Avaya IP Line Fundamentals User Manual

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Nortel Communication Server 1000

IP Line Fundamentals

Release: 5.0 Document Revision: 01.12

www.nortel.com

NN43100-500

Nortel Communication Server 1000 Release: 5.0 Publication: NN43100-500 Document status: Standard Document release date: 16 April 2009

Sourced in Canada

LEGAL NOTICE

While the information in this document is believed to be accurate and reliable, except as otherwise expressly agreed to in writing NORTEL PROVIDES THIS DOCUMENT "AS IS" WITHOUT WARRANTY OR CONDITION OF ANY KIND, EITHER EXPRESS OR IMPLIED. The information and/or products described in this document are subject to change without notice.

Nortel, Nortel (Logo), the Globemark, SL-1, Meridian 1, and Succession are trademarks of Nortel Networks.

All other trademarks are the property of their respective owners.

Contents

Publication history 9

Revision history 9

New in this release 13

Features 13 Other 14 Subject 14

How to get help 17

Getting help from the Nortel Web site 17 Getting help over the telephone from a Nortel Solutions Center 17 Getting help from a specialist by using an Express Routing Code 18 Getting help through a Nortel distributor or reseller 18

Description 19

Contents 19 Introduction 19

Interworking 21 Applicable systems 21 System requirements 22 System configurations 22 Software delivery 24 Required packages 25 Fax/Modem pass through 25

Voice Gateway Media Cards 28

Virtual superloops, Virtual TNs, and physical TNs 46 Licenses 47 Zones 48

April 2009 9

Structure 15

Digital Signaling Processor daughterboards 20

Modem traffic 27

Media Card 32S 37 Reliable User Datagram Protocol 44 Secure Real-time Transport Protocol 45

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Administration 49

Features 51

Contents 51 Introduction 52 Live Dialpad 54

Diagnostics 55

Unicode support 55

Pop-up and USB keyboard support 55

IP Client cookies 56

e2dsetShow () 57

New IP Phone Types 58

Unique TN Types for existing IP Phones 58 Automatic IP Phone TN conversion (Flexible Registration) 59

Manual IP Phone TN conversion 60 Active Call Failover for IP Phones 61 DSP peg counter for CS 1000E systems 84 Enhanced UNIStim Firmware Download for IP Phones 84 Firmware download using UNIStim FTP 111 NAT Traversal feature 119 Corporate Directory 136 Personal Directory, Callers List, and Redial List 137 IP Call Recording 137 pbxLink connection failure detection 146 LD 117 STAT SERV 147 IP Phone support 151

IP Phone 2001 152

IP Phone 2002 153

IP Phone 2004 154

IP Phone 2007 156

IP Audio Conference Phone 2033 157

WLAN Handsets 2210/2211/2212 158

WLAN Handsets 6120/6140 159

IP Phone 1110 160

IP Phone 1120E 161

IP Phone 1140E 162

IP Phone 1150E 163

Expansion Module for IP Phone 1100 Series 164 Element Manager support 165 Call Statistics collection 166 Programmable line/DN feature keys (self-labeled) 175 Private Zone configuration 175 Run-time configuration changes 178 Network wide Virtual Office 180

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Bandwidth Management for Network wide Virtual Office 183

Requirements 183 Branch Office and Media Gateway 1000B 183

802.1Q support 184 Data Path Capture tool 188 IP Phone firmware 188 Graceful Disable 188 Hardware watchdog timer 190 Codecs 191 Set type checking and blocking 191 Enhanced Redundancy for IP Line nodes 193

Personal Directory, Callers List, and Redial List 195

Contents 195 Introduction 195 Personal Directory 197 Callers List 198 Redial List 200 IP Phone Application Server configuration and administration 201 IP Phone Application Server database maintenance 204 Call Server configuration 208 Password administration 208

Codecs 211

Contents 211 Introduction 211 Codec configuration 213 Codec registration 214 Codec negotiation 218 Codec selection 220

Installation and configuration summary 225

Contents 225 Introduction 225 Before you begin 225 Installation summary 225 Voice Gateway Media Card installation summary sheet 228

Installation and initial configuration of an IP Telephony node231

Contents 231 Introduction 231 Equipment considerations 232 Install the hardware components 232 Initial configuration of MC 32S card 247

Configuring the Leader and Follower on the MC 32S 247 Initial configuration of IP Line data 247

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Configure the Expansion Module for IP Phone 1100 Series 261 Node election rules 264

Configuration of IP Telephony nodes using Element Manager 267

Contents 267 Introduction 267 Configure IP Line data using Element Manager 268 Transfer node configuration from Element Manager to the Voice Gateway Media

Cards 295 Upgrade the Voice Gateway Media Card software and IP Phone firmware 301 Assemble and install an IP Phone 313 Change the default IPL CLI Shell password 314 Configure the IP Phone Installer Passwords 314 Import node configuration from an existing node 314

IP Line administration 317

Contents 317 Introduction 317 IP Line feature administration 318 Password security 323 IP configuration commands 339 TLAN network interface configuration commands 339 Display the number of DSPs 340 Display IP Telephony node properties 340 Packet loss monitor 344 Transfer files using the CLI 345 Download the IP Line error log 346 Reset the Operational Measurements file 346

IP Line administration using Element Manager 347

Contents 347 Introduction 347 Element Manager administration procedures 347 Backup and restore data 358 Update IP Telephony node properties 361 Update other node properties 379 Telnet to a Voice Gateway Media Card using Virtual Terminal 379 Check the Voice Gateway Channels 380 Setting the IP Phone Installer Password 381

IP Line administration using TM 3.1 385

Contents 385 Introduction 385 TM administration procedures 385 Back up and restore TM data 399

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Add an IP Telephony node in TM by retrieving an existing node 400 IP Line CLI access using Telnet or local RS-232 maintenance port 402

Voice Gateway Media Card maintenance 405

Contents 405 Introduction 405 Faceplate maintenance display codes 406 System error messages 410 IP Line and IP Phonemaintenance and diagnostics 414 IP Line CLI commands 423 Lamp Audit and Keep Alive functions 452 Voice Gateway Media Card self-tests 456 Troubleshoot a software load failure 456 Troubleshoot an IP Phone installation 459 Maintenance telephone 459 Replace the Media Card CompactFlash 460

Voice Gateway Media Card maintenance using Element Manager 461

Contents 461 Introduction 461 Replace a Voice Gateway Media Card 461 Add another Voice Gateway Media Card 466 Access CLI commands from Element Manager 469 Access the CLI from Element Manager 479

Convert IP Trunk Cards to Voice Gateway Media Cards 481

Contents 481 Introduction 481 Before you begin 482 Convert the IP Trunk cards 482 Add the converted cards to an IP Telephony node 483

NAT router requirements for NAT Traversal feature 489

Contents 489 Description 489 Requirements 490 Natcheck output 492

I/O, maintenance, and extender cable description 495

Contents 495 Introduction 495 NTMF94EA I/O cable 495 Connector pin assignments 496 NTAG81CA maintenance cable description 499 NTAG81BA maintenance extender cable 500

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Replace the NT8D81BA cable 501

Product integrity 505

Contents 505 Introduction 505 Reliability 505 Environmental specifications 506

Subnet Mask Conversion from CIDR to Dotted Decimal Format 509

Introduction 509

Download IP Line files from Nortel Web site 511

Contents 511 Introduction 511 Download files from Nortel Web site 511

Moving Voice Gateway Media Cards between systems 513

Contents 513 Introduction 513 Reconfiguring the Voice Gateway Media Card 514

Node has Leader configured 514 Node has no Leader configured 519

Upgrading the software 525

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Publication history

Revision history

April 2009

Standard 01.12. This document is up-issued to update the LD 11 Virtual Office logon for IP Phones table.

March 2009

Standard 01.11. This document is up-issued to update the Upgrade the Voice Media Card loadware section.

February 2009

Standard 01.10. Document updated to explain why a TLAN DSP IP address is required when configuring an Voice Media Gateway Card.

January 2009

Standard 01.09. This document is up-issued to update the section Bandwidth Management for Network wide Virtual Office.

November 2008

Standard 01.08. This document is up-issued to update the warning information under section Upgrade the Voice Gateway Media Card loadware.

August 2008

Standard 01.07. This document is up-issued to add the graphic in the section Card Properties.

July 2008

Standard 01.06. This document is up-issued to update the sections IP Line Administration using Element Manager and IP Line Administration using TM3.1.

July 2008

Standard 01.05. This document is up-issued to support technical additions for CR Q01884316. Removing read-only attributes from VCGM flash memory card

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10 Publication history

July 2008

Standard 01.04. This document is up-issued to support technical additions for CR Q01896401.

May 2008

This document is up-issued to support technical additions for CR Q01877613-01.

November 2007

Standard 01.03. This document is up-issued to support Communication Server 1000 Release 5.0. This document is reflects updated technical content: removal of G729 AB codec support for the IP Audio Conference Phone 2033 and wrong command ed2setShow, and an update to IP Peer calls.

June 2007

Standard 01.02. This document is up-issued to support Communication Server 1000 Release 5.0. This document includes an update to the IP Phone firmware upgrade procedures.

May 2007

Standard 01.01. This document is issued to support Communication Server 1000 Release 5.0. This document is renamed IP Line Fundamentals (NN43100-500) and contains information previously contained in the following legacy document, now retired: (553-3001-365). This document also contains the solution for CR Q01504212.

December 2006

Standard 10.00. This document is up-issued to support CS 1000 Release

4.5. This document addresses the following CRs:

• Q01512086

•

Q01452824

July 2006

Standard 9.00. This document is up-issued to address the following CRs:

• Q01368947

• Q01349604

May 2006

Standard 8.00. This document is up-issued to reflect changes to the IPL> CLI default user name and password.

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April 2006

Standard 7.00. This document is up-issued to reflect changes in technical content due to the following CRs:

• Q01270071

• Q01285983

• Q01318230

January 2006

Standard 6.00. This document is up-issued to reflect changes in technical content due to the following CRs:

• Q01206792

• Q01259132

•

Q01131032

August 2005

Standard 5.00. This document is up-issued following the removal of regulatory data.

Revision history 11

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).

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12 Publication history

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New in this release

The following sections detail what is new in IP Line Fundamentals (NN43100-500) for CS 1000 Release 5.0.

•

“Features” (page 13)

•

“Other” (page 14)

Features

See the following sections for information about the feature changes.

Live Dialpad

Live Dialpad activates the line/DN key when the user makes a call by pressing the keys on the dialpad without lifting the handset, pressing a line/DN key, or pressing the handsfree key. For further information, see

“Live Dialpad” (page 54).

Unicode support

Unicode support provides the ability for IP Phone 1110, IP Phone 1120E, IP Phone 1140E, IP Phone 1150E, and IP Phone 2007 to display characters for languages with complex fonts. The following languages are supported on the IP Phones with Unicode capabilities: Traditional Chinese, Simplified Chinese, Arabic, Japanese, Greek, Korean, and Hebrew. For further information, see “Unicode support” (page 55).

IP client cookies

IP client cookies provide transparent data transfer between the CS 1000 and third-party applications, for example, Citrix AG. For further information, see “IP Client cookies” (page 56).

New IP Phone types

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14 New in this release

IP Phones are provided with unique Terminal Number (TN) types. IP Phones are no longer required to emulate one of five IP Phones. For further information, see “New IP Phone Types” (page 58).

Other

IP Line Fundamentals for CS 1000 Release 5.0. includes the following changes:

•

• Removed the following sections of TM 3.1, which are not supported

•

Rebranded OTM 2.2 to TM 3.1.

in CS 1000 Release 5.0:

—

Configuration of IP Telephony nodes using TM 3.1 section

— Update IP Telephony node properties using TM 3.1 — Update Voice Gateway Media Card properties — Voice Gateway Media Card maintenance using TM 3.1

Added the Media Card 32S for SRTP support. Updated Element Manager with enhancements.

Subject

•

Added support for Enterprise Common Manager (ECM).

•

Added new IP Phones, and Expansion Module for IP Phone 1100 Series.

•

Updated IP Phone firmware upgrade process, which is enhanced to be IP Telephony Node specific.

•

Removed instances of CS 1000S.

•

Removed instances of Meridian 1.

•

Removed instances of Media Card 8-port line card.

• Removed instances of IP Phone firmware download from the Voice

Gateway Media Card.

This document:

• describes the physical and functional characteristics of the IP Line

application for Nortel Communication Server (CS) 1000 Release 5.0 system and describes its use on Signaling Servers and Voice Gateway Media Cards.

• explains how to engineer, install, configure, administer, and maintain

an IP Telephony node that contains Voice Gateway Media Cards.

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Structure

This document has separate chapters that are applicable only to either Telephony Manager (TM) 3.1 or Element Manager.

