Information is subject to change without notice. Nortel Networks reserves the right to make changes in design
or components as progress in engineering and manufacturing may warrant.
Nortel, Nortel (Logo), the Globemark, This is the Way, This is Nortel (Design mark), SL-1, Meridian 1, and
Succession are trademarks of Nortel Networks.
4
Page 3 of 910
Revision history
August 2005
Standard 4.00. This document is up-issued to support Nortel Communication
Server 1000 Release 4.5.
September 2004
Standard 3.00. This document is up-issued to support Nortel Networks
Communication Server 1000 Release 4.0.
May 2004
Standard 2.00. This document is up-issued to support the
Nortel Networks Mobile Voice Client 2050 (MVC 2050).
October 2003
Standard 1.00. This document is a new NTP for Succession 3.0. It was created
to support a restructuring of the Documentation Library. This document
contains information previously contained in the following legacy document,
now retired: IP Line: Description, Installation and Operation (553-3001-
204).
Content from IP Line: Description, Installation and Operation (553-3001-
204) also appears in:
•Converging the Data Network with VoIP (553-3001-160),
•Communication Server 1000M and Meridian 1: Small System Planning and Engineering (553-3011-120), and
•Communication Server 1000M and Meridian 1: Large System Planning and Engineering (553-3021-120).
Downloading files from the Nortel web site . . . . . . . . . 903
553-3001-365Standard 4.00August 2005
30
Page 25 of 910
About this document
This document is a global document. Contact your system supplier or your
Nortel representative to verify that the hardware and software described are
supported in your area.
Subject
This document:
•describes the physical and functional characteristics of the IP Line 4.5
application for Nortel Communication Server (CS) 1000 Release 4.5 and
Meridian 1 systems and describes its use on the Voice Gateway Media
Cards.
•explains how to engineer, install, configure, administer, and maintain an
Structure
This document has separate chapters which are applicable only to either
Optivity Telephony Manager (OTM) or Element Manager.
The configuration, administration, and maintenance sections are divided into
three chapters each. For example, there is a generic configuration chapter
dealing with tasks related to installing and configuring IP Line 4.5. This
chapter is followed by two other configuration chapters, one for OTM and
another for Element Manager. The administration and maintenance chapters
have the same format.
IP Telephony node that contains Voice Gateway Media Cards.
IP LineDescription, Installation and Maintenance
Page 26 of 910 About this document
Note on legacy products and releases
This NTP contains information about systems, components, and features that
are compatible with Nortel Communication Server 1000 Release 4.5
software. For more information on legacy products and releases, click the
Technical Documentation link under Support on the Nortel home page:
www.nortel.com
Applicable systems
This document applies to the following systems:
•Communication Server 1000S (CS 1000S)
•Communication Server 1000M Chassis (CS 1000M Chassis)
•Communication Server 1000M Cabinet (CS 1000M Cabinet)
•Communication Server 1000M Half Group (CS 1000M HG)
•Communication Server 1000M Single Group (CS 1000M SG)
•Communication Server 1000M Multi Group (CS 1000M MG)
•Communication Server 1000E (CS 1000E)
•Meridian 1 PBX 11C Chassis
•Meridian 1 PBX 11C Cabinet
•Meridian 1 PBX 51C
•Meridian 1 PBX 61C
•Meridian1 PBX81
•Meridian 1 PBX 81C
Note: When upgrading software, memory upgrades may be required on
the Signaling Server, the Call Server, or both.
System migration
When particular Meridian 1 systems are upgraded to run CS 1000 Release 4.5
software and configured to include a Signaling Server, they become
553-3001-365Standard 4.00August 2005
About this document Page 27 of 910
CS 1000M systems. Table 1 lists each Meridian 1 system that supports an
upgrade path to a CS 1000M system.
Table 1
Meridian 1 systems to CS 1000M systems
This Meridian 1 system...Maps to this CS 1000M system
Meridian 1 PBX 11C ChassisCS 1000M Chassis
Meridian 1 PBX 11C CabinetCS 1000M Cabinet
Meridian 1 PBX 51CCS 1000M Half Group
Meridian 1 PBX 61CCS 1000M Single Group
Meridian 1 PBX 81CS 1000M Multi Group
Meridian 1 PBX 81CCS 1000M Multi Group
For more information, see one or more of the following NTPs:
•Communication Server 1000M and Meridian 1: Small System Upgrade Procedures (553-3011-258)
•Communication Server 1000M and Meridian 1: Large System Upgrade Procedures (553-3021-258)
Conventions
•Communication Server 1000S: Upgrade Procedures (553-3031-258)
•Communication Server 1000E: Upgrade Procedures (553-3041-258)
Terminology
In this document, the following systems are referred to generically as
“system”:
•Communication Server 1000S (CS 1000S)
•Communication Server 1000M (CS 1000M)
•Communication Server 1000E (CS 1000E)
•Meridian1
IP LineDescription, Installation and Maintenance
Page 28 of 910 About this document
The following systems are referred to generically as “Small System”:
•Communication Server 1000M Chassis (CS 1000M Chassis)
•Communication Server 1000M Cabinet (CS 1000M Cabinet)
Communication Server (CS) 1000 Release 4.5 introduces the IP Line 4.5
application.
The IP Line 4.5 application provides an interface that connects an IP Phone
to a Meridian 1 PBX and a CS 1000 Call Server.
Note: IP Line 4.5 does not operate on Meridian 1 or CS 1000 systems
running software earlier than 4.5.
IP Line 4.0 (or earlier) is not supported in CS 1000 Release 4.5.
Features
IP Line 4.5 introduces the following features:
•Active Call Failover
553-3001-365Standard 4.00August 2005
IMPORTANT!
Voice Gateway Media Cards
Interworking
Description Page 33 of 910
•DSP peg counter for CS 1000E systems
•Enhanced UNiStim firmware downloads for IP Phones
If a Media Card 32-port card, a Media Card 8-port card, or an ITG-P 24-port
card is running IP Line 4.5 software, it is known as a Voice Gateway Media
Card.
DHCP server
A Dynamic Host Configuration Protocol (DHCP) server can be used to
provide the required information to enable the IP Phone network connection
and connect to the Voice Gateway Media Card.
For more information on DHCP, refer to Converging the Data Network with
VoIP (553-3001-160) and IP Phones: Description, Installation, and
Operation (553-3001-368).
The IP Phone uses the IP network to communicate with the Voice Gateway
Media Card and the optional DHCP server. Figure 1 on page 34 shows a
diagram of the system architecture.
IP LineDescription, Installation and Maintenance
Page 34 of 910 Description
Figure 1
System architecture
CS 1000M
IP Line
LAN
Signaling Server
(Optionally Redundant)
-Terminal Proxy Server
-H.323 proxy
-Primary Gatekeeper
-Element Manager Web Server
Web Browser
for Element Manager
WAN
Branch Media Gateway
IP Trunk 3.0
or later
Signaling Server
Signaling Server
(Optionally Redundant)
-Terminal Proxy Server
-H.323 proxy
-Alternate Gatekeeper
-Element Manager Web Server
CS 1000
Call ServerSignaling Server
BCM
Requires BCM
Release 3.0 or higher
553-3001-365Standard 4.00August 2005
IP
Phones
Media streams routed
directly using IP
LAN
Media Gateway and
Media Gateway Expansion
553-AAA0400
Applicable systems
The CS 1000 and Meridian 1 systems support the Media Card 32-port line
card, Media Card 8-port line card, and ITG-Pentium 24-port line card.
Unsupported products
The following remote service products do not support the Media Card 32-port
line card, Media Card 8-port line card, and ITG-Pentium 24-port line card:
•Carrier Remote
•Mini-carrier Remote
•Fiber Remote
•Fiber Remote Multi-IPE
System requirements
CS 1000 Release 4.5 software is the minimum system software for
IP Line 4.5.
OTM 2.2 and Element Manager
Description Page 35 of 910
Optivity Telephony Manager (OTM) 2.2 and Element Manager are used
throughout this document as the primary interface for Voice Gateway Media
Cards and IP Line 4.5.
OTM 2.2 is the minimum required version.
CS 1000 systems
Either OTM 2.2 or Element Manager can be used as the configuration,
administration, and maintenance interface for IP Line 4.5 on a CS 1000
system.
If trying to use OTM 2.2 to perform an action available through Element
Manager, then OTM 2.2 launches Element Manager automatically.
OTM 2.2 is used for configuration activities not supported by Element
Manager, such as terminal administration.
IP LineDescription, Installation and Maintenance
Page 36 of 910 Description
Meridian 1
OTM 2.2 is used as the configuration, administration, and maintenance
interface for IP Line 4.5 on a Meridian 1. Element Manager cannot be used,
as Element Manager is located on a Signaling Server, and there is no
Signaling Server in a Meridian 1.
Corporate Directory
OTM 2.2 is necessary for creation of the Corporate Directory database.
SNMP and alarms
Element Manager does not provide a SNMP alarm browser, so the OTM 2.2
Alarm Manager is recommended when SNMP alarm collection is required.
System configurations
Although IP Line 4.5 can be used in different system configurations and its
use can vary in those configurations, there are four basic system
configurations. See Table 2.
Table 2
Possible system configurations
SystemSignaling Server present
1Meridian 1No
2CS 1000EYes
3CS 1000MYes
4 CS 1000SYes
IP Line 4.5 can use the Signaling Server if the Signaling Server is deployed
in the system configuration.
553-3001-365Standard 4.00August 2005
Meridian 1
A Meridian 1 system does not have a Signaling Server in its configuration.
Each Voice Gateway Media Card functions as both a UNIStim Line Terminal
Proxy Server (LTPS) and voice gateway.
In this system configuration, one Voice Gateway Media Card is configured
as the Leader. IP Phones register with individual Voice Gateway Media
Cards.
Note: If a Media Card 32-port card, a Media Card 8-port card, or an
ITG-P 24-port card is running IP Line 4.5 software, it is known as a
Voice Gateway Media Card.
CS 1000 systems
CS 1000 systems have a Signaling Server in their network configuration. The
Signaling Server is a server that provides signaling interfaces to the
IP network. The Signaling Server’s central processor drives the signaling for
IP Phones and IP Peer networking.
In IP Line 4.5, the LTPS executes on the Signaling Server and the voice
gateway executes on the Voice Gateway Media Cards. All IP Phones register
with the Signaling Server. The Voice Gateway Media Cards only provide
access to the voice gateway.
