Avaya 4600 Administrator's Guide

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Page 1

4600 Series IP Telephone LAN Administrator Guide

555-233-507

Issue 8

Page 2

Notice

While reasonable efforts were made to ensure that the infor mation in this document was complete and accurate at the time of printing, Avaya Inc. can assume no liability for any errors. Changes and corrections to the information in this document may be incorporated in future releases.

For full legal page information, please see the complete document, Avaya Legal Page for Hardware Documentation, Document number 03-600759.

To locate this document on our Web site, simply go to

http://www.avaya.com/support

the search box. Documentation disclaimer

Avaya Inc. is not responsible for any modifications, addition s, or deletions to the original published version of this documentation unless such modifications, additions, or deletions were performed by Avaya. Customer and/or End User agree to indemnify and hold harmless Avaya, Avaya's agents, servants and employees against all claims, lawsuits, demands and judgments arising out of, or in connection with, subsequent modifications, additions or deletions to this documentation to the extent made by the Customer or End User.

Link disclaimer

Avaya Inc. is not responsible for the contents or reliability of any linked Web sites referenced elsewhere within this documentation, and Avaya does not necessarily endorse the products, services, or informa tion described or o ff ered within them. We cannot guarantee that these links will work all of the time and we have no control over the availability of the linked pages.

Warranty

Avaya Inc. provides a limited warranty on this product. Refer to your sales agreement to establish the terms of the limited warran ty. In addition, Avaya’s standard warranty language, as well as information regarding support for this product, while under warranty, is available through the following Web site:

http://www.avaya.com/support

Except where expressly stated otherwise, the Product is protected by copyrigh t and other laws respecting proprietary rights. Unauthorized reproduction, transfer, and or use can be a criminal, as well as a civil, offense un der the applicable law.

Avaya support

Avaya provides a telephone number for you to use to report pro blems or t o ask questions about your product. The support telephone number is 1-800-242-2121 in the United States. For additional support telephone numbers, see the Avaya Web site:

http://www.avaya.com/support

Software License

USE OR INSTALLATION OF THE PRODUCT INDICATES THE END USER’S ACCEPTANCE OF THE TERMS SET FORTH HEREIN AND THE GENERAL LICENSE TERMS AVAIL ABLE ON T HE AVAYA WEBSITE AT

http://support.avaya.com/LicenseInfo/

YOU DO NOT WISH TO BE BOUND BY THESE TERMS, YOU MUST RETURN THE PRODUCT(S) TO THE POINT OF PURCHASE WITHIN TEN (10) DAYS OF DELIVERY FOR A REFUND OR CREDIT.

Avaya grants End User a license within the scope of the license types described below. The applicable number of licenses and units of capacity for which the license is granted will be one (1), unless a different number of licenses or units of capacity is specified in the Documentation or other materials available to End User. “Designated Processor” means a single stand-alone computing device. “Server” means a Designated Processor that hosts a software application to be accessed by multiple users. “Soft w are” means the computer programs in object code, originally licensed by Avaya and ultimately utilized by End User, whether as stand-alone Products or pre-installed on Hardware. “Hardware” means the standard hardware Products, originally sold by Avaya and ultimately utili zed by End User.

License Type(s):

Designated System(s) License (DS). End User may install and use each copy of the Software on only one Designated Processor, unless a different number of Designated Processors is indicated in the Documentation or other mat erials available to End User. Avaya may require the Designated Processor(s) to be identified by type, serial number, feature key, location or other specific designation, or to be provided by End User to Avaya through elect roni c mean s established by Avaya specifically for this purpose.

and search for the document number in

(“GENERAL LICENSE TERMS”). IF

Third-party Components

Certain software programs or portions thereof included in the Product may contain software distributed under third party agreements (“Third Party Components”), which may contain terms that expand or limit rights to use certain portions of the Product (“Third Party Terms”). Information identifying Third Party Components and the Third Party Terms that apply to them is available on Avaya’s Web site at:

http://support.avaya.com/ThirdPartyLicense/

Interference

Using a cell, mobile, or GSM telephone, or a two-way radio in close proximity to an Avaya IP Telephone might cause interference.

Security

See http://support.avaya.com/security vulnerabilities in Avaya products. See http://support.avaya.com latest software patches and upgrades. For information about secure configuration of equipment and mitigation of toll fraud threats, see the Avaya Toll Fraud and Security Handbook at http://support.avaya.com

to locate and/or report known

to locate the

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Chapter 1: Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

About This Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9

Intended Audience. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 10

Document Organization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 11

Change History . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 12

What’s New in This Release. . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

Terms Used in This Guide. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13

Conventions Used in This Guide . . . . . . . . . . . . . . . . . . . . . . . . . 17

Symbolic Conventions . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

Typographic Conventions . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

Online Documentation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 17

Related Documents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18

IETF Documents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 22

ITU Documents. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 24

ISO/IEC, ANSI/IEEE Documents . . . . . . . . . . . . . . . . . . . . . . . . . 25

Customer Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25

Chapter 2: Overview of Voice over IP (VoIP) and Network Protocols . . 27

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Overview of Voice over IP (VoIP) . . . . . . . . . . . . . . . . . . . . . . . . . . . 27

Data and Voice Network Similarities . . . . . . . . . . . . . . . . . . . . . . . 27

Delay and Jitter . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28

Tandem Coding . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

Voice Coding Standards . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

Telephony Protocols . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29

TFTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

HTTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

DNS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 30

NAT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31

QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31

SNMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 32

Network Assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

4600 Series IP Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 33

Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

DHCP and File Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34

H.323 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

Registration and Authentication . . . . . . . . . . . . . . . . . . . . . . . 35

Time-to-Service (TTS) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 35

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SIP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

Registration and Authentication . . . . . . . . . . . . . . . . . . . . . . . 36

WAN Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

Initialization Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 36

Step 1: Telephone to Network . . . . . . . . . . . . . . . . . . . . . . . . . . 36

Step 2: DHCP Server to Telephone. . . . . . . . . . . . . . . . . . . . . . . . 37

Step 3: Telephone and File Server . . . . . . . . . . . . . . . . . . . . . . . . 37

Step 4: Telephone and the Call Server . . . . . . . . . . . . . . . . . . . . . . 38

TCP/UDP Port Utilization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38

Suggestions for Installation and Configuration . . . . . . . . . . . . . . . . . . . 44

Reliability and Performance. . . . . . . . . . . . . . . . . . . . . . . . . . . . 44

IP Address Lists and Station Number Portability . . . . . . . . . . . . . . . . 45

Security. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 46

Chapter 3: Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . 49

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49

Hardware Requirements. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49

Additional Hardware Requirements . . . . . . . . . . . . . . . . . . . . . . . 51

Software Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 52

Chapter 4: Server Administration . . . . . . . . . . . . . . . . . . . . . 53

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53

Parameter Data Precedence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56

Administering H.323 and SIP IP Telephones on the Same Network . . . . . . . . 56

Administering 4600 Series IP Telephones on Avaya Media Servers (H.323 Only). 57

DEFINITY Releases 9, 9.5, 10, and Avaya

Communication Manager Software Release 1.1+ . . . . . . . . . . . . . . . 57

DEFINITY Release 8.4 . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 57

DHCP and File Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

Software Checklist. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

Required Network Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . 58

DHCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 60

Choosing a DHCP Configuration . . . . . . . . . . . . . . . . . . . . . . . . . 60

DHCP Software Alternatives . . . . . . . . . . . . . . . . . . . . . . . . . . . 60

DHCP Generic Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 61

Windows NT 4.0 DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Verifying the Installation of the DHCP Server . . . . . . . . . . . . . . . . 65

Initial Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65

Creating a DHCP Scope for the IP Telephones . . . . . . . . . . . . . . . 66

4 4600 Series IP Telephone LAN Administrator Guide

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Editing Custom Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . 67

Adding the DHCP Option . . . . . . . . . . . . . . . . . . . . . . . . . . . 68

Activating the Leases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 68

Verifying Your Configuration . . . . . . . . . . . . . . . . . . . . . . . . . 68

Windows 2000 DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . . . . 69

Verifying the Installation of the DHCP Server . . . . . . . . . . . . . . . . 69

Adding DHCP Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72

Activating the New Scope. . . . . . . . . . . . . . . . . . . . . . . . . . . 73

TFTP (H.323 Only) . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 73

TFTP Generic Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 74

TFTP Server on S8300 Media Server . . . . . . . . . . . . . . . . . . . . . . . 74

Avaya File Server Application . . . . . . . . . . . . . . . . . . . . . . . . . . 74

HTTP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75

HTTP Generic Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 75

HTTP Configuration for Backup/Restore. . . . . . . . . . . . . . . . . . . . . . . 76

For IIS Web Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 76

For Apache Web Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 77

Contents

4600 Series IP Telephone Scripts and Application Files . . . . . . . . . . . . . . 78

Choosing the Right Application File and Upgrade Script File . . . . . . . . . 80

Contents of the Upgrade Script. . . . . . . . . . . . . . . . . . . . . . . . . . 82

Contents of the Settings File . . . . . . . . . . . . . . . . . . . . . . . . . . . 82

The GROUP System Value . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 83

QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 84

IEEE 802.1D and 802.1Q. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 85

DIFFSERV . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 86

UDP Port Selection (H.323 Only) . . . . . . . . . . . . . . . . . . . . . . . . . 87

Network Audio Quality Display on 4600 Series IP Telephones. . . . . . . . . 87

RSVP and RTCP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 89

Internal Audio Parameters . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 90

VLAN Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

VLAN Tagging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

VLAN Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 91

VLAN Separation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 92

Unnamed Registration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94

IEEE 802.1X . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94

I802.1X Pass-Through and Proxy Logoff. . . . . . . . . . . . . . . . . . . . . 95

802.1X Supplicant Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . 96

Link Layer Discovery Protocol (LLDP) . . . . . . . . . . . . . . . . . . . . . . . . 97

Administering Options for the 4600 Series IP Telephones . . . . . . . . . . . . . 101

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DNS Addressing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 114

Customizing the Site-Specific Option Number (SSON) . . . . . . . . . . . . . 114

Entering Options Using the Telephone Dialpad . . . . . . . . . . . . . . . . . 114

Enhanced Local Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 115

Setting the Date and Time on SIP IP Telephones . . . . . . . . . . . . . . . . . . 116