The configuration, administration, and maintenance sections are divided into two chapters. For example, a generic configuration chapter deals with tasks related to the installation and configuration of IP Line. This chapter is followed by the configuration chapter for Element Manager. The administration and maintenance chapters have the same format.

Note on legacy products and releases

This NTP contains information about systems, components, and features that are compatible with Nortel Communication Server 1000 Release 5.0 software. For more information about legacy products and releases, click the Technical Documentation link under Support & Training on the Nortel home page: w

Related information

The following documents are referenced in this document:

• Converging the Data Network with VoIP Fundamentals (NN43001-260)

Subject 15

ww.nortel.com

•

Transmission Parameters Reference (NN43001-282)

•

Signaling Server Installation and Commissioning (NN43001-312)

•

Branch Office Installation and Commissioning (NN43001-314)

•

Telephony Manager 3.1 System Administration (NN43050-601)

•

WLAN IP Telephony Installation and Commissioning (NN43001-504)

•

Emergency Services Access Fundamentals (NN43001-613)

•

Element Manager System Reference—Administration (NN43001-632)

•

Enterprise Common Manager Fundamentals (NN43001-116).

•

IP Phones Fundamentals (NN43001-368),

•

Software Input Output Reference—System Messages (NN43001-712)

• Communication Server 1000M and Meridian 1 Small System Planning

and Engineering (NN43011-220)

• Communication Server 1000M and Meridian 1 Large System Planning

and Engineering (NN43021-220)

• Communication Server 1000M and Meridian 1 Large System

Maintenance (NN43021-700)

• Communication Server 1000E Planning and Engineering

(NN43041-220)

• IP Phone 2001 User Guide (NN43115-102)

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16 New in this release

• IP Phone 2002 User Guide (NN43116-104)

• IP Phone 2004 User Guide (NN43117-102)

• IP Phone 2007 User Guide (NN43118-100)

•

• IP Softphone 2050 User Guide (NN43119-101)

• Mobile Voice Client 2050 User Guide (NN43119-103)

• IP Phone 1110 User Guide (NN43110-101)

•

• IP Phone 1150E User Guide (NN43114-100)

• IP Phone Key Expansion Module User Guide (NN43119-102)

•

IP Audio Conference Phone 2033 User Guide (NN43111-100)

IP Phone 1120E User Guide (NN43112-103) IP Phone 1140E User Guide (NN43113-106)

Expansion Module for IP Phone 1100 Series User Guide (NN43130-101)

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How to get help

This chapter explains how to get help for Nortel products and services.

Getting help from the Nortel Web site

The best way to get technical support for Nortel products is from the Nortel Technical Support Web site:

ww.nortel.com

This site provides quick access to software, documentation, bulletins, and tools to address issues with Nortel products. From this site, you can:

• download software, documentation, and product bulletins

•

search the Technical Support Web site and the Nortel Knowledge Base for answers to technical issues

•

open and manage technical support cases

Getting help over the telephone from a Nortel Solutions Center

If you do not find the information you require on the Nortel Technical Support Web site, and you have a Nortel support contract, you can also get help over the telephone from a Nortel Solutions Center.

In North America, call 1-800-4NORTEL (1-800-466-7835). Outside North America, go to the following Web site to obtain the

telephone number for your region:

ww.nortel.com/callus

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18 How to get help

Getting help from a specialist by using an Express Routing Code

To access some Nortel Technical Solutions Centers, you can use an Express Routing Code (ERC) to quickly route your call to a specialist in your Nortel product or service. To locate the ERC for your product or service, go to:

ww.nortel.com/erc

Getting help through a Nortel distributor or reseller

If you purchased a service contract for your Nortel product from a distributor or authorized reseller, contact the technical support staff for that distributor or reseller.

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Description

Contents

This section contains the following topics:

•

“Introduction” (page 19)

• “Interworking” (page 21)

•

“Applicable systems” (page 21)

• “System requirements” (page 22)

•

“System configurations” (page 22)

• “Software delivery” (page 24)

•

“Required packages” (page 25)

• “Voice Gateway Media Cards” (page 28)

•

“Virtual superloops, Virtual TNs, and physical TNs” (page 46)

•

“Licenses” (page 47)

•

“Zones” (page 48)

•

“Administration” (page 49)

Introduction

Communication Server (CS) 1000 Release 5.0 requires the IP Line application.

The IP Line application provides an interface that connects IP Phones to a CS 1000 Call Server.

CS 1000 Release 5.0 requires a Signaling Server to operate. You must upgrade your Meridian 1 system, if your system is IP enabled to include a Signaling Server, which in turn becomes a CS 1000M system. CS 1000 Release 5.0 is supported on an analog/digital (TDM) only system without a Signaling Server if the system is not IP enabled. For information about upgrading your system, see Communication Server 1000M and Meridian 1

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20 Description

Digital Signaling Processor daughterboards

Small System Planning and Engineering (NN43011-220)or Communication Server 1000M and Meridian 1 Large System Planning and Engineering (NN43021-220).

ATTENTION

The IP Line version of software must match the Call Server version.

CS 1000 Release 5.0 introduces Digital Signaling Processor (DSP) daughterboards, which are placed on the Media Gateway Controller (MGC) to provide DSP resources for connecting IP and Time Divisional Multiplexing (TDM) devices together in a CS 1000E Media Gateway (MG 1000E) system. The following DSP daughterboards are available:

•

32 port daughterboard

•

96 port daughterboard

These daughterboards provide an optional solution to installing the Voice Gateway Media Cards within the CS 1000E Media Gateway (MG 1000E) chassis. However, Voice Gateway Media Cards are still supported within an MG 1000E with a Media Gateway Controller (MGC) and DSP daughterboards. The MGC is only used in a Media Gateway chassis or an Option 11C cabinet.

For further information about DSP resources residing on the MGC that are configured with DSP daughterboards, see Communication Server 1000E

Installation and Commissioning (NN43041-310).

Voice Gateway Media Cards

If a Media Card 32-port card, a ITG-P 24-port card, or a Media Card 32S is running IP Line software, it is known as a Voice Gateway Media Card.

In this document, Media Card 32-port card and Media Card 32S card are referred to as Media Card 32-port cards, unless explicitly stated.

DHCP server

A Dynamic Host Configuration Protocol (DHCP) server can be used to provide the required information so that the IP Phonenetwork connection can connect to the Signaling Server or Line Terminal Proxy Server (LTPS).

For more information about DHCP, see Converging the Data Network

with VoIP Fundamentals (NN43001-260)and IP Phones Fundamentals (NN43001-368).

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Interworking

The IP Phoneuses the IP network to communicate with the LTPS and the optional DHCP server. Figure 1 "System architecture" (page 21) shows a diagram of the system architecture.

Figure 1 System architecture

Applicable systems 21

Applicable systems

The CS 1000 system supports the Media Card 32-port line card, Media Card 32S card, and ITG-Pentium 24-port line card.

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22 Description

Unsupported products

The following remote service products do not support the Media Card 32-port line card, Media Card 32S card, and ITG-Pentium 24-port line card:

• Carrier Remote

• Mini-carrier Remote

• Fiber Remote

•

Fiber Remote Multi-IPE

System requirements

CS 1000 Release 5.0 software is the minimum system software for IP Line.

Element Manager and Telephony Manager 3.1

In CS 1000 Release 5.0, Element Manager is the primary interface for Voice Gateway Media Cards and IP Line.

Telephony Manager (TM) 3.1 is used only to obtain Operational Measurement (OM) reports. TM 3.1 is the minimum required version.

CS 1000systems

Element Manager is used as the configuration, administration, and maintenance interface for IP Line on a CS 1000system.

If you are trying to use TM 3.1 to perform an action available through Element Manager, then TM 3.1 launches Element Manager automatically.

Corporate Directory

TM 3.1 is necessary for creation of the Corporate Directory database.

SNMP and alarms

Element Manager does not provide an SNMP alarm browser. Nortel recommends you use TM 3.1 Alarm Manager when SNMP alarm collection is required.

System conﬁgurations

Although IP Line can be used in different system configurations and its use can vary in those configurations, there are two basic system configurations supported in CS 1000 Release 5.0 . See Table 1 "Possible system

configurations" (page 23).

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Table 1 Possible system configurations

System Signaling Server present

CS 1000E Yes CS 1000M Yes

CS 1000systems

CS 1000systems 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 central processor drives the signaling for IP Phones and IP Peer networking.

In IP Line, 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.

The Signaling Server is the node leader and, by default, acts as a Master for the node.

System configurations 23

Signaling Servers

The following Signaling Servers are available for CS 1000 Release 5.0:

• ISP1100

•

HP-DL320-G4

•

IBM-X306m

•

Common Processor Pentium Mobile (CP PM)

For further information about Signaling Server hardware platforms, see Signaling Server Installation and Commissioning (NN43001-312).

In a CS 1000 system, the Signaling Servers can be used in Leader/Follower and Primary/Alternate/Failsafe combinations.

In CS 1000 Release 5.0 , the Signaling Servers support the following applications:

• Line Terminal Proxy Server (LTPS)

• Virtual Trunk

• H.323 Gateway

• SIP Gateway

• SIP Redirect Server

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24 Description

• Network Routing Service (NRS)

• Personal Directory

• SIP Proxy Server (SPS)

•

Element Manager

Signaling Server redundancy Signaling Server redundancy ensures that telephony services can withstand single hardware and network failures. Several Signaling Servers can load share when the system contains multiple Signaling Servers or Voice Gateway Media Cards. One Signaling Server is a Leader Signaling Server that acts as the primary, or master, Terminal Proxy Server (TPS). The other Signaling Server is a Follower Signaling Server that acts as the backup, or secondary redundant TPS. There are several methods of redundancy for a Signaling Server. See Table 2 "Methods of Signaling Server redundancy" (page 24).

Table 2 Methods of Signaling Server redundancy

Stage With a Signaling Server to share the load

Without a Signaling Server to share the load

Description

A Signaling Server, which shares the load, can be configured in a normal configuration.

If the primary Signaling Server fails, the Signaling Server, which shares the load, takes over and all IP Phones register with this Signaling Server.

If the Signaling Server, which shares the load, fails, one of the Voice Gateway Media Cards is elected to be the node Master.

The IP Phones then register to the Voice Gateway Media Cards.

If there is no Signaling Server to share the load, and the primary Signaling Server fails, one of the Voice Gateway Media Cards is elected to be the node Master.

The IP Phones then register to the Voice Gateway Media Cards.

Software delivery

IP Line supports software delivery through the following formats:

1. CompactFlash

2. Signaling Server CD-ROM

3. Download from the Nortel Web site

Standalone IP Line software is not available through CD-ROM.

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Required packages

The IP Phones require the software packages listed in Table 3 "Required

packages" (page 25).

Table 3 Required packages

Package Package number

Fax/Modem pass through 25

M2000 Digital Sets (DSET) Aries Digital Sets (ARIE)

ATTENTION

To configure IP Line in groups five to seven on Option 81C CP PII or CS 1000M MG, the Fiber Network (FIBN) software package 365 is required.

Fax/Modem pass through

The Fax/Modem pass through feature provides a modem pass through allowed (MPTA) class of service (CLS) for an analog phone TN. MPTA CLS dedicates an analog phone TN to a modem or a Fax machine terminal. A connection that initiates from the dedicated TN, and/or calls that terminate at the dedicated TN through a Digital Signal Processor (DSP), use a G711 NO VAD codec on the Call Server.

Modem Pass through is a specific configuration of a G.711 VoIP channel that improves modem performance compared to standard VoIP configuration. Auto switch to Voice Band Data (VBD) is a feature of the DSP; the DSP monitors the data stream to distinguish between voice and data calls. The DSP reconfigures to modem pass through mode when it determines the call is a modem call.

170

The DSP mode on the Mindspeed DSP (MC32S/DB96/DB32) for MPTA to MPTA fax/modem calls displays ModemPT in the dspMode field of the vgwShow command. The Teology DSP (ITGSA) displays PassThrough.

For modem calls between CS 1000 systems connected by analog and digital trunks, you must configure MPTA CLS on the Call Server of each CS 1000 system for analog units connected to modems. MPTA CLS configuration is necessary because the call setup negotiation is not done end to end as it is for virtual trunks. If the analog unit on one Call Server uses MPTA CLS and the analog unit on the other Call Server uses modem pass through denied (MPTD) CLS, the modem call fails.

When MPTA CLS is configured on a TN, the T.38 protocol is no longer supported for that particular TN. Any call setup with an analog phone TN that has MPTA configured must use G711 Codec exclusively, as this is the only codec available for making calls using this TN. G711 codec is present by default in the DSP configuration.