Description Page 37 of 910
The Signaling Server is the node leader and, by default, acts as a Master for
the node.
Signaling Server redundancy
There are several methods of redundancy for a Signaling Serve. See Table 3.
Table 3
Methods of Signaling Server redundancy (Part 1 of 2)
StageDescription
With a backup Signaling Server
1A backup Signaling Server can be configured in a normal configuration.
IP LineDescription, Installation and Maintenance
Page 38 of 910 Description
Table 3
Methods of Signaling Server redundancy (Part 2 of 2)
StageDescription
2If the primary Signaling Server fails, the backup Signaling Server takes over and
all IP Phones register with the backup Signaling Server.
3If the backup Signaling Server fails, one of the Voice Gateway Media Cards is
elected to be the node Master.
4The IP Phones then register to the Voice Gateway Media Cards.
Without a backup Signaling Server
1If there is no backup Signaling Server, and the primary Signaling Server fails, one
of the Voice Gateway Media Cards is elected to be the node Master.
2The IP Phones then register to the Voice Gateway Media Cards.
Software delivery
IP Line 4.5 supports software delivery through the following formats:
1CompactFlash
2Signaling Server CD-ROM
3Download from the Nortel web site
Note: Stand-alone IP Line 4.5 software is not available through
CD-ROM.
553-3001-365Standard 4.00August 2005
The IP Line 4.5 software and related documentation (such as Readme First
documents) can be downloaded from the Nortel web site.
Required packages
The IP Phones require the software packages listed in Table 4.
Table 4
Required packages
PackagePackage number
M2000 Digital Sets (DSET)88
Aries Digital Sets (ARIE)170
Note: To configure IP Line 4.5 in groups 5-7 on Option 81C CP PII or
CS 1000M MG, the Fibre Network (FIBN) software package 365 is
required.
Description Page 39 of 910
IP LineDescription, Installation and Maintenance
Page 40 of 910 Description
IP Line package components lists
CS 1000 and Meridian 1 package components
Table 5 lists the IP Line 4.5 package components for CS 1000 and Meridian 1
systems.
Table 5
IP Line 4.0 Media Card 32-port line card package components (Part 1 of 2)
ComponentCode
Media Card 32-port - IP Line 4.5 Voice Gateway Systems Package includes
the following:
• Media Card 32-port assembly NTVQ01BB
• IP Line 4.5 Voice Gateway CompactFlash NTM403AC
• ITG EMC Shielding Kit (NTVQ83AA)
• Readme First Document
• Shielded 50-pin to Serial/ELAN/TLAN adaptor
• PC Maintenance cable (NTAG81CA)
• IP Line 4.5 NTP (CD-ROM)
• ITG-specific Meridian 1 Backplane 50-pin I/O Panel Filter Connector
(NTCW84JA) (see Note)
NTDU41FC
553-3001-365Standard 4.00August 2005
Description Page 41 of 910
Table 5
IP Line 4.0 Media Card 32-port line card package components (Part 2 of 2)
ComponentCode
IP Line 4.5 Voice Gateway NTP (CD-ROM), which includes:
• IP Line: Description, Installation, and Operation (553-3001-365)
• IP Phones: Description, Installation, and Operation (553-3001-368)
• IP Phone 2001 User Guide
• IP Phone 2001 Quick Reference Card
• IP Phone 2002 User Guide
• IP Phone 2002 Quick Reference Card
• IP Phone 2004 User Guide
• IP Phone 2004 Quick Reference Card
• IP Phone 2007 User Guide
• IP Phone 2007 Quick Reference Card
• IP Audio Conference Phone 2033 User Guide
• IP Audio Conference Phone 2033 Quick Reference Card
• IP Softphone 2050 User Guide
• Mobile Voice Client 2050 User Guide
Note: The I/O panel filter connector is not required for Meridian 1 Option 11C Cabinet,
Meridian 1 Option 11C Chassis, CS 1000M Cabinet, CS 1000M Chassis, or CS 1000S
systems.
NTDW81AG
IP LineDescription, Installation and Maintenance
Page 42 of 910 Description
IP Line 4.5 Media Card 8-port card
package components
Table 6 lists the IP Line 4.5 Media Card 8-port card package components. The
Media Card 8-port card is intended for branch office configurations. The card
is applicable to the CS 1000 and Meridian 1 systems.
Table 6
IP Line 4.5 Media Card 8-Port card package components
ComponentCode
Media Card 8-port - IP Line 4.5 Voice Gateway Systems Package includes:
• Media Card 8-port Assembly NTVQ01AB
• IP Line 4.5 CompactFlash NTM403AC
• ITG EMC Shielding Kit NTVQ83AA
• Readme First Document
• Shielded 50-pin to Serial/ELAN/TLAN adaptor
• PC Maintenance Cable NTAG81CA
• IP Line 4.0 NTP (CD-ROM) NTDW81AF
• ITG-specific Meridian 1 Backplane 50-pin I/O Panel Filter Connector
(NTCW84JA) (see Note)
Note: The I/O panel filter connector is not required for Meridian 1 Option 11C Cabinet,
Meridian 1 Option 11C Chassis, CS 1000M Cabinet, CS 1000M Chassis, or CS 1000S
systems.
NTDU41FB
553-3001-365Standard 4.00August 2005
Documentation
The following documents are available on the IP Line 4.5 CD-ROM and on
the Nortel web site:
•IP Line: Description, Installation, and Operation (553-3001-365)
•IP Phones: Description, Installation, and Operation (553-3001-368)
Voice Gateway Media Card is a term used to encompass the Media Card
32-port line card, Media Card 8-port line card, and ITG-P 24-port line card.
These cards plug into an Intelligent Peripheral Equipment (IPE) shelf in the
Meridian 1 and CS 1000M systems, into a Media Gateway 1000S and Media
Gateway 1000S Expander in the CS 1000S system, and into a Media Gateway
1000E and Media Gateway 1000E Expander in the CS 1000E system.
The ITG-P 24-port line card occupies two slots while the Media Card line
card occupies only one slot. The Media Card comes in two versions: 8-port
and 32-port.
IP LineDescription, Installation and Maintenance
Page 44 of 910 Description
The Media Card has the following features:
•32-port card’s packet processing power is greater than that of the ITG-P
24-port line card
•increases the channel density from 24 to 32 ports (for 32-port version)
•reduces the slot count from a dual IPE slot to a single IPE slot
•supports up to 128 IP Phones for the 32-port version, while 32 IP Phones
are supported on the 8-port version (if a Signaling Server is not present
in the network configuration).
The 8-port version is typically intended for the Media Gateway 1000B used
with the Branch Office feature in branch office locations.
Table 7 provides a comparison of the ITG-P 24-port line card and Media Card
32-port and 8-port line cards.
Table 7
Comparison of ITG-P 24-port and Media Card 32-port and 8-port cards (Part 1 of 2)
ITG-P 24-port
Item
Total DSP Channels24328
Number of slots the card
occupies
Operating SystemVxWorks 5.3VxWorks 5.4VxWorks 5.4
ProcessorPentiumIXP1200IXP1200
DSP8 x TI54094 x TI54211 x TI5421
Telogy version7.018.1 High Density
Number of IP Phones that
can register on each Voice
Gateway Media Card
553-3001-365Standard 4.00August 2005
card
211
96
(in a Meridian 1 –
see note)
Media Card 32-port
card
version
(8 ports for each
DSP)
128
(in a Meridian 1 –
see note)
Media Card 8-port
card
8.1 High Density
version
(8 ports for each
DSP)
32
(in a Meridian 1 –
see note)
Description Page 45 of 910
Table 7
Comparison of ITG-P 24-port and Media Card 32-port and 8-port cards (Part 2 of 2)
ITG-P 24-port
Item
Image file name prefixes
shown by swVersionShow
command
/C: driveOn board Flash 2
UpgradeTwo images filesOne image file
Note: If a Voice Gateway Media Card is used in a CS 1000 system, then the IP Phones
register to the Signaling Server instead of the Voice Gateway Media Card, and are not subject
to these restrictions. A Signaling Server can register a maximum of 5000 IP Phones.
card
IPL PIPL SAIPL SA
x 4Mb
Media Card 32-port
card
Plug-in
CompactFlash
32 Mb
(no backup)
Media Card 8-port
card
Plug-in
CompactFlash
32 Mb
One image file
(no backup)
Voice Gateway Media Cards have an ELAN network interface (10BaseT)
and a TLAN network interface (10/100BaseT) on the I/O panel.
Note: The ELAN (Embedded LAN) subnet isolates critical telephony
signaling between the Call Server and the other components. The ELAN
subnet is also known as the Management LAN subnet.
The TLAN (Telephony LAN) subnet carries telephony/voice/signaling
traffic. The TLAN subnet, also known as the Voice LAN subnet,
connects to the customer network and the PSTN.
There is an RS-232 Maintenance Port connection on the faceplates of both the
ITG-P 24-port card and the Media Card card. The ITG-P 24-port card has an
alternative connection to the same serial port on the I/O backplane.
CAUTION
Do not connect maintenance terminals to both the
faceplate and the I/O panel serial maintenance port
connections at the same time.
IP LineDescription, Installation and Maintenance
Page 46 of 910 Description
Capacity
The Virtual TN (VTN) feature allows each Voice Gateway Media Card to
support more IP Phones than there are physical bearer channels. There are 24
bearer channels on each ITG-P card and 8 or 32 channels on each Media Card.
Both cards support a 4:1 concentration of registered IP Phones (IP Phones
2001, 2002, 2004, 2007, IP Audio Conference Phone 2033, IP Softphone
2050, Mobile Voice Client (MVC) 2050, WLAN Handset 2210, WLAN
Handset 2211, and WLAN Handset 2212) to gateway channels. The ITG-P
supports 96 registered IP Phones. The Media Card supports 32 registered IP
Phones (when the card has 8 channels) or 128 registered IP Phones (when the
card has 32 channels). The IP Phones require the services of the bearer
channels only when they are busy on a call that requires a TDM circuit such
as an IP Phone-to-digital telephone/trunk/voice mail/conference. When an IP
Phone is idle or there is an IP-to-IP call, no gateway channel is required.
When the total number of IP Phones that are registered or are attempting to
register reaches the limit (96 on the ITG-P, 32 or 128 on the Media Card), the
Voice Gateway Media Card recognizes this and no more IP Phones are
assigned to the card. Each Voice Gateway Media Card is restricted to a total
of 1200 call attempts per hour distributed across all the IP Phones associated
with the card.