Setting the Dial Plan on SIP IP Telephones . . . . . . . . . . . . . . . . . . . . . 117

Customizing the 4630/4630SW IP Telephone . . . . . . . . . . . . . . . . . . . . 117

4630/4630SW Backup/Restore . . . . . . . . . . . . . . . . . . . . . . . . . . 120

Call Log Archive . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 121

Customizing 4610SW, 4620/4620SW, 4621SW, 4622SW,

and 4625SW IP Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 122

The Application Status Flag (APPSTAT) . . . . . . . . . . . . . . . . . . . . . 126

Backup/Restore for 4610SW, 4620/4620SW, 4621SW,

4622SW and 4625SW IP Telephones . . . . . . . . . . . . . . . . . . . . . . . . 127

Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 128

User Permissions for FTP Backup/Restore Files . . . . . . . . . . . . . . . . 128

File Structure. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129

Backup Processing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 129

Chapter 5: Troubleshooting Guidelines . . . . . . . . . . . . . . . . . . 133

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133

Error Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 133

The Clear Administrative Option . . . . . . . . . . . . . . . . . . . . . . . . . . . 140

The Reset Administrative Option. . . . . . . . . . . . . . . . . . . . . . . . . . . 142

Reset System Values . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 142

Restart the Telephone . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 144

The View Administration Option . . . . . . . . . . . . . . . . . . . . . . . . . . . 145

Error Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 148

Troubleshooting the 4601 IP Telephone . . . . . . . . . . . . . . . . . . . . . . . 156

Appendix A: Avaya - 46xx IP Telephone MIB . . . . . . . . . . . . . . . 159

Downloading the Avaya - 46xx IP Telephone MIB . . . . . . . . . . . . . . . . . . 159

Appendix B: Creating Web Sites for the

4630/4630SW IP Telephone . . . . . . . . . . . . . . . . . . . . . . . . . 161

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 161

General Background. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 162

Browser Features and Behavior . . . . . . . . . . . . . . . . . . . . . . . . . . . 162

Document Skeleton . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163

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Content-Based Style. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 163

Logical Style . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 164

Physical Style . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 165

Physical Spacing and Layout. . . . . . . . . . . . . . . . . . . . . . . . . . . 165

Lists and Tables . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166

Lists. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166

Tables. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 166

Images . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167

Links . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 167

Frames . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168

Forms. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 168

Character Entities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 169

Colors. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170

Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170

Cookies. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 170

Design Guidelines . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171

Fixed-Width Objects. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171

Images . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 171

Frames . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172

Fonts . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 172

Maintaining Context . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 173

User Interaction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 173

Click-to-Dial Functionality. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 174

Contents

Appendix C: Creating Web Sites for Other 4600

Series IP Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 177

Appendix D: Administering Thin Client Directories. . . . . . . . . . . . 179

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 179

Appendix E: The Push Feature . . . . . . . . . . . . . . . . . . . . . . . 181

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181

Push Content. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 181

Push Priorities . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 182

For More Information on Push . . . . . . . . . . . . . . . . . . . . . . . . . . . . 182

Appendix F: Sample Upgrade Script File . . . . . . . . . . . . . . . . . 183

Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 183

Issue 8 July 2008 7

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Contents

Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 253

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Chapter 1: Introduction

About This Guide

This guide provides a description of Vo ice over IP and describes how to administer the DHCP, TFTP, and HTTP servers. It also covers how to troubleshoot operational problems with the 4600 Series IP Telephones and the servers.

The 4600 Series IP Telephone product line supports two signaling protocols - the Session Initiation Protocol (SIP) and the H.323 protocol. The chart below shows the 4600 Series IP Telephone models and the protocol(s) they support.

IP Telephone Model H323 Protocol Supported? SIP Protocol Supported?

4601 Yes No 4601+ Yes No 4602 Yes Yes 4602SW Yes Yes 4602SW+ Yes Yes 4606 Yes No 4610SW Yes Yes 4612 Yes No 4620 Yes No 4620SW Yes Yes 4621SW Yes Yes 4622SW Yes No 4624 Yes No 4625SW Yes No 4630 Yes No 4630SW Yes No 4690 Yes No

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Introduction

Sets that support both protocols, for example, the 4610SW, do not support each protocol simultaneously. Instead, a given telephone must be loaded with software that supports one protocol or the other.

Telephones with H.323 software work only with Avaya Communication Manager call servers. Telephones with SIP software are supported only in Avaya server environments.

Note:

Note: Unless otherwise indicated, any reference to “the DEFINITY

document also refers to the Avaya Communication Manager media servers. Administration of the 4602/4602SW SIP Telephones with Release 1.x software is

not covered in this guide. See the 4602 SIP Telephone Administrator's Guide (Document Number 16-300037) for information on administering these 4602/ 4602SW SIP Telephones.

Intended Audience

This document is intended for personnel who administer:

server” in this

● DHCP, TFTP, HTTP, SIP Registration and/or other servers to support the 4600 Series SIP

IP and IP Telephones, and

● Local Area Networks.

CAUTION:

CAUTION: Avaya does not support many of the products mentioned in this document. Take

care to ensure that there is adequate technical support available for these types of servers:

- TFTP servers,

- HTTP servers,

- DHCP servers,

- SIP Registration servers,

- FTP servers,

- LDAP servers, and

- Web servers. Note: If the servers are not functioning correctly, the 4600 Series IP Telephones may

not operate correctly.

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Document Organization

The guide contains the following sections:

Chapter 1: Introduction Provides an overview of the 4600 Series IP

Document Organization

Telephone LAN Administrator document.

Chapter 2: Overview of Voice over IP (VoIP) and Network Protocols

Chapter 3: Requirements Describes the hardware and software

Chapter 4: Server Administration

Chapter 5: Troubleshooting Guidelines Describes messages that might occur

Appendix A: Avaya - 46xx IP Telephone MIB

Appendix B: Creating Web Sites for the 4630/4630SW IP Telephone

Describes VoIP and factors influencing its performance that must be considered when implementing this feature.

requirements for Avaya’s VoIP offering. Describes DHCP, TFTP, and HTTP

administration for the 4600 Series IP and SIP IP Telephones.

during the operation of the 4600 Series IP Telephones.

Provides a link to the MIB specification for the 46xx IP Telephones: 4601, 4601+, 4602/4602SW/4602SW+, 4606, 4610SW, 4612, 4620/4620SW, 4621SW, 4622SW, 4624, 4625SW, and 4630/4630SW.

Provides information on creating and customizing Web sites for viewing on the 4630/4630SW IP Telephone. Also describes the current capabilities and limitations of the 4630/4630SW’s Web Browser.

Appendix C: Creating Web Sites for Other 4600 Series IP Telephones

Appendix D: Administering Thin Client Directories

Appendix E: The Push Feature

Appendix F: Sample Upgrade Script File Provides a detailed example of a 46xx

Provides information on creating and customizing Web sites for viewing on the 4610SW, 4620/4620SW, 4621SW, 4622SW, and 4625SW IP Telephones.

Provides information on administering an LDAP directory for the 4610SW, 4620/ 4620SW, 4621SW, 4622SW, and 4625SW IP Telephones.

Provides information about the Push feature available as of Release 2.1.

upgrade script file.

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Introduction

Change History

Issue 1.0 This document was issued for the first time in November 2000. Issue 1.1 This version of the document, revised and issued in April 2001, supports through

DEFINITY

Issue 1.5 This version of the document was revised in June, 2001 to support DEFINI TY® Release

9.5.

Issue 1.6 This version of the document was revised to support DEFINITY

4630 IP Telephone.

Issue 1.7 This version of the document was revised in July, 2002 to support Avaya

Communication Manager Release 1.1 and the 4602 and 4620 IP Telephones.

Issue 1.8 This version of this document was revised in June, 2003 to support Avaya

Communication Manager Releases 1.2 and 1.3. This version also supported the 4602SW and 4630SW IP Telephones.

Issue 2.0 This version of this document was revised in December, 2003 to add support for Avaya

Communication Manager Release 2.0. This version also supported the 4610SW and 4620SW IP Telephones, and the 4690 IP Conference Telephone.

Issue 2.1 This version of this document was revised in July, 2004 to add support for Avaya

Communication Manager Release 2.1. This version also added support for the TFTP server on the Avaya S8300 Media Server, and support for the 4601 IP Telephone.

Issue 2.2 This version of this document was revised and issued in April, 2005. This version

supports through Avaya Communication Manager Release 2.2. This version also introduces the 4621SW, 4622SW, and 4625SW IP Telephones.

Issue 2.2.1 This version of this document was revised and issued in August, 2005. This version

introduced the SIP IP Telephones. This version also distinguishes between functionality that is H.323-specific and functionality that is SIP-specific.

Issue 2.3 This version of this document was revised and issued in November, 2005 to provide

support through Avaya Communication Manager Release 3.0.

Issue 3 This version of this document was revised and issued in April, 2006 to support Software

Release 2.4. This version provides VLAN separation parameters, an unnamed registration parameter, and audio customization parameters.

Issue 4 This version of this document was revised and issued in August, 2006. This version

supports Avaya Communication Manager 3.1 and Software Release 2.6. New features for 802.1X authentication, Link Layer Discovery Protocol (LLDP), and power conservation are introduced.

Issue 5 This version of this document was revised and issued in November, 2006 to support

Software Release 2.7. This version introduced additional Unicode languages, support for dialpad-activated Web links, the capability to turn off the display backlight, several new system parameters, and two ne w local p rocedures. This issue also introduced new telephone models and 4602SW+, which replace the 4601 and 4602/4602SW, respectively.

Issue 6 This version of this document was revised and issued in February 2007 to support

Software Release 2.8. This version introduced backup/restore capabilities to HTTP servers, additional security measures for TCP, new system parameters for adding IP Source Addresses, configuring HTTP ports, and using HTTP filepaths, and reduced time-to-service through Communication Manager 4.0 upgrades.

Issue 7 This version was revised and issued in September, 2007 to support Software Release

2.8.3. This version introduced new audio parameters and changes to audio parameter settings, guidance on a troubleshooting issue related to Message Waiting Indicators, and Communication Manager 3.1.3 and above support for downloading of four key administrative parameters to individual phones.

Release 9.