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26 Description

To ensure proper functioning of the MPTA CLS, the Enable Modem/Fax pass through mode check box must be selected in the Gateways section of Element Manager. This check box is selected by default in Element Manager. To enable SG3 fax calls over Teology DSP, the Enable V.21 FAX tone detection check box in Element Manager must be selected. For SMC and MC32S cards, this setting is available in the Voice Gateway and IP phone codec profile section in the Nodes summary page. In the modem pass through mode of operation, the DSP state is displayed as MPT on the MGC and MC32S cards and as PassThru on SMC cards.

MPTA CLS is not supported through Telephony Manager (TM) and Basic Client Configuration (BCC). BCM 50 only supports modem pass through over G.711 in Release M50R3.

MPT CLS is supported by the G.711 codec only; MPT CLS includes no other codecs. The packet interval for G.711 codec is set to 20 msecs in MPT.

The maximum speed supported for modem and fax is 33.6 Kbps. This limit is imposed by the analog line card.

MPT allows CS 1000 to support the following:

• modem pass through

•

Super G3(SG3) fax at V.34(33.6Kbps)

• V.34 rate (33.6 Kbps) modems

•

Fax machines that support V.17, V.27, V.29, and V.34

Note: MPT CLS is not be supported on IP trunks.

When the TN on the CS 1000 is configured with MPTA CLS, it supports V.17, V27 and V29 Fax calls. However, the DSP mode is “FaxBegin,” not “ModemPT”. The MPTA CLS forces calls originated or terminated on the TN to use the G.711 NO VAD Codec. This codec selection supersedes the existing bandwidth management strategy on the CS 1000.

When a CS 1000E is inter-connected with a system from a different vendor, the modem pass through feature works if the third party system supports the modem pass through mode of operation.

The Voice Gateway application displays different Auto-switch states (ModemPT, Passthru) in the dspMode field of the vgwShow command, based on tones detected by the DSP. These tones are generated by modem and fax machines connected in the TDM domain. The Voice Gateway application does not control Auto-switch states during fax and modem calls, and the dspMode reports tone indications from DSP. The

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Mindspeed DSP sends tone detection events to the host processor and changes to Modem Passthough and Pass-through auto-switch states (with or without Redundancy), based on the tones detected, as it is configured in Auto-switch mode.

For interface commands, responses, and definitions for MPT, see Table 4

"Interface commands and responses" (page 27).

Table 4 Interface commands and responses

Fax/Modem pass through 27

Command prompt

CLS MPTA Turn on the MPT feature. CLS MPTD Turn off the MPT feature.

User response Description

ATTENTION

CLS MPTA and CLS MPTD are included in LD 10 for analog line card units. MPTA to MPTA and MPTA to MPTD fax/modem calls succeed over the G.711

codec in ModemPT DSP mode. However; MPTD to MPTD modem calls fail. MPTD to MPTD fax calls succeed (best effort) over G.711 if the Enable

Modem/Fax pass through mode and Enable V.21 FAX tone detection options are set to ON in the gateway. If the Enable Modem/Fax pass through mode is unchecked, the fax call goes over the T.38 codec. The T.38 fax codec is recommended for fax calls over SIP and H323 trunks. To select the T.38 fax codec, the analog line units at both originating and terminating systems must be set to MPTD CLS.

For information on feature packaging requirements, see Table 5 "Feature

packaging requirements" (page 27).

Table 5 Feature packaging requirements

Package mnemonic

Package number

Package description Package type

(new, existing, or dependency)

Applicable market

Softswitch

IPMG

402

403

Modem trafﬁc

CS 1000E supports modem traffic in a campus-distributed network with the following characteristics:

• Media card configuration:

Identifies a softswitch system.

Identifies a system that is equipped with IPMGs.

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Existing All

28 Description

— G.711 codec —

20 msec packet size

•

one-way delay less than 5 msec

• low packet loss

•

V.34 rate (33.6 Kbps)

Performance degrades significantly with packet loss (must be less than

0.5%) and when the delay (round trip) is greater than 50 msec and mean jitter is greater than 5msec.

ATTENTION

Nortel has conducted extensive but not exhaustive tests of modem-to-modem calls, data transfers, and file transfers between a CS 1000E and MG 1000E, using Virtual Trunks and PRI tandem trunks. While all tests have been successful, Nortel cannot guarantee that all modem brands will operate properly over all G.711 Voice over IP (VoIP) networks. Before deploying modems, test the modem brand within the network to verify reliable operation. Contact your system supplier or your Nortel representative for more information.

Voice Gateway Media Cards

Voice Gateway Media Card is a term used to encompass the Media Card 32-port card, ITG-P 24-port line card, and the Media Card 32S card. These cards plug into an Intelligent Peripheral Equipment (IPE) shelf in the CS 1000Msystems 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 32-port and the Media Card 32S card occupy one slot.

The Media Card 32S card provides the following features:

•

Secure Real-time Transport Protocol (SRTP)

•

two Digital Signal Processors (DSP), based on an ARM processor

• channel density of 32 ports

• cost improvement over existing Media Cards

The Media Card 32-port card provides the following features:

• 32-port card 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 the 32-port

version)

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Table 6 Card comparison

Voice Gateway Media Cards 29

• reduces the slot count from a dual IPE slot to a single IPE slot

• supports up to 128 IP Phones in failover scenarios

Table 6 "Card comparison " (page 29) provides a comparison of the ITG-P

24-port line card, Media Card 32-port card, and the MC 32S card.

Item

Total DSP Channels Number of slots the

card occupies Operating System VxWorks 5.3 VxWorks 5.4 VxWorks 5.5 Processor Pentium IXP1200 ARM920T

DSP 8 x TI5409 4 x TI5421 32 DSP channels

Telogy version

ITG-P 24-port card Media Card 32-port

card

24 32 32 211

7.01

8.1 High Density version (8 ports for each DSP)

Media Card 32S card

Contains two ARM processors, one that runs VxWorks and one that runs the Mindspeed application.

provided on a second processor

DSP code is provided by Mindspeed.

Number of IP Phones that can register on each LTPS

The IP Phones register to the Signaling Server. If the Signaling Server fails, the IP Phones register to the Voice Gateway Media Card. A Signaling Server can register a maximum of 5000 IP Phones.

Image file name prefixes shown by swVersionShow command

/C: drive On board Flash 2 x

Upgrade Two images files One image file (no

96 128 128

IPL P IPL SA IPL ARM

4MB

Plug-in CompactFlash 32 MB

backup)

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CompactFlash 128 MB

One zipped package

30 Description

ATTENTION

In a CS 1000 system, 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 Embedded LAN subnet. The TLAN (Telephony LAN) subnet carries telephony, voice, and signaling traffic. The TLAN subnet, also known as the Voice LAN subnet, connects to the customer network.

Voice Gateway Media Cards have an ELAN network interface (10BaseT) and a TLAN network interface (10/100BaseT) on the I/O panel.

There is an RS-232 Maintenance Port connection on the faceplates of both the ITG-P 24-port card and the Media Cards. 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.

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 32 channels on each Media Card.

Both cards support a 4:1 concentration of registered IP Phones (IP Phone 2001, IP Phone 2002, IP Phone 2004, IP Phone 2007, IP Audio Conference Phone 2033, IP Softphone 2050, Mobile Voice Client [MVC] 2050, IP Phone 1110, IP Phone 1120E, IP Phone 1140E, IP Phone 1150E, WLAN Handset 2210, WLAN Handset 2211, WLAN Handset 2212, WLAN Handset 6120, WLAN Handset 6140 to gateway channels. The ITG-P supports 96 registered IP Phones. The Media Cards supports 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 Time Division Multiplexing (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 or 128 on the Media Cards), 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.

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Voice Gateway Media Cards 31

Voice Gateway Media Card controls, indicators, and connectors

The following sections show the faceplates and describe the faceplate components of the following Media Cards:

• Figure 2 "Media Card 32-port " (page 31)

• Figure 3 "ITG-P 24-port card faceplate" (page 34)

• Figure 5 "Media Card 32S card faceplate" (page 40)

Figure 2 Media Card 32-port

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32 Description

Faceplate components

The components on the faceplate of the Media Card 32-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 and 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.

WARNING

Do not format the PC Card by using a Windows application. As well, only format the PC Card using the type of card on which it is 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.

MAC address label

The MAC address label on the card 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:

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Voice Gateway Media Cards 33

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):

100 (100BaseT)

2. 10 (10BaseT)

3. A (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 eight-pin mini-DIN serial maintenance port connection. The faceplate on the card is labeled J2.

ITG-P 24-port card controls, indicators, and connectors

Figure 3 "ITG-P 24-port card faceplate" (page 34) shows the ITG-P 24-port

card faceplate components.

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34 Description

Figure 3 ITG-P 24-port card faceplate

Faceplate components

The components on the faceplate of the ITG-P 24-port card are described in the following sections.

NWK

The faceplate connector labeled NWK is a nine-pin, subminiature D-type connector. The connector is not used for the IP Line application.

WARNING

The NWK connector looks like a nine-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.

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Voice Gateway Media Cards 35

ITG-P LED (card status)

The red status faceplate LED indicates the enabled and 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.

MAC address label

The MAC address label on the card 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:

ETHERNET ADDRESS MOTHERBOARD 00:60:38:8c:03:d5 DAUGHTERBOARD 00:60:38:01:b3:cb

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.

There are no Ethernet status LEDs for the ELAN network interface.

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.

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36 Description

WARNING

Do not format the PC Card by using a Windows application. As well, only format the PC Card using the type of card on which it runs. 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 93 "ITG-P 24-port line card faceplate maintenance

display codes" (page 406) and Table 94 "Media Card 32-port card faceplate maintenance display codes" (page 407).

RS-232 maintenance port

The ITG-P 24-port line card faceplate provides a female eight-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.

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).

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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 "ITG-P 24-port

card physical assembly" (page 37).

Figure 4 ITG-P 24-port card physical assembly

Voice Gateway Media Cards 37

Media Card 32S

The Media Card 32S card (NTDW65AA) provides the following features:

• Secure Real-time Transport Protocol (SRTP)

• two Digital Signal Processors (DSP), based on an ARM processor

• channel density of 32 ports

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38 Description

Secure Real-time Transport Protocol

The Media Card 32S card uses Secure Real-time Transport Protocol (SRTP) to secure the IP media path between the card and the Nortel Phase II IP Phones, IP Phone 1110, IP Phone 1120E, IP Phone 1140E, IP Phone 1150E, another Media Card 32S card, or a DSP daughterboard. When Media Security is configured to On, the Call Server sends a message to the Voice Gateway software on the Media Card 32S card to activate SRTP for the media connection established for that call.

Media Security is configured by the system administrator. For information about SRTP, see IP Phones Fundamentals

(NN43001-368) , and System Management Reference (NN43001-600). Processors Two separate processors on the MC 32S card are known

as the Control and Signaling Processor (CSP) and the Media Stream Processor (MSP). The CSP runs application and signaling code, whereas the MSP processes the media streams.

Table 7 "Files downloaded to the MC 32S card" (page 38) lists the file

names and paths for files that are downloaded from the Signaling Server to the MC 32S card.

Table 7 Files downloaded to the MC 32S card

File name Path on Signaling

Server

IPL5.00XX.mc32s /u/fw /pl Software binaries bootp.tab /u/config /u/config bootp parameter file config.ini /u/config /u/config config file

Path on MC 32S card Description

The software for the MC 32S consists of five files, which are located in the IPL5.00XX.mc32s zipped file stored on the Signaling Server. All Gold versions of firmware are loaded on the MC 32S when the card is shipped from the factory. The Gold Boot Code, the upgradeable Boot Code, the Gold MSP Image, the Gold VxWorks Kernel for CSP Image, the Gold App Image, and the upgradeable Field-programmable Gate Array (FPGA) image are stored in separate areas of the MC 32S FLASH memory. A new load can be programmed into the appropriate area of FLASH memory under software control.

Table 8 Files within the zipped file

File name Description

bootrom.bin MC 32S boot code image

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Voice Gateway Media Cards 39

Table 8 Files within the zipped file (cont’d.)

mainos.sys VxWorks 5.5 Kernel image for the Control and

Signaling Processor (CSP). The CSP runs the application and signaling code.

mainos.sym mainos.sys and mainos.sym files form the CSP

image

ldvoice.axf Media Stream Processor (MSP) load. The MSP

runs the media stream processing code. This load is a binary image.

fpga.xsvf MC 32S board FPGA load

Figure 5 "Media Card 32S card faceplate" (page

40) shows the Media Card 32S faceplate.

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40 Description

Figure 5 Media Card 32S card faceplate

Faceplate components The components on the faceplate of the MC

32S card are described in the following sections.

Reset button

Use the Reset button on the faceplate to manually reset the card without cycling power to it. The Reset button is used to reboot the card after a software upgrade or to clear a fault condition.

