader, Version 1.4.5/2
Copyright (c) by AddPac Technology Co., Ltd. Since 1999.
System Bootstrap, Version 1.2
Decompressing the image:
###################################################
###################################################
###################################################
###################################################
#############################[t
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Voice over IP
Boot Loader
APOS image and password recovery and change are required at
the below conditions.
y
The password of root account is changed or lost
y
APOS image file is deleted or damaged
The users can restore or check the password at the boot loader
mode. Also, when APOS image is damaged or deleted, you can
download the image at the boot loader mode.
NOTE
In boot loader mode, IP routing feature is not available. So the
Ethernet IP address of the PC with the OS image and that of the
VoIP gateway should be on the same network.
NOTE
To enter boot loader mode, establish direct access to the gateway
via console port.
Network Diagram
Fig. 2-3 Network diagram for boot loader mode access
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Entering Boot Loader Mode
Step
Commands
Description
1
#
Restart the system by H/W reset
# reboot
(Power switch off/on) or S/W
System Reboot...
reset (reboot command).
System Boot Loader, Version 1.4.5/2
Copyright (c) by AddPac Technology Co., Ltd. Since 1999.
2
When
The "BOOT LOADER" is ready
the
initial
BOOT#
gateway
BOOT#
mode.
BOOT# ?
Check
configure
erase
exit
history
ping
reboot
show
telnet
Enter configuration mode
Easy Erase configuration data
Exit from the EXEC
Show command line history
Send echo messages
reboot system
Show running system information
Open a telnet connection
is
displayed, press “ctrl-C” & “ctrlX” by turns.
3
message
Then the VoIP
enters
the
boot
loader
commands
supported at the boot loader
mode.
BOOT#
Checking Password
Step
Commands
Description
1
BOOT#
Check the current password.
BOOT# show password
Password = ”router”
BOOT#
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Password change and verification
Step
Commands
Description
1
BOOT#
Enter
BOOT# config
mode.
the
boot
loader
command
BOOT(config)#
2
BOOT(config)#
Enter the new password twice to
BOOT(config)# password abcd abcd
change the password.
password change
BOOT(config)#
3
BOOT(config)#
Exit from the boot loader command
BOOT(config)# exit
mode.
BOOT#
4
Verify the newly configured password.
BOOT#
BOOT# show password
Password = ”abcd”
BOOT#
IP Address Checking & Recovery
Step
Commands
Description
1
BOOT#
Check
BOOT# show interface
statistical information
Interface Configuration : ether0.0
IP address : 172.17.103.10 netmask : 255.255.0.0 mtu = 1500
Ethernet Address : 00 02 a4 ff ff 1a
Ethernet0 is DOWN, Line protocol is DOWN
Bandwdith : 10000 Kbit
Operating mode : HALF-DUPLEX
Operating speed : 10 Mbps
0 packets input, 0 bytes, 0 no buffers
Received 0 runts, 0 giants
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
0 input packets with dribble condition detected
0 packets output, 0 bytes, 0 drops
0 output errors, 0 collision, 0 interface resets
0 underruns, 0 late collisions, 0 deferred
0 lost carrier, 0 no carrier
the IP address assigned on
the
interfaces,
and
the Ethernet interface 0.0.
BOOT#
2
BOOT# config
Assign the IP address to the
BOOT(config)# address 192.168.1.2 255.255.255.0
interface.
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Voice over IP
BOOT(config)#
Exit from the boot loader
BOOT(config)# exit
command mode.
BOOT#
4
BOOT#
Check
the
interfaces,
BOOT# show interface
statistical information
Interface Configuration : ether0.0
IP address : 192.168.1.2 netmask : 255.255.0.0 mtu = 1500
Ethernet Address : 00 02 a4 ff ff 1a
Ethernet0 is DOWN, Line protocol is DOWN
Bandwdith : 10000 Kbit
Operating mode : HALF-DUPLEX
Operating speed : 10 Mbps
0 packets input, 0 bytes, 0 no buffers
Received 0 runts, 0 giants
0 input errors, 0 CRC, 0 frame, 0 overrun, 0 ignored
0 input packets with dribble condition detected
0 packets output, 0 bytes, 0 drops
0 output errors, 0 collision, 0 interface resets
0 underruns, 0 late collisions, 0 deferred
0 lost carrier, 0 no carrier
verifies the new IP address
and
assigned on the Ethernet
interface 0.0.
BOOT#
APOS Image File Download
The APOS image file download procedure is same as that of APOS
image upgrade via FTP. Please note that IP routing feature is not
supported at the boot loader mode and this should be done at the
same IP netmask. The IP address setting can be done at both the
boot loader command mode and APOS command mode but the
commands are not identical.
Step
Commands on PC
Description
1
F:\test>
Check the APOS image
F:\test>dir
on PC.
2003-08-08
2003-08-08
2003-08-08
04:43p
<DIR>
04:43p
<DIR>
04:43p
2 Files(s)
2 Dir(s)
2
1,142,532
1,541,978
5,221,683,200
.
..
ap1100rom_v6_120.bin
bytes
byte free
F:\test>
Access
to
the
F:\test>ftp 192.168.1.2
gateway via FTP.
VoIP
Connected to 192.168.1.2.
220 router FTP server (Version 1.12) ready.
User (192.168.1.2:(none)): root
331 Password required for root.
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Password:*****
230 User root logged in ok.
F:\test>
3
4
5
ftp>bin
Set the APOS image as
200 Type set to I.
binary.
ftp> put ap1100 rom_v6_120.bin
Upgrade APOS image
200 PORT command successful.
150 BINARY data connection for ap1100rom_v6_120.bin
(194.168.1.2,1826).
226 BINARY Transfer complete.
1142532
bytes
sent
in
1.06
seconds
(1075.83
Kbytes/sec)
from
ftp> quit
Exit from FTP mode.
PC
gateway
to
VoIP
with
“PUT”
command.
F:\test>
F:\test>
APOS Configuration Initialization
At boot loader mode, the default APOS configuration can be
restored.
Step
Commands
Description
1
BOOT#
Restore
BOOT# erase
configuration at the boot loader
Do you want to ERASE configuration ? [y|n] y
command mode.
the
default
APOS
Erasing configuration....done
BOOT#
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Voice over IP
Chapter 3. VoIP Network
Configuration
This chapter provides information on network interface configuration
of VoIP products (ex. VoIP gateway, router and etc.). These are real
network application examples which can be applied to general
customer environment.
Before you begin, carefully review this
chapter.
PPPoE Network Application
PPPoE application is for the users of PPPoE broadband network
environment using ADSL modem.
NOTE
PPPoE Network Application is supported by AddPac Technology’s
all VoIP products along with VoIP gateway.
Network Diagram
Fig. 3-1 VoIP network diagram on ADSL Network
APOS command script
interface ether0.0
no ip address
encapsulation pppoe
ppp authentication pap callin
ppp pap sent-username addpac password 1234
ppp ipcp ms-dns
ppp ipcp default-route
!
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Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
y
Access ID: “AddPac”
y
Access password: “1234”
y
get DNS IP (option)
y
get default-router IP (option)
To configure PPPoE network application, follow this procedure.
Step
Commands
Description
1
#
Enter
# config
Enter
configuration
line. End with CNTL/Z
APOS
global
configuration
mode.
commands,
one
per
(config)#
2
3
(config)# interface ether0.0
Enter
the interface configuration
(config-ether0.0)#
mode.
(config-ether0.0)# no ip address
Do not assign an IP address to the
interface.
4
(config-ether0.0)# encapsulation pppoe
Assign encapsulation type.
5
(config-ether0.0)# ppp authentication pap
callin
(config-ether0.0)# ppp pap sent-username
addpac password 1234
Assign PAP as PPPoE authentication.
6
Configure PAP User ID and Password.
In this example, the user ID is
“AddPac”
and
the
password
is
“1234”.
7
(config-ether0.0)# ppp ipcp ms-dns
Configure to get default router IP
from PPP Server.
8
(config-ether0.0)# ppp ipcp default-route
Configure to get DNS IP from PPP
Server.
9
10
(config-ether0.0)# exit
Exits from the interface configuration
(config)#
mode.
(config)# exit
Exits from APOS global configuration
#
mode.
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Voice over IP
DHCP Client Application
DHCP Client application is for the users of the DHCP Server
broadband network environment using Cable Modem.
NOTE
DHCP Client Application is supported by AddPac Technology’s all
VoIP products along with VoIP Gateway.
Network Diagram
At the below diagram, a VoIP gateway interoperates with Cable
Modem, broadband networking equipment.
Fig. 3-2 VoIP network diagram on DHCP network
APOS command script
!
interface ether0.0
ip address dhcp
!
Related APOS commands & structure
No parameters are required for this application
Step
Commands
Description
1
#
Enter
# config
Enter
configuration
line. End with CNTL/Z
APOS
global
configuration
mode.
commands,
one
per
(config)#
2
(config)# interface ether0.0
Enter
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Voice over IP
(config-ether0.0)#
mode.
(config-ether0.0)# ip address dhcp
DHCP server assigns the IP address.
(config-ether0.0)#
4
5
(config-ether0.0)# exit
Exit from the interface configuration
(config)#
mode..
(config)# exit
Exit from APOS global configuration
#
mode.
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Voice over IP
Fixed IP Application
On fixed IP environment, VoIP network includes WAN router. At least
two Ethernet interfaces (LAN0, LAN1) are required for this application.
NOTE
Fixed IP Application is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway.
Network Diagram
Fig. 3-3 VoIP network diagram on fixed IP Network
APOS command script
!
interface ether0.0
ip address 192.168.1.2 255.255.255.0
!
route 0.0.0.0 0.0.0.0 192.168.1.1
!
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
y
IP address for LAN 0 interface: 192.168.1.2
y
Net mask: 255.255.255.0
y
IP address of default router: 192.168.1.1
To configure Fixed IP Application, follow this procedure.
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Step
Commands
Description
1
#
Enter
# config
Enter
configuration
line. End with CNTL/Z
APOS
global
configuration
mode.
commands,
one
per
(config)#
2
3
4
(config)# interface ether0.0
Enter
the interface configuration
(config-ether0.0)#
mode.
(config-ether0.0)# ip address 192.168.1.2
Assign
255.255.255.0
interface.
(config-ether0.0)# route 0.0.0.0 0.0.0.0
Assign the default router.
the
IP
address
to
the
192.168.1.1
5
6
(config-ether0.0)# exit
Exit from the interface configuration
(config)#
mode.
(config)# exit
Exit from APOS global configuration
#
mode.
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Voice over IP
Bridge Mode Application
Bridge mode is implemented when WAN Router environment (PPP,
HDLC, Frame Relay, ATM and etc.) requires traffic priority control for
the traffic from local network to IP network.
Also, when the QoS
feature of WAN Router is not sufficient and VoIP gateway should
offer priority control between voice and data traffic, the bridge
mode is recommended.
At lease two Ethernet interfaces (LAN0, LAN1) are required for Bridge
mode application.
NOTE
Bridge Mode Application is supported by AddPac Technology’s all
VoIP products along with VoIP Gateway.
Network Diagram
Fig. 3-4 VoIP network diagram of Ethernet Bridge Network
APOS command script
!
no ip routing
!
no bridge spanning-tree
!
!
interface ether0.0
ip address 192.168.1.2 255.255.255.0
qos-control 200 150
bridge
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!
interface ether1.0
no ip address
bridge
!
route 0.0.0.0 0.0.0.0 192.168.1.1
!
Related APOS commands & structure
At the above network diagram, the PC connected to LAN Switch
regards VoIP Gateway as a transmission path. So the IP address of
default router should be the Ethernet IP address of the Leased line
router. Also, the VoIP Gateway only accepts the traffic which has the
IP address of the VoIP gateway as the destination IP. The QoS is
applied to Up-Link interface, so the priority and bandwidth control of
all the traffic coming from the network under VoIP Gateway (LAN1)
to the Internet (LAN 0) including VoIP traffic is possible.
Configure the below parameters appropriate for the network
environment.
y
IP address of the gateway: 192.168.1.2
y
Net Mask: 255.255.255.0
y
IP address of the default router: 192.168.1.1
y
Bridge mode for LAN 0 & LAN 1 interface
y
QoS configuration for LAN0 interface
y
No IP routing required
To configure bridge mode application, follow this procedure.
Step
Commands
Description
1
#
Enter
# config
Enter configuration commands,
line. End with CNTL/Z
APOS
global
configuration
mode.
one
per
2
(config)# no ip routing
Disable IP routing features.
3
(config)# no bridge spanning-tree
No BPDU Exchange feature is required.
4
(config-ether0.0)# interface ether0.0
Enter
(config-ether0.0)#
mode.
(config-ether0.0)# ip address 192.168.1.2
255.255.255.0
Assign the IP address to the interface.
5
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6
(config-ether0.0)# qos-control 200 150
Voice over IP
Configure the QoS.
Set the RX
bandwidth and PPS as “20Kbps ~
150Kpbs”.
7
(config-ether0.0)# bridge
Activate the bridge mode for the
interface.
8
(config-ether0.0)# interface ether1.0
Enter
the
interface
configuration
(config-ether1.0)#
mode.
9
(config-ether1.0)# no ip address
No IP routing is required.
10
(config-ether1.0)# bridge
Activate the bridge mode for the
interface.
11
(config-ether1.0)# route 0.0.0.0 0.0.0.0
192.168.1.1
Assign the default router.
(config-ether1.0)#
12
13
(config-ether1.0)# exit
Exits from the interface configuration
(config)#
mode.
(config)# exit
Exits from APOS global configuration
#
mode.
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Voice over IP
NAT/PAT Environment Application
NAT(Network Address Translation) or PAT(Port Address Translation)
environment of VoIP network is implemented when the IP based
network (PPP, HDLC, Frame Relay, ATM and etc.) of WAN Router or IP
sharer assigns private IP addresses to its local network. This part
explains how to configure a gateway on a private network under IP
sharer. NAT (Network Address Translation) Server and PAT (Port
Address Translation) Server applications are explained below.
NOTE
NAT/PAT
Environment
Application
is
supported
by
AddPac
Technology’s all VoIP products along with VoIP Gateway.
Network Diagram of NAT Application
At NAT environment application, the WAN router or IP sharer
connecting the VoIP gateway to exterior network has its own public
IP Pool and dynamically converts a private IP to the public IP before
the packets are forwarded onto the outside network.