553-3001-365Standard 4.00August 2005
Description Page 47 of 910
Media Card controls, indicators, and connectors
Figure 2 shows the Media Card 32-port and 8-port card faceplate.
Figure 2
Media Card faceplate
Reset
MC
A:
E T
NTVQ01AA
J2
Reset button
Enable LED
PC Card slot (Drive /A:)
MAC address label
(TLAN and ELAN network interface addresses)
100
10
Ethernet activity LEDs
A
HEX display
RS-232 maintenance port
Lock latches
553-SMC0001
IP LineDescription, Installation and Maintenance
Page 48 of 910 Description
Faceplate components
The components on the faceplate of the Media Card 32-port and 8-port card
are described in the following sections.
Reset button
Use the Reset button on the faceplate to manually reset the Media Card. This
enables the card to be reset without cycling power to it. The Reset button is
used to reboot the card after a software upgrade or to clear a fault condition.
Enable LED
The faceplate red LED indicates the following:
•the enabled/disabled status of the card
•the self-testing result during power up or card insertion into an
operational system
PC Card slot
This slot accepts the Type I or Type II standard PC Flash Cards, including
ATA Flash cards (3 Mb to 170 Mb). The slot is labeled /A:.
Nortel supplies PC Card adaptors that enable CompactFlash cards to be used
in the slot.
WAR NING
Do not format the PC Card using a Windows application.
As well, only format the PC Card using the type of card on
which it will be running. For example, a PC Card
formatted using a Small System Controller (SSC) card is
only readable by the SSC card. It is not readable by the
ITG-P 24-port card or the Media Card. A PC Card
formatted using a Voice Gateway Media Card (ITG-P
24-port card or Media Card) is only readable by another
Voice Gateway Media Card. It is not readable by the SSC
card.
553-3001-365Standard 4.00August 2005
Description Page 49 of 910
MAC address label
The MAC address label on the card’s faceplate is labeled ETHERNET
ADDRESS. It shows the TLAN and ELAN network interface MAC
addresses. The Management /ELAN network interface MAC address for each
card is assigned during manufacturing and is unchangeable. The MAC
address label on the Media Card is similar to the following example:
ETHERNET ADDRESS
TLAN
00:60:38:BD:C9:9C
ELAN
00:60:38:BD:C9:9D
Ethernet activity LEDs
The faceplate contains six Ethernet activity LEDs: three for the ELAN
network interface and three for the TLAN network interface. The LEDs
indicate the following links on the ELAN network interface and TLAN
network interface (in order from the top):
1100 (100BaseT)
210 (10BaseT)
3A (Activity)
Maintenance hex display
This is a four-digit LED-based hexadecimal display that provides the role of
the card. It also provides an indication of fault conditions and the progress of
PC Card-based software upgrades or backups.
RS-232 Maintenance Port
The Media Card faceplate provides a female 8-pin mini-DIN serial
maintenance port connection. The faceplate on the card is labeled J2.
IP LineDescription, Installation and Maintenance
Page 50 of 910 Description
ITG-P 24-port card controls, indicators, and connectors
Figure 3 shows the ITG-P 24-port card faceplate components.
Figure 3
ITG-P 24-port card faceplate
NWK
not used
ITG-P LED (card status)
Reset button
RS-232
Maintenance Port
ITG-P
Reset
NWK
Status
A:
NTVQ55AA
Maint
Port
MAC address label
(motherboard and daughterboard addresses)
The components on the faceplate of the ITG-P 24-port line card are described
in the following sections.
NWK
The faceplate connector labeled NWK is a 9-pin, sub-miniature D-type
connector. The connector is not used for the IP Line 4.0 application.
WARNING
The NWK connector looks like a 9-pin serial connector.
Do not connect a serial cable or any other cable to it. If a
cable is connected to the NWK connector, the TLAN
network interface is disabled.
ITG-P LED (card status)
The red status faceplate LED indicates the enabled/disabled status of the
24 card ports. The LED is on (red) during the power-up or reset sequence.
The LED remains lit until the card is enabled by the system. If the LED
remains on, the self-test failed, the card is disabled, or the card rebooted.
Reset button
Press the Reset button to reset the card without having to cycle power to the
card. This button is normally used after a software upgrade to the card or to
clear a fault condition.
IP LineDescription, Installation and Maintenance
Page 52 of 910 Description
MAC address label
The MAC address label on the card’s faceplate shows the motherboard and
daughterboard addresses. The ELAN network interface address corresponds
to the Management MAC address. The Management MAC address for each
card is assigned during manufacturing and is unchangeable. The ELAN
network interface MAC address is the MOTHERBOARD Ethernet address
found on the label. The MAC address label on the ITG-P 24-port line card is
similar to the following example:
TLAN network interface activity LEDs (labeled NWK Status LEDs)
The two NWK Status LEDs display TLAN network interface activity.
•Green – the LED is on if the carrier (link pulse) is received from the
TLAN network interface switch.
•Yellow – the LED flashes when there is data activity on the TLAN
network interface. During heavy traffic, the yellow LED can stay
continuously lit.
Note: There are no Ethernet status LEDs for the ELAN network
interface.
553-3001-365Standard 4.00August 2005
Description Page 53 of 910
PC Card slots
The ITG-P 24-port card has one faceplate PC Card slot (designated
Drive /A:). It is used for optional maintenance. The ITG-P 24-port card also
has one unused inboard slot (designated Drive /B:). The PC Card slots support
high-capacity PC flash memory cards.
WAR NING
Do not format the PC Card using a Windows application.
As well, only format the PC Card using the type of card on
which it will be running. For example, a PC Card
formatted using a Small System Controller (SSC) card is
only readable by the SSC card. It is not readable by the
ITG-P 24-port card or the Media Card. A PC Card
formatted using a Voice Gateway Media Card (ITG-P
24-port card or Media Card) is only readable by another
Voice Gateway Media Card. It is not readable by the SSC
card.
Matrix maintenance display
A four-character, LED-based dot matrix display shows the maintenance
status fault codes and other card state information. For a list of the fault codes,
see Table 79: “ITG-P 24-port line card faceplate maintenance display codes”
on page 689 and Table 80: “Media Card faceplate maintenance display
codes” on page 691.
RS-232 maintenance port
The ITG-P 24-port line card faceplate provides a female 8-pin mini-DIN
serial maintenance port connection, labeled Maint Port. An alternative
connection to the faceplate serial maintenance port exists on the NTMF94EA
I/O panel breakout cable.
CAUTION
Do not connect maintenance terminals or modems to the
faceplate and I/O panel DB-9 male serial maintenance
port at the same time.
IP LineDescription, Installation and Maintenance
Page 54 of 910 Description
Backplane interfaces
The backplane provides connections to the following:
•ELAN network interface
•TLAN network interface
•alternate connection to the DS-30X serial maintenance port
•Card LAN interface connectors
DS-30X voice/signaling
The DS-30X serial maintenance port carries Pulse Code Modulation (PCM)
voice and proprietary signaling on the IPE backplane between the ITG-P
24-port card and the Intelligent Peripheral Equipment Controller (XPEC).
Card LAN
The card LAN carries card polling and initialization messages on the IPE
backplane between the ITG-P 24-port card and the Intelligent Peripheral
Equipment Controller (XPEC).
Assembly description
The ITG-P 24-port card assembly is a two-slot motherboard and
daughterboard combination. A PCI interconnect board connects the
motherboard and the DSP daughterboard. See Figure 4 on page 55.
553-3001-365Standard 4.00August 2005
Figure 4
ITG-P 24-port card physical assembly
Description Page 55 of 910
Functional description of the Voice Gateway Media Cards
The Media Card and ITG-P 24-port line cards can perform two separate
functions, depending on the system in which the card is located:
1The card acts as a gateway between the circuit-switched voice network
and the IP network.
2The card acts as a Line Terminal Proxy Server (LTPS) or “virtual line
card” for the IP Phones, based on whether a Signaling Server is used in
the configuration or not.
IP LineDescription, Installation and Maintenance
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Gateway functional description
The Gateway performs the following functions:
•registers with the system using the TN Registration messages
•accepts commands from the system to connect/disconnect audio channel
•uses Realtime Transport Protocol/Realtime Conferencing Protocol
(RTP/RTCP) protocol to transport audio between the gateway and the IP
Phone
•encodes/decodes audio from PCM to and from the IP Phone’s format
•provides echo cancellation for the speaker on IP Phones for echoes
originating in the circuit-switched voice network (not applicable to the IP
Softphone 2050 or MVC 2050 as they have no handsfree capability)
Gateway functionality on the Meridian 1
Since there is no Signaling Server, each Voice Gateway Media Card
functions as both the LTPS and Voice Gateway.
The Gateway portion of the card connects to the Meridian 1 through the
DS-30X backplane. The Gateway portion also receives call speech-path setup
and codec selection commands through the ELAN network interface. The IP
Phone connects to both the Gateway and the LTPS functions through the
TLAN network interface.
Gateway functionality on the CS 1000 systems
A Signaling Server is always present in the CS 1000 systems. The LTPS
executes on the Signaling Server and the Voice Gateway executes on the
Voice Gateway Media Cards. The Voice Gateway Media Cards only provide
the voice gateway access.
Active Master
The LTPS maintains a count of the number of IP Phones registered to the
card. Each IP Telephony node has one active Master. The active Master
broadcasts to all Voice Gateway Media Cards and requests a response if it has
room for another IP Phone.
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The Election function uses a selection process to determine the node’s
Master. The Census function determines the Voice Gateway Media Cards
within an IP Telephony node.
IP Phone registration
IP Phone registration on a Meridian 1 system
Table 8 describes the maximum number of IP Phones that can be registered
to each type of line card in a Meridian 1 system.
Table 8
Maximum number of IP Phones that can register to a Voice Gateway
Media Card in a Meridian 1
Card typeMaximum number
Media Card 32-port 128
Media Card 8-port32
ITG-P 24-port96
For more information, refer to “System capacities” in Communication
Server 1000M and Meridian 1: Large System Planning and Engineering
(553-3021-120), Communication Server 1000M and Meridian 1:
Small System Planning and Engineering (553-3011-120), Communication
Server 1000S: Planning and Engineering (553-3031-120), and
Communication Server 1000E: Planning and Engineering (553-3041-120).