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Voice Gateway Media Cards 41

Enable LED

The faceplate red LED indicates the following:

• the enabled and disabled status of the card

• the self-testing result during power-up or card insertion into an

operational system

Ethernet port

The Ethernet port is used for debugging. It connects to the six-ports layer-2 switch through port three on the card and is mirrored to any other ports of the layer 2-switch.

Four character LED display

Ethernet activity LEDs

The faceplate contains two Ethernet activity LEDs: one for the ELAN network interface and one for the TLAN network interface. The LEDs indicate the following links on the ELAN network interface and TLAN network interface:

•

Link Activity

• Speed

RS-232 Maintenance port

The Media Card 32S card faceplate provides a female eight-pin mini-DIN serial maintenance port connection.

Functional description of the Voice Gateway Media Cards

The Media Cards and ITG-P 24-port line card perform three separate functions, depending on the system in which the card is located, and the type of card:

1. The card acts as a gateway between the circuit-switched voice network

and the IP network.

2. The card acts as a Line Terminal Proxy Server (LTPS) or virtual line

card for the IP Phones.

3. The Media Card 32S card provides Secure Real-time Transport

Protocol (SRTP) to secure the IP media path between the DSP channels and the card.

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42 Description

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 or disconnect the audio channel

• uses Real-time Transport Protocol (RTP) and Real-time Control

Protocol (RTCP) to transport audio between the gateway and the IP Phoneor another gateway

• encodes and decodes audio from PCM to and from the IP Phoneformat

• provides echo cancellation for the speaker on IP Phones for echoes

that originate in the circuit-switched voice network (not applicable to the IP Softphone 2050or MVC 2050, as they have no handsfree capability)

Gateway functionality on the CS 1000systems

A Signaling Server is always present in the CS 1000systems. The LTPS executes on the Signaling Server, and the Voice Gateway executes on the Voice Gateway Media Cards and DSP daughterboards. 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 LTPS and requests a response if it has room for another IP Phone.

IP Phone registration

This section describes the maximum number of IP Phones that can register to an LTPS in a CS 1000 system.

IP Phoneregistration on a CS 1000system

On a CS 1000system, the IP Phones register with the LTPS on the Signaling Server. If more than one Signaling Server exists, the IP Phoneregistrations are distributed equally among the Signaling Servers to aid in load balancing.

If the primary Signaling Server fails, a secondary Signaling Server takes over (if it exists), and the IP Phones that are registered with the failed Signaling Server re-register with the LTPS on secondary Signaling Server. If no other Signaling Servers exist or if the Signaling Servers fail, the IP Phones register with the LTPS on the Voice Gateway Media Cards.

ATTENTION

Each Signaling Server supports the registration of up to 5000 IP Phones.

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For more information about Signaling Server failure and redundancy, see

Communication Server 1000M and Meridian 1 Small System Planning and Engineering (NN43011-220) and Signaling Server Installation and Commissioning (NN43001-312).

Interactions with IP Phones

The following information describes the process by which an IP Phoneregisters and unregisters with a CS 1000system.

Registration

Table 9 "Registration process" (page 43) describes the registration

process.

Table 9 Registration process

Voice Gateway Media Cards 43

Step

Description

The IP Phonereceives the IP address of the Connect Server (co-located with the LTPS) through either DHCP or manual configuration.

The IP Phonecontacts the Connect Server.

The Connect Server instructs the IP Phoneto display a message on its display screen requesting the customer IP Telephony node number and TN.

The node number and TN are entered. The Connect Server redirects the IP Phoneto the Node Master.

The IP Phonecontacts the Node Master. The Node Master redirects the IP Phoneto the LTPS.

The IP Phonecontacts the LTPS.

If the IP Phoneis valid, the LTPS registers it with the system.

Unregistration

Table 10 "Unregistration process" (page 43) describes the unregistration

process.

Table 10 Unregistration process

Step

Description

If the LTPS detects a loss of connection with one of its registered IP Phones, it logs the event.

The LTPS then sends an unregister message to the system for that IP Phone.

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44 Description

Signaling and messaging

Signaling protocols

Reliable User Datagram Protocol

The IP Line application sends Scan and Signaling Distribution (SSD) messages to the Call Server through the system ELAN subnet. When tone service is provided, the service is signaled to the LTPS by using new SSD messages sent through the ELAN subnet.

The signaling protocol between the IP Phoneand the IP Telephony node is the Unified Networks IP Stimulus Protocol (UNIStim). The Reliable User Datagram Protocol (RUDP) is the transport protocol.

Reliable User Datagram Protocol (RUDP) is used for:

•

signaling between the Call Server and the LTPS

• signaling between the IP Telephony node and the IP Phones

For more information, see “ELAN TCP transport” (page 45).

Description

Signaling messages between the LTPS and IP Phones use RUDP. Each RUDP connection is distinguished by its IP address and port number. RUDP is another layer on top of User Datagram Protocol (UDP). RUDP is proprietary to Nortel.

The features of RUDP are as follows:

•

provides reliable communication system over a network

•

packets 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

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

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packet, is sent to A. A retransmits the first 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 Phoneand an LTPS on the Voice Gateway Media Card or Signaling Server.

Secure Real-time Transport Protocol

The Media Card 32S card uses Secure Real-time Transport Protocol (SRTP) to secure the IP media path between the card and the Nortel Phase II IP Phones, IP Phone 1110, IP Phone 1120E, IP Phone 1140E, or IP Phone 1150E, or another Media Card 32S card, or a DSP daughterboard. When Media Security is configured to On, the Call Server sends a message to the Voice Gateway software on the Media Card 32S card to activate SRTP for the media connection established for that call.

Voice Gateway Media Cards 45

Media Security is configured by the system administrator. For information about SRTP, see IP Phones Fundamentals

(NN43001-368), and System Management Reference (NN43001-600).

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 that RUDP messages are exchanged to maintain the link status between the Call Server and the Signaling Server or 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.

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 CPU. TCP transports messages, whereas RUDP establishes and maintains the link.

The IP Line software version must match the Call Server software version; otherwise, IP Line terminates the link and logs an error message.

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Virtual superloops, Virtual TNs, and physical TNs

Virtual TNs (VTN) enable configuration of service data for an IP Phone, such as key layout and class of service, without requiring the IP Phoneto be dedicated (hard-wired) to a given TN on the Voice Gateway Media Card.

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 TN instead of virtual TN. Calls are made between an IP Phoneand circuit-switched telephone or trunks using the full CS 1000 feature set. Digital Signal Processor (DSP) channels are allocated dynamically for this type of call to perform the encoding or decoding required to connect the IP Phoneto the circuit-switched network.

The IP Phones (virtual TN) are defined on virtual superloops. To create an IP Phoneusing VTNs, create a virtual superloop in LD 97 or in Element Manager. To create the virtual superloop in Element Manager, click System > Core Equipment > Superloops in the Element Manager navigator.

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 TN)-to-IP Phone calls.

You can configure up to 1024 VTNs on a single virtual superloop for Large Systems, CS 1000M Cabinetand CS 1000M Chassissystems, and CS 1000E systems.

Each ITG-P 24-port card provides 24 physical TNs, and each Media Card 32-port card provides 32 physical TN. The physical TN are the gateway channels (DSP ports), which provide 128 channels. The channels (ports) on the Voice Gateway Media Cards are pooled resources.

Configure the physical TNs (IPTN) in LD 14. They appear as VGW data blocks.

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Licenses

Licenses 47

There are three types of licenses:

•

Temporary IP User Licence for IP Phones configured for Branch Office or Network wide redundancy

• Basic IP User License for the IP Phone 2001, IP Audio Conference

Phone 2033, and IP Phone 1110

• IP User License for the IP Phone 2002, IP Phone 2004, IP Phone

2007, IP Phone 1120E, IP Phone 1140E, IP Phone 1150E, IP Softphone 2050, Mobile Voice Client (MVC) 2050, WLAN Handset 2210, WLAN Handset 2211, WLAN Handset 2212, WLAN Handset 6120, and WLAN Handset 6140

If insufficient Temporary IP User Licenses are available, Basic IP User License and IP User License can be used.

If insufficient Basic IP User Licenses are available for the IP Phone 2001, IP Audio Conference Phone 2033, and IP Phone 1110, then the IP User License can also be used.

If there are no Basic IP User Licenses available for the IP Phone 2001, IP Audio Conference Phone 2033, and IP Phone 1110, and IP User Licenses are used, 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 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 Phoneinstalled on CS 1000 system. A new license uses the existing keycode to enable the IP Phonein the system software. The default is zero.

To expand the license limits for the IP Phones, order and install a new CS 1000 keycode. See Features and Services Fundamentals—Book 4 of 6 (NN43001-106).

ATTENTION

Individual licenses are not supported on Functional Pricing. With Functional Pricing, licenses are provisioned in blocks of eight.

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License limits

Zones

The total number of TN configured with Temporary IP User Licenses must not exceed 100. The total number of TN configured with Basic IP User Licenses must not exceed 32 767. The total number of TN configured with IP User Licenses must not exceed 32 767. The total number of IP phones configured within the system must not exceed the allowable 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, which indicates the zone to which they belong.

When a call is made, the codecs that are used vary, depending on which zones 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.

Access zones in Element Manager by clicking IP Network > Zones in the Element Manager navigator.

Configure each zone 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 shared and private zones, see “Private Zone

configuration” (page 175). For more information about zones and virtual

trunks, see IP Trunk Fundamentals (NN43001-563). For more information about zones and branch office locations, see Branch Office Installation and Commissioning (NN43001-314). For information about Adaptive Bandwidth Management and Alternative Call Routing, see Branch Office Installation and Commissioning (NN43001-314).

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Administration

The Voice Gateway Media Card is administered using multiple management interfaces, including the following:

• Web browser interface provided by Element Manager—Element

•

• Administration and maintenance overlays of Call Servers.

•

TM 3.1

IP Line uses TM 3.1 to obtain OM reports only.

Element Manager

Element Manager is a resident Web-based user interface used to configure and maintain CS 1000 components. Element Manager Web interface enables IP Line to be configured and managed from a Web browser.

Administration 49

Manager is used to administer Voice Gateway Media Cards in the systems that use a Signaling Server.

Command Line Interface (CLI)—The CLI prompt, which displays depends on the type of Voice Gateway Media Card in the system. IPL> prompt displays for the ITG-P 24-port line card, and Media Card 32-port. oam> or LBD> prompt displays for the Media Card 32S card.

IP Line application GUI provided by TM 3.1. TM 3.1 is used to obtain OM reports only.

The Element Manager Web server resides on the Signaling Server or within Enterprise Common Manager (ECM) framework. For further information about Element Manager residing on a Signaling Server, see Element Manager System Reference—Administration (NN43001-632).

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, and D-channels)

• maintenance commands, system status inquiries, backup and restore

functions

• software download, patch download, patch activation

• configuration of SNMP parameters (such as SNMP community strings,

and SNMP trap destinations

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Command Line Interface

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 are traditionally available through multiple overlays.

• Parameters are presented in logical groups to increase ease-of-use

and speed-of-access.

•

Administrators see information that relates directly to the task at hand by using the hide or show information option.

• Full-text descriptions of parameters and acronyms help administrators

reduce configuration errors.

• Configuration screens offer preselected defaults, drop-down lists,

check boxes, and range values to simplify response selection.

You can access Element Manager directly through a Web browser or Telephony Manager 3.1. The TM navigator includes integrated links to each network system and their respective instances of Element Manager.

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 Teletype (TTY) or PC to the card serial port or Telnet through the ELAN or TLAN network interface IP address.

For more information about the CLI commands, see “IP Line CLI

commands” (page 423).

Overlays

For information about the overlays, see Software Input Output Administration (NN43001-611).

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Features

Contents

This section contains the following topics:

•

• “Active Call Failover for IP Phones” (page 61)

•

• “Enhanced UNIStim Firmware Download for IP Phones” (page 84)

•

• “NAT Traversal feature” (page 119)

•

• “IP Call Recording” (page 137)

•

“Introduction” (page 19)

“DSP peg counter for CS 1000E systems” (page 84)

“Firmware download using UNIStim FTP” (page 111)

“Personal Directory, Callers List, and Redial List” (page 137)

“pbxLink connection failure detection” (page 146) “LD 117 STAT SERV” (page 147)

•

“IP Phone support” (page 151)

•

“Corporate Directory” (page 136)

•

“Element Manager support” (page 165)

•

“Call Statistics collection” (page 166)

•

“Programmable line/DN feature keys (self-labeled)” (page 175)

• “Private Zone configuration” (page 175)

• “Run-time configuration changes” (page 178)

• “Network wide Virtual Office” (page 180)

• “Branch Office and Media Gateway 1000B” (page 183)

• “802.1Q support” (page 184)

• “Data Path Capture tool” (page 188)

• “IP Phone firmware” (page 188)

• “Graceful Disable” (page 188)

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• “Hardware watchdog timer” (page 190)

• “Codecs” (page 191)

• “Set type checking and blocking” (page 191)

•

Introduction

Table 11 "IP Line feature support" (page 52) outlines the IP Line features

available for CS 1000systems with CS 1000 Release 5.0 software.