Fig. 3-5 VoIP network diagram of NAT application
However, when exterior network tries direct access to a specific
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internal address, the public IP cannot address the private IP address
matched. Then, the WAN router or IP sharer operates as NAT Server
and it forcefully converts a private IP to one of the IP address at the
its public IP Pool. That is, there is a call attempt from an exterior
network to the gateway, the setup message can be reached to the
internal IP because of the static map configured at the NAT Server.
Network Diagram of PAT Application
At PAT environment application, the WAN router or IP sharer
connecting the VoIP gateway to exterior network has a public IP
address and dynamically assigns a public IP to the private IP
forwarded to WAN.
However, NAT and PAT application is a little bit different. For NAT
environment, number of public IP addresses can be mapped to
number of private IP addresses. However, for PAT environment, only
one public IP address is available.
Fig. 3-6 VoIP network diagram of PAT application
PAT server shares one public IP address by offering dynamic
mapping of local and remote TCP and UDP ports for the internal IPs
forwarded to WAN.
So more than one private IP addresses can
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share one public IP address.
Same as NAT application, without this feature, there is a problem
when exterior network tries direct access to a specific internal IP
address. To solve this problem, Packets destined for an external
address have their private IP address plus port number translated to
the router's external IP address before the IP packet is forwarded to
the WAN.
When, there is a call attempt from an exterior network to
the gateway, the setup message can be reached to the internal IP
because of the static map configured at the PAT Server.
The
configuration is same as that of NAT Server.
APOS commands & structure
To configure IP address on NAT/PAT environment, follow this
procedure.
Public IP address configuration under NAT/PAT environment
Step
Commands
Description
1
#
Enter
# config
Enter configuration commands, one per line.
End with CNTL/Z
(config)#
mode.
(config)# gateway
Enters gateway configuration mode.
2
APOS
global
configuration
(config-gateway)#
3
4
5
(config-gateway)# public-ip 192.168.1.9
Configures the public IP address for
(config-gateway)#
NAT/ PAT application.
(config-gateway)# exit
Assign
(config)#
interface.
(config)# exit
Exit from APOS global configuration
#
mode.
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IP
address
to
the
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VoIP network under Firewall environment (VoIP Port Minimize)
The below configuration example is for the network environment with
Firewall.
Firewall restricts the number of TCP/UDP ports for
communication. That’s why it is necessary to reduce the number of
ports used by VoIP gateway.
The number of TCP and UDP ports required by VoIP call connection is
minimized.
That is, the LISTEN and SOURCE ports of TCP and UDP
packets can be configured.
Refer to the above configuration with the VoIP network of PAT Server
application.
Step
Commands
Description
1
#
Enter
# config
Enter configuration commands, one per line. End
with CNTL/Z
(config)#
configuration mode.
(config)#
Enter
(config)# voice service voip
configuration mode.
2
APOS
global
VoIP
gateway
(config-vservice-voip)#
3
(config-vservice-voip)#
(config-vservice-voip)# minimize-voip-ports ?
multiply
service
port pool = channel number x multiply
Assign port per each service
(config-vservice-voip)#
4
(config-vservice-voip)#
(config-vservice-voip)# minimize service ?
5
6
signal-tcp-src
set H.225 signalling source port range
Configure the no. of ports.
control-tcp-src
set H.245 control source port range
Configure the no. of ports.
control-tcp-listen
set H.245 control listen port range
Configure the no. of ports.
rtp-udp-listen
set RTP/RTCP port range
Configure the no. of ports.
(config-vservice-voip)# exit
Exit
(config)#
configuration mode.
(config)# exit
Exit
#
configuration mode.
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NOTE
Voice over IP
“Minimize multiply” and “minimize service command” cannot be
configured at the same time.
The configuration values can be
overlapped.
APOS command script (Configuration Verification)
(config)#
(config)# voice service voip
(config-vservice-voip)# minimize-voip-ports multiply 2
(config-vservice-voip)# show gateway
System Information
status = init 2 (waiting for setting IP address on a VoIP interface)
product name = AddPac VoIP
product version = 7.00
endpoint type = gateway
Gatekeeper Registration Information
H.323 id =
gatekeeper registration option = disabled
gatekeeper security option = disabled
Gatekeeper registration status :
not registered.
last registration reject information from gatekeeper
ConfigAsNoRegistration (Oct 30 18:25:20)
Gatekeeper list :
Local aliases
Technical prefixes
-- more -Gateway Information
discovery (send GRQ) = disabled
ARQ option = arq default
LRQ option = no lrq
lightweight IRR = disabled
TTL margin = 20 %
public ip = 192.168.1.9
h323 call start mode = fast
h323 call tunneling mode = enabled
h323 call channel mode = late
h323 response msg = default
system fax mode = t38
system fax rate (bps) = 9600
system T.38 fax redundancy = 0
force to send startH245 = enabled
dialPeer hunt algorithm = longest - preference - random
translate voip incoming called number = -1
translate voip incoming calling number = -1
local ringback tone = normal
end of digit = #
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ip address prefix = *
-- more -permit unregistered h323 incoming call to FXO =
yes
voice confirmed connect on FXO/E&M = disabled
number
number
number
number
number
number
number
number
of
of
of
of
of
of
of
of
ports = 8
pots peers = 1
voip peers = 0
number expansions = 0
codec classes = 0
user classes = 0
alternate gatekeepers = 0
current calls = 0
Announcement Option
language = korean
element : delayed dial = disabled
element : wrong number = disabled
element : connection fail = disabled
element : enter password = disabled
element : pstn reroute = disabled
element : all lines busy = disabled
element : dial number = disabled
-- more -Timer & Counter parameter value
tinit (initial digit timer) = 10 sec.
tring (ring timer) = 30 sec.
t301 (alert -> connect) = 180 sec.
t303 (setup -> alert) = 20 sec.
tras (RAS msg ack timer) = 6 sec.
tttl (RAS Time To Live timer) = 60 sec.
tidt (inter digit timer) = 3 sec.
treg (GK Registration retry timer) = 20 sec.
treg2 (GK Registration retry timer : long period by RRJ) = 120 sec.
tohd (On Hook Delay Time) = 0 sec.
tpoll (polling timer on trunk or polling type connection) = 180 sec.
dtmf duration = 150 msec.
dtmf guard time = 100 msec.
cras (RAS retry counter) = 3
Remote Call Log (syslog)
primary server =
secondary server =
interval = 0 minutes
cdr format type = 0
-- more -Assigned VoIP TCP/UDP ports
minimized assign = yes
multiply = 2
Q.931 signalling port (TCP listen) = 1720
SIP signalling port (UDP listen) = 5060
H.245 control port (TCP src) = 10000 - 10015
Q.931 signalling port (TCP src) = 14000 - 14015
H.245 control port (TCP listen) = 18000 - 18015
RAS port and IRR port (UDP listen) = 22000, 22001
RAS GK src (UDP) port = 22002
RTP/RTCP port (UDP listen) = 23000 - 23031
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Voice over IP
IP Sharing Application
In IP sharing application, the public IP address of VoIP gateway is
shared with the devices of local network such as personal computers.
It is different from NAT (network Address Translation)/PAT (Port
Address Translation) converting the public IP address to private ones.
Currently, ordinary houses or SOHO users use dynamic or fixed IP for
broadband Internet access. In case of dynamic IP address, a new
IP address is assigned every time connecting Internet via ADSL
Modem or Cable Modem.
On the other hands, for the fixed IP
Internet access, ADSL modem or dedicated line is assigned with fixed
IP from ISP.
For dynamic IP access, VoIP Gateway is assigned with a dynamic &
public IP address with PPPoE and DHCP application.
Then the
public IP is shared with the local network users. For fixed IP access,
the fixed IP assigned by network service providers or ISPs is shared by
the VoIP Gateway and the PC s of the local network.
With dynamic IP access, assign the dynamic IP to Ethernet 0.0 (LAN
0) and configure Ethernet 1.0 (LAN 1) as DHCP Server without
assigning IP address.
With fixed IP address, assign the IP to the
Ethernet 0.0 (LAN 0) and do not assign IP address to Ethernet 1.0 (LAN
1).
For IP sharing function, more than two Ethernet Interfaces (LAN0,
LAN1) are required.
NOTE
IP Sharing Application is supported by AddPac Technology’s all
VoIP products along with VoIP Gateway.
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Network Diagram
Fig. 3-7 VoIP network diagram of IP sharing application
At the view of packet transmission path, IP sharing is same as that of
Bridge mode. QoS configuration of Ethernet 0.0 Interface (LAN 0) is
required to allow higher priority for the voice traffic. With the QoS
configuration, the VoIP Gateway can offer priority and bandwidth
control for all the data coming through Ethernet 1.0 (LAN 1) and VoIP
traffic as well, which realizes optimized voice quality.
Basically, changing existing user environment is not recommendable.
So if the up-link port is PPPoE Client, assign the local interface as PPP
Server. If up-link port is DHCP Client, assign the local interface as
DHCP Server.
Also, if up-link interface is assigned with Static IP,
configure the local interface as static.
Up-link Interface (LAN 0)
Local Interface (LAN 1)
Configurability
DHCP
DHCP
O
PPP
O
Static
X
DHCP
O
PPP
O
Static
X
DHCP
X
PPP
X
PPP
Static
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Static
O
Related APOS commands & structure
The below parameters should be configured at the VoIP Gateway for
the above application.
y
IP address configuration of LAN 0 & LAN 1 interface: PPPoE, DHCP,
Static
y
IP address of default router: Optional
y
Traffic QoS configuration for LAN 0 interface: Optional
y
IP sharing configuration
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Voice over IP
DHCP environment with public IP address assigned
DHCP environment application is for the users of broadband network
using cable modems.
APOS command script
!
dhcp-list 0 type server
dhcp-list 0 address server interface ether0.0
dhcp-list 0 option dhcp-lease-time 600
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address dhcp
mac-address 00:02:a5:00:00:00
qos 200 150
!
interface ether1.0
no ip address
ip dhcp-group 0
!
Step
Commands
Description
1
#
Enter
# config
Enter configuration commands,
line. End with CNTL/Z
(config)#
2
APOS
global
configuration
mode.
one
per
(config)# dhcp-list 0 type server
configure the VoIP gateway as DHCP
server.
3
(config)# dhcp-list 0 address server
interface ether0.0
Assign the IP address of the interface
4
(config)# dhcp-list 0 option dhcp-leasetime 600
The public IP address from Cable
as the IP address of DHCP server.
network is refreshed periodically.
The
internal PCs check for the IP address at
every 300 seconds (600/2).
recommend
to
configure
It is
“dhcp-
lease-time” as “10 min”.
5
(config)# ip-share enable
Enable IP sharing
6
(config)# ip-share interface net-side
Assign the public IP address to the
ether0.0
7
(config)# ip-share interface local-side
ether1.0
Ethernet interface 0.0.
Connect Internal PCs or other devices
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Voice over IP
to the Ethernet Interface 1.0.
8
(config)# interface ether0.0
Enter
the
interface
configuration
mode.
9
(config-ether0.0)# ip address dhcp
Assign the IP address with DHCP.
10
(config-ether0.0)# mac-address 00:02:a5:
00:00:00
Change the MAC address of the
Ethernet
0
as
“00:02:a5:00:00:00.”
Some cable modems ask for the MAC
address of the internal PC for the
authentication.
Use the MAC address
of the internal PC for the Ethernet
interface 0.0. (The MAC address of the
VoIP gateway is changed temporary
and the original address is recovered
when the command is removed.)
Use this command only when it is
necessary.
11
(config-ether0.0)# qos 200 150
Configure QoS.
12
(config-ether0.0)# interface ether1.0
Enter
the
interface
configuration
mode.
13
(config-ether1.0)# no ip address
Do not assign an IP address to the
interface.
14
(config-ether1.0)# ip dhcp-grou 0
To share a dynamically allocated IP
address, configure the interface as
DHCP Server interface.
15
16
(config-ether1.0)# exit
Exit from the interface configuration
(config)#
mode.
(config)# exit
Exit from APOS global configuration
#
mode.
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Voice over IP
PPPoE environment with public IP assigned
PPPoE environment application is for the users of broadband network
using ADSL modems.
APOS command script
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
no ip address
encapsulation pppoe
ppp authentication pap callin
ppp pap sent-username addpac password test
ppp echo interval 20
ppp ipcp ms-dns
ppp ipcp default-route
qos 200 150
!
interface ether1.0
no ip address
encapsulation pppoe
ppp authentication pap callin
ppp pap sent-username addpac password test
ppp echo interval 20
ppp ipcp ms-dns
ppp ipcp default-route
ppp role server
!
Step
Commands
Description
1
#
Enter
# config
Enter configuration commands,
line. End with CNTL/Z
(config)#
APOS
global
configuration
mode.
one
per
2
(config)# ip-share enable
Enable IP sharing.
3
(config)# ip-share interface net-side
Configure IP sharing features on the
ether0.0
Ethernet interface 0.0, the interface for
external access.
4
(config)# ip-share interface local-side
ether1.0
Configure IP sharing features on the
Ethernet interface 1.0, the interface for
internal access.
5
6
(config)# interface ether0.0
Enter
(config-ether0.0)#
mode.
(config-ether0.0)# no ip address
Do not assign an IP address to the
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interface.
7
(config-ether0.0)# encapsulation pppoe
Configure encapsulation type.
8
(config-ether0.0)# ppp authentication pap
callin
(config-ether0.0)# ppp pap sent-username
addpac password test
Configure PPP authentication as PAP.
9
Configure
the
PAP
User
ID
as
“addpac”
and
the
password
as
“1234”.
10
(config-ether0.0)# ppp echo interval 20
11
(config-ether0.0)# ppp ipcp ms-dns
Configure to get default router IP from
PPP Server.
12
(config-ether0.0)# ppp ipcp default-route
Configure to get DNS IP from PPP
Server.
13
(config-ether0.0)# qos 200 150
14
(config-ether0.0)# interface ether1.0
Enter
(config-ether1.0)#
mode.
(config-ether1.0)# no ip address
Do not assign an IP address to the
15
the
interface
configuration
interface.
16
(config-ether1.0)# encapsulation pppoe
Configure encapsulation type.
17
(config-ether1.0)# ppp authentication pap
callin
(config-ether0.0)# ppp pap sent-username
addpac password test
Configure PPP authentication as PAP.
18
Configure
the
PAP
User
ID
as
“addpac”
and
the
password
as
“1234”.
19
(config-ether1.0)# ppp echo interval 20
20
(config-ether1.0)# ppp ipcp ms-dns
Configure to get default router IP from
PPP Server.