Description Page 57 of 910
IP Phone registration on a CS 1000 system
On a CS 1000 system, the IP Phones register with the LTPS on the Signaling
Server. If a secondary Signaling Server exists, the IP Phone registrations are
split between the primary and secondary Signaling Servers to aid in load
balancing. In that case, the IP Phone registrations alternate between the
primary and secondary Signaling Servers.
IP LineDescription, Installation and Maintenance
Page 58 of 910 Description
If the primary Signaling Server fails, the secondary Signaling Server takes
over (if it exists) and the IP Phones that were registered with the failed
Signaling Server reregister with the LTPS on the secondary Signaling Server.
If there is no secondary Signaling Server or the secondary Signaling Server
fails, the IP Phones register with the LTPS on the Voice Gateway Media
Cards.
Each Signaling Server supports the registration of up to 5000 IP
Phones.
For more information on Signaling Server failure and redundancy, see
Communication Server 1000S: Planning and Engineering (553-3031-120),
Communication Server 1000E: Planning and Engineering (553-3041-120), and
Signaling Server: Installation and Configuration (553-3001-212).
Virtual Terminal Manager
The Virtual Terminal Manager (VTM) performs the following functions:
•arbitrates application access to the IP Phones
IMPORTANT!
•manages all the IP Phones between the applications and the UNIStim
messaging to the IP Phone
•maintains context-sensitive states of the IP Phone (for example, display
or lamp state)
•isolates IP Phone-specific information from the applications (for
example, the number of display lines, number of characters for each
display line, tone frequency, and cadence parameters)
Interactions with IP Phones
The following information describes the process by which an IP Phone
registers and unregisters with a Meridian 1 or CS 1000 system.
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Description Page 59 of 910
Registration
Table 9 describes the registration process.
Table 9
Registration process
StepDescription
1The IP Phone receives the IP address of the Connect Server
(co-located with the LTPS) through either DHCP or manual
configuration.
2The IP Phone contacts the Connect Server.
3The Connect Server instructs the IP Phone to display a
message on its display screen requesting the customer’s IP
Telephony node number and TN.
4The node number and TN are entered. The Connect Server
redirects the IP Phone to the Node Master.
5The IP Phone contacts the Node Master. The Node Master
redirects the IP Phone to the LTPS.
6The IP Phone contacts the LTPS.
7If the IP Phone is valid, the LTPS registers it with the system.
Unregistration
Table 10 describes the unregistration process.
Table 10
Unregistration process
StepDescription
1If the LTPS detects a loss of connection with one of its
registered IP Phones, it logs the event.
2The LTPS then sends an unregister message to the system
for that IP Phone.
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Signaling and messaging
The IP Line 4.5 application sends Scan and Signaling Distribution (SSD)
messages to the Call Server through the system’s ELAN subnet. When tone
service is provided, the service is signaled to the LTPS using new SSD
messages sent through the ELAN subnet.
Signaling protocols
The signaling protocol between the IP Phone and the IP Telephony node is
the Unified Networks IP Stimulus Protocol (UNIStim). The Reliable User
Datagram Protocol (RUDP) is the transport protocol.
RUDP
RUDP is used for:
•signaling between the Call Server and the Voice Gateway Media Cards
•signaling between the IP Telephony node and the IP Phones
Description
Signaling messages between the Voice Gateway Media Card and IP Phones
use RUDP. Each RUDP connection is distinguished by its IP address and port
number. RUDP is another layer on top of UDP. RUDP is proprietary to
Nortel.
The features of RUDP are as follows:
•provides reliable communication system over a network
•packages are resent if an acknowledgement message (ACK) is not
received following a time-out
•messages arrive in the correct sequence
•duplicate messages are ignored
•loss of contact detection
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Description Page 61 of 910
When a data sequence is packetized and sent from source A to receiver B,
RUDP adds a number to each packet header to indicate its order in the
sequence.
•If the packet is successfully transmitted to B, B sends back an ACK to A,
acknowledging that the packet has been received.
•If A receives no message within a configured time, it retransmits the
packet.
•If B receives a packet without having first received its predecessor, it
discards the packet and all subsequent packets, and a NAK (no
acknowledge) message which includes the number of the missed packet
is sent to A. A retransmits the missed packet and continues.
UNIStim
The Unified Network IP Stimulus protocol (UNIStim) is the single point of
contact between the various server components and the IP Phone.
UNIStim is the stimulus-based protocol used for communication between an
IP Phone and an LTPS on the Voice Gateway Media Card or Signaling
Server.
ELAN TCP transport
Although TCP is used for the signaling protocol between the Call Server and
the Voice Gateway Media Card, RUDP remains for the Keep Alive
mechanism for the link. This means RUDP messages are exchanged to
maintain the link status between the Call Server and the Voice Gateway
Media Card.
There is no change to UNIStim signaling. IP Phones continue to use the
RUDP transport protocol to communicate with the Voice Gateway Media
Card.
The TCP protocol enables messages to be bundled. Unlike the RUDP
transport that creates a separate message for every signaling message (such as
display updates or key messages), the TCP transport bundles a number of
messages and sends them as one packet.
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Handshaking is added to the Call Server and IP Line software so that the TCP
functionality is automatically enabled. A software version check is performed
by the IP Line application each time before it attempts to establish a TCP link
with the CS 1000 and Meridian 1 CPUs. TCP transports messages, while
RUDP establishes and maintains the link.
If the version does not satisfy the minimum supported version, a RUDP link
is used instead to maintain the link and all signaling.
Virtual superloops, virtual TNs, and physical TNs
Virtual TNs (VTNs) enable configuration of service data for an IP Phone,
such as key layout and class of service, without requiring the IP Phone to be
dedicated (hard-wired) to a given TN on the Voice Gateway Media Card.
Calls are made between an IP Phone and circuit-switched telephone/trunks
using the full CS 1000 and Meridian 1 feature set. Digital Signal Processor
(DSP) channels are allocated dynamically for this type of call to perform the
encoding/decoding required to connect the IP Phone to the circuit-switched
network.
To create an IP Phone using VTNs, create a virtual superloop in LD 97 or in
Element Manager. To create the virtual superloop in Element Manager, click
System > Superloops in the Element Manager navigator.
•Up to 1024 VTNs can be configured on a single virtual superloop for
Large Systems, CS 1000M Cabinet and CS 1000M Chassis systems, and
CS 1000E systems
•Up to 128 VTNs can be configured on a single virtual superloop for
Meridian 1 Option 11C Cabinet and Meridian 1 Option 11C Chassis
systems, leading to support for a maximum of 640 VTNs for each of
these systems.
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Description Page 63 of 910
•Up to 1024 VTNs can be configured on a single virtual superloop for
CS 1000S systems. Table 11 describes the virtual superloop and virtual
card mapping on a CS 1000S system. Each superloop has two ranges of
cards.
Table 11
Virtual superloop/virtual card mapping for CS 1000S
SUPLCard
9661-6481-84
10065-6885-88
10469-7289-92
10873-7693-96
11277-8097-99
Each ITG-P 24-port card provides 24 physical TNs and each Media Card
32-port card provides 32 physical TNs. The physical TNs are the gateway
channels (DSP ports).
Configure the physical TNs (IPTN) in LD 14. They appear as TIE trunks
without a Route Data Block (RDB).
Virtual TNs
Virtual TNs enable service data to be configured for an IP Phone, such as key
layout and class of service, without requiring a physical IP Phone to be
directly connected to the Call Server.
The concentration of IP Phones is made possible by dynamically allocating a
port (also referred to as a physical TN) of the Voice Gateway Media Card for
a circuit-switched- to-IP Phone call. All system speech path management is
done with physical TNs instead of virtual TNs.
The channels (ports) on the Voice Gateway Media Cards are pooled
resources.
The IP Phones (virtual TNs) are defined on virtual superloops.
IP LineDescription, Installation and Maintenance
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A virtual superloop is a hybrid of real and phantom superloops. Like phantom
superloops, no hardware (for example, XPEC or line card) is used to define
and enable units on a virtual superloop. As with real superloops, virtual
superloops use the time slot map to handle IP Phone (virtual TNs)- to-IP
Phone calls.
Licenses
There are two types of licenses:
•Basic IP User License for the IP Phone 2001 and IP Audio Conference
Phone 2033
•IP User License for the IP Phone 2002, IP Phone 2004, IP Phone 2007,
IP Softphone 2050, Mobile Voice Client (MVC) 2050, WLAN Handset
2210, WLAN Handset 2211, and WLAN Handset 2212
Note: If insufficient Basic IP User Licenses are available for the IP
Phone 2001 and IP Audio Conference Phone 2033, then the IP User
License can also be used for the IP Phone 2001 and IP Audio Conference
Phone 2033.
If there are no Basic IP User Licenses available for the IP Phone 2001 and IP
Audio Conference Phone 2033, and IP User Licenses are used, then an error
message is generated.
“SCH1976: Basic IP User License counter has reached its maximum
value. IP User License was used to configure <data> basic IP Phone(s)
type 2001. Action: (Recommended) Purchase additional Basic IP User
Licenses for IP Phones type 2001, instead of using higher-priced IP User
Licenses.”
Each time an IP Phone is configured, the system TN ISM counter is
decremented.
Customers must purchase one License for each IP Phone installed on CS 1000
and Meridian 1 systems. A new License uses the existing keycode to enable
the IP Phone in the system software. The default is zero.
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Zones
Description Page 65 of 910
To expand the License limits for the IP Phones, order and install a new
Meridian 1 or CS 1000 keycode. Refer to the Incremental Software
Management feature module in the Features and Services (553-3001-306)
NTP.
Note: Individual Licenses are not supported on Functional Pricing. With
Functional Pricing, Licenses are provisioned in blocks of eight.
License limits
The total number of TNs configured with Basic IP User Licenses must not
exceed 32767. The total number of TNs configured with IP User Licenses
must not exceed 32767. The total number of IP phones configured within the
system must not exceed the allowed system capacity limit controlled by
customer keycodes).
To optimize IP Line traffic bandwidth use between different locations, the
IP Line network is divided into “zones”, representing different topographical
areas of the network. All IP Phones and IP Line ports are assigned a zone
number indicating the zone to which they belong.
When a call is made, the codecs that are used vary, depending on which
zone(s) the caller and receiver are in.
By default, when a zone is created in LD 117 or in Element Manager:
•codecs are selected to optimize voice quality (BQ - Best Quality) for
connections between units in the same zone.
•codecs are selected to optimize voice quality (BQ - Best Quality) for
connections between units in different zones.