Release 10 and the

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Document Organization

Issue 8 This is the current version of this document, revised and issued in July 2008 to support

Software Release 2.9. This version introduces the capability to disable the 802.1X supplicant operation, offers enhanced LLDP MED Network Policy TLV processing, and allows immediate access to the manual addressing local administrative procedure during startup. For information, see What’s New in This Release

What’s New in This Release

New material in this issue to support Release 2.9 software includes:

● The ability to download and use third-party trusted certificates to authenticate file servers

that download configuration files and for encryption of backup/restore file parameters.

● New system parameter BRAUTH can be used to control whether registration credentials

are sent to a backup/restore server so that the telephone requesting the file can be authenticated. For information, see Backup/Restore for 4610SW , 4620/4620SW, 4621SW,

4622SW and 4625SW IP Telephones.

● New system parameter TRUSTCERTS is used to specify the file names of trusted

Certificate Authority certificates the telephone should download to authenticate server certificates.

● The manual addressing local administrative procedure (ADDR) can now be accessed

during telephone initialization, as described in Chapter 3 of the 4600 Series IP Telephone Installation Guide. When saving statically programmed values, the telephone now resets if

the system parameter PROCSTAT is set to "1" (access to administrative procedures allowed).

● 802.1X supplicant operation can now be disabled using the new system parameter

DOT1XSTAT

. As covered in Chapter 3 of the 4600 Series IP Telephone Installation Guide, the "Set the 802.1X Operational Mode" local administrative procedure has been modified to reflect changes in pass-thru and supplicant operation.

● LLDP MED Network Policy TLV processing has been modified, as described in Link Layer

Discovery Protocol (LLDP).

● Certain parameters and sections in this guide relating to HTTP Backup and Restore have

been revised to correct documentation errors in describing Release 2.8 processing. The affected parameters and sections are: HTTP Configuration for Backup/Restore

, Backup/

Restore for 4610SW, 4620/4620SW, 4621SW, 4622SW and 4625SW IP Telephones, BRURI

, HTTP, and HTTPS.

For detailed information regarding installation-related updates and revisions, see the 4600 Series IP Telephone Installation Guide (Document Number 555-233-128).

Terms Used in This Guide

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Introduction

802.1D

802.1Q

802.1Q defines a layer 2 frame structure that supports VLAN identification and a QoS mechanism usually referred to as 802.1D.

802.1X Authentication method for a protocol requiring a network device to authenticate with a back-end Authentication Server before gaining network access. Applicable 4600 Series IP telephones support IEEE 802.1X as a Supplicant with the EAP-MD5 authentication method.

ARP Address Resolution Protocol, used, for example, to verify that the IP Address

provided by the DHCP server is not in use by another IP telephone.

CELP Code-Excited Linear-Predictive. Voice compression requiring only 16 kbps of

bandwidth.

CLAN Control LAN, type of Gatekeeper circuit pack. CNA Converged Network Analyzer, an Avaya product to test and analyze network

performance.

DHCP Dynamic Host Configuration Protocol, an IETF protocol used to automate IP Address

allocation and management.

DiffServ Differentiated Services, an IP-based QoS mechanism. DNS Domain Name System, an IETF standard for ASCII strings to represent IP

Addresses.

EAP Extensible Application Protocol. Gatekeeper H.323 application that performs essential control, administrative, and managerial

functions in the media server. Sometimes called CLAN in Avaya documents.

H.323 A TCP/IP-based protocol for VoIP signaling. HTTP Hypertext Transfer Protocol, used to request and transmit pages on the World Wide

Web.

HTTPS A secure version of HTTP. IETF Internet Engineering Task Force, the organization that produces standards for

communications on the internet.

LAN Local Area Network. LDAP Lightweight Directory Access Protocol, an IETF standard for database organization

and query exchange.

LLDP Link Layer Discovery Protocol. All IP telephones with an Ethernet interface support

the transmission and reception of LLDP frames on the Ethernet line interface in accordance with IEEE standard 802.1AB.

MAC Media Access Control, ID of an endpoint. Media

Channel

Encryption of the audio information exchanged between the IP telephone and the call server or far end telephone.

Encryption NAPT Network Address Port Translation. NAT Network Address Translation.

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Document Organization

OPS Off-PBX Station. PAE Port Access Entity. The protocol entity associated with a port. The PAE supports the

protocol functionality associated with the authenticator, supplicant, or both.

PHP Hypertext Preprocessor, software used to assist in the format and display of Web

pages.

PoE Power over Ethernet. PSTN Public Switched Telephone Network, the network used for traditional telephony. QoS Quality of Service, used to refer to several mechanisms intended to improve audio

quality over packet-based networks.

Registration Server

A SIP server that accepts REGISTER requests. The Registration Server places the information received in the requests into the location service for the domain the server handles.

RSVP Resource ReSerVation Protocol, used by hosts to request resource reservations

throughout a network.

RTCP RTP Control Protocol, monitors quality of the RTP services and can provide real-time

information to users of an RTP service.

RTP Real-time Transport Protocol. Provides end-to-end services for real-time data such

as voice over IP.

SDP Session Description Protocol. A well-defined format for conveying sufficient

information to discover and participate in a multimedia session.

Signaling Channel Encryption

Encryption of the signaling protocol exchanged between the IP telephone and the call server. Signaling channel encryption provides additional security to the security provided by media channel encryption.

SIP Session Initiation Protocol. An IETF standard protocol for IP communication. SIP

enables IP telephony gateways, client endpoints, PBXs, and other communication systems or devices to communicate with each other. SIP mainly addresses the call setup and tear down mechanisms of sessions and is independent of the transmission of media streams between the caller and the called party. SIP is an alternative to H.323 for VoIP signaling.

SMTP Simple Mail Transfer Protocol. An IETF standard pr otocol (RFC 2821) for e-mail. Part

of the TCP/IP protocol suite. SMTP sets the message format and message transfer agent that stores and forwards the email.

SNMP Simple Network Management Protocol. The Internet standard protocol, defined in

STD 15, RFC 1157, developed to manage nodes on an IP network.

SNTP Simple Network Time Protocol. An adaptation of the Network Time Protocol used to

synchronize computer clocks in the internet.

Supplicant An entity at one end of a point-to-point LAN segment that is being authenticated by

an authenticator at the other end.

TCP/IP Transmission Control Protocol/Internet Protocol, a network-layer protocol used on

LANs and internets.

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Introduction

TFTP Trivial File Transfer Protocol, used to provide downloading of upgrade scripts and

application files to the IP telephones.

Time-toService (TTS)

A new feature with Communication Manager 4.0, IP Endpoint Time-to-Service (TTS) decouples gatekeeper H.323 registration from TCP socket establishment for call signaling, thus reducing the time for the endpoint to come into service.

TLS Transport Layer Security, an enhancement of Secure Sockets Layer (SSL). TLS is

compatible with SSL 3.0 and allows for privacy and data integrity between two communicating applications.

TLV Type-Length-Value elements transmitted and received as part of Link Layer

Discovery Protocol (LLDP).

UDP User Datagram Protocol, a connectionless transport-layer protocol. Unnamed

Registration

Registration with Avaya Communication Manager by an IP telephone with no extension. Unnamed registration is typically used to limit incoming calls.

VLAN Virtual LAN. VoIP Voice over IP, a class of technology for sending audio data and signaling over LANs. WML Wireless Markup Language, used by any 4600 Series IP Telephones that can

communicate with WML servers.

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Conventions Used in This Guide

This guide uses the following textual, symbolic, and typographic conventions to help you interpret information.

Symbolic Conventions

Note:

Note: This symbol precedes additional information about a topic. This information is not

required to run your system.

CAUTION:

CAUTION: This symbol emphasizes possible harm to software, possible loss of data, or

possible service interruptions.

Typographic Conventions

Online Documentation

This guide uses the following typographic conventions:

command Words printed in this type are commands that you enter into your

system. message Words printed in this type are system messages. device Words printed in this type indicate parameters associated with a

command for which you must substitute the appropriate value. For

example, when entering the mount command, device must be

replaced with the name of the drive that contains the installation disk. Administrative Words printed in bold type are menu or screen titles and labels. W ords

printed in bold type can also be items on menus and screens that you

select or enter to perform a task, i.e., fields, buttons, or icons. Bold

type also provides general emphasis for words or concepts. italics Italic type indicates a document that contains additional information

about a topic.

Online Documentation

The online documentation for the 4600 Series IP Telephones is located at the following URL:

http://www.avaya.com/support

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Introduction

Customer Support

● IEEE Std. 802.1X-2004, IEEE Standard for Local and Metropolitan Area Networks -

Port-Based Network Access Control

● IEEE Std. 802.1AB-2005, IEEE Standard for Local and Metropolitan Area Networks:

Station and Media Access Control Connectivity Discovery

● For more information about 802.1AB, see:

http://www.standards.ieee.org/getieee802/download/802.1AB-2005.pdf

● For more information about 802.1X, see:

http://www.standards.ieee.org/getieee802/download/802.1X-2004.pdf

Customer Support

Call the Avaya support number provided to you by your Avaya representative or Avaya reseller for 4600 Series IP Telephone support.

Information about Avaya products can be obtained at the following URL:

http://www.avaya.com/support

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Introduction

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Chapter 2: Overview of Voice over IP (VoIP) and

Network Protocols

Introduction

This chapter describes the differences between data and voice networks, and the factors that influence the performance of VoIP. The installation and administration of 4600 Series IP Telephones on Avaya Media Servers, and the installation and configuration of DHCP and TFTP are addressed.

Overview of Voice over IP (VoIP)

The 4600 Series IP Telephones allow enterprises to use Voice over IP (VoIP). VoIP uses packet-switched networks over the Public Switched Telephone Network (PSTN) instead of telephony. However, using data networks to transmit voice packets poses a problem. Data networks were not designed for the specific qualities required by voice traffic.

Data and Voice Network Similarities

Data and voice networks share similar functions because of the nature of networking.

● Signaling: establishes a connection between two endpoints.

In a voice network, signaling helps identify who the calling party is trying to call and where the called party is on the network. Traditional telephony uses terminals with fixed addresses. Traditional telephony establishes a fixed connection for the communication session between two such terminals, allocating fixed bandwidth resources for the duration of the call.

IP communications constitute a connectionless network, having neither fixed addresses nor fixed connections.