21
(config-ether1.0)# ppp ipcp default-route
Configure to get DNS IP from PPP
Server.
22
(config-ether1.0)# ppp role server
Set to PPPoE Server
23
(config-ether1.0)# exit
Exit the interface configuration mode.
(config)#
24
(config)# exit
Exit from APOS global configuration
#
mode.
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Voice over IP
Fixed IP environment with public IP assigned
Fixed IP environment with a public IP address is for the users of
broadband network using a WAN router (PPP, HDLC, Frame-Relay,
ATM and etc.).
Configurations (static)
!
ip-share enable
ip-share interface net-side ether0.0
ip-share interface local-side ether1.0
!
interface ether0.0
ip address 192.168.1.2 255.255.255.0
!
interface ether1.0
no ip address
!
route 0.0.0.0 0.0.0.0 192.168.1.1
!
Step
Commands
Description
1
#
Enter
# config
Enter configuration commands, one per line.
End with CNTL/Z
(config)#
mode.
2
(config)# ip-share enable
Enable IP sharing feature.
3
(config)# ip-share interface net-side
ether0.0
Configure IP sharing features on the
APOS
global
configuration
Ethernet interface 0.0, the interface
for external access.
4
(config)# ip-share interface local-side
ether1.0
Configure IP sharing features on the
Ethernet interface 1.0, the interface
for internal access.
5
6
7
(config)# interface ether0.0
Enter
(config-ether0.0)#
mode.
(config-ether0.0)# ip address
192.168.1.2 255.255.255.0
Assign
(config-ether0.0)# interface ether1.0
Enter
(config-ether1.0)#
mode.
the interface configuration
the
IP
address
to
the
interface.
the interface configuration
8
(config-ether1.0)# no ip address
9
(config-ether1.0)# route 0.0.0.0
0.0.0.0 192.168.1.1
Assign the default router.
10
(config-ether1.0)# exit
Exits from the interface configuration
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11
Voice over IP
(config)#
mode.
(config)# exit
Exits from APOS global configuration
#
mode.
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Voice over IP
PAT Server (VoIP Gateway) Application
In this application, the VoIP gateway operates as a PAT server. The
VoIP gateway connected to the external network is assigned with a
public IP address and shares it with the equipment on the internal
network. This application is available for both dynamic IP address
environment via ADSL Modem or Cable Modem and fixed IP address
environment via ADSL modem or leased line.
NOTE
PAT Server Application is supported by AddPac Technology’s all
VoIP products along with VoIP Gateway
Network Diagram
A VoIP gateway is assigned with a public & dynamic IP through
PPPoE or DHCP.
Then it shares the public IP address with the
equipment of the internal network by using port mapping method.
Fig. 3-8 VoIP network diagram of VoIP gateway operating as PAT server
This is the typical application of VoIP gateway operating as PAT
server. In this case, VoIP network configuration and PAT static map
for address translation are also required.
The VoIP gateway offers both VoIP gateway function and PAT server
function. Thus the static TCP/UDP map configuration explained at
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Voice over IP
the previous chapter should be done on the gateway.
At the view of packet transmission path, this application is same as
that of Bridge mode. So QoS configuration of Ethernet 0.0 Interface
(LAN 0) of the VoIP Gateway is possible to allow higher priority for the
voice traffic.
With this QoS configuration, the VoIP Gateway can
offer priority and bandwidth control for all the data coming through
Ethernet 1.0 (LAN 1) and VoIP traffic as well, which realizes optimized
voice quality.
If the customer network is not allowed to change, the “IP sharing”
application is recommended.
APOS command script
!
nat-list
nat-list
nat-list
nat-list
nat-list
nat-list
nat-list
1
1
1
1
1
1
1
pat
pat
pat
pat
pat
pat
pat
static-entry tcp 1720 local
static-entry udp 5060 local
group-static-entry udp 22000
group-static-entry udp 23000
group-static-entry tcp 10000
group-static-entry tcp 14000
group-static-entry tcp 18000
22001
24999
10999
14999
18999
local
local
local
local
local
nat-list 1 pat static-entry tcp 23 local
nat-list 1 pat group-static-entry tcp 20 21 local
nat-list 1 pat group-static-entry udp 67 68 local
nat-list 1 pat static-entry icmp ping local
interface ether0.0
ip address dhcp
!
interface ether1.0
ip address 10.1.1.1 255.255.255.0
ip nat-group 1 pat ether0.0
ip dhcp-group 0
!
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
y
IP configuration for LAN0 : (DHCP or PPPoE or static)
y
IP address of default router: Optional
y
QoS configuration of LAN 0 Ethernet interface
y
NAT static map
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Voice over IP
y
NAT configuration binding in local interface (e1.0)
y
VoIP configuration
Step
Commands
Description
1
#
Enter
# config
Enter configuration commands,
line. End with CNTL/Z
(config)#
2
4
global
configuration
mode.
one
per
(config)#
H.323/Q.931
(config)# nat-list 1 pat static-entry
(TCP1720) for incoming calls.
tcp 1720
3
APOS
signaling
listen
port
local
(config)# nat-list 1 pat static-entry
listen Port(UDP 5060). SIP signaling listen
udp 5060 local
port (UDP5060) for incoming calls.
(config)# nat-list 1 pat group-static-
RAS and IRR listening port for GK
entry udp 22000 22001 local
5
6
(config)# nat-list 1 pat group-static-
RTP/RTCP
source
port
for
entry udp 23000 24999 local
communication
(config)# nat-list 1 pat group-static-
TCP source port for H.245 control
voice
entry tcp 10000 10999 local
7
(config)# nat-list 1 pat group-static-
Q931 Signaling Source Port
entry tcp 14000 14999 local
8
(config)# nat-list 1 pat group-static-
TCP listen port for H245 control
entry tcp 18000 18999 local
9
(config)# nat-list 1 pat static-entry
tcp 23
10
TCP listen Port (Telnet)
local
(config)# nat-list 1 pat group-static-
TCP listen Port (FTP)
entry tcp 20 21 local
11
(config)# nat-list 1 pat group-static-
TCP listen Port (BOOTP- for DHCP
entry udp 67 68 local
client).
When the public IP is assigned
by DHCP
12
(config)# nat-list 1 pat static-entry
icmp ping
TCP listen Port (ICMP - for Ping)
local
13
(config)# interface ether0.0
14
(config-ether0.0)# ip address dhcp
15
(config-ether0.0)# interface ether1.0
Enter
the
interface
configuration
mode.
16
(config-ether1.0)# ip address 10.1.1.1
Assign the IP address to the interface.
255.255.255.0
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17
Voice over IP
(config-ether1.0)# ip nat-group 1 pat
Share the public IP of LAN 0.0 with the
ether0.0 ip dhcp-group 0
local devices of LAN 1.0.
Invalid input command - (0)
18
19
(config-ether1.0)# exit
Exits from the interface configuration
(config)#
mode.
(config)# exit
Exits from APOS global configuration
#
mode.
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Voice over IP
Chapter 4. VoIP Network
Configuration
This chapter provides information for configuring Call Routing, E.164
and Gatekeeper related parameters along with additional features.
For more detailed information on APOS commands which are not
mentioned on this guide refer to APOS Operation Guide.
Point-to-Point Application
This application is recommended for the companies with only small
number of remote offices.
Each VoIP Gateway should have the
routing information such as dial-peer which is the called party
telephone number to be connected.
NOTE
Point-to-Point Application is supported by AddPac Technology’s all
VoIP products along with VoIP Gateway.
Network Diagram
Fig. 4-1 VoIP network diagram of peer-to-peer communication
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Voice over IP
APOS command script (Head Office)
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
!
dial-peer voice 0 pots
destination-pattern 8225683848
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 8325683848
session target 193.158.1.2
dtmf-relay h245-alphanumeric
!
dial-peer voice 1001 voip
destination-pattern 84T
session target 194.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Branch A)
!
hostname BA
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
!
dial-peer voice 0 pots
destination-pattern 8325683848
port 1/0
!
dial-peer voice 1000 voip
destination-pattern 82……..
session target 192.158.1.2
dtmf-relay h245-alphanumeric
!
dial-peer voice 1001 voip
destination-pattern 8425683848
session target 194.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Branch B)
!
hostname BB
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Voice over IP
!
interface ether0.0
ip address 194.168.1.1 255.255.255.0
!
!
dial-peer voice 0 pots
destination-pattern 8425683848
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 8225683848
session target 192.158.1.2
dtmf-relay h245-alphanumeric
!
dial-peer voice 1001 voip
destination-pattern 8325683848
session target 193.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
y
IP address of VoIP gateway
y
Default router
y
Dial-peer VoIP
y
Dial-peer POTS
y
VoIP interface
To configure point-to-point application, follow this procedure.
Step
1
Commands
Description
HO(config-ether0.0)# dial-peer voice 0 pots
Create a pots peer to group
HO(config-dialpeer-pots-0)#
destination
pattern
and
a
specific physical voice interface.
The tag number “0” is assigned
for the pots peer. (The valid tag
number range is “0 ~ 65,535”
and typically it starts from “0”.)
2
HO(config-dialpeer-pots-0)# destinationpattern 8225683848
Define
the
full
E.164
phone
number to be used for the dial
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peer.
3
HO(config-dialpeer-pots-0)# port 0/0
Associate a POTS dial peer with
a specific voice port. (The no. of
voice ports and their kinds are
different by each device.)
4
HO(config-dialpeer-pots-0)# dial-peer voice
1000 voip
HO(config-dialpeer-voip-1000)#
Create a VoIP dial peer for VoIP
call
setup.
The
tag
number
“1000” is assigned for the VoIP
peer. (The valid tag number
range
is
“0
~
65,535”
and
typically it starts from “1000”.)
5
6
7
8
9
HO(config-dialpeer-voip-1000)# destinationpattern 8325683848
Assign the called party number
HO(config-dialpeer-voip-1000)# session target
Send the VoIP call connection
193.158.1.2
messages to the gatekeeper.
HO(config-dialpeer-voip-1000)# dtmf-relay
h245-alphanumeric
Define the DTMF transmission
HO(config-dialpeer-voip-1000)# dial-peer voice
1001 voip
Create a VoIP dial-peer for VoIP
HO(config-dialpeer-voip-1001)# destinationpattern 84T
Assign the called party number
for the VoIP peer.
type as “H. 245 Alphanumeric”.
call setup.
stating with “84” for the VoIP
dial-peer.
10
11
12
HO(config-dialpeer-voip-1001)# session target
194.158.1.2
Send the VoIP call connection
HO(config-dialpeer-voip-1001)# dtmf-relay
h245-alphanumeric
Define the DTMF transmission
HO(config-dialpeer-voip-1001)# voip-interface
ether0.0
VOIP_INTERFACE_DOWN : (192.168.1.1)
VOIP_INTERFACE_UP : (192.168.1.1)
Gatekeeper shutdowned.
Assign VoIP interface.
message to the gatekeeper.
type as “H. 245 Alphanumeric”.
HO(config)#
13
HO(config)# exit
Exit
HO#
configuration mode.
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Voice over IP
Gatekeeper Interoperating Application
The VoIP network environment with Gatekeeper is recommended for
the middle and large scale enterprises or individual users using
Internet telephony services provided by ITSPs (Internet Telephony
Service Provider). Each VoIP Gateway registers its ID (a telephone
number) and establishes VoIP calls.
Thus, the VoIP Gateway
configuration is much simpler.
NOTE
Gatekeeper Interoperating Application is supported by AddPac
Technology’s all VoIP products along with VoIP Gateway.
Network Diagram
Fig. 4-2 VoIP network diagram of Gatekeeper interoperating application
APOS command script (Head office)
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
!
dial-peer voice 0 pots
destination-pattern 8225683848
port 0/0
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Voice over IP
!
dial-peer voice 1000 voip
destination-pattern 8325683848
session target ras
dtmf-relay h245-alphanumeric
!
dial-peer voice 1001 voip
destination-pattern 84T
session target ras
dtmf-relay h245-alphanumeric
!
gateway
h323-id addpac
gkip 199.168.1.1 1719 128
register
!
voip-interface ether0.0
!
APOS command script (Branch A)
!
hostname BA
!
interface ether0.0
ip address 193.168.1.1 255.255.255.0
!
!
dial-peer voice 0 pots
destination-pattern 8325683848
port 1/0
!
dial-peer voice 1000 voip
destination-pattern T
session target ras
dtmf-relay h245-alphanumeric
!
gateway
h323-id addpac
gkip 199.168.1.1 1719 128
register
!
voip-interface ether0.0
!
APOS command script (Branch B)
!
hostname BB
!
interface ether0.0
ip address 194.168.1.1 255.255.255.0
!
!
dial-peer voice 0 pots
destination-pattern 8425683848
port 0/0
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Voice over IP
!
dial-peer voice 1000 voip
destination-pattern 8T
session target 192.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Configuration Verification)
HO# show gateway
Gatekeeper Registration Information
H.323 id = addpac
gatekeeper registration option = enabled
gatekeeper security option = disabled
Gatekeeper registration status :
registered.
last registration reject information from gatekeeper
ConfigAsNoRegistration (Aug 9 03:02:43)
Gatekeeper list :
199.168.1.1
1719
priority(128)
by user
Local aliases
[1] H323ID : addpac
[2] 8225683848
Technical prefixes
Gateway Information
status = init 1 (waiting for setting IP address on a VoIP interface)
product name = AddPac VoIP
product version = 6.12
endpoint type = gateway
discovery (send GRQ) = disabled
ARQ option = arq default
LRQ option = no lrq
lightweight IRR = disabled
TTL margin = 20 %
h323 call start mode = fast
h323 call tunneling mode = enabled
h323 call channel mode = late
h323 response msg = default
system fax mode = t38
system fax rate (bps) = 9600
system T.38 fax redundancy = 0
force to send startH245 = enabled
dialPeer hunt algorithm = longest - preference - random
translate voip incoming called number = -1
translate voip incoming calling number = -1
local ringback tone = normal
end of digit = #
ip address prefix = *
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Voice over IP
voice confirmed connect on FXO/E&M = disabled
number
number
number
number
number
number
number
of
of
of
of
of
of
of
ports = 1
pots peers = 1
voip peers = 2
number expansions = 0
codec classes = 0
alternate gatekeepers = 1
current calls = 0
Announcement Option
language = korean
element : delayed dial = disabled
element : wrong number = disabled
element : connection fail = disabled
Timer & Counter parameter value
tinit (initial digit timer) = 10 sec.
tring (ring timer) = 30 sec.
t301 (alert -> connect) = 180 sec.
t303 (setup -> alert) = 20 sec.
tras (RAS msg ack timer) = 6 sec.
tttl (RAS Time To Live timer) = 60 sec.
tidt (inter digit timer) = 3 sec.
treg (GK Registration retry timer) = 20 sec.
treg2 (GK Registration retry timer : long period by RRJ) = 120 sec.
tohd (On Hook Delay Time) = 0 sec.
tpoll (polling timer on trunk or polling type connection) = 180 sec.
dtmf duration = 150 msec.
dtmf guard time = 100 msec.
cras (RAS retry counter) = 3
Remote Call Log (syslog)
primary server =
secondary server =
interval = 0 minutes
cdr format type = 0
Assigned VoIP TCP/UDP ports
minimized assign = no
Q.931 signalling port (TCP listen) = 1720
SIP signalling port (UDP listen) = 5060
H.245 control port (TCP src) = 10000 - 10999
Q.931 signalling port (TCP src) = 14000 - 14499
H.245 control port (TCP listen) = 18000 - 18999
RAS port and IRR port (UDP listen) = 22000, 22001
RTP/RTCP port (UDP listen) = 23000 - 24999
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
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Voice over IP
y
IP address of VoIP gateway
y
Default router
y
E.164 number for VoIP gateway registration
y
H.323 ID
y
IP address of VoIP Gatekeeper
To configure Gatekeeper Interoperating Application, follow this
procedure.