Table 11 IP Line feature support

“Enhanced Redundancy for IP Line nodes” (page 193)

Feature

Support for Media Card Yes Yes Support for Element Manager Yes Yes Support for Signaling Server Yes Yes Support for the following IP Phones:

•

IP Phone 2001

CS 1000M CS 1000E

Yes Yes

• IP Phone 2002

• IP Phone 2004

• IP Phone 2007

• IP Audio Conference Phone 2033

• IP Phone 1110

• IP Phone 1120E

• IP Phone 1140E

•

IP Phone 1150E

•

WLAN Handset 2210

• WLAN Handset 2211

• WLAN Handset 2212

• WLAN Handset 6120

• WLAN Handset 6140

Node level patching is not provided by TM 3.1. The patching CLI command of the Media Card 32-port card, MC 32S card, and ITG-Pentium 24-port line card can be used.

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Table 11 IP Line feature support (cont’d.)

Introduction 53

Feature

Support for the following software clients:

CS 1000M CS 1000E

Yes Yes

• IP Softphone 2050

•

Mobile Voice Client (MVC) 2050 Support for the IP Phone Key Expansion Module (KEM) Yes Yes Support for the Expansion Module for IP Phone 1100 Series

(Expansion Module) Live Dialpad Yes Yes Unicode support Yes Yes IP Client cookies Yes Yes New IP Phone Types Yes Yes Active Call Failover Yes Yes DSP peg counter for the CS 1000E No Yes Enhanced UNIStim firmware downloads for IP Phones Yes Yes Support for external server applications Yes Yes Enhanced VLAN support on Phase II IP Phones; support for

Voice VLAN hardware filter providing enhanced traffic control on IP Phone and PC port

Yes Yes

Network Address Translation (NAT) Traversal Yes Yes Personal Directory, Callers List, and Redial List with

password protection UNIStim File Transfer Protocol (UFTP) for IP Phone firmware

downloads IP Call Recording Yes Yes pbxLink connection failure detection Yes Yes Dynamic Loss Plan Yes Yes Network-wide Virtual Office Yes Yes Patching Partial Yes

802.1Q support Yes Yes Corporate Directory Yes Yes

Node level patching is not provided by TM 3.1. The patching CLI command of the Media Card 32-port card, MC 32S card, and ITG-Pentium 24-port line card can be used.

Yes Yes

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Table 11 IP Line feature support (cont’d.)

Feature

Data Path Capture tool Yes Yes Self-labeled line/programmable feature keys Yes Yes Private Zone Yes Yes Graceful TPS Disable Yes Yes Run-time download Yes Yes Watchdog Timer Yes Yes Password Guessing Protection Yes Yes Ringer and buzzer volume adjustment Yes Yes Set-based installation Yes (Small

Maintenance Audit enhancement Yes Yes Multilanguage support Yes Yes Enhanced Redundancy for IP Line nodes Yes Yes IP Softphone 2050user-selectable codec (not applicable to

MVC 2050 as it only supports G.711 codec)

Node level patching is not provided by TM 3.1. The patching CLI command of the Media Card 32-port card, MC 32S card, and ITG-Pentium 24-port line card can be used.

CS 1000M CS 1000E

Yes

Systems only)

Yes Yes

Live Dialpad

IP Line provides support for the Live Dialpad feature. Live Dialpad activates the primary line/DN key when the user makes a call by pressing the keys on the dialpad without lifting the handset, pressing a line/DN key or the handsfree key.

The Live Dialpad feature is supported on the following IP Phones:

• IP Phone 2001

• IP Phone 2002

• IP Phone 2004

• IP Phone 2007

• IP Audio Conference Phone 2033

• IP Softphone 2050

• WLAN Handset 2210/2211/2212/6120/6140

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• IP Phone 1110

• IP Phone 1120E

• IP Phone 1140E

•

IP Phone 1150E

Live Dialpad is enabled and disabled in the Telephone Options menu on the IP Phone. Feature processing is performed on the LTPS. The on or off state for the Live Dialpad feature is stored on the Call Server.

Diagnostics

Output of vxWorksShell command e2dsetShow() contains a state of the Live Dialpad feature.

Unicode support

IP Line provides Unicode capabilities on the IP Phone 2007, IP Phone 1110, IP Phone 1120E, IP Phone 1140E, and IP Phone 1150E. Using the Unicode feature the Call Server can easily display multilingual text on the IP Phones for the following languages:

Unicode support 55

•

Japanese

• Greek

•

Traditional Chinese

•

Simplified Chinese

•

Arabic

•

Korean

•

Hebrew

These languages are not available on IP Phones without Unicode capabilities. Japanese is available in Katakana version.

If an IP Phone without Unicode support registers to a TN with Unicode-only language configured, the IP Phone falls back to English. The information stored on the Call Server is not changed unless the user explicitly changes the language.

Personal Directory, Callers List, Redial List, and user-defined feature key labels support Unicode. Corporate Directory and Calling Party Name Display (CPND) do not support Unicode.

Pop-up and USB keyboard support

Only a limited subset of Unicode characters can be input using the dialpad. The Special Characters Input Screen includes the character set which is used by the current language. For languages with a large amount of

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56 Features

characters, such as Traditional Chinese, Japanese, and Korean, localized input is not supported. The Pop-up and USB keyboard supports English input only.

Language synchronization

Language synchronization is handled strictly through UNIStim messaging. If the IP Phone receives an Assign IT Language from the Call Server, the IP Phone changes its local prompts to match the specified language. If a user sets the IP Phone language using the Telephone Options menu, the IP Phone sends a UNIStim Assign NI Language message to the Call Server. The Call Server then synchronizes its language with the IP Phone. The Call Server can use the Query IT Language message at any time to determine the current language of the IP Phone. If there is no default or programmed mapping for a language specified by the Call Server, then the IP Phone uses the same language that is previously used. It is up to the Call Server to only specify languages for which the IP Phone has fonts and font mappings. The Call Server retrieves a list of language codes using the Display Manager Query Supported IT Languages.

Virtual Office interaction

Virtual Office log on can be performed from an IP Phone without Unicode support to an IP Phone with Unicode entries in the Personal Directory. <Unicode name> is displayed instead of the name of the entry. The entry cannot be edited, but it can be deleted.

IP Client cookies

IP Client cookies provide a transparent transfer of data from the Call Server to third-party applications, for example, Citrix AG. The cookies are a set of UTF-8 variable names and values, which are duplicated and synchronized between the LTPS and the IP Phone. IP Line uses public cookies which are visible to both the IP Phone and third-party applications. IP Client cookies are not supported on Nortel Phase I IP Phones and third-party IP Phones, such as WLAN Handset 2210/2211/2212/6120/6140, and IP Audio Conference Phone 2033.

Table 12 Cookie definitions

Cookie name Description

PrimeDN A string of digits containing the current primary DN of the IP Phone. AgentPosition A string of digits containing the current agent position of the IP Phone. This

string is empty if the agent is not logged on.

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IP Client cookies 57

Table 12 Cookie definitions (cont’d.)

CallState A UTF-8 string indicating the current call processing state of the IP Phone.

The following are possible values:

•

BUSY—an active call is established

• IDLE—no active calls (there can be calls on hold)

• RINGING—IP Phone is ringing

CustNo A string of digits indicating the IP Phone customer number. Zone A string of digits indicating the IP Phone zone. VPNI A string of digits containing the Virtual Private Network Identifier configured

for the IP Phone.

TN UFT-8 string containing the TN currently associated with the IP Phone in

hexadecimal format. The maximum number of TNs is FFFF.

Although cookie names and values are UTF-8 strings, it is not necessary for the IP Phone to support Unicode.

Output of the e2dsetShow() command shows the contents of the IP Phone cookie storage, as well as all display lines, soft keys, feature keys (including feature keys on KEM and Expansion Module) and programmable line/DN keys associated with the IP Phone. This command is only available from the VxWorksShell.

Figure 6 dsetShow

e2dsetShow ()

The e2dsetShow () function expects a pointer to a DSET emulator, which can be obtained by dsetShow () function.

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New IP Phone Types

IP Line introduces support for new TN types. This feature provides the following functionality:

• unique TN types for each IP Phone

• special emulation mode for IP Phones that are not known to the TPS

• automatic and manual IP Phone TN type conversion

•

existing IP Phone TN types are renamed to match the brand name of the IP Phone

• enhanced Model Names support is accessed from LD 20 and TM 3.1

Unique TN Types for existing IP Phones

The names of existing IP Phone TN types are updated with a naming convention to match the brand name of the IP Phone.

Table 13 CS 1000 Release 5.0 TN Type naming convention

IP Phone model name CS 1000 Release 4.5

TN_TYPE

IP Phone 2001 i2001 2001P2 IP Phone 2002 Phase I i2002 2002P1 IP Phone 2002 Phase II i2002 2002P2 IP Phone 2004 Phase 0/I i2004 2004P1 IP Phone 2004 Phase II i2004 2004P2 IP Audio Conference Phone

2033 IP Softphone 2050 i2050 2050PC Mobile Voice Client 2050 i2050 2050MC WLAN Handset 2210 i2004 2210 WLAN Handset 2211 i2004 2211 WLAN Handset 2212 i2004 2212 WLAN Handset 6120 WLAN Handset 6140 IP Phone 2007 i2004 2007

i2001 2033

Not applicable Not applicable

CS 1000 Release 5.0 and later TN_TYPE

6120 6140

IP Phone 1110 Not applicable 1110 IP Phone 1120E i2002 1120

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Table 13 CS 1000 Release 5.0 TN Type naming convention (cont’d.)

IP Phone 1140E i2004 1140 IP Phone 1150E IPACD 1150

Note: WLAN Handsets 6120/6140 did not have any TYPE naming

conventions in CS1000 Release 4.5, but from CS1000 Relase 5.0 onwards it is given a TN_TYPE naming convention as TYPE = 6120/6140. WLAN sets 2210/2211/2212 were programmed as TYPE i2004 in CS1000 Release 4.5, but from Release 5.0 onwards its convention for TYPE = 2210/2211/2212.

Emulation Mode

During IP Phone registration, the LTPS determines the IP Phone TN Type (TN_TYPE) by looking up its User Interface capabilities (UI_TYPE) and Firmware ID (FW_ID) in a mapping table. The mapping table is used to map the IP Phone UI_TYPE and FW_ID with TN_TYPE. If an IP Phone has a known UI_TYPE but an unknown UI_TYPE and FW_ID combination, the IP Phone registers in Emulation Mode.

New IP Phone Types 59

Use the isetShow command or LD 20 to list the IP Phones registered in Emulation Mode.

Automatic IP Phone TN conversion (Flexible Registration)

Flexible Registration Class of Service (CLS) for all IP Phones is configured in LD 11. Flexible Registration CLS can be set to one of the following values:

• FRA-Flexible Registration Allowed (default)

•

FRU-Flexible Registration on Upgrade

•

FRD-Flexible Registration Denied

Use LD 81 to list the IP Phone TN which have Flexible Registration Allowed (FRA), Flexible Registration on Upgrade (FRU), and Flexible Registration Denied (FRD) CLS.

When the LTPS attempts to register an IP Phone with the Call Server, the following occurs:

1. If the TN has FRD CLS, the Call Server checks the IP Phone type

against the TN type. Registration is rejected if the types do not match.

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Furthermore, the Call Server checks the Emulation Flag and blocks registration in the Emulation Mode.

If the TN has FRA CLS, the Call Server checks the IP Phone type against the TN type. If the types are compatible, the TN is converted, and the IP Phone registers.

3. If the TN has FRU CLS, the Call Server checks the IP Phone type

against the VTN type. If the types are compatible, the TN is converted and the IP Phone registers. After the TN is converted, the Flexible Registration CLS is set to FRD. The Call Server checks the Emulation Flag and blocks registration in the Emulation Mode.

Manual IP Phone TN conversion

Manual IP Phone TN conversion lowers the administrative effort required to replace an IP Phone with another model while preserving the IP Phone features. Use LD 11 to convert an IP Phone TN to another IP Phone TN while preserving the IP Phone features.