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Overview of Voice over IP (VoIP) and Network Protocols

● Addressing: a unique address that must identify each terminal on a network.

In a voice network, the unique address is a permanent attribute, based on any combination of:

- international numbering plans,

- national numbering plans,

- local telephone company practices,

- internal customer-specific codes.

In IP communications, dial plans track extension numbers assigned to terminals. No fixed connection path is needed.

● Routing: related to addressing and allows connections to be established between

endpoints.

Although these functions are common to data and voice networks, the implementations differ.

Delay and Jitter

Data traffic is usually short and comes in burst s. Dat a networks like the Internet are designed to manage these bursts of traffic from many sources on a first-come, first-served basis. Data packets are sent to multiple destinations, often without any attempt to keep them in a particular order.

Voice networks are designed for continuous transmission during a call. The traffic is not bursty, and the conversation uses a specific amount of bandwidth between the two ends during the call.

Several features of data networks are unsuitable for voice telephony:

● Data network design delivers data at the destination, but not necessarily within a certain

time, producing delay (latency). In data networks, delay tends to be variable. For voice messages, variable delay results in jitter, an audible choppiness in conversations.

● Variable routing also can result in loss of timing synchronization, so packets are not

received at the destination in the proper order.

● Data networks have a strong emphasis on error correction, resulting in repeated

transmissions.

Data network concepts include prioritization of traffic types to provide some form of greater traffic reliability, for example, for interactive transactions. However, data requirements tend to not be as strict as most voice requirements.

The 4600 Series IP Telephones include a dynamic jitter buffer. This feature automatically smooths jitter to improve audio quality.

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Tandem Coding

Tandem coding, also called transcoding, refers to converting a voice signal from analog to digital and back again. When calls are routed over multiple IP facilities, they can be subject to multiple transcodings. The multiple conversions between analog and digital coding result in a deterioration in the voice quality. Avoid tandem coding wherever possible in any compressed voice system, for example, by minimizing analog trunking on the PBX.

Voice Coding Standards

There are several voice coding standards. Avaya 4600 Series IP Telephones offer these options:

● G.711, which describes the 64 kbps PCM voice coding technique. G.711-encoded voice is

already in the correct format for digital voice delivery in the public telephone network or through PBXs.

● G.726 ADPCM at 32Kbps.

● G.729A and G.729B, both of which describe adaptive code-excited, linear-predictive

(CELP) compression that allows voice to be coded into 8 kbps streams.

Overview of Voice over IP (VoIP)

Telephony Protocols

There are two major protocols used for Voice over IP (VoIP) signaling - Session Initiation Protocol (SIP) and H.323. The two protocols provide connection control and call progress signaling, but in very different ways. These protocols can be used simult aneously over the same network, but in general, no endpoint supports both protocols at the sa me time. Neith er protocol is necessarily superior, but each offers some unique advantages. SIP telephones, for example, do not require centralized call servers, and can route telephone calls when a URL identifies the destination. H.323 telephones leverage the call server’s presence into the potential availability of hundreds of telephone-related features that a standalone SIP telephone cannot provide.

DHCP

Dynamic Host Configuration Protocol (DHCP) allows a server to assign IP Addresses and other parameters to devices like the 4600 Series IP Telephones on an as-needed basis. DHCP eliminates the need to configure each end station with a static IP Address. The DHCP application also passes information to the 4600 Series IP Telephone. The DHCP application identifies the PBX and the file server’s IP Addresses. The application also identifies the paths to the upgrade script and the application file on the file server.

For further information, see DHCP and File Servers

on page 58 and DHCP on page 60.

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Overview of Voice over IP (VoIP) and Network Protocols

TFTP

The Avaya 4600 IP Telephones can get useful application information from the TFTP server. The telephones also can upgrade themselves using files stored on the TFTP server. After downloading software, the Avaya 4600 Series IP Telephones can operate without a file server. However, some functionality can be lost if the file server is not available for a telephone reset. For further information, see:

● DHCP and File Servers on page 58,

● TFTP (H.323 Only) on page 73, and

● Table 1: File Servers and Compatible Telephone Software.

HTTP

HTTP is potentially a more secure alternative to TFTP, particularly when Transport Layer Security (TLS) is used to create HTTPS (Secure HTTP). You can store the same application software, script file, and settings file on an HTTP server as you can on the TFTP server. With proper administration, the telephone seeks out and uses that material appropriately. However, not all 4600 Series IP Telephones support HTTP, as indicated in Table 1

Table 1: File Servers and Compatible Telephone Software

DNS

IP Telephone Software

File Server IP Telephone Models

HTTP, HTTPS, or TFTP 4601+, 4602SW+, 4625SW

4601, 4602, 4602SW, 4610SW, 4620,

Release Number

R2.7 R2.2+

4620SW, 4621SW, 4622SW HTTP, HTTPS or TFTP 4690 R 2.3 TFTP 4606, 4612, 4624, 4625SW, 4630,

All releases

4630SW

For Release 2.8 and future releases only, use HTTPDIR

and HTTPPORT to configure backup/

restore and file retrieval operations. These must be set via DHCP for retrieving script files. As with TFTP, some functionality might be lost by a reset if the HTTP server is not available.

For more information, see DHCP and File Servers

on page 58 and HTTP on page 75.

The Domain Name System (DNS) is a distributed Internet directory service. DNS is used mostly to translate between domain names and IP Addresses. Release 1.5 and later Avaya IP Telephones can use DNS to resolve names into IP Addresses. In DHCP, TFTP, and HTTP files, DNS names can be used wherever IP Addresses were available as long as a valid DNS server is identified first. See DNS Addressing

on page 114.

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NAT

Overview of Voice over IP (VoIP)

A Network Address Translation (NAT) is an application that can be administered between your network and the Internet. The NAT translates network layer IP Addresses so your local intranet IP Addresses can duplicate global, Internet addresses. A detailed discussion of NAT is beyond the scope of this document. Note that NAT use can lead to problems that affect the consistency of addressing throughout your network. In Release 1.6 and earlier releases of the 4600 Series IP Telephones, NAT is not recommended for networks handling IP-based telephony traffic. As of Release 1.7, all 4600 Series H.323 IP Telephones support NAT interworking. Therefore, no problems exist with NAT and these H.323 IP telephones. Note that support for NAT does not imply support for Network Address Port Translation (NAPT). Specifically, the H.323 IP telephones do not support communication to the PBX through any NAPT device. SIP IP telephones do not support NAT.

NAT requires specific administration on the media server. The capability to have a direct Avaya IP Telephone-to-Avaya IP Telephone call with NAT, also called “NAT shuffling,” requires Avaya Communication Manager Release 1.3 software. See the Administration for Network Connectivity document listed in Related Documents

on page 18.

QoS

Quality of Service (QoS) is a term covering several initiatives to maximize the voice quality heard at both ends of a call that originates or terminates on an IP-based telephone. These initiatives include various prioritization schemes to offer voice packets a larger or prioritized share of network resources. These schemes include standards such as:

● IEEE’s 802.1D and 802.1Q,

● the Internet Engineering Task Force’s (IETF’s) “Differentiated Services,”

● RTP Control Protocol (RTCP),

● Resource ReSerVation Protocol (RSVP), and

● port-based priority schemes such as UDP port selection.

Documentation for your LAN equipment details the extent to which your network can support any or all of these initiatives. See Server Administration

on page 53, for some implications of

QoS for the 4600 Series IP Telephones. As of Release 1.7, the 4620, 4630, and 4630SW IP Telephones provided network audio quality

information to the end user. This network audio quality information might be useful to the LAN Administrator. As of Release 1.8, all 4600 Series IP Telephones provide some level of detail about network audio quality. For specific information, see Network Audio Quality Display on

4600 Series IP Telephones on page 87.

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Overview of Voice over IP (VoIP) and Network Protocols

SNMP

Simple Network Management Protocol (SNMP) is a family of standards-based protocols and procedures to allow vendor-independent data network management. Using a simple set of protocol commands, an SNMP-compliant device stores information in stan dard format in one or more Management Information Bases (MIBs). Usually, devices support the standards-specific MIB termed MIB-II. In addition, devices can define one or more “custom MIBs” that contain information about the device’s specifics.

As of Release 1.1, the 4600 Series IP Telephones are fully compatible with SNMPv2c, a later version of SNMP, and with Structure of Management Information Version 2 (SMIv2). The telephones respond correctly to queries from entities that comply with earlier versions of SNMP, such as SNMPv1. “Fully compatible” means that the telephones respond to queries directed either at the MIB-II or the Custom MIB. The 4600 Series IP Telephone Custom MIB is read-only. Read-only means that the values therein cannot be changed externally by means of network management tools.

You can restrict which IP Addresses the telephone accepts SNMP queries from. You can also customize your community string with system values SNMPADD and SNMPSTRING, respectively, as indicated in Chapter 4: Server Administration

Telephone Customizable System Parameters.

, Table 10: 4600 Series IP

Important:

Important: SNMP has been enabled by default since Release 1.1. However, as of Release

2.6, the SNMP default changed to Null (Off). To activate SNMP, you must set SNMPSTRING to a non-null value by means of either the 46xxsettings file or DHCP Option 176 (SSON). If you use static programming, you cannot enable SNMP as of Release 2.6.

As of Release 2.8, Communication Manager Release 3.1.3 and above supports downloading of SNMPADD and SNMPSTRING settings upon registration of the phones.

To find more information about SNMP and MIBs, see the IETF references listed in Related

Documents on page 18. The Avaya Custom MIB for the 4600 Series IP Telephones is available

for download in *.txt format on the Avaya support Web site. This Custom MIB is common to both H.323 and SIP IP telephones. Objects that are not relevant to a given telephone have Null dat a.

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Network Assessment

The current technology allows optimum network configurations to deliver VoIP with perceived voice quality close to that of the Public Switched Telephone Network (PSTN). Not all networks can take advantage of packet voice transmissions. Some data networks have insufficient residual capacity for even compressed voice traffic. In addition, the usual approach to developing data networks by integrating products from many vendors requires testing the components for Voice over IP traffic compatibility.

Avaya assumes that your organization has performed a network assessment with or without Avaya’s assistance before attempting to inst all V oice over IP. The network assessment provides a high degree of confidence that the existing data network has the capacity to carry voice packet traffic. The network assessment assures that the existing data network is compatible with the required technology.