Step
Commands
Description
1
HO(config-ether0.0)# dial-peer voice 0 pots
Create a pots peer to group
destination
pattern
and
a
specific physical voice interface.
The tag number “0” is assigned
for the pots peer. (The valid tag
number range is “0 ~ 65,535”
and typically it starts from “0”.)
2
HO(config-dialpeer-pots-0)# destinationpattern 8225683848
Define
the
full
E.164
phone
number to be used for the dial
peer.
3
HO(config-dialpeer-pots-0)# port 0/0
Associate a POTS dial peer with
a specific voice port. (The no. of
voice ports and their kinds are
different by each device.)
4
HO(config-dialpeer-pots-0)# dial-peer voice
1000 voip
Create a VoIP dial peer for VoIP
call
setup.
The
tag
number
“1000” is assigned for the VoIP
peer. (The valid tag number
range
is
“0
~
65,535”
and
typically it starts from “1000”.)
5
6
7
8
HO(config-dialpeer-voip-1000)# destinationpattern 8325683848
Assign the called party number
HO(config-dialpeer-voip-1000)# session target
ras
Send the VoIP call connection
HO(config-dialpeer-voip-1000)# dtmf-relay
h245-alphanumeric
Define the DTMF transmission
HO(config-dialpeer-voip-1000)# dial-peer voice
1001 voip
Create a VoIP dial-peer for VoIP
for the VoIP peer.
message to the gatekeeper.
type as “H. 245 Alphanumeric”.
call setup.
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9
HO(config-dialpeer-voip-1001)# destinationpattern 84T
Voice over IP
Assign the called party number
stating with “84” for the VoIP
dial-peer.
10
11
12
HO(config-dialpeer-voip-1001)# session target
ras
Send the VoIP call connection
HO(config-dialpeer-voip-1001)# dtmf-relay
h245-alphanumeric
Define the DTMF transmission
HO(config-dialpeer-voip-1001)# gateway
Enter
message to the gatekeeper.
type as “H. 245 Alphanumeric”.
the
gatekeeper
configuration mode.
13
HO(config-gateway)# h323-id addpac
Assign H.323 ID.
14
HO(config-gateway)# gkip 199.168.1.1 1719 128
Assign the IP address of the
gatekeeper.
15
HO(config-gateway)# register
Register to the gatekeeper.
16
HO(config-gateway)# voip-interface ether0.0
VOIP_INTERFACE_DOWN : (192.168.1.1)
VOIP_INTERFACE_UP : (192.168.1.1)
Gatekeeper shutdowned.
Assign VoIP interface.
HO(config)#
17
HO(config)# exit
Exit
HO#
configuration mode.
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Voice over IP
Number Translation Feature
This part provides information about prefixing or digit stripping
number translation of called party and calling party telephone
numbers at the VoIP gateway.
NOTE
Number Translation is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway.
Network Diagram
Fig. 4-3 VoIP gateway number translation feature diagram
APOS command script
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 8225683848
port 0/0
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target ras
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Voice over IP
dtmf-relay h245-alphanumeric
translate-outgoing called-number 0
!
! Gateway configuration.
!
gateway
h323-id addpac
gkip 199.168.1.1 1719 128
register
!
!
! Translation Rule configuration.
!
translation-rule 0
rule 0
[1-9]T
02%99
!
Number Translation Example
rule 0 1234T %01%03%99
Translated numbers
1234
Æ 134
12345678 Æ 1345678
1235678
Æ 1235678 (the rule is not applied.)
rule 0 T %04%03%98
Translated numbers
1235
Æ 53
1235678
Æ 53
1245678
Æ 54
rule 0 T 999%03%04%99
Translated numbers
1236
Æ 99936
12345678 Æ 999345678
1235678
Æ 99935678
rule 0 [1-3]T 000%99
Translated numbers
1234
Æ 0001234
2345678
Æ 0002345678
4567890
Æ 4567890 (the rule is not applied.)
rule 0 [1-3]T %01%02%03%98
Translated numbers
1234
Æ 123
2345678
Æ 234
4567890
Æ 4567890 (the rule is not applied.)
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Voice over IP
APOS command script (Configuration Verification)
HO(config)# show
translation-rule
translation-rule 0
rule 0
[1-9]T
HO(config)# show
02%99
translation-rule 0 1234
The translation result is (021234)
HO(config)# show
translation-rule 0 021234
The translation result is (021234)!
Related APOS commands & structure
At the above diagram, VoIP gateway prefixes “02” for all the called
party number. However, if the called party number starts with “0”,
there is no prefixing.
Please note the configuration of translation rules and how the rule is
applied to the VoIP peer.
Configure the below parameters appropriate for the network
environment.
y
E.164 number for registration of VoIP Gateway or VoIP router
y
H.323 ID (at gatekeeper interoperating mode)
y
IP address of the gatekeeper (at gatekeeper interoperating
mode)
y
ID number of the gatekeeper (at gatekeeper interoperating
mode)
y
Number translation rules
To configure the feature, follow this procedure.
Step
Commands
Description
1
BB(config-dialpeer-voip-1000)# translateoutgoing called-number 0
Apply the Translation rule 0 to
the called party number of the
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dial-peer 1000.
2
BB(config-gateway)# translation-rule 0
3
BB(config-translation-rule#0)# rule 0
02%99
[1-9]T
Prefix “02” if the number starts
with the digit among “1-9”
Ex.) 12345678
-Æ 0212345678
“%99” refers to the rest digits
except the first digit.
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Voice over IP
Call Pickup & Transfer Feature
The call pick-up feature allows the user to answer a call that comes
in on a number other than his/her own. Also the users can transfer
an established call to other numbers with the call transfer feature.
NOTE
Call Pickup & Transfer is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway.
Network Diagram
The below is the network diagram of Call-pickup feature.
Fig. 4-4 VoIP gateway Call-pickup feature
According to the above examples, the user of the telephone 1000 at
the head office tries a call to the telephone 2000 at the branch
office A.
When the user of telephone 2000 is absent, the telephone
2001 picks up the call by pressing special keys “##”.
NOTE
The special key (“##”) used here is an example, and the VoIP
Gateway operators are allowed to choose any keys.
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Voice over IP
Fig. 4-5 VoIP gateway Call-transfer feature
In the above example, the user of the telephone 1000 at the head
office calls to the telephone 2000 at the branch office. The called
party picks up the phone and finishes the conversation. When the
called party tries to transfer the call to the user of telephone 2001,
the called party presses the special key for call transfer (“Hook-flash”)
and “2001”. Then the telephone 2001 rings and with the hook-off of
the telephone 2001, the user of telephone 2000 hangs up. Then the
call between the telephone 1000 and the telephone 2001 is
established.
NOTE
The special keys can be not changed by the VoIP Gateway
operator.
APOS command script (Call pick-up & transfer configuration)
!
hostname BB
!
interface ether0.0
ip address 194.168.1.1 255.255.255.0
!
!
dial-peer voice 0 pots
destination-pattern 2000
port 0/0
!
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Voice over IP
dial-peer voice 1 pots
destination-pattern 2001
port 0/1
!
dial-peer voice 1000 voip
destination-pattern 1…
session target 192.158.1.2
dtmf-relay h245-alphanumeric
!
dial-peer call-pickup ##
dial-peer call-transfer h
!
voip-interface ether0.0
!
APOS command script (Call pick-up & transfer configuration Verification)
Branch-A# show gateway
Gatekeeper Registration Information
H.323 id = addpac
gatekeeper registration option = enabled
gatekeeper security option = disabled
Gatekeeper registration status :
not registered.
last registration reject information from gatekeeper
ConfigAsNoRegistration (Aug 9 03:02:43)
Gatekeeper list :
Local
[1]
[2]
[3]
aliases
H323ID : addpac
2000
2001
Technical prefixes
Gateway Information
status = init 1 (waiting for setting IP address on a VoIP interface)
product name = AddPac VoIP
product version = 6.12
endpoint type = gateway
discovery (send GRQ) = disabled
ARQ option = arq default
LRQ option = no lrq
lightweight IRR = disabled
TTL margin = 20 %
h323 call start mode = fast
h323 call tunneling mode = enabled
h323 call channel mode = late
h323 response msg = default
system fax mode = t38
system fax rate (bps) = 9600
system T.38 fax redundancy = 0
force to send startH245 = enabled
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Voice over IP
dialPeer hunt algorithm = longest - preference - random
translate voip incoming called number = -1
translate voip incoming calling number = -1
local ringback tone = normal
end of digit = #
ip address prefix = *
voice confirmed connect on FXO/E&M = disabled
call pickup digits = ##
call transfer = enabled (hookflash)
number
number
number
number
number
number
number
of
of
of
of
of
of
of
ports = 1
pots peers = 3
voip peers = 2
number expansions = 0
codec classes = 0
alternate gatekeepers = 1
current calls = 0
Announcement Option
language = korean
element : delayed dial = disabled
element : wrong number = disabled
element : connection fail = disabled
Timer & Counter parameter value
tinit (initial digit timer) = 10 sec.
tring (ring timer) = 30 sec.
t301 (alert -> connect) = 180 sec.
t303 (setup -> alert) = 20 sec.
tras (RAS msg ack timer) = 6 sec.
tttl (RAS Time To Live timer) = 60 sec.
tidt (inter digit timer) = 3 sec.
treg (GK Registration retry timer) = 20 sec.
treg2 (GK Registration retry timer : long period by RRJ) = 120 sec.
tohd (On Hook Delay Time) = 0 sec.
tpoll (polling timer on trunk or polling type connection) = 180 sec.
dtmf duration = 150 msec.
dtmf guard time = 100 msec.
cras (RAS retry counter) = 3
Remote Call Log (syslog)
primary server =
secondary server =
interval = 0 minutes
cdr format type = 0
Assigned VoIP TCP/UDP ports
minimized assign = no
Q.931 signalling port (TCP listen) = 1720
SIP signalling port (UDP listen) = 5060
H.245 control port (TCP src) = 10000 - 10999
Q.931 signalling port (TCP src) = 14000 - 14499
H.245 control port (TCP listen) = 18000 - 18999
RAS port and IRR port (UDP listen) = 22000, 22001
RTP/RTCP port (UDP listen) = 23000 - 24999
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Related APOS commands & structure
y
E.164 number for registration of VoIP Gateway or VoIP router
y
H.323 ID (at gatekeeper interoperating mode)
y
IP address of the gatekeeper (at gatekeeper interoperating
mode)
y
ID number of the gatekeeper (at gatekeeper interoperating
mode)
y
call Transfer configuration
y
call pick-up configuration
To configure the feature, follow this procedure.
Step
Commands
Description
1
BB(config-dialpeer-voip-1000)# dial-peer callpickup ##
Enable the Call pick-up features.
(“##” is a special key randomly
assigned for the feature.)
2
BB(config)# dial-peer call-transfer h
Enable Call transfer feature. (“h”
means “hook-flash”)
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Voice over IP
Chapter 5. VoIP Protocol
Configuration
This chapter provides information on configuring VoIP signaling
protocols. AddPac Technology’s VoIP Gateway supports H.323, SIP
and MGCP protocols. H.323 is mainly explained at this chapter. SIP
and MGCP related configuration information is also included.
NOTE
H.323, SIP and MGCP VoIP signaling protocols are supported by
AddPac Technology’s all VoIP products along with VoIP Gateway.
VoIP Protocol
AddPac’s VoIP products supports below VoIP signaling protocols.
H.323 Protocol Application
The APOS configuration examples of the guide are based on H.323
VoIP
protocol.
For
detailed
H.323
VoIP
signaling
protocol
configuration, refer to the each related chapter.
NOTE
H.323 VoIP signaling protocol is supported by AddPac Technology’s
all VoIP products along with VoIP Gateway.
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Voice over IP
SIP Protocol (Direct Call) Application
VoIP calls with SIP protocol have two kinds of call connection types;
direct connection and indirect connection via SIP Proxy Server. The
below is the configuration example of Point-to-Point calls in SIP direct
call mode.
NOTE
SIP Protocol (Direct Call) application is supported by AddPac
Technology’s all VoIP products along with VoIP Gateway.
Network Diagram
Fig. 5-1 VoIP network diagram of SIP direct call configuration
APOS command script (Head office)
!
hostname HO
!
interface ether0.0
ip address 192.168.1.2 255.255.255.0
!
route 0.0.0.0 0.0.0.0 192.168.1.1
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 8225683848
port 0/0
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
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destination-pattern T
session target 194.168.1.2
session protocol sip
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Branch A)
!
hostname BA
!
interface ether0.0
ip address 194.168.1.2 255.255.255.0
!
route 0.0.0.0 0.0.0.0 194.168.1.1
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 8425683848
port 0/0
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target 192.168.1.2
session protocol sip
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
y
IP address of the VoIP gateway
y
Default router
y
E.164 for VoIP gateway registration
y
IP address of DNS
y
IP address of VoIP Peer
To configure the application, follow this procedure.
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Voice over IP
Step
Commands
Description
1
# config
Enter
(config)#
configuration mode.