Note: Support for zones in Element Manager is accessed by clicking IP
Telephony > Zones in the Element Manager navigator.
IP LineDescription, Installation and Maintenance
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Each zone can be configured to:
•optimize either voice quality (BQ) or bandwidth usage
(BB - Best Bandwidth) for calls between users in that zone
•optimize either voice quality or bandwidth usage within a zone and all
traffic going out of a zone
For more information about zones, refer to the following:
•Shared and Private zones (see “Private Zone configuration” on page 225)
•Zones and Virtual Trunks (see IP Trunk: Description, Installation, and Operation (553-3001-363))
•Zones and branch office locations (see Branch Office: Installation and Configuration (553-3001-214))
Administration
The Voice Gateway Media Card is administered using multiple management
interfaces, including the following:
•the IP Line 4.5 application GUI provided by OTM 2.2
•a Command Line Interface (CLI)
•administration and maintenance overlays of Call Servers
•a web browser interface provided by Element Manager. Element
Manager is used for administering Voice Gateway Media Cards in the
systems that use a Signaling Server
IP Line 4.5 application in OTM 2.2
For Meridian 1 systems, OTM 2.2 is required for IP Line 4.5. OTM 2.2 is
used for tasks such as the following:
•creating a node
•adding Voice Gateway Media Cards to the node
•transmitting loadware to the Voice Gateway Media Cards
•upgrading loadware
553-3001-365Standard 4.00August 2005
•defining SNMP alarms
•selecting codecs
Element Manager
Element Manager is a resident web-based user interface used to configure and
maintain CS 1000 components. Element Manager’s web interface enables IP
Line to be configured and managed from a web browser.
Description
Element Manager is a simple and user-friendly web-based interface that
supports a broad range of system management tasks, including:
•configuration and maintenance of IP Peer and IP Telephony features
•configuration and maintenance of traditional routes and trunks
•configuration and maintenance of numbering plans
•configuration of Call Server data blocks (such as configuration data,
customer data, Common Equipment data, D-channels)
•maintenance commands, system status inquiries, backup and restore
functions
Element Manager has many features to help administrators manage systems
with greater efficiency. Examples are as follows:
•Web pages provide a single point-of-access to parameters that were
traditionally available through multiple overlays.
•Parameters are presented in logical groups to increase ease-of-use and
speed-of-access.
•The “hide or show information” option enables administrators to see
information that relates directly to the task at hand.
•Full-text descriptions of parameters and acronyms help administrators
reduce configuration errors.
•Configuration screens offer pre-selected defaults, drop-down lists,
checkboxes, and range values to simplify response selection.
IP LineDescription, Installation and Maintenance
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The Element Manager web server resides on the Signaling Server and can be
accessed directly through a web browser or Optivity Telephony Manager
(OTM). The OTM navigator includes integrated links to each network system
and their respective instances of Element Manager.
Command Line Interface
Definition
The Command Line Interface (CLI) provides a text-based interface to
perform specific Signaling Server and Voice Gateway Media Card
installation, configuration, administration, and maintenance functions.
Access
Establish a CLI session by connecting a TTY or PC to the card serial port or
Telnet through the ELAN or TLAN network interface IP address.
In the case of an IP Telephony node with no Signaling Server, the CLI
must be used to configure the Leader card of the IP Telephony node.
This enables OTM 2.2 and Element Manager to communicate with the
Leader card and the node.
IMPORTANT!
For more information about the CLI commands, see “IP Line CLI
commands” on page 710.
Overlays
For information on the overlays, refer to Software Input/Output:
Administration (553-3001-311).
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252
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Features
Contents
This section contains information on the following topics:
Enhanced Redundancy for IP Line nodes . . . . . . . . . . . . . . . . . . . . . . . 251
IP LineDescription, Installation and Maintenance
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Introduction
Table 12 outlines the IP Line features available for CS 1000 and Meridian 1
systems with CS 1000 Release 4.5 software.
Table 12
IP Line 4.5 feature support (Part 1 of 4)
FeatureMeridian 1CS 1000M CS 1000SCS 1000E
Support for Media
Card
Support for Element
Manager
Support for Signaling
Server
Support for the
following IP Phones:
• IP Phone 2001
• IP Phone 2002
• IP Phone 2004
• IP Phone 2007
•IP Audio
Conference Phone
2033
• WLAN Handset
2210
• WLAN Handset
2211
Ye sYe sYe sYes
NoYe sYe sYes
Ye sYe sYe sYes
Ye sYe sYe sYes
• WLAN Handset
2212
a. Node level patching is not provided by OTM 2.2. The patching CLI command of the Media Card 32-port
line card, Media Card 8-port line card, and ITG-Pentium 24-port line card can be used.
* introduced in IP Line 4.5
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Features Page 73 of 910
Table 12
IP Line 4.5 feature support (Part 2 of 4)
FeatureMeridian 1CS 1000M CS 1000SCS 1000E
Support for the
Ye sYe sYe sYes
following software
clients:
• IP Softphone 2050
• Mobile Voice Client
(MVC) 2050
Support for the IP
Ye sYe sYe s
Phone Key Expansion
Module (KEM)
Active Call Failover *Ye sYe sYe sYes
DSP peg counter for
NoNoNoYe s
the CS 1000E *
Enhanced UNIStim
NoYe sYe sYes
firmware downloads for
IP Phones *
Support for external
Ye sYe sYe sYes
server applications
Enhanced VLAN
Ye sYe sYe sYes
support on Phase II IP
Phones; support for
Voice VLAN hardware
filter providing
enhanced traffic
control on IP Phone
and PC port
a. Node level patching is not provided by OTM 2.2. The patching CLI command of the Media Card 32-port
line card, Media Card 8-port line card, and ITG-Pentium 24-port line card can be used.
* introduced in IP Line 4.5
IP LineDescription, Installation and Maintenance
Page 74 of 910 Features
Table 12
IP Line 4.5 feature support (Part 3 of 4)
FeatureMeridian 1CS 1000M CS 1000SCS 1000E
Network Address
NoYe sYe sYes
Translation (NAT)
Tr av e r sa l
Personal Directory,
NoYe sYe sYes
Callers List, and Redial
List with password
protection
UNIStim File Transfer
Ye sYe sYe sYes
Protocol (UFTP) for IP
Phone firmware
downloads
IP Call RecordingYe sYe sYe sYes
pbxLink connection
Ye sYe sYe sYes
failure detection
Dynamic Loss PlanYe sYe sYe sYe s
Network-wide Virtual
Ye sYe sYe sYes
Office
PatchingPar tialPartialYesYe s
802.1Q supportYe sYe sYe sYe s
Corporate DirectoryYe sYe sYe sYes
Data Path Capture toolYe sYe sYe sYe s
User-defined Feature
Ye sYe sYe sYes
Key Labels
Private ZoneYe sYe sYe sYes
a. Node level patching is not provided by OTM 2.2. The patching CLI command of the Media Card 32-port
line card, Media Card 8-port line card, and ITG-Pentium 24-port line card can be used.
* introduced in IP Line 4.5
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Features Page 75 of 910
Table 12
IP Line 4.5 feature support (Part 4 of 4)
FeatureMeridian 1CS 1000M CS 1000SCS 1000E
Graceful TPS DisableYe sYe sYe sYe s
Run-time downloadYe sYe sYe sYe s
Watchdog TimerYe sYe sYe sYe s
Password Guessing
Protection
Ringer and buzzer
volume adjustment
Set-based installationYes (Small
Maintenance Audit
enhancement
Multi-language supportYe sYe sYe sYe s
Enhanced
Redundancy for IP
Line nodes
IP Softphone 2050
user-selectable codec
(not applicable to MVC
2050 as it only
supports G.711 codec)
a. Node level patching is not provided by OTM 2.2. The patching CLI command of the Media Card 32-port
line card, Media Card 8-port line card, and ITG-Pentium 24-port line card can be used.
Ye sYe sYe sYes
Ye sYe sYe sYes
Yes (Small
Systems
only)
Ye sYe sYe sYes
Ye sYe sYe sYes
Ye sYe sYe sYes
Systems
only)
Ye sYes
* introduced in IP Line 4.5
Active Call Failover for IP Phones
CS 1000 Release 4.5 introduces the Active Call Failover (ACF) feature for IP
Phones.
IP LineDescription, Installation and Maintenance
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The ACF feature for IP Phones allows active IP calls to survive the following
failures:
Note: IP/IP calls means both parties are IP Phones. IP/TDM calls means
one party is an IP Phone and the other party is a TDM telephone or trunk.
•IP and IP/TDM calls survive Signaling Server restarts.
Note: The IP/TDM call does not survive if the Voice Gateway Media
Card with the DSP resource used for the call fails.
•IP and IP/TDM calls survive LTPS ELAN subnet failures.
•IP calls survive a Call Server cold start and Call Server failures in system
configurations with a redundant Call Server of the following types (see
Note 1 on page 77):
— CS 1000S systems with an alternate Call Server when the primary
— Media Gateway 1000B for a branch office configuration
— Geographic Redundancy Secondary Call Server. The feature
Call Server fails
addresses the Primary Call Server failures.
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Features Page 77 of 910
Note 1: IP Phone to IP Phone calls survive the Call Server failures listed
above.
IP Phone to Media Gateway calls that are connected to media services
and switched-circuit line and trunk terminals are dropped on the TDM
side of the Media Gateway when the CS 1000S Alternate Call Server
performs a cold restart in order to come into service upon failure of the
Primary Call Server, and dropped again when the Primary Call Server
comes back into service.
IP Phone to Media Gateway calls through IP Peer virtual trunk routes are
preserved on the TDM side of the Media Gateway, in some cases, when
the IP Phone is redirected in ACF mode from the main office CS 1000 to
the MG 1000B at the branch office location, or from the Geographic
Redundancy Primary to the Secondary Call Server.
IP Phone to Media Gateway calls are preserved if the Media Gateway to
which the call is established is not affected by the failure, or if there is
cold restart of the Call Server that controls the Media Gateway where the
IP Peer virtual trunk call is established.
•For Call Server call processor types CP PII and CP PIV:
— IP/IP calls survive a cold start on all systems.
— IP/IP and IP/TDM calls survive a warm start on all systems.
— Graceful switchover and graceful failover to the redundant Logical
Call Processor (LCP) side of the Call Server makes the failure
transparent and allows all the calls to survive without any loss.