A network assessment should include:

● A network audit to review existing equipment and evaluate its capabilities, including its

ability to meet planned voice and data needs.

Network Assessment

● A determination of network objectives, including the dominant traffic type, selection of

technologies, and setting voice quality objectives.

The assessment should leave you confident that the implemented network will have the capacity for the foreseen data and voice traffic, and can support H.323, SIP, DHCP, TFTP, HTTP, and jitter buffers in all applications.

It is important to distinguish between compliance with the minimal VoIP standards and QoS support, the latter being a requirement to run VoIP on your configuration.

4600 Series IP Telephones

The 4600 Series IP Telephones support either of two signaling protocol families - H.323 and Session Initiation Protocol (SIP).

The H.323 standard, developed by ITU-T, provides for real time audio, video, and data communications transmission over a packet network. An H.323 telephone protocol stack comprises several protocols:

● H.225 for registration, admission, status (RAS), and call signaling,

● H.245 for control signaling,

● Real Time Transfer Protocol (RTP), and

● Real Time Control Protocol (RTCP).

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SIP was developed by the IETF. Like H.323, SIP provides for real time audio, video, and data communications transmission over a packet network. SIP uses various messages, or methods, to provide:

● Registration (REGISTER),

● Call signaling (INVITE, BYE)

● Control signaling (SUBSCRIBE, NOTIFY)

SIP also supports RTP and RTCP using the Session Description Protocol. A telephone is loaded with either H.323 or SIP software as part of its initial script file

administration and initialization.

Software

As shipped from the factory , the 4600 Series IP Telephones may not contain the latest sof tware. When the telephone is first plugged in, a software download from a TFTP or HTTP server start s to give the telephone its proper functionality.

For downloads of H.323 software upgrades, the PBX provides the capability for a remote restart of the 4600 Series IP Telephone. As a consequence of restarting, the telephone automatically restarts reboot procedures. If new software is availa ble on the serve r, the telephone downloads it as part of the reboot process.

A 4602, 4602SW, 4602SW+, 4610SW, 4620SW, 4621SW, or 4625SW IP Telephone can support either H.323 or SIP software, but not both at the same time. All telephones come from the factory with H.323 software by default. You can convert a telephone from H.323 to SIP, or from SIP to H.323 by administering your server and settings file. For more information, see “Converting Software on Avaya 4600 Series IP Telephones” in the 4600 Series IP Telephone Installation Guide (Document Number 555-233-128).

DHCP and File Servers

The DHCP server provides the following information to the 4600 Series IP Telephone:

● IP Address of the 4600 Series IP Telephone

● IP Address of the Gatekeeper board on the Avaya Media Server, applicable only to H.323

IP telephones

● IP Address of the TFTP server if applicable, otherwise the HTTP or HTTPS server

● The subnet mask

● IP Address of the router

● DNS Server IP Address

Administer the LAN so each IP telephone can access a DHCP server containing the IP Addresses and subnet mask listed.

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4600 Series IP Telephones

The IP telephone cannot function without an IP Address. The failure of a DHCP server at boot time leaves all the affected voice terminals unusable. A user can manually assign an IP Address to an IP telephone. This can cause a problem when the DHCP server finally returns because the telephone never looks for a DHCP server unless the static IP data is unassigned manually. In addition, manual entry of IP data is an error-prone process. Avaya therefore strongly recommends that a DHCP server be available when the IP telephone reboots. If a DHCP server is not available at remote sites during WAN failures, the IP telephone is not available after a reboot.

A minimum of two DHCP servers are recommended for reliability. Avaya strongly recommends that a DHCP server be available at remote sites if WAN failures isolate IP telephones from the central site DHCP server(s).

The file server provides the 4600 Series IP Telephone with a script file and, if appropriate, new or updated application software. See Step 3: Telephone and File Server

Initialization Process

telephone parameters for your specific environment. See Administering Options for the 4600

Series IP Telephones on page 101.

. In addition, you can edit an associated settings file to customize

on page 37 under

H.323

Registration and Authentication

The Avaya Media Server supports using the extension and password to register and authenticate 4600 Series IP Telephones. For further information, see Related Documents page 18.

Time-to-Service (TTS)

The IP Endpoint Time-to-Service (TTS) feature introduced in the R2.8 software release, along with the Communication Manager 4.0 release, changes the way IP endpoints register with their gatekeeper, reducing the time to come into service. Currently, IP endpoints are brought into service in two steps, which are coupled (1) H.323 registration and (2) TCP socket establishment for call signaling. The TTS feature de-couples these steps. In CM 4.0, IP endpoints can be enabled for service with just the registration step. TCP sockets are established later, as needed. The TTS feature also changes the direction of socket establishment. With TTS, Communication Manager, rather than the endpoint, initiates socket establishment, which further improves performance.

In CM 4.0, TTS is enabled by default, but can be disabled for all IP endpoints in a given IP network regions by changing the IP Network form. TTS applies only to IP endpoints whose firmware has been updated to support this feature. It does not apply to the following endpoints: third party H.323, SIP, DCP, BRI, and analog. Please refer to Administrator Guide for Avaya Communications Manager, # 03-300509 for further information.

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SIP

Registration and Authentication

A 4600 Series SIP IP Telephone requires an off-PBX station (OPS) extension on the Avaya Communication Manager and a login and password on the Registration Server to register and authenticate it. Registration is described in the Initialization process, in Step 4: Telephone and

the Call Server on page 38. For further information, see the Converged Communication Server

Release 3.0 Installation and Administration Guide (555-245-705), available on the Avaya support Web site, http://www.avaya.com/support

WAN Considerations

QoS is harder on a WAN than a LAN. A LAN assumes no bandwidth concerns. A WAN assumes a finite amount of bandwidth. Therefore, QoS considerations are more significant when the IP telephony environment includes a WAN. In addition, there are administrative and hardware compatibility issues unique to W ANs. WAN administration is beyond the scope of this document.

Initialization Process

These steps offer a high-level description of the information exchanged when the telephone initializes and registers. This description assumes that all equipment is properly administered ahead of time. This description can help you understand how the 4600 Series IP Telephones relate to the routers and servers in your network.

Step 1: Telephone to Network

The telephone is appropriately installed and powered. After a short initialization process, the telephone identifies the LAN speed. If applicable to your network and telephone model, 802.1X Supplicant authentication occurs at this time, where an 802.1X ID and password have to be submitted to proceed. Then the telephone sends a DHCP message out into the network, identifying itself and requesting further information. Depending on your network structure, the message goes directly to the DHCP server or a network router receives the message and relays it to the appropriate DHCP server.

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Step 2: DHCP Server to Telephone

The DHCP file server provides information to the telephone, as described in DHCP and File

Servers on page 58. Among other data passed to the telephone is the IP Address of the TFTP

or HTTP server, which is crucial for the next step.

Step 3: Telephone and File Server

Many 4600 Series IP Telephones can download script files, application files, and settings files from either a TFTP, HTTP, or HTTPS server. For specific telephone/server compatibility, see

Table 1

You need TFTP servers for upgrades if you have a release prior to R2.2 on a telephone. Currently, only Avaya servers support 4600 Series IP Telephone file transfers using HTTPS. The reason is 4600 Series IP Telephones only establish encrypted TLS connections with servers that use Avaya-signed digital certificates.

. If you have a mixture of telephones, you can use either:

● TFTP servers only.

● Both TFTP and HTTP servers, with TFTP running telephones with older releases and

HTTP for telephones capable of using HTTP.

Initialization Process

A telephone that supports HTTP will attempt to access the HTTP server (if a dministered), and, if successful, will not attempt to access the TFTP server (if administered).

The script files, application files, and settings files discussed in this section are identical for HTTP and TFTP servers. The general downloading process for those files is essentially the same. One exception is that when you use an HTTPS server, a TLS server is contacted first. Therefore, we use the generic term “file server” here to mean both “TFTP server” and “HTTP server.”

The telephone queries the file server, which transmits a script file to the telephone. This script file, at a minimum, tells the telephone which application file the telephone must use. The application file is the software that has the telephony functionality, and can be easily updated for future enhancements.

The telephone uses the script file to determine if it has the proper application file. A newly installed telephone may have no application file, and therefore would not have the proper one. A previously installed telephone might not have the proper application file. If the telephone determines the application file indicated in the script file is missing, the telephone requests a download of the proper application file from the file server. The file server then downloads the file and conducts some checks to ensure that the file was downloaded properly. If the telephone determines it already has the proper file, the telephone proceeds to the next step without downloading the application file again.

After checking and loading the application file, the 4600 Series IP Telephone, if appropriate, uses the script file to look for a settings file. The settings file can contain options you have administered for any or all of the 4600 Series IP Telephones in your network. For more information about this settings file, see Administering Options for the 4600 Series IP

Telephones on page 101.

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Overview of Voice over IP (VoIP) and Network Protocols

Step 4: Telephone and the Call Server

The call server referred to in this step differs depending on whether the telephone is H.323 or SIP. For H.323 IP telephones, the call server is the Avaya Media Server. For SIP IP telephones, the call server is the Registration Server.

In this step, the telephone and the call server exchange a series of messages, which cause th e display on the telephone to prompt the user. For a new installation and for full service, the user must enter the telephone’s extension and the call server password. For a restart of an existing installation, this information is already stored on the telephone, but the user may have to confirm the information. The telephone and the call server exchange more messaging , with th e expected result being that the telephone is appropriately registered.

An exception to the requirement to enter an extension and pa ssword is for H.323 IP telephones running R2.3 and later software. These telephones support a feature called Unnamed Registration. Unnamed Registration allows a telephone to register with the Avaya Media Server without an extension, assuming the Avaya Media Server also supports this feature. To invoke Unnamed Registration, take no action—just let the Extension... prompt display for 60 seconds without making an entry. The telephone will automatically attempt to register by means of Unnamed Registration. A telephone registered with Unnamed Registration has the following characteristics:

● only one call appearance,

● no administrable features,

● can make only outgoing calls, subject to call server Class of Restriction/Class of Service

limitations, and

● can be converted to normal “named” registration by entering a valid extension and

password.