(config)#
Enter
(config)# sip-ua
Configuration mode. Enter
(config-sip-ua)# ?
“?” to check the possible
2
APOS
SIP
global
User
Agent
commands.
no
register
signalling-port
sip-server
sip-username
sip-password
timeout
end
exit
3
4
set to default configuration
try registration to sip registrar
set SIP signalling port (default 5060)
Configure a SIP Server Interface
Set Username of SIP User Agent
Set Password of SIP User Agent
Set timeout value
Go to Top menu
Exit from the EXEC
(config-sip-ua)# exit
Exit from SIP User Agent
(config)#
Configuration mode.
(config)# exit
Exits
#
configuration mode.
from
APOS
global
Step
Commands
Description
1
HO(config-dialpeer-pots-0)# dial-peer voice
1000 voip
Create a VoIP dial peer for VoIP
call
setup.
The
tag
number
“1000” is assigned for the VoIP
peer. (The valid tag number
range
is
“0
~
65,535”
and
typically it starts from “1000”.)
2
3
4
5
HO(config-dialpeer-voip-1000)#
pattern T
HO(config-dialpeer-voip-1000)#
194.168.1.2
HO(config-dialpeer-voip-1000)#
protocol sip
HO(config-dialpeer-voip-1000)#
h245-alphanumeric
destinationsession target
session
dtmf-relay
Define the DTMF transmission
type as “H. 245 Alphanumeric”.
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Voice over IP
SIP Protocol (Indirect, Proxy Server) Application
VoIP calls with SIP signaling protocol have two kinds of call
connection type; direct connection and indirect connection via SIP
Proxy Server. The below is the configuration example of Point-topoint SIP indirect calls made via SIP Proxy Server.
NOTE
SIP Protocol (Indirect, Proxy Server) application is supported by
AddPac Technology’s all VoIP products along with VoIP Gateway.
Network Diagram
Fig. 5-2 VoIP network diagram of SIP indirect calls via SIP Proxy server
APOS command script (Head office)
!
hostname HO
!
interface ether0.0
ip address 192.168.1.2 255.255.255.0
!
route 0.0.0.0 0.0.0.0 192.168.1.1
!
dnshost nameserver 199.168.1.2
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 8225683848
port 0/0
!
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Voice over IP
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
dtmf-relay h245-alphanumeric
!
!! Gateway configuration.
!
!
! SIP UA configuration.
!
sip-ua
sip-username 8225683848
sip-password AddPac-HO
sip-server proxy.addpac-test.com
register e164
!
voip-interface ether0.0
!
APOS command script (Branch A)
!
hostname BA
!
interface ether0.0
ip address 194.168.1.2 255.255.255.0
!
route 0.0.0.0 0.0.0.0 194.168.1.1
!
dnshost nameserver 199.168.1.2
!
! Pots peer configuration.
!
dial-peer voice 0 pots
destination-pattern 8425683848
port 0/0
!
!
!
! Voip peer configuration.
!
dial-peer voice 1000 voip
destination-pattern T
session target sip-server
session protocol sip
dtmf-relay h245-alphanumeric
!
!! Gateway configuration.
!
!
! SIP UA configuration.
!
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Voice over IP
sip-ua
sip-username 8425683848
sip-password AddPac-A
sip-server proxy.addpac-test.com
register e164
!
voip-interface ether0.0
!
Related APOS commands & structure
This application is similar to H.323 application using GK, which is
typical configuration of commercial VoIP network, or middle and
large scale enterprise VoIP network.
Each end point SIP terminal
requires authentication from SIP Server to establish calls.
To use
domain name instead of IP address, Domain Name Server (DNS) is
required.
The below example uses DNS to establish calls.
Configure the below parameters appropriate for the network
environment.
y
IP address of VoIP gateway
y
Default router
y
E.164 for registering gw
y
IP address of DNS
y
IP address of SIP Proxy Server
y
SIP user name
y
SIP password
To configure the application, follow this procedure.
Step
Commands
1
HO(config)# dnshost nameserver 199.168.1.2
2
HO(config-dialpeer-pots-0)# dial-peer voice
1000 voip
Description
Create a VoIP dial peer for
VoIP
call
setup.
The
tag
number “1000” is assigned for
the VoIP peer. (The valid tag
number range is “0 ~ 65,535”
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and typically it starts from
“1000”.)
3
4
5
6
HO(config-dialpeer-voip-1000)#
pattern T
HO(config-dialpeer-voip-1000)#
sip-server
HO(config-dialpeer-voip-1000)#
sip
HO(config-dialpeer-voip-1000)#
Alphanumeric
destinationsession target
session protocol
dtmf-relay h245-
Define the DTMF transmission
type
as
“H.
245
Alphanumeric”.
7
HO(config-dialpeer-voip-1000)# sip-ua
8
HO(config-sip-ua)# sip-username 8225683848
9
HO(config-sip-ua)# sip-password AddPac-HO
10
HO(config-sip-ua)# sip-server proxy.addpactest.com
11
HO(config-sip-ua)# register e164
Resiger E.164 number
Username/Password Registration of SIP Dial-Peer
A separate username and password can be assigned for each dialpeer. Until now, the gateway with multiple E.164 numbers is only
assigned with one username and password, and the separate
authentication of each E.164 is not applicable. However, APOS v 7.0
supports username and password registration function.
That is, if the user assigns e.164 100 at dial-peer 1, e.164 200 at dialpeer 2, and also assigns usernames and passwords for each dialpeer, then the gateway sends Registration Request to SIP server two
different times for each dial-peer. Thus separate registration process
is possible for each dial-peer.
This newly added command is the sub-command of dial-peer
command, and the same command already exists as the subcommand of the sip-ua command. That’s why the users are
requested to pay attention to the priority. When the user name and
password is configured at both dial-peer command and sip-ua
command, the sip-ua command is only applied due to its higher
priority. Thus the user name and password setting of the dial-peer
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Voice over IP
command is ignored.
That means, if sip-username and sip-password of sip-ua is assigned,
and sip-username and sip-password of dial-peer is also assigned at
the same time, APOS gives the higher priority to the global
configuration that affects the entire gateway. Therefore, the
username and password setting at dial-peer is ignored.
Related APOS commands & structure
dial-peer command
(config)# dial-peer voice 0 pots
(config-dialpeer-pots-0)#
user-name
set username of dial peer
user-password
set password of dial peer
(config-dialpeer-pots-0)# user-name <string>
(config-dialpeer-pots-0)# user-password <string>
sip-ua command
(config)# sip-ua
(config-sip-ua)#
sip-username
Set Username of SIP User Agent
sip-password
Set Password of SIP User Agent
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Voice over IP
MGCP Protocol Application
This chapters provides information on APOS commands of MGCP
VoIP protocol. For further details, refer to APOS Operation Guide.
NOTE
MGCP Protocol application is supported by AddPac Technology’s
all VoIP products along with VoIP Gateway.
Network Diagram
Fig. 5-3 VoIP network diagram based on MGCP protocol
Main APOS Commands for MGCP Protocol
Enters MGCP configuration mode
Step
Commands
Description
1
# config
Enter
(config)#
configuration mode.
(config)#
Enter MGCP Configuration
(config)# MGCP
mode.
(config-MGCP)# ?
commands by entering “?”.
2
no
register
call-agent
default-package
dtmf-relay
restart-delay
timeout
end
APOS
global
Check the related
set to default configuaration
Enable MGCP
Specify address of call-agent
Select the Default Package Capability
Configure mgcp dtmf-realy
Specify the Restart Delay timer value
Set timeout value
Go to Top menu
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exit
3
4
Voice over IP
Exit from the EXEC
(config-MGCP)# exit
Exit
from
(config)#
configuration mode.
(config)# exit
Exit
#
configuration mode.
from
MGCP
APOS
global
MGCP Register Command
Step
Commands
Description
1
# config
Enter
(config)#
configuration mode.
(config)#
Enter MGCP Configuration
(config)# MGCP
mode.
2
APOS
global
(config-MGCP)#
3
(config-MGCP)# register ?
<0-65536>
<cr>
Enable MGCP with user specified UDP port number
Register
command
is
for
MGC registration. It sends
(config-MGCP)#
RSIP (restart) message to
(config-MGCP)# exit
MGC. Also, the local port
(config)#
no. of MG(Media Gateway)
(config)#
can be configured.
(config)#
4
(config)# exit
Exit
from
APOS
global
#
configuration mode.
MGCP Call Agent Command
Step
Commands
Description
1
# config
Enter
(config)#
configuration mode.
(config)#
Enter MGCP Configuration
(config)# MGCP
mode.
2
APOS
global
(config-MGCP)#
3
(config-MGCP)# call-agent ?
alias
set Hostname or IP address of the call-agent
(config-MGCP)# call-agent 1.1.1.1 ?
<0-65536>
<cr>
port number (default 2427)
Assign IP address or domain
name
of
MGC
or
Soft
Switch. Also, the port no. of
MGC can be configured.
(config-MGCP)# call-agent 1.1.1.1 2427
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<0-254>
Voice over IP
priority (default 128)
(config-MGCP)# exit
(config)#
(config)#
4
(config)# exit
Exit
from
APOS
#
configuration mode.
global
MGCP Package Command
Step
Commands
Description
1
# config
Enter
(config)#
configuration mode.
(config)#
Configure
(config)# MGCP
package
2
(config-MGCP)# default-package ?
as-package
dtmf-package
gm-package
hs-package
line-package
trunk-package
3
4
Select the Announcement Server Package
Select the DTMF Package
Select the Generic Media Package
Select the Handset Package
Select the Line Package
Select the Trunk Package
APOS
global
the
for
default
the
Media
Gateway.
Default: Line-package
(config-MGCP)# exit
Exit
from
(config)#
configuration mode.
(config)# exit
Exit
#
configuration mode.
from
MGCP
APOS
global
MGCP DTMF Relay Command
Step
Commands
Description
1
# config
Enter
(config)#
configuration mode.
(config)#
Enter MGCP Configuration
(config)# MGCP
mode.
2
APOS
global
(config-MGCP)#
3
(config-MGCP)# dtmf-relay ?
rtp-2833
DTMF relay by RTP payload defined by RFC 2833
out-of-band
DTMF relay by out-of-band signal
Select DTMF Relay type.
(config)#
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4
5
6
Voice over IP
(config-mgcp)# no dtmf-relay
Assign DTMF relay as in-
(config)#
band type.
(config-mgcp)# dtmf-relay rtp-2833
Assign DTMF relay according
(config)#
to the RFC-2833 standard.
(config-mgcp)# dtmf-relay out-of-band
Assign DTMF relay as out-of-
(config)#
band
(config)#
transmitted
type.
DTMF
is
with
NTFY
APOS
global
message.
7
(config)# exit
Exit
from
#
configuration mode.
MGCP Restart Relay command
Step
Commands
Description
1
# config
Enter
(config)#
configuration mode.
(config)#
Enter MGCP Configuration
(config)# MGCP
mode.
2
APOS
global
(config-MGCP)#
3
Configure RSIP message
(config-MGCP)# restart-delay ?
<0 - 500>
Select the Restart Delay timer value (sec)
transmission delay after
(config-MGCP)#
executing register
(config-MGCP)# exit
command. For examples, if
(config)#
the delay is “10sec”, the
(config)#
RSIP message is sent in 10
(config)#
seconds after executing
register command at MG.
( Default: 5sec)
4
(config)# exit
Exit
from
APOS
#
configuration mode.
global
MGCP Timeout Command
MGCP Timeout commands are: Tretry, Tmax, Thist. Trtry configures
message retransmission time, and the default is 4 sec.
configures the maximum Tretry time.
Tmax
The message retransmission
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time should not be longer than Thist time. The default value is 20 sec.
The message is retransmitted at every 4 seconds within the Tmax time
(20 seconds). Thist configures the max. retransmission time. The
default is 30 sec.
After Tmax timer is expired, it stands by for 30
seconds.
Step
Commands
Description
1
# config
Enter APOS global configuration
(config)#
mode..
(config)#
Configure MGCP Timeout values.
2
(config)# MGCP
(config-MGCP)# timeout ?
tretry
thist
tmax
3
4
set MGCP retry timeout value (msec)
set MGCP hist timeout value (sec)
set MGCP max timeout value (sec)
Message retry timeout
Max. message retransmission time
Max. Tretry time
(config-MGCP)# exit
Exits
from
MGCP
(config)#
mode.
(config)# exit
Exits
#
configuration mode.
from
configuration
APOS
global
MGCP Voice port configuration command
Step
Commands
Description
1
# config
Enter APOS global configuration
(config)#
mode.
(config)#
Enter
(config)# MGCP
mode.
2
MGCP
Configuration
(config-MGCP)#
3
(config-MGCP)# dial-peer voice 0 pots
(config-dialpeer-pots-0)#
4
(config-dialpeer-pots-0)# port 0/0
Assign a voice port operating
(config-dialpeer-pots-0)# application mgcpapp
with MGCP.
(config-dialpeer-pots-0)#
5
6
(config-dialpeer-pots-0)# exit
Exit
(config)#
Configuration mode.
(config)# exit
Exit
#
configuration mode.
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MGCP End-point ID configuration command
The
MGCP
End-point
number@domain-name”.
format
is
“aaln/slot-number/port-
The APOS command for hostname
configuration can be used for domain name configuration.
With the domain name, “111.222.333.444”, the End-point ID of voice
port 0/0 is aaln/0/0@111.222.333.444.
Step
Commands
Description
1
# config
Enter APOS global configuration mode..
(config)#
2
(config)#
Inquire for Hastname command.
(config)# hostname ?
<hostname> Hostname of this system
(config-MGCP)#
3
(config)# hostname 111.222.333.444
Assign the domain name.
111.222.333.444(config)#
111.222.333.444(config)# exit
4
111.222.333.444# exit
Exit from APOS global configuration
#
mode..
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Chapter 6. Voice Interface
Configuration
This chapter provides information on VoIP Gateway voice interface
configuration of gain/tone control and various voice interfaces such
as FXS, FXO and E&M.
Input & Output Gain configuration
This part provides information on APOS commands and parameters
commonly used for voice interface configuration. Make sure to
consider all the equipment including PBX on the network when
configuring input and output gain of the VoIP gateway.
At the calling party’s viewpoint, the input gain can be considered as
the volume of a microphone. If the voice volume on the called
party is too loud, reduce the input gain of the gateway. On the
other hands, the out put gain can be considered as the volume of a
speakerphone. If the volume of the phone or PBX connected to
VoIP gateway is too loud, reduce the output gain.