Table 14 LD 11 IP Phone interface commands

Prompt Response Description

REQ CHGTYP

TYPE 2004P1, 2004P2, 2002P1,

2002P2, 2001P2, 2050PC, 2050MC, 2033, 2210, 2211, 2212,6120, 6140, 2007, 1120, 1140,1150

TN lscu

NEWTYP 2004P1, 2004P2, 2002P1,

2002P2, 2001P2, 2050PC, 2050MC, 2033, 2210, 2211, 2212, 6120, 6140, 2007, 1110, 1120, 1140,1150

PROCEED YES

Change the IP Phone TN type Type of TN block to convert

For Large Systems TN_TYPE to convert

The Call Server lists the features that are lost if the administrator proceeds.

Perform or reject the IP Phone TN conversion

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Active Call Failover for IP Phones

The Active Call Failover (ACF) for IP Phones feature allows active IP calls to survive the following failures:

• IP/IP calls and IP/TDM calls survive signaling path TLAN subnet

failures.

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.

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

— Media Gateway 1000B for a branch office configuration —

Geographic Redundancy Secondary Call Server. The feature addresses the Primary Call Server failures.

Active Call Failover for IP Phones 61

IP Phone to IP Phone calls survive the Call Server failures listed above.

ATTENTION

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, CP PIV, and CP-PM:

— 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 re-registers, the call data is rebuilt if the Call Server does not know about the call, using the internal IP Phone information.

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Minimum requirements

ACF mode

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.

The ACF feature for IP Phones has the following minimum requirements:

• Call Server must be running CS 1000 Release 4.5, or later 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.)

The ACF feature for IP Phones enables an IP Phone to re-register 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

All other elements (the self-labeled line/programmable feature keys, context-sensitive 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 can exist where it takes a long time to fix a failure and no failover Call Server is available. During this time, if the user released the call by pressing the Release key or hanging up the telephone, the call-associated resources are not used. The call-associated resources still exist on the Call Server because they are not released. To prevent this, the 10-minute Call Server ACF timer is introduced for each call. The

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ACF scenarios

Table 15 ACF behaviors

Active Call Failover for IP Phones 63

timer prevents call processing-related resources from being unnecessarily used when an IP Phone that had an active call unregisters and never re-registers.

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.

Table 15 "ACF behaviors" (page 63) describes ACF behavior in different

scenarios.

Scenario

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 re-register.

• 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 re-register with the

node again.

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 re-register.

Result

The call is not lost as the IP Phones re-register.

In this scenario, the call exists on the Call Server during the failover time and has the following transitions: UNREGISTERED ->HALF-REGISTERED -> NO ACF

The call is not lost as the IP Phones re-register.

The scenario is similar to the TLAN subnet failure, but the ACF call transition on the Call Server is instantaneous, because Offline events are generated in a group as the ELAN subnet goes down.

•

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 re-register with the

node again.

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Table 15 ACF behaviors (cont’d.)

Scenario 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.

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

•

Branch IP Phones A and B register with the

Media Gateway 1000B and are redirected to

the main office.

•

IP Phones A and B registered with the main

office establish a call.

Result

The call is not lost.

The call is rebuilt after the warm restart and has the following transitions: UNREGISTERED->HALF REGISTERED->N O ACF. The transition is almost instantaneous because the Online messages are sent in a group as a response to the Sync Request.

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.

•

A serious main office failure occurs. The active

Branch IP Phones cannot re-register with the

main office and re-register with the branch

office in local mode. IP Phone A re-registers in

local mode first.

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Table 15 ACF behaviors (cont’d.)

Active Call Failover for IP Phones 65

Scenario Main office failure for branch office (scenario

•

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 re-register

with the main office and they re-register with

the Branch office in local mode. IP Phone A

re-registers in local mode first.

Primary Call Server failure (WAN geographical ly 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 re-registered with the

secondary site. IP Phone A re-registers first.

Result

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:

1. Site 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.

2. IP 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 Call Server does not have the RLM entries for the re-registering 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 15 ACF behaviors (cont’d.)

Scenario Virtual Office logon 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 re-register with the system when the TLAN subnet comes back up. IP Phone A re-registers first and then IP Phone B.

Virtual Office logon 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 re-register with the

system when the TLAN comes back up. IP Phone A re-registers first and then IP Phone B.

Result

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

Virtual Office logon 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.

Network TLAN subnet failure

• IP Phone A has an IP Peer call with a remote

user over a virtual trunk.

• IP Phone A TLAN subnet connection fails.

• Active IP Phone A re-registers with the Call

Server when the TLAN subnet comes back up.

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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. See "Virtual Office login failure (scenario 1)"

(page 66) and "Virtual Office login failure (scenario 2)" (page 66) .

The call is not lost.

The scenario is the same as if the far end were a local IP Phone. See "TLAN subnet

failure" (page 63) .

Table 15 ACF behaviors (cont’d.)

Active Call Failover for IP Phones 67

Scenario 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 re-registers with the Call

Server as the TLAN subnet comes back up.

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 re-registers with the Call

Server as the TLAN subnet comes back up.

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.

Result

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" (page 64) .

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)" (page 65) . 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 re-register 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 TLAN subnet connection fails.

•

Active IP Phone A re-registers with the Call Server as the TLAN subnet comes back up.

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The call is not lost.

The scenario is the same as "TLAN subnet

failure" (page 63) and "Network TLAN subnet failure" (page 66) . The call has the following

transitions: NO ACF -> HALF REGISTERED

-> UNREGISTERED.

68 Features

Table 15 ACF behaviors (cont’d.)

Scenario 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 re-registers with the Call

Server as the TLAN subnet comes back up.

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 re-registers 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 has IP Phones with a mixture of firmware versions registered with it. The firmware can be downloaded later, after the idle IP Phone registers again or can be downloaded manually using appropriate CLI commands.

Result

The call is not lost.

The scenario is same as if the far end were a local IP Phone. See "Call Server warm

restart" (page 64) .

The call is lost as the Call Server comes back up.

WLAN Handsets 2210/2211/2212/6120/6140

The Wireless LAN (WLAN) Handsets 2210/2211/2212/6120/6140 support Active Call Failover in the same manner as Phase II IP Phones if their firmware supports UNIStim 2.9.

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.

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Active Call Failover for IP Phones 69

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.

The zone bandwidth usage in the zone table remains zero for all IP Peer calls on this side; zone bandwidth usage is cleared for all calls as the Call Server comes back up after the warm start. In this case, Network Bandwidth Management information is lost and the Call Server is unable to restore the correct zone bandwidth usage for IP Peer calls.

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.

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.

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” (page 68).

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NAT devices

The ACF feature cannot handle the case of a NAT device changing the media path mapping between the IP Phone 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 re-registers 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) re-registers, a firmware download can occur, if needed. If that IP Phone actually had calls 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 kb/s

• 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.

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Active Call Failover for IP Phones 71

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 re-registers with its home TN, the active call is released as IP Phone A re-registers.

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 re-registers 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

This section shows the ACF feature interactions with Virtual Office and Branch Office.

Branch Office

When the first failed IP Phone re-registers in local mode, the branch office Call Server looks up 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 re-registers in local mode. Even if the main office IP Phone survives the

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72 Features

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.

Virtual Office

It is possible that active IP Phone A, that was logged on to IP Phone B before the failure, cannot re-register with the Call Server, because IP Phone C performed a Virtual Office logon and uses IP Phone A 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 SRG 50 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 re-registers. 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 re-registers. 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 submenu the user is currently in) is lost when the failover occurs and the IP Phone re-registers.

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), a Converged Desktop call is maintained when the involved IP Phone re-registers to the system. If the Call Server loses the call information or the SIP Gateway Signaling Server reboots, the Converged Desktop call is impacted.

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Active Call Failover for IP Phones 73

A Converged Desktop consists of a telephone and multimedia PC Client (PCC) software.

The following are scenario examples. Example 1: The IP Phone TLAN subnet fails and the IP Phone

re-registers 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 re-registers. The multimedia applications still work: the presence is updated on PCC after the telephone re-registers.

If the unregistered converged IP Phone releases the call during the TLAN subnet failure, the Presence status is updated on PCC as the idle converged IP Phone re-registers.

Example 2: The IP Phone Signaling Server fails and the IP Phone re-registers 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 ELAN subnet fails and the IP Phone re-registers 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 re-registers 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 AML 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.

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74 Features

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:

•

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, because the ACD feature is implemented on the Call Server.

CS 1000 base features

No feature works when the active IP Phone is disconnected and trying to re-register with the Call Server. All the features can be used in the context of the failover call after the IP Phone re-registers (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 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 re-registers. 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 re-register with the Call Server. When the Call Server comes back up and the IP Phones

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Active Call Failover for IP Phones 75

re-register, 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, because 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 re-registers. Only the LTPS display is lost when the IP Phone re-registers.

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 re-register with the server.

The TLAN comes up again and the IP Phone re-registers. 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 DN key.

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 re-registers. 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 re-registers

• the active IP Phone re-registers and then the call is released

The records include the failover time as well. This means that the user can 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” (page 69) and

Table 15 "ACF behaviors" (page 63).

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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 IP Phones and CallPilot is dropped due to INI, which can be initiated by, for example, cold start, warm start, or ungraceful switchover.

Note that during any failure, user input is not passed to CallPilot. The user must resume entering responses after the IP Phone re-registers.

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

Contact Center Management Server

The feature interacts with the Contact Center Management Server (CCMS) 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” (page 74).

— If the active unregistered ACD agent changes the call state during

the failure period (for example, releases the call), the status

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message is sent to the Symposium and CTI applications as the idle agent re-registers with the system.

•

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 re-registers 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 feature for IP Phones requires no installation. It is active by default on any CS 1000 system running the CS 1000 Release 4.5, or later software.

Active Call Failover for IP Phones 77

On a system running CS 1000 Release 4.5, or later software, every node running the CS 1000 Release 4.5 or later LTPS software has the ACF feature enabled for the IP Phones that register to it.

Conﬁgurable RUDP Timeout and Retries Count

When a network failure occurs and the IP Phone 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 ms). 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. Now the time-out and number of retries can be configured in the OAM and PDT shells of the Signaling Server. See Table 16 "RUDP Timeout and Retries Count commands"

(page 78).

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Table 16 RUDP Timeout and Retries Count commands

Command

usiSetPhoneRudpRetries

usiGetPhoneRudpRetries Display the RUDP Retries Count maximum for

usiSetPhoneRudpTimeout

usiGetPhoneRudpTimeout Display the RUDP Timeout value (in ms) for IP

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 from being reset due to significant network delay.

Description

Configure the RUDP Retries Count maximum for IP Phones

1 – (10) – 20

IP Phones Configure the RUDP Timeout value (in ms) for

IP Phones

50 – (500) – 1000 in increments of 50 milliseconds

Phones

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:

……

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.

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Overlay and command modiﬁcations

Because call failover is an exceptional situation, ACF information is output only if it exists.

Status deﬁnitions

UNREG

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 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.

Active Call Failover for IP Phones 79

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.

ATTENTION

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.

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LD 32 STAT command

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.

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

• REB for rebuilt calls

•

PREB for partially-rebuilt calls

where <timer> is:

•

an integer value if the timer exists for the call

•

N/A if there is no Call Server ACF timer attached

See Figure 7 "LD 32 STAT output with ACF example" (page 80).

Figure 7 LD 32 STAT output with ACF example

LD 80 TRAC command

If ACF information exists for the requested IP Phone, it is output as follows:

ACF STATUS <status> TMR <timer> ORIG <orig_state> TERM <term_state>

where <status> is:

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• UNREG for unregistered calls

• HREG for half-registered calls

• REB for rebuilt calls

•

PREB for partially-rebuilt calls

where <timer> is:

•

an integer value if the timer exists for the call

• N/A if there is no Call Server ACF timer attached

ORIG <orig_state> and TERM <term_state> can be REGISTERED or UNREGISTERED.

Figure 8 "LD 80 TRAC with ACF example" (page 81) is a sample output for

IP Phones involved in UNREGISTERED and PARTIAL-REBUILT calls.

Figure 8 LD 80 TRAC with ACF example

LD 117 STIP ACF command

A sub-command ACF is added to the existing LD 117 STIP command.

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Table 17 LD 117 STIP ACF command

Command

STIP ACF <status>

Description

Displays the Active Call Failover (ACF) information.

<status> – optional parameter. Specifies the status to be output. Outputs all IP Phones involved in the following types of calls:

•

UNREG - UNREGISTERED calls

•

HREG - HALF-REGISTERED calls

• REB - REBUILT calls

•

HREB - HALF-REBUILT calls

•

PREB - PARTIAL-REBUILT calls

•

ALL – all types of ACF calls

If no status parameter is entered, all types of ACF calls are output.

Output

The output is similar to the existing LD 117 STIP output, with the addition of a column titled ACF STATUS. If the call is in an inactive state, the value of the Call Server ACF timer follows that status, separated by a colon (:).