More details about the installation process are available in the 4600 Series IP Telephone Installation Guide and in Chapter 3: Requirements

TCP/UDP Port Utilization

Like most network equipment, the 4600 Series IP Telephones use a variety of protocols, particularly TCP and UDP, to communicate with other equipment in that network—numerous different types of servers, routers, other telephones, etc. Part of this communication identifies which TCP and/or UDP ports each piece of equipment uses to support each protocol and each task within the protocol.

Depending on your network, you might need to know what ports or ranges are used in the 4600 Series IP Telephones’ operation. Knowing these ports or ranges allows you to appropriately administer your networking infrastructure. In this case, you will find the following material useful.

of this document.

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TCP/UDP Port Utilization

In Figure 1, Figure 2, and Figure 3:

● The box on the left always represents the 4600 Series IP Telephone.

● Depending on the diagram, the boxes on the right refer to various pieces of network

equipment with which the telephone can (or will) communicate.

● Open-headed arrows (for example, ) represent the direction(s) of communication

initialization.

● Closed-headed arrows (for example, ) represent the

direction(s) of data transfer.

● The text the arrows point to identifies the port or ports that the 4600 Series IP Telephones

support for the specific situation. Brackets identify ranges when more than one port applies. In addition, the text indicates any additional qualifications or clarifications. In man y cases, the ports used are the ones called for by IETF or other standards bodies.

● Many of the diagrams’ explanations refer to system parameters or options settings,

for example, IRSTAT or DIRSRVR. See Administering Options for the 4600 Series IP

Telephones in Chapter 4: Server Administration for more information on parameters and

settings.

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Overview of Voice over IP (VoIP) and Network Protocols

Figure 1: Signaling, Audio and Management Diagram

Signaling, Audio and Management

4600 Series IP Telephone

Port: [49300-49309]

Default is 49300

Port: [1500–6500]

randomly selected

Port: 1720

Port: 5060

Port: [2048–3028]

randomly selected;

range may be changed via

Gatekeeper administration;

always an even number

Port: RTP port +1

(only active during a call if

RTCP is enabled)

H.323 Gatekeeper

H.323 RAS (UDP/IP)

H.323 Signaling (TCP/IP)

Port: 1719

Port: 1720 Port: [61440-61444]

source port range defined on the in-network-region form

SIP (UDP/IP)

SIP Registrar/Proxy

Port: 5060

Media Gateway or another IP endpoint

RTP Audio (UDP/IP)

Port selected from the audio port range administered for the

RTCP (UDP/IP)

network region Port: audio port +1

Port: RTP port +1

(only active during a call if

RTCP is enabled)

Port:161

Note: SNMP is disabled by

default.

Port: (any unused)

Port: (50000)

RTCP (UDP/IP)

SNMP (UDP/IP)

CNA Registration (TCP/IP)

CNA Test Requests (UDP/IP)

CNA Test Results (UDP/IP)

(4610SW, 4620SW, 4621SW,

4622SW, 4625SW)

RTCP Monitor

Port: RTCPMON

SNMP MIB Viewer

Port depends on MIB viewer admin

Avaya Converged Network Analyzer

Port: CNAPORT Port: any

Port: as specified in the test request

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Figure 2: Initialization and Address Resolution Diagram

Initialization and Address Resolution

TCP/UDP Port Utilization

4600 Series IP T elephone

Port: 68

Port: [1024 - 5000]

Operating System

–selected

Port: [1024 - 5000]

Operating System –

selected

Port: [1024 - 5000]

Operating System

–selected (a new port

is used for each file

requested)

DHCP (UDP/IP)

HTTPS (TCP/IP)

(4601, 4601+, 4602, 4602SW,

4602SW+, 4610SW, 4620, 4620SW,

4621SW, 4622SW, & 4625SW)

4602SW+, 4610SW, 4620, 4620SW,

TFTP Read Request (UDP/IP)

TFTP Data, ACKs & Errors (UDP/IP)

HTTP (TCP/IP)

(4601, 4601+, 4602, 4602SW,

4621SW, 4622SW, & 4625SW)

DHCP Server

Port: 67

HTTPS Server

Port:411 R2.8+: TLSPORT

HTTP Server

Port: 80 (81 on CM)

R 2.8+: HTTPPORT

TFTP Server

Port: 69 Port: Operating

System - selected (a new port is used for each file transferred)

Port: [1024 - 5000]

Operating System

–selected

DNS Server

DNS (UDP/IP)

Port: 53

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Overview of Voice over IP (VoIP) and Network Protocols

Figure 3: Applications Diagram

Applications

4600 Series IP Telephone

Port: [1024 – 5000]

Operating System –

selected (only active if

DIRSRVR is non-null)

Port: [1024 – 5000]

Operating System –

selected (only active if

WEBHOME (4630),

VMLHOME (4630) or

WMLHOME (4620) is

non-null

Port: 21

(only active if user enters

FTP server IP Address)

Port: 20

(only active during

a backup or restore)

Port: [49714 - 49721]

49721 unless changed

via CTIUDPPORT

(only active if

CTISTAT is 1,

phone is fully registered,

and signaling-channel

encryption is not being

used)

4621SW, 4622SW, 4625SW,

LDAP (TCP/IP)

(4630 only)

HTTP (TCP/IP)

(4610SW, 4620, 4620SW,

4621SW, 4622SW, 4625SW,

and 4630 only)

HTTP over SSL (TCP/IP)

(4630 only)

FTP control (TCP/IP)

(4610SW, 4620, 4620SW,

and 4630 only)

FTP data (TCP/IP)

(4610SW, 4620, 4620SW,

4621SW, 4622SW, 4625SW,

and 4630 only)

CTI Discovery (UDP/IP))

(CTI is not supported by the

4601, 4601+, or 4602)

Directory Server

Port: 389, or as set by DIRLDAPPORT

Web or Proxy Server

Port: Usually 80 for Web servers and 8000 for proxy servers, but URLs may specify other ports as well

Port: 443

FTP Server

Port:21

Port: 20

IP Softphone

Port: [50000 – 51000] OS-selected

Port: 49722

(only active if CTISTAT is 1,

phone is fully registered,

and signaling-channel

encryption is not being

42 4600 Series IP Telephone LAN Administrator Guide

used)

CTI Data (TCP/IP)

(CTI is not supported by the

4601, 4601+, or 4602)

Port: [1024 – 5000] randomly selected

Page 43

Applications, continued

TCP/UDP Port Utilization

4600 Series IP Telephone

Port: 5000

(only active if

IRSTAT is 1)

Port: [1024 - 5000]

Operating System -

selected (only active if

SMTPSRVR is non-null

and if IRSTAT is 1)

Port: [80]

(for Pre-Release 2.6)

Port: PUSHPORT

(for Release 2.6 and later)

Port: [any unused]

Another 4600 Series IP Telephone

IrOBEX (UDP/IP)

Port: 5000

(IR Object Relay is only

supported by the 4620 and

4620SW)

Mail Server

SMTP (TCP/IP)

Port: 25

(IR Object Relay is only

supported by the 4620 and

4620SW)

Push Server

HTTP (TCP/IP)

Traceroute (UDP/IP)

Port: any

Port: [33434 +1 for each hop]

Port: [any unused]

Syslog (UDP/IP)

Syslog Server

Port: [514]

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Suggestions for Installation and Configuration

Reliability and Performance

There is a cost/performance trade-off associated with Voice over IP. Greater reliability and improved performance can be obtained through server redundancy and components with higher bandwidth capabilities.

The reliability and performance of the traditional PBX systems are very high to date. Much of the LAN is outside the PBX’s control. However, consider the points in the next paragraph to enhance the reliability and performance of the IP telephone network.

All 4600 Series IP Telephones support the tools “ping” and “traceroute.” These are standard LAN/WAN tools to identify:

● whether two points on a network can communicate with each other, and

● what path a sample communication takes as it traverses the network from one point to the

other.

As of R2.6, the telephone will not respond to a traceroute if Internet Control Message Protocol Destination Unreachable transmission (ICMPDU

All 4600 Series IP Telephones respond appropriately to a ping or a traceroute message sent from the Communication Manager or any other source on your network. These telephones will not, in general, initiate a ping or traceroute. Release 1.6 of the 4600 Series IP Telephones supporting H.323 introduced “remote ping” and “remote traceroute” support. The switch can instruct such a 4600 Series IP Telephone to initiate a ping or a traceroute to a specified IP Address. The telephone carries out that instruction and sends a message to the switch indicating the results. See your Communication Manager Administration documentation for more details.

As of Release 1.8, if applicable, 4600 Series IP Telephones test whether the network Ethernet switch port supports IEEE 802.1D/q tagged frames by ARPing the router with a tagged frame. See VLAN Considerations your router must respond to ARPs for VLAN tagging to work properly.

on page 91. If your LAN environment includes Virtu al LANs (VLANs),

) has been set to zero.

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Suggestions for Installation and Configuration

IP Address Lists and Station Number Portability

Release 1.5 of the 4600 Series IP Telephones provided the capability to specify IP Address list s in either dotted decimal or DNS format. Release 1.5 allowed key network elements to have multiple IP Addresses, rather than being restricted to just one address for each element. You can specify up to 127 total characters in each list of the following devices:

● router/gateways,

● DHCP/TFTP/HTTP servers, and

● the media server.

Upon startup or a reboot, the 4600 Telephone attempts to establish communication with these various network elements in turn. The telephone starts with the first address on the respective list. If the communication is denied or times out, the telephone procee ds to the next address on the appropriate list and tries that one. The telephone does not report failure unless all the addresses on a given list fail.

Obviously, this capability significantly improves the reliability of IP telephony. Multiple IP Addresses maximize the telephone’s likelihood to communicate with backup equipment if the primary equipment is not operating or is not accessible. For example, alternate communication would be needed during a limited network outage.

However, this capability also has the advantage of making station number portability easier. Assume a situation where the company has multiple locations in London and New York, all sharing a corporate IP network. Users want to take th eir telephones from their offices in Lo ndon and bring them to New York. When users start up their telephones in the new location, the local DHCP server will generally route them to the local call server. In this case the call server for H.323 is the Avaya Media Server and the call server for SIP is the Registration Server. But the local call server denies service because it knows nothing about these new users. With proper administration of the local DHCP server, the telephone knows to try a second call server IP Address, this one in London. The user can then be automatically registered with the London call server.

Chapter 4: Server Administration

contains details on administration of DHCP servers for lists of alternate media servers, router/gateways, and TFTP servers. For specific information, see DNS

Addressing on page 114.