The default value is “0”. However, considering the natural decrease
on PSTN, set the value “+3dB” or “+6dB”.
NOTE
The default value doesn’t consider specific network condition of
each user. If the voice volume is too loud or there is echo and
noise, decrease the input and output gain to eliminate the
background noise.
NOTE
This application is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway.
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Network Diagram
Fig. 6-1 VoIP Gateway Input/Output gain
Input gain increases or decreases the voice volume coming to the
VoIP gateway from the voice ports.
Also, the out gain increases or
decreases the voice volume coming from the IP network to the VOIP
gateway.
The default input/output gain value is 0dB.
The valid range is “-
18dB~ +8dB”.
APOS command script
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
!
voice-port 1/0
input gain 2
output gain 3
!
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
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y
Voice over IP
Input and output gain value
To configure the gain, follow this procedure.
Step
Commands
Description
1
HO(config-ether0.0)# voice-port 1/0
Configure the voice port 1/0.
2
HO(config-voice-port-1/0)# input gain 2
Increase the input again by 2
HO(config-voice-port-1/0)#
dB.
HO(config-voice-port-1/0)# output gain 3
Increase the output gain by 3dB.
3
HO(config-voice-port-1/0)#
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Tone Configuration
Various tones such as dial tone, busy tone, reoder tone, ringback
tone, linelock tone and etc can be configured by APOS commands.
At this guide, the reorder tone configuration is provided as an
example.
NOTE
This application is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway.
Network Diagram
Fig. 6-2 VoIP gateway tone setting
The above figure is an APOS configuration example of reorder tone
between PBX and a head office. Reorder tone is a tone used to
terminate calls between analog ports of PBX or VoIP Gateway. With
the wrong tone values, the call cannot be terminated. Generally,
PBXs and PSTN follow the tone standard by the government.
However, tone values of some PBXs or the extension lines of PBXs are
non-standard, so the call cannot be terminated while interoperating
with VoIP GW. In this case, modify the tone values of VoIP GW.
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When the reorder tone is set, FXO interface detects the tone, and
FXS interface plays the tone.
Use the “tone” command and its
options for the configuration of various tones.
APOS command script
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
!
!
dial-peer voice 0 pots
destination-pattern 5683847
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 5683848
session target 193.158.1.2
dtmf-relay h245-alphanumeric
!
! Tones
voice class reorder-tone 350 520 350 300 0 0 -12
!
voip-interface ether0.0
!
APOS command script (Tone Configuration Verification)
#
# show tone
Tag
Low(Hz)
High(Hz)
On1(ms)
Off1(ms)
On2(ms)
Off2(ms)
dBm
Description
---------------------------------------------------------------------------------------------------
350
440
10000
0
0
0
-18
Dial tone
-
440
480
1000
2000
0
0
-12
RingBack tone
-
480
620
500
500
0
0
-12
LineBusy tone
-
350
520
350
300
0
0
-12
Reorder tone
-
1400
2060
100
100
0
0
0
LineLock tone
#
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
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y
Voice over IP
The frequency of reorder tone
To configure polarity inverse, follow this procedure.
Step
Commands
1
HO(config-dialpeer-voip-1000)# voice class
reorder-tone 350 520 350 300 0 0 -12
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E1/T1 Voice Interface Configuration/ ISDN-PRI
This chapter offers information about the common APOS commands
for E1/T1 configuration.
For more detailed configuration and for
parts are not mentioned here, refer to APOS Operation Guide.
The common and basic commands related to E1/T1 ISDN-PRI
configuration are mentioned below.
NOTE
This configuration is applied to AddPac Technology’s all VoIP
products along with VoIP Gateway which can be equipped with
digital E1/T1 voice interface module.
Network Diagram
Fig. 6-3 VoIP gateway digital E1/T1 ISDN-PRI
The digital E1/T1 voice interface configurations can be divided by
the signaling types; ISDN –PRI, R2 & DTMF. At the example, the PBX
and VoIP gateway at the head office are operated with ISDN-PRI
signaling type. In case of ISDN-PRI, the interface protocol between
PBX and PBX or between PBX and VoIP gateway should be a pair of
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“network side” and “user”.
As you can see from the above
example, PBX operates as ”user side” and the VoIP at the head
office operates as “network.” Also, the compand-type of PBX and
the VoIP GW should be configured same according to the PCM type
(A-law or U-law).
The below is the default parameters of digital E1 ISDN-PRI of the VoIP
Gateway.
y
Signaling type: Un-defined
y
Clock: Master
y
No. of channels: None
y
Compand-type : A-law
y
Protocol: network
APOS command script
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
! PRI controller configuration.
!
controller e1(t1) 0/0
signalling-type isdn
channel-group timeslots 1-31 0
isdn protocol-emulate network
!
voice-port 0/0
! E1(t1)
compand-type u-law
!
dial-peer voice 0 pots
destination-pattern 99T
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 5683848
session target 193.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Configuration Verification)
HO# show controller 0/0
Controller T1 slot(0)/port(0)
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T1 Link is UP
No Alarm detected.
Applique type is Channelized T1.
Framing is SF, Line Code is AMI, Cable Length is Short 110.
Signalling type is ISDN PRI.
0 Line Code Violations, 0 Framing Bit Errors
0 Out Of Frame Errors, 0 Bit Errors
6 Frames Received, 6 Frames Transmitted
signalling type = isdn
clock source = master
channel group 0 = 1-24
1
2
3
allocated timeslots = YYYYYYYYYYYYYYYYYYYYYYYNNNNNNNN
outgoing barred channel group =
channel order = descending
b-channel negotiation = exclusive
overlap receiving = enabled
protocol side = user
R2 get calling number = disabled
ISDN virtual connect = disabled
T1 cable length = short 110
T1 framing = sf
T1 line code = ami
T1 CAS type = immediate
ISDN Layer 2 is UP
ISDN Values
ISDN Layer 2 values
k
= 7
N200 = 3
N201 = 260
T200 = 1 seconds
T203 = 10 seconds
ISDN Layer 3 values
T301 = 180 seconds
T302 = 15 seconds
T303 = 4 seconds
T305 = 30 seconds
T306 = 30 seconds
T308 = 4 seconds
T310 = 10 seconds
T313 = 4 seconds
T316 = 120 seconds
T309 = 90 seconds
N303 = 1
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
Signaling-type
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No. of Channel-groups
Clock type
Compand-type
Protocol type
To configure the application, follow this procedure.
Step
Commands
1
HO(config-ether0.0)# controller e1(t1) 0/0
2
HO(config-ether0.0)# signalling-type isdn
3
HO(config-ether0.0)# channel-group timeslots 1-31
Description
0
4
HO(config-ether0.0)# isdn protocol-emulate
network
5
HO(config-ether0.0)# voice-port 0/0
Enter
Voice
Port
Configuration mode.
6
HO(config-voice-port-0/0)# compand-type u-law
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E1/T1 Voice Interface Configuration/ R2 DTMF
The popular E1/T1- R2/DTMF configuration commands are explained
at this chapter.
NOTE
This application is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway which can be equipped with
digital E1/T1 voice interface module.
Network Diagram
Fig. 6-4 VoIP gateway digital E1/T1 R2/DTMF
In the above example, the PBX and VoIP GW at the head office are
operated with R2/DTMF signaling type.
In case of R2/DTMF, only
signal and channel configuration is required, and the preconfigured
ISDN-PRI becomes invalid.
Also, the compand-type of PBX and the VoIP GW should be
configured same according to the PCM type (A-law or U-law).
The below is the default parameters of digital E1 ISDN-PRI of the VoIP
Gateway.
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y
Signaling type: Un-defined
y
No. of channels: None
y
Compand-type : A-law
APOS command script
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
! PRI controller configuration.
!
controller e1(t1) 0/0
signalling-type r2/dtmf
Clock slave
channel-group timeslots 1-31 0
!
voice-port 0/0 0
! E1(t1)
compand-type u-law
!
dial-peer voice 0 pots
destination-pattern 99T
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 5683848
session target 193.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Configuration Verification)
HO# show controller 0/0
Controller T1 slot(0)/port(0)
T1 Link is UP
No Alarm detected.
Applique type is Channelized T1.
Framing is SF, Line Code is AMI, Cable Length is Short 110.
Signalling type is R2-MFC.
7967 Line Code Violations, 2 Framing Bit Errors
1 Out Of Frame Errors, 2 Bit Errors
signalling type = r2
clock source = slave
channel group 0 = 1-24
1
2
3
allocated timeslots = YYYYYYYYYYYYYYYYYYYYYYYYNNNNNNN
outgoing barred channel group =
channel order = descending
b-channel negotiation = exclusive
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overlap receiving = enabled
protocol side = network
R2 get calling number = disabled
ISDN virtual connect = disabled
T1 cable length = short 110
T1 framing = sf
T1 line code = ami
T1 CAS type = immediate
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
y
Signaling type
y
No. of channel groups
y
Clock type
y
Compand type
y
Protocol type
To configure the application, follow this procedure.
Step
Commands
1
HO(config-ether0.0)# controller e1(t1) 0/0
2
HO(config-ether0.0)# signalling-type r2/dtmf
3
HO(config-ether0.0)# Clock slave
4
HO(config-ether0.0)# channel-group timeslots 131 0
5
HO(config-ether0.0)# voice-port 0/0
6
HO(config)# compand-type u-law
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FXS/FXO Voice Interface configuration for caller ID
This part is related to FXO/FXO voice interface configuration. Even
though this is not a commonly required configuration, it needs to be
done at the initial configuration of VoIP gateway. For more detailed
information on this configuration, refer to APOS Operation Guide.
The general information on Caller ID is provided below. FXS voice
interface only detects caller ID and FXO voice interface generates
caller ID.
NOTE
This application is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway which can be equipped with
FXS/FXO voice interface and modules.
Network Diagram
Fig. 6-5 VoIP gateway caller- ID feature
This part explains Caller ID detection on FXS analog interface and
Caller ID generation of FXO analog interface. At the head office,
the FXO port of the VoIP Gateway connected to PBX, detects caller
ID and sends it to branch office A. On the other hand, the FXS voice
interface generates caller ID according to the received information.
The caller ID message is displayed on the end terminal such as
telephones.
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APOS command script (Head Office)
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
!
voice-port 0/0
! FXO
caller-id enable
!
!
dial-peer voice 0 pots
destination-pattern 5683847
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 5683848
session target 193.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Head Office without forward digits)
!
hostname HO
!
interface ether0.0
ip address 194.168.1.2 255.255.255.0
!
!
voice-port 0/0
! FXO
caller-id enable
!
!
dial-peer voice 0 pots
destination-pattern T
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 5678
session target 193.168.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Branch A)
!
hostname BA
!
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interface ether0.0
ip address 193.168.1.1 255.255.255.0
!
!
!
voice-port 0/0
! FXS
caller-id enable
!
!
dial-peer voice 0 pots
destination-pattern 5683848
port 1/0
!
dial-peer voice 1000 voip
destination-pattern 99T
session target 192.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Head Office-Configuration Verification)
HO# show voice port 0/0
Voice port slot(0)/port(0)
line type = FXS
status = Idle
input gain = 0 db
output gain = 0 db
ring frequency = 25 Hz
ring cadence = 1000 msec on, 2000 msec off
polarity inverse = disabled
tie connection = none
description =
translate incoming called-number = -1
translate incoming calling-number = -1
comfort noise generation = enabled
dial tone generation = enabled
echo cancellation = enabled
announcement = enabled
low dtmf gain = -8
high dtmf gain = -5
caller ID = enabled
caller ID type = bellcore
caller ID NAME = enabled
busyout action = none
associated call number = -1
Related APOS commands & structure
Configure the below parameters appropriate for the network
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environment.
None
To configure the application, follow this procedure.
Step
Commands
1
HO(config-ether0.0)# voice-port 0/0
2
HO(config-voice-port-0/0)# caller-id enable
Enable Caller-ID.
3
HO(config-voice-port-0/0)# dial-peer voice 0
Pots
HO(config-dialpeer-pots-0)# forward-digit from
For example, the received
4
0
Description
number is 1234, the gateway
transfers the number to the
PBX.
4
5
HO(config-dialpeer-pots-0)# destination-pattern
1234
HO(config-dialpeer-pots-0)# port 0/0
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FXS/FXO Voice Interface configuration for polarityinverse
Polarity-Inverse feature of FXS port initiates inversion signal to PBX and
the PBX starts billing when the inversion signal is detected.
NOTE
This application is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway which can be equipped with
FXS/FXO voice interface and modules.
Network Diagram
Fig. 6-6 VoIP gateway polarity inverse feature on FXS port
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Fig. 6-7 VoIP gateway polarity inverse feature on FXO port
At the above example, PBX requires polarity inversion signal from
trunk line to start billing and the inversion signal is sent from the head
office. The signal is generated when the called party VoIP gateway
receives connect message.
APOS command script
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
!
voice-port 0/0
! FXO
polarity-inverse
!
dial-peer voice 0 pots
destination-pattern 5683847
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 5683848
session target 193.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
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APOS command script (Configuration Verification)
HO# show voice port 0/0
Voice port slot(0)/port(0)
line type = FXS
status = Idle
input gain = 0 db
output gain = 0 db
ring frequency = 25 Hz
ring cadence = 1000 msec on, 2000 msec off
polarity inverse = enabled
tie connection = none
description =
translate incoming called-number = -1
translate incoming calling-number = -1
comfort noise generation = enabled
dial tone generation = enabled
echo cancellation = enabled
announcement = enabled
low dtmf gain = -8
high dtmf gain = -5
caller ID = enabled
caller ID type = bellcore
caller ID NAME = enabled
busyout action = none
associated call number = -1
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
None
To configure the application, follow this procedure.
Step
Commands
Description
1
HO(config-ether0.0)# voice-port 0/0
Select
the
voice
port
to
configure.
2
HO(config-voice-port-0/0)# polarity-inverse
Enable
the
polarity inverse
feature.
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E&M Voice Interface Configuration
This part provides information on general E&M configuration and
related commands.
For more detailed information, refer to APOS
Operation Guide.
NOTE
For E&M voice interface hardware configuration, refer to Chapter 7.
Appendix.
NOTE
This application is supported by AddPac Technology’s all VoIP
products along with VoIP Gateway which can be equipped with
E&M voice interface modules.