LD 117 STIP ACF in Element Manager

Support for the STIP ACF command in LD 117 is provided by Element Manager. Click System > Maintenance . Select LD 117 - Ethernet and Alarm Management. Select Ethernet Diagnostics. The Ethernet Diagnostics window appears.

Figure 9 "LD 117 STIP ACF in Element Manager" (page 83) illustrates the

placement of the STIP ACF command with the other STIP commands.

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Figure 9 LD 117 STIP ACF in Element Manager

A list of command parameters are made available after the STIP ACF command is selected. The ALL command parameter is displayed as the default.

Click the Submit button after selecting one of these available parameters to execute the command. The output from the command is displayed in the text box located in the lower portion of the Web page.

Online Help describes the various parameters available for the STIP ACF command.

isetShow command

If the ACF status exists for the requested IP Phone, it is provided in the State field of the isetShow command output. The ACF status is separated from the state by dash (-). The ACF status is any value described in the LD 80 output. The Call Server ACF timer value is not provided in the output.

See Figure 10 "isetShow command output with ACF example" (page 83).

Figure 10 isetShow command output with ACF example

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DSP peg counter for CS 1000E systems

The conversion of TDM voice to IP packets is performed by Digital Signaling Processor (DSP) resources residing on a Voice Gateway Media Card in the IP Media Gateway (IPMG) of a CS 1000E system. The Voice Gateway Media Cards have a limited number of DSP resources that actually perform the conversion. When all DSP resources are busy, IP-to-TDM calls and TDM-to-TDM calls between different IPMGs are blocked. IP-to-IP calls are not blocked.

The DSP Peg Counter feature provides three counters. The first peg counter provides a count of the number of attempts to allocate a DSP resource on an IPMG. The second provides a count of the number of times calls were blocked on an IPMG due to a lack of DSP resources. If the call failed due to a lack of bandwidth, this is reflected in the third peg counter. The counters are a part of customer traffic measurement in LD 2.

For more information, see Traffic Measurement: Formats and Output

Reference (NN43001-750), and Software Input Output Administration (NN43001-611).

Enhanced UNIStim Firmware Download for IP Phones

Prior to CS 1000 Release 5.0, firmware files were downloaded to the Voice Gateway Media Cards. In CS 1000 Release 5.0, firmware files are stored on and are downloaded from the Signaling Server.

The Enhanced UNIStim Firmware Download for IP Phones provides a method of delivering new firmware for Nortel IP Phones.

Specifically, this feature provides the following functionality:

• Enhanced firmware file header that includes the IT_TYPE and name

string for each IP Phone type. Element Manager and the LTPS can read this information and automatically display the mapping to the administrator.

•

Revised definition of the IP Client IP Phone identification.

• Maintenance Mode for the Signaling Server that allows simultaneous

firmware downloads from the UFTP server. Maintenance Mode (Turbo Mode) is manually initiated by the administrator in which premarked node Signaling Servers utilize all possible resources for processing firmware upgrade jobs.

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• Identification of the registered IP Phones using string names and

providing more detailed identification of IP Phones that register in Emulation Mode.

•

UNIStim IP Phones are allowed to register with an older version of firmware if the UFTP servers are busy, then periodically offers the option to start the firmware upgrade to the IP Phone user.

• Introduction of missing firmware file retrieval to the Branch Office from

the Main Office.

System management commands are provided to collect information about registered IP Phones, their models, and their firmware.

Operating parameters

Enhanced UNIStim Firmware Download feature is supported on the following systems running CS 1000 Release 4.5 or later software.

•

CS 1000M HG

•

CS 1000M SG

•

CS 1000M MG

Enhanced UNIStim Firmware Download for IP Phones 85

• CS 1000E

The Enhanced UNIStim Firmware Download feature has the following operating parameters:

•

It supports only firmware downloads performed by the UFTP server to the UNIStim IP Phones supporting the UFTP download protocol.

•

Enhanced functionality is provided only if the recommended commands are used. For example, use of the VxWorks shell copy command instead of the firmwareFileGet command bypasses the other features and is therefore not supported.

•

Firmware retrieval mechanism described for the Branch Office LTPS retrieves only firmware files it finds missing. It does not compare the list of firmware on the Branch Office LTPS and Main Office LTPS to determine whether the Branch Office has the latest firmware, or perform any automatic compare and update operations. The Branch Office LTPS only receives firmware files when the umsUpgradeAll command was issued on the Main Office LTPS.

Feature interactions

Active Call Failover for IP Phones

The Active Call Failover feature handles cases when an IP Phone registers with an active RTP stream (has a call active at the time of registration). The check of IP Phone firmware is skipped in this case, and the IP Phone registers with the LTPS.

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System view

Table 18 System response

The Active Call Failover scenario is the same as the postponed firmware upgrade scenario described in Table 22 "IP Phone registration and

download scenarios" (page 89). After the call ends, the user is prompted

to start the firmware upgrade. For more information about Active Call Failover for IP Phones, see “Active

Call Failover for IP Phones” (page 61).

IP Phone firmware upgrades

Each IP Phone registering with the LTPS is queried for its firmware ID and IT_TYPE. The system response depends on the results of the query. See

Table 18 "System response" (page 86).

Query result

LTPS software supports the reported IT_TYPE (see Table 19 "Supported IT_TYPES" (page

87)) and the Upgrade Manager has firmware for

the given firmware ID. LTPS software supports the reported IT_TYPE,

but the Upgrade Manager has no firmware for the given firmware ID.

LTPS software does not support the IT_TYPE reported.

The branch office IP Phone is upgraded at the branch office before the IP Phone is redirected to the main office.

Firmware file management

To manage available firmware, the following information is collected about each firmware file on the Signaling Server or Voice Gateway Media Card:

• firmware ID

• firmware version

• applicable IT_TYPE (see Table 19 "Supported IT_TYPES" (page 87))

Response

Registration of the IP Phone continues. The IP Phone firmware upgrade is performed if possible.

Registration of the IP Phone continues with no firmware download.

Registration of the IP Phone is rejected.

If the branch office does not have the necessary firmware file, an attempt is made to download the firmware file from the main office.

• applicable model names

IT_TYPEs

Table 19 "Supported IT_TYPES" (page 87) lists the IT_TYPES supported

by the Upgrade Manager for CS 1000 Release 5.0

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Table 19 Supported IT_TYPES

IT_TYPE IP Phone

0x02

0x03 0x04

IP Phone 2004, IP Phone 2007, WLAN 2210/2211/2212/6120/6140, IP Phone 1140E

IP Phone 2002, IP Phone 1120E IP Phone 2001, IP Audio Conference Phone 2033, IP Phone 1110

Enhanced UNIStim Firmware Download for IP Phones 87

0x20 0x06

IP Softphone 2050, Mobile Voice Client 2050 IP Phone 1150E

Two events trigger data about firmware files to be updated by the LTPS:

LTPS reboot

2. new firmware file upload from either the LTPS Command Line Interface

(CLI) or Element Manager

In the first case, the LTPS explores possible locations of firmware files and collects information about found files in its internal database. In the second case, when a new firmware file is uploaded, the LTPS updates the internal database with information extracted from the file.

Element Manager uses data from the firmware file to provide information about the firmware file and the IP Phones to which it can be downloaded.

Firmware file names

Firmware file names are originally in the format SSFFYxx.bin. See Table

20 "Original firmware file name format" (page 87).

Table 20 Original firmware file name format

Designator Definition Values

SS Site code where firmware

was built

FF Firmware type 02 – Phase 0/1 IP Phone 2004

Y Alpha character A – 0

XX Release number 2-digit decimal integer (for

06 – Calgary 30 – Ottawa

03 – Phase 1 IP Phone 2002 04 – Phase 2 IP Phone 2001/2002/2004

B–1 C–2 D–3 (and so on)

example,.38)

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The files are renamed according to the following rules:

•

Phase 0/1 IP Phone 2004 firmware is renamed to x00.fw

• Phase 1 IP Phone 2002 firmware is renamed to x01.fw

•

All other firmware files are renamed to xFF.fw, where:

—

x emphasizes that FF is a hexadecimal number

— FF is the firmware ID for that file

Table 21 Firmware file naming conventions

x00.fw IP Phone 2004 Phase 0/1 x00.fw IP Phone 2002 Phase 1 x02.fw IP Phone 2004 Phase 2 x02.fw IP Phone 2002 Phase 2 x02.fw IP Phone 2001 Phase 2 x10.fw IP Phone 2033 (Conference Phone) x21.fw IP Phone 2007 Phase 2 x23.fw IP Phone 1110 x24.fw IP Phone 1120E x25.fw IP Phone 1140E x27.fw IP Phone 1150E x2A.fw IP Phone 1200

The xFF.fw format also applies to the firmware file for the Phase 2 IP Phone 2001, IP Phone 2002, and IP Phone 2004. The file was named IPP2SET.fw but is renamed to x02.fw to conform to the naming convention.

Download maximums

The following modifications are available on the Signaling Server to the Upgrade Manager:

• The default number of allowed simultaneous downloads is increased

to 100.

• Maintenance Mode (Turbo Mode) that is manually initiated by the

administrator is available in which premarked node Signaling Servers utilize all possible resources for processing firmware upgrades. The following commands are used to manage the Maintenance Mode:

— uftpTurboMode — uftpTurboModeTimeoutSet — uftpTurboModeShow

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The uftpTurboMode command is used in conjunction with RST FW (hard-resets all IP Phones with specified F/W ID) and RST ZONE (hard-resets all IP Phones) commands in LD 117. For more information about commands in a specified zone, see Table 23 "Maintenance

Mode commands" (page 91).

For more information about Maintenance Mode, see “Maintenance

Mode” (page 90).

Immediate and delayed ﬁrmware downloads

The IP Phones display various messages to indicate the status of IP Phone registration and firmware downloads. Table 22 "IP Phone

registration and download scenarios" (page 89) lists some scenario

examples with the resulting IP Phone displays.

Table 22 IP Phone registration and download scenarios

Scenario

Normal firmware download for known IP Phone type

Postponed firmware upgrade

Result

IP Phone displays message that IP Phone is connecting to the LTPS.

IP Phone displays message that firmware download is initiated.

If download is successful, IP Phone continues with normal registration.

IP Phone displays message that IP Phone is connecting to the LTPS.

IP Phone cannot download firmware. It is allowed to proceed with registration using old firmware.

At the completion of call (if download resources are available), IP Phone displays message Upgrade F/W now?

IP Phone displays Yes and No soft keys to use to select choice. If Yes is selected, firmware download begins. If no choice is made, IP Phone proceeds with firmware download after timer expiration.

If No is selected, IP Phone display returns to idle state. Off-hook dialing, on-hook dialing, and external events such as an incoming call imply a No response.

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Table 22 IP Phone registration and download scenarios (cont’d.)

Scenario

Unknown firmware ID for known IT_TYPE

Unknown IT_TYPE

Branch Office LTPS determines IP Phone requires firmware upgrade

Result

IP Phone displays message that IP Phone is connecting to the LTPS.

No firmware upgrade is performed, but IP Phone is allowed to register.

IP Phone has no display. The IP Phone just resets continuously.

IP Phone registration is not allowed.

Log message is sent to LTPS administrator. IP Phone displays message that firmware download is

initiated. IP Phone is placed into local mode.

Message is displayed until firmware is downloaded. IP Phone upgrade process is initiated.

If firmware download is unsuccessful after 10 retries, IP Phone remains in local mode.

Maintenance Mode

When a Signaling Server is placed into Maintenance Mode, the allowable maximum number of simultaneous firmware downloads increases. Maintenance Mode enables the UFTP server to utilize most of its processing resources to deal with the downloads.

The actual number of simultaneous downloads is determined by measuring the CPU idle time, so each new firmware download session is launched if there are less than 100 download sessions for the Signaling Server already taking place and one of the following is true:

• there are less than five download sessions currently active

• Signaling Server is in regular mode (not in Maintenance Mode) and

its CPU usage is less than 85%

• Signaling Server is in Maintenance Mode and its CPU usage is less

than 100%

The UMS tries to launch a pending download session every 5 seconds.

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ATTENTION

When Maintenance Mode is enabled, call processing signaling could be impacted by the UFTP download processes.

After Maintenance Mode is enabled, it can be exited in several ways:

• manually, by using the uftpTurboMode "stop" command

• automatically, after the Upgrade Manager is idle for MM minutes after

at least one download has been started This prevents a time-out from occurring while the system is being configured and the downloads start. After a download starts, if MM minutes pass with no new firmware upgrade jobs starting, the normal mode of operation resumes. The idle timeout timer is configured using the uftpTurboModeTimeoutSet command.