When the IP Phone with an active call reregisters, the call data is rebuilt if the
Call Server does not know about the call, using the internal IP Phone
information.
The ACF feature for IP Phones meets Joint Interoperability Test Command
(JITC) requirements if the LAN/WAN network is engineered to provide full
redundancy: that is, if a LAN/WAN network component fails, an alternate
path between the clients and LTPS server is provided.
IP LineDescription, Installation and Maintenance
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Minimum requirements
The ACF feature for IP Phones has the following minimum requirements:
•Call Server must be running CS 1000 Release 4.5 software.
•LTPS must be running IP Line 4.5 software.
•IP Phones (including IP Softphone 2050) must support Unistim version
2.9. (Use the isetShow command to determine the Unistim version. One
of the columns in the isetShow output is UNIStimVsn.)
ACF mode
The ACF feature for IP Phones enables an IP Phone to reregister in the ACF
mode during a supported system failure.
The ACF mode preserves the following:
•active media session
•LED states of the Mute, Handsfree, and Headset keys
•DRAM content
Note: All other elements (the feature keys, soft keys and text areas) are
retained until the user presses a key or the connection with the Call
Server is resumed. If the user presses a key during the failover, the
display is cleared and a localized “Server Unreachable” message is
displayed.
The IP Phone uses this new mode of re-registration only when the Call Server
explicitly tells the IP Phone to do so. IP Phones clear all call information if
they register to a Call Server or LTPS that does not support the ACF feature.
IP Phone ACF timer
It is possible that there may be an LTPS supporting the ACF feature and an
LTPS that does not support the feature in the same system.
A situation could exist where it takes a long time to fix a failure and no
failover Call Server is available. During this time, the user may have released
the call by pressing the Release key or hanging up the telephone. In this case,
553-3001-365Standard 4.00August 2005
the call-associated resources are not used, but they still exist on the Call
Server since they are not released. To prevent this, the ten-minute Call Server
ACF timer is introduced for each call. The timer prevents call
processing-related resources from being unnecessarily used when an IP
Phone that had an active call unregisters and never reregisters.
The timer is set if:
•the ACF call status is UNREGISTERED; that is, when both parties go
offline.
•only one of the parties is offline, and the other party does not support
disconnect supervision.
ACF scenarios
Table 13 describes ACF behavior in different scenarios.
Table 13
ACF behaviors (Part 1 of 8)
ScenarioResult
Features Page 79 of 910
TLAN subnet failure:
• A call is established between IP Phones A
and B registered with the same node.
• TLAN subnet goes down.
• The IP Phones detect the connection is
lost and periodically try to reregister.
• The TLAN subnet is up shortly (less than
10 minutes), or an election is called and
another accessible LTPS node acquires
the node IP address. The IP Phones
reregister with the node again.
The call is not lost as the IP Phones
reregister.
In this scenario, the call exists on the Call
Server during the failover time and has the
following transitions:
UNREGISTERED ->HALF-REGISTERED ->
NO ACF
IP LineDescription, Installation and Maintenance
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Table 13
ACF behaviors (Part 2 of 8)
ScenarioResult
Signaling Server/Voice Gateway Media
Card platform failure:
• A call is established between IP Phones A
and B registered with the same node.
• The LTPS node goes down.
• The IP Phones detect the connection is
lost and periodically try to reregister.
• The LTPS node is up shortly (less than 10
minutes), or an election is called and
another accessible LTPS node acquires
the node IP address. The IP Phones
reregister with the node again.
Call Server warm restart:
• A call is established between IP Phones A
and B registered with the same Call
Server.
• The Call Server warm restart (INI) occurs.
• The users of IP Phones A and B do not go
on-hook or press any keys during the Call
Server restart.
The call is not lost as the IP Phones
reregister.
The scenario is similar to the TLAN subnet
failure, but the ACF call transition on the Call
Server is instantaneous, since Offline events
are generated in a group as the ELAN subnet
goes down.
The call is not lost.
The call is rebuilt after the warm restart and
has the following transitions:
UNREGISTED->HALF REGISTERED->NO
ACF.
The transition is almost instantaneous since
the Online messages are sent in a group as a
response to the Sync Request.
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Table 13
ACF behaviors (Part 3 of 8)
ScenarioResult
Features Page 81 of 910
Call Server cold restart:
• A call is established between IP Phones A
and B registered with the same Call
Server.
• The Call Server cold restart (SYSLOAD)
occurs.
• The users of IP Phones A and B do not go
on-hook or press any keys during the Call
Server warm restart.
Main office failure for branch office
(scenario 1):
• Branch IP Phones A and B register with the
Media Gateway 1000B and are re-directed
to the main office.
• IP Phones A and B registered with the
main office establish a call.
• A serious main office failure occurs. The
active Branch IP Phones cannot reregister
with the main office and reregister with the
branch office in local mode. IP Phone A
reregisters in local mode first.
The call is not lost.
The call cannot be rebuilt after the SYSLOAD.
The PARTIAL REBUILT -> REBUILT
transition is almost instant since the Online
messages are sent in a group as a response
to the Sync Request.
The call is not lost.
The HALF REBUILT -> REBUILT transition
occurs since the far end is known to the Call
Server gateway to the Media Gateway 1000B.
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Table 13
ACF behaviors (Part 4 of 8)
ScenarioResult
Main office failure for branch office
(scenario 2):
• IP Phones A and B register with the Media
Gateway 1000B and are redirected to the
main office.
• Branch office warm or cold starts.
• Branch users A and B registered with the
main office establish a call.
• A serious main office failure occurs so the
active branch IP Phones cannot reregister
with the main office and they reregister
with the Branch office in local mode. IP
Phone A reregisters in local mode first.
Primary Call Server failure (WAN
geographically redundant system):
• A call is established between IP Phones A
and B that are registered with the primary
site in the geographically redundant
system.
• The primary site fails.
• The IP Phones are reregistered with the
secondary site. IP Phone A reregisters
first.
The call is not lost.
Although the branch office LTPS wrote the IP
Phones A and B data to its RLM table when it
redirected the IP Phones to the main office,
the RLM data is lost and cannot be restored
when the branch office restarts. The transition
is similar to a Call Server cold start: PARTIAL
REBUILT -> REBUILT.
The call is not lost.
IP Phones can be configured in 2 ways:
1Site 1 is the secondary site and Site 2 is
not configured.
In this case the scenario is the same as
main office failure for branch office
(scenario 1): the HALF REBUILT->
REBUILT transition.
2IP Phones have Site 1 defined as the
primary site while Site 2 is defined as the
secondary site. Registration by Site 1
fails.
In this case, the secondary site’s Call
Server does not have the RLM entries for
the reregistering IP Phones and the
scenario is the same as main office failure
for branch office (scenario 2): the
PARTIAL REBUILT -> REBUILT
transition.
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Table 13
ACF behaviors (Part 5 of 8)
ScenarioResult
Features Page 83 of 910
Virtual Office login failure (scenario 1):
• IP Phone A logs into IP Phone C and
establishes a call with IP Phone B. All three
IP Phones are registered with the same
Call Server.
• TLAN subnet failure occurs. IP Phone A
goes offline first, then IP Phone B.
• Active IP Phones A and B reregister with
the system when the TLAN subnet comes
back up. IP Phone A reregisters first and
then IP Phone B.
Virtual Office login failure (scenario 2):
• IP Phone A logs into IP Phone C and
establishes a call with IP Phone B. All three
IP Phones are registered with the same
Call Server.
• TLAN subnet failure occurs. IP Phone B
goes offline first, then IP Phone A.
• Active IP Phones A and B reregister with
the system when the TLAN comes back
up. IP Phone A reregisters first and then IP
Phone B.
The call is not lost.
The following ACF transitions occur:
NO ACF -> PARTIAL REBUILT -> IDLE ->
HALF REBUILT -> REBUILT
The call is not lost.
The following ACF transitions occur:
NO ACF -> HALF REGISTERED -> IDLE ->
HALF REBUILT -> REBUILT
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Table 13
ACF behaviors (Part 6 of 8)
ScenarioResult
Virtual Office login failure (scenario 3):
• IP Phone A logs into IP Phone C and
establishes a call with IP Phone B. All three
IP Phones are registered with the same
Call Server.
• TLAN subnet failure occurs. IP Phones A
and B fail and IP Phone C does not fail.
• IP Phone C tries to log into its home TN
before IP Phones A and B go offline.
• IP Phone A has an IP Peer call with a
remote user over a virtual trunk.
• IP Phone A’s TLAN subnet connection
fails.
• Active IP Phone A reregisters with the Call
Server when the TLAN subnet comes back
up.
Network-wide operation —
network Call Server warm start
• IP Phone A has an IP Peer call with a
remote user over a virtual trunk.
IP Phone C cannot log into its home TN if
another active IP Phone is logged on its TN.
IP Phone C can log into its home TN only
when the call register is released or becomes
PARTIAL REBUILT.
Refer to Virtual Office login failure scenarios 1
and 2 on page 83.
The call is not lost.
The scenario is the same as if the far end
were a local IP Phone. See “TLAN subnet
failure:” on page 79.
The call is not lost.
The scenario is the same as if the far end
were a local IP Phone. See “Call Server warm
restart:” on page 80.
• The Call Server warm starts.
• Active IP Phone A reregisters with the Call
Server as the TLAN subnet comes back
up.
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Table 13
ACF behaviors (Part 7 of 8)
ScenarioResult
Features Page 85 of 910
Network-wide operation — network Call
Server cold start:
• IP Phone A has an IP Peer call with a
remote user over a virtual trunk.
• The Call Server cold starts.
• Active IP Phone A reregisters with the Call
Server as the TLAN subnet comes back
up.
Network-wide operation — network branch
office:
• Branch IP Phones A and B belong to
different branches – Branch A and Branch
B respectively. IP Phones A and B are
registered on the main office Call Server.
• A call is established between IP Phones A
and B.
• Main office Call Server failure occurs and
IP Phones A and B register with their
branches in local mode.
The call is lost as the Call Server comes up.
The call is not lost.
The scenario for each branch is the same as
the first 3 steps of “Main office failure for
branch office (scenario 2):” on page 82.
Branch A does not know about IP Phone B
and Branch B does not know about IP Phone
A. Therefore, each branch builds the
PARTIAL REBUILT call.
Two local PARTIAL REBUILT calls exist on
the branches as the IP Phones reregister in
local mode. The calls are never transitioned to
the REBUILT state and exist until the IP
Phones release the call.