Network Diagram
WAN
(IP Network)
Router
FXS
E&M
AddPac
Analog Phone
PBX
Access Code : 8
DID Number : 100
VoIP Gateway
(Head Office)
Analog Phone
E.164 : 568-3848
Voice Port 0/0
E&M 2-Wire
Signaling Type : Immediate
E&M Type : 5
IP Address : 192.168.1.
Fig. 6-8 VoIP gateway E&M interface
This part provides information on E&M 2/4-wire configuration
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between PBX and VoIP gateway(Head Office). Information for E&M
wire type and Dip switch configuration of AddPac’s E&M voice
interface module, refer to Chapter 7. Appendix.
APOS command script
!
hostname HO
!
interface ether0.0
ip address 192.168.1.1 255.255.255.0
!
!
voice-port 0/0
! E&M
operation 2-wire
signal immediate
type 5
!
!
dial-peer voice 0 pots
destination-pattern 5683847
port 0/0
!
dial-peer voice 1000 voip
destination-pattern 5683848
session target 193.158.1.2
dtmf-relay h245-alphanumeric
!
voip-interface ether0.0
!
APOS command script (Configuration Verification)
HO# show voice port 0/0
Voice port slot(0)/port(0)
line type = E&M
status = Idle
input gain = 0 db
output gain = 0 db
tie connection = none
description =
E&M operation = 2-wire
E&M signal = immediate
E&M type = 5
E&M non-confirmed connect = disabled
E&M delay duration = 2000 msec
E&M delay start = 300 msec
E&M dialout delay = 300 msec
E&M wait wink = 550 msec
E&M wink duration = 200 msec
E&M wink wait = 200 msec
translate incoming called-number = -1
translate incoming calling-number = -1
comfort noise generation = enabled
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dial tone generation = enabled
echo cancellation = enabled
announcement = enabled
low dtmf gain = -8
high dtmf gain = -5
busyout action = none
associated call number = -1
Related APOS commands & structure
Configure the below parameters appropriate for the network
environment.
y
E&M Signaling type
y
E&M wire type
y
E&M type
To configure the application, follow this procedure.
Step
Commands
Description
1
HO(config-ether0.0)# voice-port 0/0
Enter
the
voice
port
configuration mode.
2
HO(config-voice-port-0/0)# operation 2-wire
Configure 2-wire E&M voice
interface.
3
HO(config-voice-port-0/0)# signal immediate
4
HO(config-voice-port-0/0)# type 5
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Chapter 7. Appendix
E&M Voice Interface Dip Switch setting
E&M voice interface is equipped with jumper switches for E&M type
selection.
NOTE
E&M voice interface dip switch setting is supported by AddPac
Technology’s all VoIP products along with VoIP Gateway which
can be equipped with E&M voice interface modules.
E&M Voice Interface Module Jumper Switch
Each jumper switch of E&M voice interface module is marked at the
below picture. Fourteen (14) jumper switches consist of 7 groups.
J2
J3
J1
J5
J6
J7
J4
J8
J9
J11
J10
JP1
J13
J12
Fig. 7-1 E&M voice interface module jumper switch image
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E&M Voice Interface Jumper Switch Description
E&M voice interface jumper switch group operates with a specific
purpose. More detailed descriptions are provided below.
E&M voice interface groups and purposes
Type
Jumper
Purpose
Wire type
JP1, J3, J5, J6, J7, J8, J9, J11, J13
2/4-wire type jumper switches
Module board
J2
E&M voice interface module board ID
ID
setting.
The setting is fixed at shipment and
unable to modify.
E&M type
J1, J4, J10, J12
E&M
voice
interface
type
setting.
Selectable from 5 types of channels.
E&M voice interface channels and jumper switches
Channels
Jumper
Description
Channel 0
J1
Supports voice channel line 0 of E&M voice interface module.
Channel 1
J4
Supports voice channel line 1 of E&M voice interface module.
Channel 2
J10
Supports voice channel line 2 of E&M voice interface module.
Channel 3
J12
Supports voice channel line 3 of E&M voice interface module.
☞ E&M channels supported by E&M voice interface jumper switches are provided below.
Fig. 7-2 E&M voice interface module front view
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E&M Voice Interface Type and Jumper Setting
E&M voice interface type and jumper setting
Type
Jumper
TypeⅠ
J1, J4, J10, J12
Setting
Description
Jumper switch setting for
E&M
voice
type1.
Connect only A3-
B3,
A5-B5
interface
and
leave
others open.
Type Ⅱ
J1, J4, J10, J12
Jumper switch setting for
E&M
voice
type2.
Connect only A2-
B2,
A5-B5,
interface
A6-B6
and
leave others open.
Type Ⅲ
J1, J4, J10, J12
Jumper switch setting for
E&M voice interface type
3.
Connect only A1-B1,
A3-B3, A5-B5, A6-B6 and
leave others open.
Type Ⅳ
J1, J4, J10, J12
Jumper switch setting for
E&M voice interface type
4.
Connect only A1-B1,
A2-B2, A3-B3, A4-B4 and
leave others open
Type Ⅴ
J1, J4, J10, J12
Jumper switch setting for
E&M voice interface type
5.
Connect only A3-B3,
A4-B4 and leave others
open
☞ J1, J4, J10, J12 jumper switch points 7 separated (A1~A7, B1~B7) jumper switch groups.
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E&M Voice Interface Wire Type and Jumper Setting
2-Wire E&M Voice Interface Jumper Setting
2-wire E&M Voice Interface Jumper Setting
Type
Jumper
Main jumper
JP1
Setting
Description
Main jumper for 2-wire type E&M
voice interface setting. Connect
A1-B1.
Wire
setting
jumper
J3, J5, J6, J7 J8,
Jumper for 2-wire type E&M
J9, J11, J13
voice interface setting. Connect
A2-B2,
A4-B4.
Each
jumper
switch refers to the 4 difference
jumper switch groups.
☞ Wire setting jumper J3/J5 and E&M channel line 0, J6/J7 and E&M channel line 1, J9/J11
and E&M channel line 2, J8/J13 and E&M channel line 3 are separately mapped to each
other.
4-Wire E&M Voice Interface Jumper Setting
4-wire E&M Voice Interface Jumper Setting
Type
Jumper
Main jumper
JP1
Setting
Description
Main jumper for 4-wire type E&M
voice interface setting. Open
A1-B1.
Wire
jumper
setting
J3, J5, J6, J7 J8,
Jumper for 4-wire type E&M
J9, J11, J13
voice interface setting. Connect
A1-B1,
A3-B3.
Each
jumper
switch refers to the 4 difference
jumper switch groups.
☞ Wire setting jumper J3/J5 and E&M channel line 0, J6/J7 and E&M channel line 1, J9/J11
and E&M channel line 2, J8/J13 and E&M channel line 3 are separately mapped to each
other.
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Glossary
Terms
Definition & Description
An acronym for Asymmetric Digital Subscriber Line, ADSL is a method of
transmitting data over traditional copper telephone lines. Data can be
ADSL
downloaded at speeds of up to 1.544 Megabits per second and
uploaded at speeds of 128 Kilobits per second (asymmetric).
An acronym for VoIP Plug & Play Management Software.
AddPac
Technology developed integrated management software for VoIP
AP-VPMS
product
remote
installation,
real-time
monitoring,
network
management on Graphic User Interface (GUI).
An acronym for Application Programming Interface, an Interface which
API
is used for accessing an application or a service from a program.
An acronym for AddPac Internetworking Operation System,
APOS
AddPac
Technology developed operating system for network devices.
An acronym for Asynchronous Transfer Mode. It an International Cell
Relay standard sending various service such as voice, video and data
as fixed size (53bytes) cells. With the fixed size cells, the cell processing is
ATM
mainly done by hardware, so the transmission delay is significantly
reduced. ATM is designed for high transmission media such as E3,
SONET, T3.
Starting from ‘1993, ATM information Super-highway was established to
ATM
Information
highway
Super-
offer data service and internet service to public offices by the Korean
government. Data service includes ATM, Dedicated line, packet
switching, Frame relay and Internet service includes Internet compound
service and internet service via ATM access lines.
Establish by Cisco Systems, NET/ADAPTIVE, Northern Telecom, Sprint in
ATM Forum
‘1991 for the development and
acceleration of ATM technology star
nards. It encompasses the standard by ANSI and ITU-T, and further
develops the agreed terms of ATM standard.
Authentication ensures that digital data transmissions are delivered to
Authentication
the intended receiver. Authentication also assures the receiver of the
integrity of the message and its source (where or whom it came from).
BNC Connector
Boot Loader
A standard connector connecting IEEE 802.3 10Base-2 coaxial cable to
MAU(Media Access Unit).
The built-in chip on the printed circuit board generating booting
command of network equipment.
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Bps
Voice over IP
Bits per second. Refer to: bit rate.
A modem designed to operate over cable TV lines. Because the
coaxial cable used by cable TV provides much greater bandwidth than
Cable Modem
telephone lines, a cable modem can be used to achieve more
bandwidth.
Cable
network
also
requires
modularization
and
demutualization process while sending the data.
A call center is a central place where customer and other telephone
calls are handled by an organization, usually with some amount of
computer automation. Typically, a call center has the ability to handle
a considerable volume of calls at the same time, to screen calls and
Call Center
forward them to someone qualified to handle them, and to log calls.
Call
centers
are
used
by
mail-order
catalog
organizations,
telemarketing companies, computer product help desks, and any large
organization that uses the telephone to sell or service products and
services.
A feature that displays the name and/or number of the calling party on
the phone's display when an incoming call is received. Virtually all
digital phones - as well as many analog phones - have this capability.
Caller ID
While typically only the number is received, most phones will display the
name, if the number matches an entry in the phone's built-in phone
book.
unshielded twisted pair (UTP) cabling. An Ethernet network operating at
Category 5 cabling
10 Mbits/second (10BASE-T) will often tolerate low quality cables, but at
100 Mbits/second (10BASE-Tx) the cable must be rated as Category 5,
or Cat 5 or Cat V, by the Electronic Industry Association (EIA).
Constant Bit Rate. A data transmission that can be represented by a
non-varying, or continuous, stream of bits or cell payloads. Applications
CBR
such as voice circuits generate CBR traffic patterns. CBR is an ATM
service type in which the ATM network guarantees to meet the
transmitter's bandwidth and Quality of Service requirements
An acronym for Circuit Emulation Service. enables users to multiplex or
CES
to concentrate multiple circuit emulation streams for voice and video
with packet data on a single, high-speed ATM link without a separate
ATM access multiplexer.
A computed value which is dependent upon the contents of a packet.
Checksum
This value is sent along with the packet when it is transmitted. The
receiving system computes a new checksum based upon the received
data and compares this value with the one sent with the packet. If the
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two values are the same, the receiver has a high degree of confidence
that the data was received correctly.
A cable with a single inner conductor with foam insulation and braided
Coaxial cable
shield. There are two types of this cable; 50Ω cable for digital signaling
process and 75Ω cable for analog signal process and high speed
digital signal process.
An
CODEC
acronym
for
COder-DECoder
1.
Built-in
circuit
device
for
coding/decoding of analog signal to bit stream with Pulse Code
Modulation method. 2. DSP software algorithm for compressing/
decompressing voice or audio signal
Console
DTE interface whether the command is delivered to the host.
Class of Service (CoS) is a way of managing traffic in a network by
grouping similar types of traffic (for example, e-mail, streaming video,
voice, large document file transfer) together and treating each type as
CoS
a class with its own level of service priority. Unlike Quality of Service
(QoS) traffic management, Class of Service technologies do not
guarantee a level of service in terms of bandwidth and delivery time;
they offer a "best-effort."
Decryption
The process of converting encrypted data back into its original form, so
it can be understood.
Dynamic Host Configuration Protocol. A protocol which allows a host to
obtain configuration information, such as its IP address and the default
DHCP
router from a server. This simplifies network administration because the
software keeps track of IP addresses. With DHCP device can have a
different IP address every time it connects to the network
DNS
DS-3
DSP
DTMF
Domain Name Server, an Internet service that translates domain names
into IP addresses.
Digital signal level 3, A line capable of delivering 44.7 Mbps (44,700
Kbps) in both directions
Digital Signal Processor.
Dedicated microprocessor for digital signal
process.
Dual Tone Multi-Frequency. Using two types of voice-band tones for
dialing.
An acronym for recEive and transmit or
E&M
ear and mouth. E&M interface
uses a RJ-48 telephone cable to connect remote calls from an IP
network to PBX trunk lines (tie lines) for local distribution. It is a signaling
technique for two-wire and four-wire telephone and trunk interfaces.
E1
The basic building block for European multi-megabit data rates, with a
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bandwidth of 2.048Mbps.
Encryption
the manipulation of a packet's data in order to prevent any but the
intended recipient from reading that data.
Broadband LAN standard initiated by Xerox Corporation and co-
Ethernet
developed by Intel and DEC.
Utilizing CSMA/CD and the various
cables of 10Mbps are used. It is similar to IEEE 802.3. Refer to: 10Base-2,
10Base5, 10Base-F, 10Base-T, 10Broad-36, Fast Ethernet, IEEE 802.3.
Short for "FACSimile." In essence, a fax machine sends an electronic
"facsimile" or copy of the document. An optical scanner in the machine
FAX
scans the document and the resulting bit stream is then sent to the
receiving machine via telephone line. The transmission and the
reproduction at a distance of still pictures printed matter and similar
documented material
data that is transmitted between network points as a unit complete
with addressing and necessary protocol control information. A frame is
Frame
usually transmitted serial bit by bit and contains a header field and a
trailer field that "frame" the data. (Some control frames contain no
data.)
Frame-Relay
Switching type Data Link Layer Protocol. Using HDLC capsule, process
multi-number of virtual circuits between devices.
an acronym for File Transfer Protocol, a very common method of
FTP
transferring one or more files from one computer to another. Defined at
RFC 959.
Foreign Exchange Office. An FXO interface connects to the Public
FXO
Switched Telephone Network (PSTN) central office and is the interface
offered on a standard telephone.
FXS
Foreign Exchange Station. An FXS interface connects directly to a
standard telephone and supplies ring, voltage, and dial tone.
Describes the 64-kbps PCM voice coding technique. In G.711, encoded
G.711
voice is already in the correct format for digital voice delivery in the
PSTN or through PBXs.
Describes a compression technique that can be used for compressing
speech or audio signal components at a very low bit rate as part of the
G.723.1
H.324 family of standards. This CODEC has two bit rates associated with
it: 5.3 and 6.3 kbps. The higher bit rate is based on ML-MLQ technology
and provides a somewhat higher quality of sound. The lower bit rate is
based on CELP and provides system designers with additional flexibility.