•

automatically, after expiration of the Maintenance Mode period Active firmware upgrade jobs are not cancelled when the Maintenance Mode exits. No new jobs are added until the number of active jobs is below the default value. Maintenance Mode can be enabled only on the Signaling Server. Maintenance Mode affects only Signaling Servers designated for Maintenance Mode. This allows some Signaling Servers in the node to operate in Maintenance Mode while others do not. The Signaling Server is designated for Maintenance Mode with the uftpTurboMode "on" command. The Maintenance Mode designation is saved and maintained even if the Signaling Server is power-cycled or is rebooted. Call processing for Signaling Servers operating in normal mode is not impacted by the firmware download process. Postponed firmware upgrades are not performed when at least one Signaling Server is in Maintenance Mode.

Table 23 "Maintenance Mode commands" (page 91) lists the commands

used for Maintenance Mode.

Table 23 Maintenance Mode commands

Command

uftpTurboMode <"HH:MM/start/stop/on/off">, <MM> <"show">

Description

Configures Maintenance Mode

"HH:MM" – time to enter Maintenance Mode in 24-hour format

"start" – enter Maintenance Mode immediately

"stop" – stop Maintenance Mode

"on" – allow Signaling Server to enter Maintenance Mode

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Command

uftpTurboModeTimeoutSet <MM>

Description

"off" – do not allow Signaling Server to enter Maintenance Mode

MM – optional parameter that defines the length of time in minutes that Maintenance Mode is to be maintained

"show" – displays the same output as uftpTurboModeSh ow

If no parameter is entered, Upgrade Manager defaults to uftpturboMode "start".

Configures the idle timeout timer for Maintenance Mode

MM – optional parameter that defines the number of minutes the Upgrade Manager waits after the last firmware download job is started before returning the Signaling Server to normal mode If this parameter is configured as 0 (zero), the Upgrade Manager never exits Maintenance Mode unless the umsUpgradeModeSet command is issued with the "stop" parameter.

If no parameter is entered, then the current timeout setting is displayed.

uftpTurboModeShow Displays current status of Maintenance Mode.

The following is an example of output when Maintenance Mode is to start at 11 p.m.

oam> uftpTurboMode "23:00" oam> 28/07/04 08:23:56 LOG0006 shell: F/W upgrade Maintenance Mode will start after 52564 seconds

Call Server commands

LD 20

A response ISET is introduced to the LD 20 TYPE prompt. When ISET is entered, the prompt MODEL_NAME is displayed. The MODEL_NAME prompt allows a user to specify the Short Model Name mnemonic for filtering the output of TN blocks. If only the ISET response is used, printed TN blocks contain the long IP Phone Model Name in the output.

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Table 24 LD 20 Listing or printing TN blocks of specified IP Phone model

Prompt Response Description

REQ

LTN

PRT

List TN blocks.

Print TN blocks. TYPE ISET Enable filtering by IP Phone model name.

... ...

MODEL_NAME

xxxxxx

IP Phone model

For example, 2004P2

... ...

The following is an example of the input and output.

>ld 20 REQ: PRT TYPE: ISET TN CUST TEN DATE PAGE DES MODEL_NAME: 2004P2 KEM_RANGE IP_PHONE_MODEL: IP PHONE 2004 PHASE2 DES FAKE TN 064 0 00 00 VIRTUAL TYPE 2004P2 CDEN 8D CUST 0 ZONE 000 FDN TGAR 1 LDN NONCOS 0 SGRP 0RNPG 0 SCI 0 SSU XLST SCPW 6400 SFLT NOCAC_CIS 3 CAC_MFC 0 CLS CTD FBD WTA LPR MTD FND HTD ADD HFD CRPD MWD LMPN RMMD SMWD AAD IMD XHD IRD NID OLD VCE DRG1 POD DSX VMD CMSD SLKD CCSD SWD LND CNDD CFTD SFD MRD DDV CNID CDCA MSID DAPA BFED RCBD ICDD CDMD LLCN MCTD CLBD AUTU

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GPUD DPUD DNDD CFXD ARHD CLTD ASCD CPFA CPTA HSPD ABDD CFHD FICD NAID DNAA RDLA BUZZ AGRD MOAD UDI RCC HBTD AHD IPND DDGA NAMA MIND PRSD NRWD NRCD NROD DRDD EXR0 USMD USRD ULAD CCBD RTDD RBDD RBHD PGND FLXD FTTC DNDY DNO3 MCBN FDSD NOVD VOLA VOUA CDMR CPND_LANG ENG HUNT PLEV 02 CSDN AST IAPG 0 AACS NO ITNA NO DGRP MLWU_LANG 0 DNDR 0 KEY 00 SCR 640 0 MARP ANIE 0 01 02 03 04 05 06 07 08 09 10 11 12 13 14 15 16 17 TRN 18 AO6 19 CFW 16 20 RGA 21 PRK 22 RNP23 24 PRS 25 CHG 26 CPN 27 28 29 30 31 DATE 8 JUL 2004 NACT

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Table 25 LD 117 commands

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LD 117

LD 117 commands are as follows:

•

STIP FW <XX> <A> <BB> <FF> – list IP Phones with specified firmware ID and, optionally, firmware version. If no parameters are entered, output is a list of available model names.

• STIP MODL <MMMM> – list IP Phones of specified model name

•

RST ZONE <ZoneNumber> <START/STOP> <HH:MM> – reset IP Phones in specified zone

• RST FW <FWID> <START/STOP> <HH:MM> – reset IP Phones with

specified F/W ID

See Table 25 "LD 117 commands" (page 95).

Command

STIP FW <XX> <A> <BB> <FF>

Description

Displays information from the Resource Locator Module (RLM) for IP Phones with specified firmware ID and running specified firmware version.

<XX> – firmware ID

<A> – major version designator

<BB> – minor version designator

<FF> – filter to apply on firmware version; can be one of the following:

= – equal to

~ – not equal to

< – less than

> – greater than

Only the XX parameter is required.

STIP FW <XX> <A> <BB> is equivalent to STIP FW <XX> <A> <BB> EQ.

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Command

STIP MODL <MMMM>

RST ZONE <ZoneNumber>

RST ZONE <ZoneNumber> <START/STOP> <HH:MM>

Description

STIP FW <XX> <A> lists all registered IP Phones with firmware ID equal to <XX> and major version designator equal to <A>.

STIP FW <XX> lists all registered IP Phones with firmware ID equal to <XX>.

Displays information from the RLM for all IP Phones of the specified model, where:

• MMMM = IP Phone model

If the <MMMM> parameter is omitted, a table of existing model names and associated mnemonics is displayed.

Immediately hard-resets all IP Phones, where:

•

ZoneNumber = zone number

Schedule or cancel hard-resets of all IP Phones in specified zone.

<ZoneNumber> – zone number in which to reset IP Phones

RST FW <FWID> <START/STOP> <HH:MM>

START/STOP – IP Phones reset, where:

•

START – configures reset time schedule

•

STOP – cancels scheduled reset

If START is specified and the last parameter is omitted, then IP Phones are reset immediately.

<HH:MM> – hour and minute when IP Phones are to be reset

With only the first parameter, or no parameters, the schedule of IP Phones resets is printed.

Hard-resets all IP Phones with specified firmware ID.

<F/W ID> – firmware ID of IP Phones that should be reset

<START/STOP> – schedules or cancels IP Phones hard-reset. If START is specified and the last parameter is omitted, then IP Phones are reset immediately.

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Command

LTPS CLI commands

LTPS CLI commands are as follows:

•

• uftpAutoUpgradeTimeoutSet

•

See Table 26 "LTPS CLI commands" (page 97).

Table 26 LTPS CLI commands

Command

Description

<HH:MM> – hour and minute when IP Phones should be reset

With only the first parameter, or with no parameters specified, the schedule of IP Phones resets is printed.

firmwareFileGet

isetFWShow isetFWGet

Description

firmwareFileGet <"ServerIP">, <"UserID">, <"Password">, <"/path/to/file">, <"file name">

Initiates a firmware download from a specified FTP server. After the download is completed, the downloaded file is checked for Enhanced Header (or proper naming). If the file is considered a valid firmware file, the UMS database is updated accordingly.

ServerIP – FTP server IP address from where the firmware is retrieved

UserID, Password – credentials for logging on to the FTP server

/path/to/file – absolute or relative path to the firmware file (does not include the file name itself)

file name – name of the firmware file on the FTP server

Use the firmwareFileGet command instead of firmwareFileGetI2004, firmwareFileGetI2002, and firmwareFileGetIPP2.

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Table 26 LTPS CLI commands (cont’d.)

Command

uftpAutoUpgradeTimeoutSet <MM>

isetFWShow Displays the status of IP Phones firmware. isetFWGet <filter>

Description

Configures the length of time the IP Phone waits for a user response after the "Upgrade F/W now?" message is displayed before automatically beginning the firmware upgrade.

MM – user response timeout in minutes. A value of 0 (zero) means "Print current settings".

If no parameter is entered, the current value is printed.

Filters the output of the isetFWShow command by one of that command output field names.

Field names include:

• IP

• Model Name

•

Type

• FWID

• Supported

•

FWVsn

• UNIStim

•

firmwareFileGet example

pdt>firmwareFileGet "192.168.0.1","admin1","0000 ","/u/fw","0604D45.BIN"

firmwareFilePut example

pdt>firmwareFilePut "192.168.0.1","admin1","0000 ","/u/fw","0604D45.BIN"

uftpAutoUpgradeTimeoutSet output example

pdt> uftpAutoUpgradeTimeoutSet 4 pdt> 25/08/04 06:22:23 LOG0006 shell: New value of auto F/W upgrade timeout is 240 seconds. pdt> uftpAutoUpgradeTimeoutSet pdt> 25/08/04 06:22:23 LOG0006 shell: Current value of auto F/W upgrade timeout is 240 seconds.

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isetFWShow output example

oam> isetFWShow Set Information

--------------IP Address Model ItType FWID Supported FWVsn UNIStimVsn TN

------------------ ------------------- ---------- -------

-------------- ------------ ---------------- ---------

192.168.29.56 Polycom 2033 2033 0x10 No A.10 2.9 064-00 Total sets = 1 oam>

isetFWGet output example

oam> isetFWGet "FWID==0x10" Set Information

--------------IP Address Model ItType FWID Supported FWVsn UNIStimVsn TN

------------------ ------------------- ---------- -------

-------------- ------------ ---------------- ---------

192.168.29.56 Polycom 2033 2033 0x10 No A.10 2.9 064-00 Total sets = 1 oam>

Modified LTPS CLI commands

The output of the following commands has been changed to print IP Phone model name (long or short), firmware ID, firmware version, and so on:

• isetShow

•

uftpShow

•

umsPolicyShow

•

isetGet

Short model name example is "2004P2". Long model name example is "IP Phone 2004 Phase 2".

isetShow output example

The output has been modified to display the IP Phone Model Name and firmware version in ABB format.

oam> isetShow Set Information

--------------IP Address NAT Model ItType RegType State Up Time Set-TN Regd-TN HWID FWVsn UNIStimVsn SrcPort DstPort

------------------ ---- ------------------- ---------- -------

-------- -------------- ------------ ------------

-------------------------- ------- ---------- ------- -------

192.168.29.56 IP Phone 2004 2004P2 Regular online 0 00:00:32 064-00 064-00 18-006038ddc6b6-6600 B.65 2.8 5100 5000

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Total sets = 1 oam>

uftpShow output example

The output has been modified to display the IP Phone Model Name, firmware ID, and firmware version in ABB format.

oam> uftpShow

------------ UFTP Server Configuration -------------

UFTP Server IP address.......... 192.168.29.42 [port: 5105]

Concurrent downloading limit.... 15 sets

Total firmware = 5 FW ID FWVsn Model PolicyName file name

---------- ---------- -------------------- -------------

---------------- 0x00 B.65 IP Phone 2004 DEFAULT /ums/i2004.fw 0x00 B.65 IP Phone 2002 DEFAULT /ums/i2002.fw 0x02 D.44 IP Phone 2001 DEFAULT /ums/x02.fw 0x02 D.44 IP Phone 2002 Ph2 DEFAULT /ums/x02.fw 0x02 D.44 IP Phone 2004 Ph2 DEFAULT /ums/x02.fw

------------------- Run Time Data ------------------

Last UFTP reset................. 1/14/2096 08:38:19

Cumulation Period............... 0004 01:55:01

Successful downloads............ 1

Fail downloads............ 0

---------------- Active Downloads -----------------

Current downloading sets........ 0

Model IP Address Downloaded[KByte]

umsPolicyShow output example

The output has been modified to display the IP Phone Model Name, firmware ID, and firmware version in ABB format.

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Avaya IP Line Fundamentals User Manual

Specifications and Main Features

Frequently Asked Questions

User Manual