IP/TDM call with TLAN subnet failure:
• IP Phone A has a call with a TDM
telephone or trunk B.
• IP Phone A’s TLAN subnet connection
fails.
• Active IP Phone A reregisters with the Call
Server as the TLAN subnet comes back
up.
The call is not lost.
The scenario is the same as “TLAN subnet
failure:” on page 79 and “Network-wide
operation — network TLAN subnet failure:” on
page 84. The call has the following transitions:
NO ACF -> HALF REGISTERED ->
UNREGISTERED.
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Table 13
ACF behaviors (Part 8 of 8)
ScenarioResult
Network-wide operation — network Call
Server warm start:
• IP Phone A has an IP Peer call with a
remote user over a virtual trunk.
• The Call Server warm starts.
• Active IP Phone A reregisters with the Call
Server as the TLAN subnet comes back
up.
Network-wide operation — network Call
Server cold star:
• IP Phone A has an IP Peer call with a
remote user over a virtual trunk.
• The Call Server cold starts.
• Active IP Phone A reregisters with the
server as the TLAN subnet comes back up.
Firmware downloads
If the IP Phone has an active media stream, the LTPS does not request the
firmware download in order to avoid resetting the IP Phone and losing the
call. Therefore, it is possible that a system might have IP Phones with a
mixture of firmware versions registered with it. The firmware can be
downloaded later when the idle IP Phone registers again or can be
downloaded manually using appropriate CLI commands.
The call is not lost.
The scenario is same as if the far end were a
local IP Phone. See “Call Server warm
restart:” on page 80.
The call is lost as the Call Server comes back
up.
WLAN Handsets 2210/2211/2212
The Wireless LAN (WLAN) Handsets 2210/2211/2212 support Active Call
Failover in the same manner as Phase 2 IP Phones if their firmware supports
UNIStim 2.9.
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Operating parameters
IP Peer calls
IP Peer calls survive the following failure types:
•TLAN subnet failures.
•Signaling Server platform failures/restarts.
When the Signaling Server reboots after the failure, all sessions are lost.
Therefore, when the local IP Phone or far-end telephone releases the call,
no RELEASE message is sent to the other party. The other party must go
on-hook to become idle.
•Call Server warm starts.
IP Peer calls do not survive the Call Server cold start; all virtual trunks are
idled as the Call Server comes back up after the cold start. In this case, the
local IP Phone must go on-hook to become idle.
IP/TDM calls
IP/TDM calls do not survive a Call Server cold start; all DSP channels are
closed as the Call Server comes back up after the cold start. In this case, the
local IP Phone must go on-hook to become idle.
Features Page 87 of 910
Dialing state
Only established calls survive failures. All calls having the DIALING state
on the Call Server are released when an LTPS or signaling failure occurs that
causes an IP Phone to unregister.
Calls that are ringing are handled as follows:
•If the IP Phone originating the ringing call unregisters, the call is released
by the Call Server.
•If the IP Phone receiving the call unregisters, the call receives CFNA
treatment if possible.
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Held calls
From the ACF feature perspective, held calls are considered to be established.
This means that the call is preserved on the Call Server despite TLAN subnet
or LTPS failure. The IP Phone itself is unaware of the state of any held call.
Phase 0/1 IP Phones
Phase 0/1 phones do not support ACF.
Feature key labels
If user-defined feature key labels have been changed but no datadump has
been performed, the changes are lost if there is a Call Server failure.
SIP telephones
SIP telephones appear as IP Peer endpoints to the system. See “IP Peer calls”
on page 87.
NAT devices
The ACF feature cannot handle the case of a NAT device changing the media
path’s mapping between the IP Phone's private address and public address
during the failover period. There is no way to discover the mapping while the
port is in use. For instance, if a main office failure occurs and the user
reregisters in local mode, NAT mapping is changed and the active call cannot
survive.
Control messages
The LTPS sends the Audio Stream Control and LEDs Control commands in
separate messages. If a failure occurs in the time between the two messages,
the Audio Stream and LEDs states may not be synchronized. For example, it
is possible for the Audio Stream to be muted and a network failure to occur
at just the right moment to prevent the LED Control message for the mute
LED from being received by the IP Phone.
Held Calls
When an idle IP Phone (one without an active speech path) reregisters, a
firmware download may occur if needed. If that IP Phone actually had calls
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Features Page 89 of 910
on hold, this means the held calls cannot be retrieved until after the firmware
download is finished.
Voice Gateway Media Cards
The ACF feature does not handle failures of the Voice Gateway functionality
of the Voice Gateway Media Cards.
ELAN and TLAN subnet failures that affect the signaling with the IP Phones
registered to a Voice Gateway Media Card are addressed in the same manner
as failures affecting the Signaling Server. However, if there is a failure
affecting the speech path to an IP Phone, such as when a PBX link failure
occurs and the 10-minute PBX link timer expires, the Voice Gateway calls are
released.
Codecs
Not all the codec properties are restored for the failed-over call. The
following default codec properties are used for the active failover call:
•VAD is OFF
•G.723 Working Rate is 5.3 kbps
•G.729 Annex is Annex A
QoS monitoring
The QoS monitoring is always disabled for the failover call. This is only for
the period of the failover call; for all subsequent calls, the QoS monitoring
works as configured.
Virtual Office
Active Call Failover is not supported for the active call from an IP Phone
logged on another IP Phone to a TDM resource or virtual trunk. Such a call is
released when the LTPS detects that the connection to the IP Phone is lost.
For example, IP Phone A is logged on to IP Phone B and talking to a TDM
resource or a virtual trunk. If a TLAN subnet failure occurs and IP Phone A
reregisters with its home TN, the active call is released as IP Phone A
reregisters.
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Handsfree
Scenario: IP Phone A has handsfree denied and IP Phone B has handsfree
allowed. IP Phone A is logged on IP Phone B and talks to IP Phone C using
handsfree.
If a TLAN subnet failure occurs and IP Phone A reregisters with its home TN
(with handsfree disabled), the handsfree functionality is turned off and IP
Phone A must go off-hook to continue the conversation.
ELAN subnet failure
The ACF state cannot be determined on the LTPS side during an ELAN
subnet failure. This is because the ACF state is stored on the Call Server and
it is not possible to send the ACF state on the LTPS side when the ELAN
subnet has failed.
When the ELAN subnet is down, the isetShow command always outputs the
ACF state as UNKNOWN for all established calls (the state is shown as
busy-UNK).
Feature interactions
Virtual Office and Branch Office
Branch Office
When the first failed IP Phone reregisters in local mode, the branch office
Call Server look ups the far-end branch IP Phone local TN using the specified
far-end IP address and builds a local call.
The call can be rebuilt only if both the IP Phones are branch users of the same
branch office.
Example: A regular main office IP Phone talks to the branch IP Phone
registered with the main office. A failure occurs on the main office, so that
the branch IP Phone cannot register in normal mode again, and reregisters in
local mode. Even if the main office IP Phone survives the failure, the call
cannot be rebuilt because the call becomes an IP Peer call between the branch
office and main office. This call becomes Partial Rebuilt and exists until
released.
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Features Page 91 of 910
Virtual Office
It is possible that active IP Phone A, that was logged into IP Phone B before
the failure, cannot reregister with the Call Server, because IP Phone C
performed a Virtual Office login and uses IP Phone A’s TN. In this case, the
Signaling Server/Voice Gateway Media Card locally handles the Release,
Onhook and Mute events coming from IP Phone A in the Logged Out state.
Survivable Remote Gateway
The Survivable Remote Gateway (SRG) 1.0 and SRG50 do not support ACF.
If the IP Phone is an SRG user, the active call, either in normal mode or local
mode, does not survive a failure.
NAT
The NAT discovery is delayed for an IP Phone with an active call when it
reregisters. NAT discovery messages are sent through the port used for the
RTP stream. NAT discovery is not initiated if the LTPS detects that the IP
Phone has an active RTP stream.
Personal Directory, Callers List, Redial List
The display content is cleared and the Personal Directory/Callers List/Redial
List applications are reset when the active call failover process starts. The
applications can be used again only after the IP Phone reregisters. A user that
is using one of the Personal Directory/Callers List/Redial List menus sees the
display clear and loses any data in that transaction that was not selected or
saved with the Personal Directory/Callers List/Redial List feature.
ACF implementation does not maintain data present only on the Signaling
Server/Voice Gateway Media Card. Transient data (for example, the Services
key sub-menu the user is currently in) is lost when the failover occurs and the
IP Phone reregisters.
Converged Desktop
If the Call Server maintains the active call information during the active call
failover, and the SIP Gateway maintains the link and information with the
MCS 5100 (the SIP Gateway has not failed or is not on the Signaling Server
that reboots if that is the failure mode), then a Converged Desktop call is
maintained when the involved IP Phone reregisters to the system. If the Call
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Server loses the call’s information or the SIP Gateway’s Signaling Server
reboots, the Converged Desktop call is impacted.
Note: A Converged Desktop consists of a telephone and multimedia
PC Client (PCC) software.
The following are scenario examples.
Example 1: The IP Phone’s TLAN subnet fails and the IP Phone reregisters
with the same or a different TPS.
In this case, both the voice and multimedia sessions survive: if a SIP call is
established with the other party in the SIP domain, the call is not released as
the IP Phone reregisters. The multimedia applications still work: the presence
is updated on PCC after the telephone reregisters.
If the unregistered converged IP Phone releases the call during the TLAN
subnet failure, then the Presence status is updated on PCC as the idle
converged IP Phone reregisters.
Example 2: The IP Phone’s Signaling Server fails and the IP Phone
reregisters with the same or a different TPS (active converged IP Phone and
SIP Gateway are on different Signaling Servers in the same node).
In this case, both the voice and multimedia sessions survive; the scenario is
the same as the TLAN subnet failure in Example 1.
Example 3: The IP Phone’s ELAN subnet fails and the IP Phone reregisters
with the same or a different TPS.
The voice session survives. If the ELAN subnet comes back up before the
IP Phone changes the call state (that is, releases the call), then the multimedia
session is not impacted.
If the IP Phone releases the call when the ELAN subnet is still down, the PCC
status update happens when the idle converged IP Phone reregisters with the
system.
If the call is released by the supervisory timer, the status is updated on PCC
after the ELAN subnet comes back up and the Converged Desktop’s AML
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ELAN subnet link is enabled (the CSA104 message is output on the Call
Server when this happens).