G.726
Describes ADPCM coding at 40, 32, 24 and 16 kbps. ADPCM encoded
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voice can be interchanged between packet voice, PSTN, and PBX
networks if the PBX networks are configured to support ADPCM.
Described in the ITU-T standard in its G-series recommendations.
Describes a 16 kbps low-delay variation of CELP voice compression.
G.728
CELP voice coding must be translated into a public telephony format
for delivery to or through the PSTN. Described in the ITU-T standard in its
G-series recommendations..
The component of an H.323 conferencing system that performs call
address
resolution,
admission
control,
and
subnet
bandwidth
management. H.323 entity on a LAN that provides address translation
and control access to the LAN for H.323 terminals and gateways. The
Gatekeeper
gatekeeper can provide other services to the H.323 terminals and
gateways, such as bandwidth management and locating gateways. A
gatekeeper maintains a registry of devices in the multimedia network.
The devices register with the gatekeeper at startup and request
admission to a call from the gatekeeper.
An International Telecommunication Union (ITU-T) standard
H.225
for H.225.0
session control and packetization. It defines various protocols of RAS,
Q.931, RTP and etc.
H.245
H.323
HBD3
An International Telecommunication Union (ITU-T) standard for H.245
end-point control.
An International Telecommunication Union (ITU-T) standard that
describes packet-based video, audio, and data conferencing.
Line code type of E1 line.
An acronym for High-Level Data Link Control. A transmission protocol for
the Data Link Layer. In HDLC, data is organized into a unit (called a
frame) and sent across a network to a destination that verifies its
HDLC
successful arrival. Variations of HDLC are also used for the public
networks that use the X.25 communications protocol and for frame
relay, a protocol used in both and wide area network, public and
private.
Short on-hook period usually generated by a telephone-like device
during a call to indicate that the telephone is attempting to perform a
Hookflash
dial-tone recall from a PBX. Hookflash is often used to perform call
transfer.
HTTP
IPSec
An acronym for Hypertext Transfer Protocol. A file transfer protocol used
by web browser or web server for transmitting text or graphic files.
Internet Protocol Security protocol, a framework for a set of protocols
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for security at the network or packet processing layer of network
communication. Earlier security approaches have inserted security at
the Application layer of the communications model. IPsec is said to be
especially useful for implementing virtual private networks and for
remote user access through dial-up connection to private networks. A
big advantage of IPsec is that security arrangements can be handled
without requiring changes to individual user computers. Cisco has been
a leader in proposing IPsec as a standard (or combination of standards
and technologies) and has included support for it in its network routers.
IPv6 (Internet Protocol Version 6) is the latest level of the Internet
Protocol (IP) and is now included as part of IP support in many products
including the major computer operating systems. IPv6 has also been
called "IPng" (IP Next Generation). Formally, IPv6 is a set of
IPv6
specifications from the Internet Engineering Task Force (IETF). IPv6 was
designed as an evolutionary set of improvements to the current IP
Version 4. Network hosts and intermediate nodes with either IPv4 or IPv6
can handle packets formatted for either level of the Internet Protocol.
Users and service providers can update to IPv6 independently without
having to coordinate with each other.
An ISP (Internet service provider) is a company that provides individuals
and other companies access to the Internet and other related services
such as Web site building and virtual hosting. An ISP has the equipment
and the telecommunication line access required to have a point-of-
ISP
presence on the Internet for the geographic area served. The larger
ISPs have their own high-speed leased lines so that they are less
dependent on the telecommunication providers and can provide
better service to their customers. Among the largest national and
regional ISPs are AT&T WorldNet, IBM Global Network, MCI, Netcom,
UUNet, and PSINet.
The ITU-T (for Telecommunication Standardization Sector of the
International Telecommunications Union) is the primary international
ITU-T
body for fostering cooperative standards for telecommunications
equipment and systems. It was formerly known as the CCITT. It is located
in Geneva, Switzerland
Interactive Voice Response (IVR) is a software application that accepts
IVR
a combination of voice telephone input and touch-tone keypad
selection and provides appropriate responses in the form of voice, fax,
callback, e-mail and perhaps other media. IVR is usually part of a larger
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application that includes database access. Common IVR applications
include: Bank and stock account balances and transfers.
A local area network is a group of computers and associated devices
that share a common communications line and typically share the
LAN
resources of a single processor or server within a small geographic area
(for example, within an office building). LAN standard defines cable
connection and signal processing on Physical Layer and Data Link
Layer.
Network communication channels consisting of sending and receiving
Link
devices, circuits, transmission path.
Usually refer to WAN connection.
Referred as Line, or transmission link.
A loopback test is a test in which a signal in sent from a
Loopback test
communications device and returned (looped back) to it as a way to
determine whether the device is working right or as a way to pin down
a failing node in a network.
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
MAC Address
update routing tables and data structures. MAC addresses are 6 bytes
long and are controlled by the IEEE. Also known as a hardware address,
MAC-layer address, and physical address. Compare with network
address.
MAN
A data network designed for a town or city. MANs are considered
larger than LANs but smaller than WANs. Compare with: LAN, WAN.
MGCP, also known as H.248 and Megaco, is a standard protocol for
handling the signaling and session management needed during a
multimedia conference. The protocol defines a means of
MGCP
communication between a media gateway, which converts data from
the format required for a circuit-switched network to that required for a
packet-switched network and the media gateway controller. MGCP
can be used to set up, maintain, and terminate calls between multiple
endpoints. Megaco and H.248 refer to an enhanced version of MGCP
NAT (Network Address Translation) is the translation of an Internet
NAT
Protocol address (IP address) used within one network to a different IP
address known within another network. One network is designated the
inside network and the other is the outside.
NTP
Network Time Protocol (NTP) is a protocol that is used to synchronize
computer clock times in a network of computers. In common with
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similar protocols, NTP uses Coordinated Universal Time (UTC) to
synchronize computer clock times to a millisecond, and sometimes to a
fraction of a millisecond.
Private Automatic Branch Exchange. A telephone switch for use inside
a corporation. It connects offices (internal extensions) with each other
PABX
and provides access (typically by dialing an access number such as 9)
to the public telephone network PABX is the preferred term in Europe,
PBX is used in the USA.
Packets contain a source and destination address as well as the actual
Packet
message. Packets also known as Datagrams.
A PBX (private branch exchange) is a telephone system within an
enterprise that switches calls between enterprise users on local lines
PBX
while allowing all users to share a certain number of external phone
lines.
Packet INternet Groper, a packet (small message) sent to test the
PING
validity / availability of an IP address on a network
Basic connection type.
Point to Point Connection
In ATM, point to point connection is half
duplex connection between two ATM end systems or full duplex
connection.
Basic connection type. In ATM, point to multipoint connection is half
Pont to Multipoint
duplex connection among one sending end system (root node) and
Connection
multiple
receiving
end
system.
Compare
with:
point-to-point
connection.
POTS
Plain Old Telephone Service.
Compare with: PSTN.
The most popular method for transporting IP packets over a serial link
between the user and the ISP. Developed in 1994 by the IETF and
PPP
superseding the SLIP protocol, PPP establishes the session between the
user's computer and the ISP using its own Link Control Protocol (LCP).
PPP supports PAP, CHAP and other authentication protocols as well as
compression and encryption.
Any set of communication protocols, such as TCP/IP, that consists of
Protocol Stack
two or more layers of software and hardware. It's called a stack
because each layer builds on the functionality in the layer below
PSTN
PVC
Public Switched Telephone Network – term for the entire, world-wide
telephone network.
Sometimes refers to as POTS.
Permanent
Circuit
Virtual
or
permanent
virtual
connection.
A
continuously available communications path that connects two fixed
end points.
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Q.931 Signaling
Voice over IP
ITU-T specification for network layer of ISDN. Q.931 uses out-of-band
signaling on the D-channel to control calls.
This refers to the assumption that data transmission rates, error rates,
and other characteristics can be measured, improved, and to some
QoS
degree, guaranteed in advance. Basically, QoS describes a collective
measure of the level of service a provider delivers to its customers or
subscribers.
Random-Access Memory, a non-retentive memory, whose contents get
RAM
lost after a switch-off or reset. Application programs run in the random
access memory and data is stored and processed.
Registration Admission Status protocol. The communication protocol
RAS
used to convey registration, admission and status messages between
H.323 endpoints and the gatekeeper.
RISC
Reduced Instruction Set Computing
On the Internet, a router is a device or, in some cases, software in a
computer, that determines the next network point to which a packet
should be forwarded toward its destination. The router is connected to
at least two networks and decides which way to send each information
Router
packet based on its current understanding of the state of the networks
it is connected to. A router is located at any gateway (where one
network meets another), including each Internet point-of-presence. A
router is often included as part of a network switch. Compare with:
gateway. Refer to: relay.
RS-232
Most common Physical Layer interface. Known as EIA/TIA-232.
Real-time Control Protocol (RTCP) is a companion protocol of RTP that is
RTCP
used to maintain quality of service. Refer to: RTP(Real-Time Transport
Protocol).
1. Routing Table Protocol, VINES routing protocol based on RIP.
Distributes network topology, and aids VINES servers in finding
neighboring clients, servers, and routers. Uses delay as a routing metric.
Refer to: SRTP.
2. Rapid Transport Protocol. Provides pacing and error recovery for
RTP
APPN data as it crosses the APPN network. With RTP, error recovery and
flow control are done end-to-end rather than at every node. RTP
prevents congestion rather than reacts to it.
3. Real-Time Transport Protocol. Commonly used with IP networks. RTP is
designed to provide end-to-end network transport functions for
applications transmitting real-time data, such as audio, video, or
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simulation data, over multicast or unicast network services. RTP provides
such services as payload type identification, sequence numbering,
time-stamping, and delivery monitoring to real-time applications.
The Session Initiation Protocol (SIP) is an Internet Engineering Task Force
(IETF) standard protocol for initiating an interactive user session that
involves multimedia elements such as video, voice, chat, gaming, and
virtual reality.
Like HTTP or SMTP, SIP works in the Application layer of the Open Systems
Interconnection (OSI) communications model. The Application layer is
the level responsible for ensuring that communication is possible. SIP
can establish multimedia sessions or Internet telephony calls, and
modify, or terminate them. The protocol can also invite participants to
unicast or multicast sessions that do not necessarily involve the initiator.
Because the SIP supports name mapping and redirection services, it
SIP
makes it possible for users to initiate and receive communications and
services from any location, and for networks to identify the users
whatever they are. SIP is a request-response protocol, dealing with
requests from clients and responses from servers. Participants are
identified by SIP URLs. Requests can be sent through any transport
protocol, such as UDP, SCTP, or TCP. SIP determines the end system to
be used for the session, the communication media and media
parameters, and the
communication.
Once
called party's desire
these
are
assured,
to engage
SIP
in the
establishes
call
parameters at either end of the communication, and handles call
transfer and termination. The Session Initiation Protocol is specified in
IETF Request for Comments [RFC] 2543.
The real-time monitoring, statistical data search and management GUI
SmartViewer
based software developed by AddPac Technology for AP-GK1000, APGK2000, AP-GK3000 models.
Simple
Network
Management
Protocol.
Network
management
protocol used almost exclusively in TCP/IP networks. SNMP provides a
SNMP
means to monitor and control network devices, and to manage
configurations, statistics collection, performance, and security. Refer to:
SGMP, SNMP2.
A TDM physical transmission standard consisting of two twisted wire pairs
T1
and related equipment capable of carrying a 1.544 Mbps DS-1 signal.
Term often used interchangeably with DS-1. Refer to: AMI, B8ZS, DS-1.
TCP/IP
Transmission Control Protocol/Internet Protocol, The protocol suit
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developed by DoD (USA) in 1970s for the worldwide inter-network
development. TCP & IP is the most well known protocols of the suite.
Refer to: IP, TCAP.
Telephone Company, referring to the company offering telephone
Telco
service to customers.
Typically, it refers to an individual company such
as Bell operating company offering local telephone service, however,
sometimes local telephony service providers are included.
Standard Terminal Emulation program covered by TCP/IP protocol
Telnet
stack. Used for remote terminal connection.
Via Telnet, users can log-
in to the system and operate the resources as working on the local
system.
Defined on RFC 854.
the address or label of a VC; a value stored in a field in the ATM cell
VCI
header that identifies an individual virtual channel to which the cell
belongs. VCI values may be different for each data link hop of an ATM
virtual connection.
New DSL technology that accepts bandwidths of up to 27 Mbps over
relatively short distances. VDSL, in the process of being standardized,
allows symmetric or asymmetric throughputs that are much higher than
other xDSL standards (up to 27 Mbps when downloading and 3 Mbps
VDSL
when uploading under asymmetric or 14 Mbps in symmetric), as well as
the simultaneous transport of ISDN (Numeris) services but with much
shorter ranges that do not exceed 900 m to 1 km. In practice, this
technique may require the deployment of optical remotes and the
setting up of active equipment in the local loop. Compare with: ADSL,
HDSL, SDSL.
Voice Over ATM. Voice over ATM enables an ATM switch to carry voice
VoATM
traffic (for example, telephone calls and faxes) over an ATM network.
When sending voice traffic over ATM, the voice traffic is encapsulated
using AAL1/AAL2 ATM packets.
Voice Over Frame Relay. Voice over Frame Relay enables a router to
carry voice traffic (for example, telephone calls and faxes) over a
VoFR
Frame Relay network. When sending voice traffic over Frame Relay, the
voice traffic is segmented and encapsulated for transit across the
Frame Relay network using FRF.12 encapsulation.
Voice Over HDLC. Voice over HDLC enables a router to carry live voice
VoHDLC
traffic (for example, telephone calls and faxes) back-to-back to a
second router over a serial line.
VoIP
VoIP (Voice delivered using the Internet Protocol) is a term used in IP
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telephony for a set of facilities for managing the delivery of voice
information using the Internet Protocol (IP). In general, this means
sending voice information in digital form in discrete packets rather than
in the traditional circuit-committed protocols of the public switched
telephone network (PSTN). A major advantage of VoIP and Internet
telephony is that it avoids the tolls charged by ordinary telephone
service.
Virtual Private Network, VPN allows IP traffic to travel securely over a
VPN
public TCP/IP network by encrypting all traffic from one network to
another. A VPN uses "tunneling" to encrypt all information at the IP level.
A network that covers a large geographical area. Typical WAN
WAN
technologies include point-to-point, X.25 and frame relay. Compare
with: LAN, MAN.
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