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Security

In VoIP, physical wire is replaced with an IP connection. The connection is more mobile. Unauthorized relocation of the IP telephone allows unauthorized users to send and receive calls as the valid owner. For further details on toll fraud, see the DEFINITY Communication Manager documents mentioned in Related Documents

Any equipment on a data network, including a 4600 Series IP Telephone, can be the target of a Denial of Service attack. Usually, such an attack consists of flooding the network with so many messages that the equipment either:

● spends so much time processing the messages that legitimate tasks are n ot processed, or

● the equipment overloads and fails.

The 4600 Series IP Telephones cannot guarantee resistance to all Denial of Service attacks. However, each Release has increasing checks and protections to resist such attacks while maintaining appropriate service to legitimate users.

All 4600 Series IP Telephones that run R2.2 or greater software support Transport Layer Security (TLS). This standard allows the telephone to establish a secure connection to a HTTPS server, in which the telephone’s upgrade and settings file can reside. This setup adds security over the TFTP alternative.

or Avaya

on page 18.

You also have a variety of optional capabilities to restrict or remove how crucial network information is displayed or used. These capabilities are covered in more detail in

Chapter 4: Server Administration

● As of Release 2.8, 4600 Series IP T elephones use IP source address filtering to improve

, and include:

their resiliency to denial of service attacks when only services that require messages from known IP Addresses are enabled (i.e., when ICMPDU is 0, where RTCPMON is null, when CNASRVR is null, when CTISTAT is 0, and when SNMPSTRING is null or when SNMPSTRING and SNMPADDR are both non-null).

Additional IP source addresses can be explicitly excluded from filtering, if necessary, through use of the FILTERLIST

● As of Release 2.8, 4600 Series IP Telephones require that any DNS names used as

parameter, which is also new in Release 2.8.

values of TPSLIST be fully-qualified to improve the security of the Push feature. For more details on the Push feature, Appendix E: The Push Feature

● As of Release 2.8, 4600 Series IP T elephones support IETF RFC 1948 (Defending Against

Sequence Number Attacks).

● As of Release 2.7, the 4602SW+ and 4625SW IP Telephones support IEEE 802.1X as a

Supplicant with the EAP-MD5 authentication method. The functionality is identical to other 4600 Series SW IP Telephones supporting this feature.

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Suggestions for Installation and Configuration

● As of Release 2.6, the 4610SW, 4620SW, 4621SW, and 4622SW IP Telephones support

IEEE 802.1X as a Supplicant with the EAP-MD5 authentication method. The modes supported are as follows:

- Unicast Supplicant operation only with PAE multicast pass-through, with and without proxy Logoff, and

- Unicast or multicast Supplicant operation without PAE multicast pass-through or proxy Logoff.

Note:

Note: The 4601 and 4601+ IP Telephones do not support 802.1X as a Supplicant.

● As of Release 2.6, SNMP is disabled by default. You must enable SNMP through DHCP or

the 46xxsettings file.

● As of Release 2.3, 4600 Series IP Telephones support VLAN separation.

● As of Release 2.3, 4600 Series H.323 IP Telephones support signaling channel encryption

while registering, and when registered, with appropriately administered Avaya Media Servers.

● As of Release 2.0, a 4600 Series IP T elephone’ s response to SNMP queries is restricted to

only IP Addresses on a list you specify.

● As of Release 2.0, an SNMP community string is specified for all SNMP messages sent by

the telephone.

● As of Release 1.8, dialpad access to Local Administration Procedures, such as specifying

IP Addresses, is restricted by a password.

● Dialpad access to most Local Administration Procedures was removed.

● The end user’s ability to use a telephone Options application to view network data is

restricted.

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Chapter 3: Requirements

Introduction

The 4600 Series IP Telephones use Internet Protocol (IP) technology with Ethernet line interfaces. The IP telephones supplement the existing Avaya IP Solutions platform. This feature provides the user with the capability to natively administer and maintain new 4600 Series IP Telephones.

The 4600 Series IP Telephones provide support for DHCP, TFTP, and HTTP over IPv4/UDP, which enhance the administration and servicing of the phones. These phones use DHCP to obtain dynamic IP Addresses and TFTP or HTTP to download new versions of software for the phones.

Using either a built-in hub or a switched port, the 4600 Series IP Telephones offer one desktop connection for both the telephone set and the PC.

Note:

Note: In all cases, references to “Avaya Communication Manager” or “CM” apply only

to 4600 Series IP Telephones supporting H.323.

Hardware Requirements

Before plugging in the 4600 Series IP Telephone, verify that all the hardware requirements are met. Failure to do so prevents the telephone from working and might have a negat ive impact on your network.

The following hardware is required for 4600 Series IP Telephones supporting H.323 to work properly.

Note:

Note: The recommended configuration is the latest PBX software and the latest IP

telephone firmware. In the event your site does not have the latest PBX software, follow the recommendations in the table immediately following.

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Requirements

Media Server Release (H.323 only)

Avaya Communication Manager 4.0

Avaya Communication Manager 3.1

Avaya Communication Manager 3.0

Avaya Communication Manager 1.3+

Avaya Communication Manager 1.1,

Avaya Communication Manager 1.2

R10, Avaya Communication Manager 1.1,

Avaya Communication Manager 1.2

Avaya IP Telephone

IP Telephone Release

Notes

All telephones R2.8+

4601+, 4602+,

R2.7

and 4625SW All telephones

R2.6 except 4601, 4602, 4602SW, and 4625SW

All telephones R1.8+ Use the latest release.

All telephones

R1.8+ Use the latest release. except 4630

4630 R1.74 Upgrade to Avaya

Communication Manager Release 1.3 or later before installing R1.8 on 4630 Telephones.

R10 4606,

4612,

R1.8+ The 4602 and 4620 are not

supported.

4624

R9.5 4606,

4612, 4624

R1.8+ The 4620, 4602, and 4630 are

not supported. R1.5 is the minimum 4600 IP Telephone vintage.

R9 4612,

4624

R8.4 4612,

4624

R1.1 R1.1 is the only supported

4600 IP Telephone vintage.

R1.0 R1.0 is the only supported

4600 IP Telephone vintage.

4600 Series IP Telephones supporting SIP need Avaya Converged Communications Server (CCS) Release 3.0, which includes SIP Enablement Services (SES), to work properly. Avaya Communication Manager is considered a “feature server” behind SES that provides Off-PBX Station (OPS) features.

50 4600 Series IP Telephone LAN Administrator Guide

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Additional Hardware Requirements

● Ensure that the appropriate circuit pack(s) are administered on your media server. See the

media server’s hardware guide for more detail.

Important:

Important: IP telephone firmware Release 2.3 or greater requires TN799C V3 or greater

Control-LAN (C-LAN) circuit pack(s). For more information, see the Communication Manager Software and Firmware Compatibility Matrix on the Avaya support Web site

http://www.avaya.com/support

● A Category 5e LAN. If the telephones are to be powered from the LAN, the power supply

must be designed to the IEEE 802.3af-2003 standard for LAN powering.

● Electrical power provided to each telephone by one of the following two sources:

- A Telephone Power Module, also called the DC power jack. You must order this module separately, except for the 4630 and the 4690 phones. The 4630 comes with its own power brick and the 4690 has its own power interface module. The 4630SW does not come with a power brick. For the 4630SW, you must order the power brick separately if LAN powering will not be used for that particular telephone model.

Hardware Requirements

- IEEE 802.3af-2003, if the LAN supports this powering scheme. Note that the 4630 and 4690 cannot be powered this way, but the 4630SW can be powered this way.

● Verify that the 4600 Series IP Telephone package includes the following components:

- 1 telephone set.

- 1 telephone handset. Note that the 4622SW and the 4690 telephones do not come with handsets.

- 1 H4DU 4-conductor coiled handset cord that is 9 feet long when extended, plugged into the telephone and the handset. The handset cord for the 4601 and 4601+ is 6 feet long. Not applicable for the 4622SW and 4690 IP Conference Telephones.

- 1 Category 5 modular line cord for the connection from the IP telephone to the Ethernet wall jack.

- 4600 Series IP Telephone Safety Instructions (555-233-779).

- Power Interface Module for the 4690 IP Conference Telephone only.

- Power Brick for 4630 IP Telephones only.

- Stylus for 4630/4630SW IP Telephones only.

● You might need a Category 5e modular line cord for the connection from the 4600 Series

IP Telephone to the PC.

Note:

Note: See the 4600 Series IP Telephone Installation Guide.

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Requirements

Software Requirements

The following software is required for 4600 Series IP Telephones to work properly:

● The DHCP server and application must be installed and properly administered, as

described in DHCP

CAUTION:

CAUTION: A DHCP server is not mandatory, but static addressing is necessary when a

DHCP server is unavailable. Because of difficulties associated with static addressing, we very strongly recommend that a DHCP server be installed and that static addressing be avoided.

● The TFTP and/or HTTP file server and application must be properly administered, as

described in TFTP (H.323 Only)

CAUTION:

CAUTION: A file server does not need to be available for the Avaya IP Telephones to

operate. The Avaya IP Telephones obtain important information from the script files on the file server and depend on the application file for software upgrades. If the file server is not available when the Avaya IP Telephones reset, the telephones will register with the media server and operate. Some features may not be available, and restoring those features requires resetting the Avaya IP Telephone(s) when the file server is available.

● For 4630 and 4630SW IP Telephone environments, if users are to have access to LDAP

directories or corporate Web sites, the appropriate servers must be in place. The 4630/ 4630SW Telephones must be appropriately administered in accordance with Server

Administration on page 53.

CAUTION:

CAUTION: 4630 IP Telephone Release 1.72 introduced significant software architecture

changes. Thus, unlike most 4600 Series IP Telephones software releases, 4630 IP Telephone Release 1.72 and later cannot be downgraded to a release earlier than 1.72. Attempting to do so renders the 4630 (and 4630SW, if Release 1.8 or greater) set inoperable. In addition, if you are upgrading a 4630 from a release prior to Release 1.61, you must first upgrade to Release 1.61. Then you must upgrade to the newer Release. You cannot upgrade directly from a pre-1.61 Release to a post-1.61 Release for the 4630.