Example 4: Call Server warm start.
The voice and multimedia sessions survive. The Presence status is updated on
PCC as the converged IP Phone releases the call after the warm start.
Example 5: Call Server cold start.
The voice and multimedia sessions are closed as the Call Server comes up.
The Presence status becomes “Connected - Idle” even if the call is rebuilt and
active after the Call Server cold start.
IP Phone firmware downloads
The firmware is not downloaded to an IP Phone that has an active RTP stream
open when it registers with the failover system. The firmware is downloaded
later when the idle IP Phone registers again or by using appropriate CLI
commands.
IP Phone as ACD agent or supervisor telephone
If an IP Phone is used as an ACD agent (or supervisor) and the Call Server
fails, then:
•In the case of a Call Server warm start (INI), the active calls are retained
on the agent telephone.
•In the case of a Call Server cold start (SYSLOAD), the active calls are
dropped and the agents are logged out.
This applies to both the In-calls (PRIMARY) key and any secondary DN key
on the ACD telephone.
TPS failures do not impact general ACD functionality, since it is
implemented on the Call Server.
CS 1000 base features
No feature works when the active IP Phone is disconnected and trying to
reregister with the Call Server. All the features can be used in the context of
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the failover call after the IP Phone reregisters (if it is not a PARTIAL
REBUILT call).
The feature context is lost on the Call Server if the Call Server fails.
The feature context is not lost on the Call Server in a case of TLAN/ELAN
subnet failure. Only the feature data on the IP Phone’s display is lost.
Feature context in Call Server failures
The context of any feature is lost on the Call Server in cases of Call Server
failure (Call Server warm or cold start). The LTPS IP Phone display is lost as
the IP Phone reregisters. This means if a feature is activated and the Call
Server fails, all the user input and data is lost.
Example: IP Phone A is in a call; the user presses the Transfer key and starts
dialing a DN. The Call Server cold or warm starts. Therefore, IP Phone A
does not accept the user input and tries to reregister with the Call Server.
When the Call Server comes back up and the IP Phones reregister, IP Phone
A does not have the Call Transfer activated. The held call is also lost: it is not
rebuilt after INI or by the ACF feature, since the call is not active.
TLAN/ELAN subnet and LTPS failures
When a network or Signaling Server/Voice Gateway Media Card failure
occurs and the active IP Phone has some feature activated, the feature context
and data is not lost on the Call Server. The user can proceed with the feature
after the IP Phone reregisters. Only the LTPS display is lost when the IP
Phone reregisters.
Example: IP Phone A is in a call; the user presses the Transfer key, and starts
dialing a DN. A TLAN subnet failure occurs when the first digit is dialed. The
user is unaware of the failure and continues dialing the DN. The digits dialed
after the failure are ignored, the IP Phone detects the failure, clears the
display, and tries to reregister with the server.
The TLAN comes up again and the IP Phone reregisters. Although the IP
Phone is now idle and the display is cleared, the IP Phone can resume dialing
the DN starting from the second digit. The IP Phone can also return to the held
call by pressing the held call’s DN key.
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CDR
No ACF-specific information is added to the Call Detail Record (CDR)
records.
In the case of Call Server failure, the CDR records for the call before the
failure occurred are lost. CDR is restarted as the active IP Phone reregisters.
Therefore, the records are generated only for the “post-failure” period of time.
In the case of the LTPS or network failure, CDR continues. The CDR is then
stopped only if:
•the Call Server supervisory timer expires
•the IP Phone is idle when it reregisters
•the active IP Phone reregisters and then the call is released
The records include the failover time as well. This means that the user may
be “under-charged” in case of Call Server failure and “over-charged” in a
case of LTPS/network failure.
CallPilot
ACF considers CallPilot to be a TDM resource and interaction of an IP Phone
with CallPilot as an IP/TDM call. See “IP/TDM calls” on page 87 and
Table 13: “ACF behaviors” on page 79.
Example: IP Phone A calls telephone B and is redirected to CallPilot on no
answer. The IP/TDM call is established between the IP Phone A and
CallPilot.
The media session between CallPilot and the IP Phone survives any type of
failures except a Call Server cold start.
Note that during any failure, user input is not passed to CallPilot. The user
must resume entering responses after the IP Phone reregisters.
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Interactions considered as IP/TDM calls
The ACF feature also considers interaction of an IP Phone with the following
to be an IP/TDM call:
•CallPilot Mini
•Meridian Mail
•Meridian Mail Card Option
•Companion DECT Telephones (DMC8 version)
•Remote Office 9150
•Mini Carrier Remote
•Carrier Remote
•Periphonics Open IVR (VPS/is)
•Integrated Call Assistant
•Integrated Conference Bridge
•Integrated Recorded Announcer
•Integrated Personal Call Director
•Integrated Voice Services
Symposium Call Center Server
The ACF feature interacts with the Symposium Call Center Server (SCCS)
environment in the following cases:
•Acquired ACD agent is an IP Phone.
— If a failure occurs when the IP Phone is active, the ACD IP Phone
behaves as described in “IP Phone as ACD agent or supervisor
telephone” on page 93.
— If the active unregistered ACD agent changes the call state during
the failure period (for example, releases the call), the status message
is sent to the Symposium and CTI applications as the idle agent
reregisters with the system.
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•Associated non-ACD telephone is an IP Phone.
— If a failure occurs when the IP Phone is active, the ACD IP Phone
behaves as any other IP Phone. If the active associated IP Phone
changes the call state during the failure period (for example, releases
the call), the status message is sent to the Symposium and CTI
applications as the idle telephone reregisters with the system.
MCS 5100
The SIP calls between the CS 1000 IP Phone and a SIP party on the
MCS 5100 side are considered to be IP Peer calls. Such calls survive any type
of failure except a Call Server cold start.
Installation and configuration
The AFC feature for IP Phones requires no installation. It is active by default
on any CS 1000 system running the CS 1000 Release 4.5 software.
On a system running CS 1000 Release 4.5 software, every node running the
CS 1000 Release 4.5 LTPS software has the ACF feature enabled for the IP
Phones that register to it.
Features Page 97 of 910
Configurable RUDP Timeout and Retries Count
When a network failure occurs and the IP Phone's connection is lost, the
IP Phone does not instantly start the failover process. The IP Phone waits for
a length of time for a reply from the server (the length of time is the value of
“RUDP timeout” in msecs). If the IP Phone does not receive a reply from the
server in that length of time, the IP Phone retransmits the message. The IP
Phone retransmits the message for the number of times of the “Retries count”
value, and then starts the failover process: the IP Phone tries to reconnect to
S1, then to S2 and so on.
Previously, the “RUDP timeout” was hard-coded to 500 msec, which meant
that the IP Phone detected the connection failure after a 5-second delay, and
“Retries count” was hard-coded to 10 retries. During that time, the IP Phone
appeared frozen to the user.
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Now the time-out and number of retries can be configured in the OAM and
PDT shells of the Signaling Server. See Table 14.
Table 14
RUDP Timeout and Retries Count commands
CommandDescription
usiSetPhoneRudpRetriesConfigure the RUDP Retries Count maximum
for IP Phones
1 – (10) – 20
See Note 1.
usiGetPhoneRudpRetriesDisplay the RUDP Retries Count maximum for
IP Phones
usiSetPhoneRudpTimeoutConfigure the RUDP Timeout value (in
msecs) for IP Phones
50 – (500) – 1000in increments of 50
milliseconds
See Note 1.
usiGetPhoneRudpTimeoutDisplay the RUDP Timeout value (in msecs)
for IP Phones
Note 1: If the customer has a network with low network delays, one or both parameters can be
reduced to make an IP Phone more responsive to failures. If the network delay values are high,
the parameters can be increased to prevent the IP Phones phones from being reset due to
significant network delay
The RUDP Timeout and Retries Count commands are found in the usi group.
If Help is typed at the OAM prompt, the following is output.
oam> help
For help on a particular command group type: help 'group'
Available command groups are:
……
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DLOG f/w download log file commands
usi UNISTIM related commands
vte Virtual Terminal Emulator related
commands
The configured values are saved in the [usiLib] section of the TPS.ini file and
downloaded to all UNiStim IP Phones registered to the Signaling Server or
Voice Gateway Media Card where the value was configured. When a
supported IP Phone registers with the Signaling Server or Voice Gateway
Media Card, the IP Phone downloads the new values.
It is necessary to configure these values on every Signaling Server and Voice
Gateway Media Card in the node.
Overlay and command modifications
Since call failover is an exceptional situation, ACF information is output only
if it exists.
Status definitions
UNREG
Features Page 99 of 910
The ACF call is UNREGISTERED (UNREG). This occurs when both parties
go offline. This state is always monitored by the 10-minute ACF timer. The
call is released if the Call Server ACF timer expires.
HREG
The ACF call is HALF-REGISTERED (HREG). This occurs when one of the
telephones involved in the call is registered with the Call Server, but the other
telephone fails or is not connected to the Call Server. The CS ACF timer is
started only if the other party does not support disconnect supervision.
HREB
The ACF call is HALF-REBUILT (HREB). This is when no call-associated
data was found and the Call Server creates the data. HREB happens when the
first of the two telephones involved registers with the Call Server, while
IP LineDescription, Installation and Maintenance
Page 100 of 910 Features
another telephone is still not connected to the Call Server. When the far-end
telephone registers, the partially-rebuilt call is promoted to REBUILT state.
PREB
The ACF call is PARTIAL-REBUILT (PREB). This is when no
call-associated data is found. The far-end IP address is not known on the Call
Server, or the far-end IP address is translated to the virtual trunk TN or Voice
Gateway TN. The Call Server creates the data leaving the far-end TN
undefined.
This scenario happens when:
•the far-end telephone is a local telephone, but while it was registered with
the remote Call Server, the local Call Server was cold-started and
TN-to-IP address associations were lost.
•the far-end telephone is a remote telephone.
The terminating-party TN in the PREB call is 0.
Note: No signaling is passed to the far-end telephone involved in the
HREG, HREB, and PREB calls. This means any features that involve
both parties do not work with such calls.
REB
The ACF call is REBUILT (REB). This means the calls have both parties
available, but all call data except bandwidth and connected transducers is lost.
LD 32 STAT command
If ACF information exists for the requested IP Phone, it is output as follows:
ACF STATUS <status> TMR <timer>
where <status> is:
•UNREG for unregistered calls
•HREG for half-registered calls
553-3001-365Standard 4.00August 2005
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