● For 4610SW/4620/4620SW/4621SW/4622SW/4625SW IP Telephone environments, if

users are to have access to LDAP directories or corporate WML Web sites, the appropriate servers must be in place. You must download the LDAP Directory Application software from the Avaya support Web site. You must appropriately administer the telephones in accordance with Server Administration

Note:

Note: Ensure that all required parameters are configured correctly. For Avaya Media

Server information, see your administration documentation. For the DHCP and file servers, see Chapter 4: Server Administration

on page 60.

on page 73 and HTTP on page 75.

on page 53.

52 4600 Series IP Telephone LAN Administrator Guide

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Chapter 4: Server Administration

Introduction

When a 4600 Series IP Telephone is plugged in and powered, it automatically negotiates with its associated LAN to determine the Ethernet speed. From that point on, the telephone’s actions depend largely on network administration prior to telephone installation, and on any actions the installer takes. This chapter details the parameters and other data the telephone needs to operate, and the alternatives to deliver that information to the telephone, where appropriate. Recommendations and specifications for alternatives to certain parameters are also provided.

The parameters under which the telephone needs to operate are summarized as follows:

● Telephone Administration on the call server. The call server for H.323 is the Avaya Media

Server, while the call server for SIP is the Registration Server.

● IP Address management for the telephone.

● Tagging Control and VLAN administration for the telephone, if appropriate.

● Quality of Service (QoS) administration for the telephone, if appropriate.

● Site-specific Option Number (SSON) setting of DHCP servers, if appropriate.

● Protocol administration, for example, Simple Network Management Control (SNMP) and

Link Layer Discovery Protocol (LLDP).

● Interface administration for the telephone, if appropriate.

● Application-specific telephone administration, if appropriate. For example, Directory- or

Web-specific information required for these optional applications.

The delivery mechanisms are:

● Maintaining the information on the call server.

● Manually entering the information using the telephone dialpad.

● Administering the DHCP server.

● Editing the settings file on the applicable TFTP or HTTP file server.

These parameters can be administered in a variety of ways, as indicated in Table 2

. Note that

not all parameters can be administered on all delivery mechanisms.

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Table 2: Administration Alternatives and Options for 4600 Series IP Telephones

Administrative

Parameter(s)

Telephone Administration

Mechanisms

Avaya call server (applies only to H.323)

For More Information See:

Administering 4600 Series IP Telephones on Avaya Media Servers (H.323 Only) on page 57

and Related Documents

IP Addresses DHCP

(strongly recommended) Manual administration at

the telephone

Tagging and

LLDP See Link Layer Discovery Protocol (LLDP)

VLAN

DHCP and File Servers on page 58, and

especially DHCP “Static Addressing Installation” in Chapter 3 of

the 4600 IP Telephone Installation Guide. page 97.

DHCP DHCP and File Servers

Administering Options for the 4600 Series IP Telephones on page 101. Also see VLAN Considerations on page 91 for background

information.

Settings file (strongly recommended)

DHCP and File Servers Administering Options for the 4600 Series IP Telephones on page 101.

Quality of Service

Manual administration at the telephone

Avaya call server (applies only to H.323)

“Static Addressing Installation” in Chapter 3 of the 4600 IP Telephone Installation Guide.

Administering 4600 Series IP Telephones on Avaya Media Servers (H.323 Only) on page 57

and Related Documents on page 18.

DHCP DHCP and File Servers on page 58, and

Administering Options for the 4600 Series IP Telephones on page 101.

Settings file (strongly recommended)

DHCP and File Servers on page 58, and Administering Options for the 4600 Series IP Telephones on page 101.

Manual administration at the telephone

See “QoS Option Setting” in Chapter 3 of the 4600 IP Telephone Installation Guide.

Interface DHCP DHCP and File Servers

Administering Options for the 4600 Series IP Telephones on page 101.

Settings file (strongly recommended)

DHCP and File Servers Administering Options for the 4600 Series IP Telephones on page 101.

Manual administration at the telephone

See “Secondary Ethernet (Hub) Interface Enable/Disable” in Chapter 3 of the 4600 IP Telephone Installation Guide.

on page 18.

on page 60.

on page 58, and

on page 58 and

on page 58, and

1 of 2

54 4600 Series IP Telephone LAN Administrator Guide

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Table 2: Administration Alternatives and Options for 4600 Series IP Telephones (continued)

Administrative

Parameter(s)

Mechanisms

For More Information See:

SSON DHCP Customizing the Site-Specific Option Number

(SSON) on page 114. DHCP and File Servers on

Settings file (strongly recommended)

page 58, and especially DHCP

Customizing the Site-Specific Option Number (SSON) on page 114. DHCP and File Servers on

on page 60.

page 58, and especially TFTP Generic Setup page 74 and HTTP Generic Setup

Manual administration at the telephone

802.1X DHCP IEEE 802.1X Settings file IEEE 802.1X Manual administration at

the telephone

“Site-Specific Option Number Setting ” in Chapter 3 of the 4600 IP Telephone Installation Guide.

on page 94. on page 94.

“802.1X Supplicant Authentication” and “Set the

802.1X Operational Mode” in the 4600 Series IP

Telephone Installation Guide.

SNMP DHCP DHCP on page 60.

Settings file 4600 Series IP Telephone Scripts and

Application Files on page 78 and 4600 Series IP Telephone Customizable System Parameters on

page 102.

Application specific parameters

DHCP DHCP and File Servers

especially DHCP on page 60. Also, Customizing the 4630/4630SW IP

on page 58, and

Telephone on page 117 and Customizing 4610SW , 4620/4620SW , 4621SW , 4622SW , and 4625SW IP Telephones on page 122.

Settings file (strongly recommended)

DHCP and File Servers TFTP (H.323 Only)

on page 73. Also,

on page 58, especially

Customizing the 4630/4630SW IP Telephone on

page 117 and Customizing 4610SW, 4620/

4620SW, 4621SW, 4622SW, and 4625SW IP Telephones on page 122.

LLDP Link Layer Discovery Protocol (LLDP)

page 97.

Introduction

on page 75.

2 of 2

General information about administering DHCP servers is covered in DHCP and File

Servers on page 58, and more specifically, DHCP on page 60. General information about

administering TFTP servers is covered in DHCP and File Servers

, and more specifically, TFTP

(H.323 Only) on page 73. General information about administering HTTP servers is covered in DHCP and File Servers

, and more specifically, HTTP. Once you are familiar with that material, you can administer telephone options as described in Administering Options for the 4600 Series

IP Telephones on page 101.

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Parameter Data Precedence

If a given parameter is administered in multiple places, the last server to provide the parameter has precedence. The precedence, from lowest to highest, is:

● LLDP,

● manual administration, with the two exceptions described for the system parameter

STATIC

● DHCP,

● TFTP/HTTP,

● the call server (meaning the Avaya Media Server for H.323, and the Registration Server

for SIP), and finally,

● FTP/HTTP backup files (if administered and if permitted).

Settings the IP telephone receives from backup files or the media server overwrite any previous settings including manual settings. The only exception to this sequence is in the case of VLAN IDs. In the case of VLAN IDs, LLDP settings of VLAN IDs are the absolute authority. Then the usual sequence applies through TFTP or HTTP as appropriate. If the VLAN ID is not zero, any VLAN ID from the media server is ignored.

on page 112,

Administering H.323 and SIP IP Telephones on the Same Network

Both H.323- and SIP-based telephones can run on the same LAN or VLAN without difficulty. You can even mix H.323 phones and SIP phones having the same model, for example, the 4620SW. However, any given telephone supports only one signaling protocol at a time. You must therefore ensure that the proper application files (the software that runs in the telephone) are sent to the appropriate telephones. For information on downloading the software to the telephones, see Choosing the Right Application File and Upgrade Script File chapter.

In general, H.323 and SIP telephones use the same administration mechanisms, with the exception that H.323 phones get some telephony administration from the Avaya call server. If you intend to mix H.323 and SIP telephones on the same network, you can use common information in either DHCP or in a common script file for those settings commo n to both types of sets. Because settings not applicable to a given telephone set are ignored, you can send SIP-specific settings to an H.323 telephone and H.323 settings to a SIP telephone without creating problems.

later in this

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Administering 4600 Series IP Telephones on Avaya Media Servers (H.323 Only)

DEFINITY Releases 9, 9.5, 10, and Avaya Communication Manager Software Release 1.1+

DEFINITY® Releases 9 and 9.5 provide support for the 4606, 4612, and 4624 IP Telephones.

DEFINITY

Release 10 adds support for the 4630 and 4630SW IP Telephones. Avaya Communication Manager Software Release 1.1 adds support for the 4602/4602SW and 4620/4620SW IP Telephones. Avaya Communication Manager 3.1 supports the 4601+ and 4602SW+ IP Telephones. Avaya Communications Manager 4.0 adds a Time -to-Service feature that decouples registration and TCP socket establishment (for H.323 only. Administration of a 4612 and 4624 IP Telephone is identical to a 6424 IP softphone. The 4610SW an d 4690 are not natively supported, but can be aliased as 4620 IP Telephones. See Related Documents page 18, particularly the Administration for Network Connectivity and the Administrator Guides. Follow these guidelines:

● On the Customer Options form, verify that the IP Stations field is set to “y” (Yes). If it is

not, contact your Avaya sales representative.

● The IP Softphone field does not have to be set to “y” (Yes).

DEFINITY Release 8.4

Note:

Note: DEFINITY

DEFINITY

Release 8.4 supports the 4612 and 4624 IP Telephones. The 4612 and 4624 IP Telephones are aliased as 6424 Telephones, administered as IP Softphones. The administrative forms for the 6424 IP Softphone are used for the two IP telephones. See Related

Documents on page 18, particularly the Administration for Network Connectivity and the

Administrator Guides. Follow these guidelines:

Release 8.4 is very old. We do not recommend using this release.

● Alias the IP telephone as a 6424D+ DCP set, with the IP Softphone field set to “y” (Yes).

● Administer a Media Complex Ext for the audio channel.

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DHCP and File Servers

Dynamic Host Configuration Protocol (DHCP) provides a means by which configuration parameters can be automatically assigned to clients on a TCP/IP network. DHCP minimizes a 4600 Series IP Telephone network’s maintenance by removing the need to individually assign and maintain IP Addresses and other parameters for each IP telephone on the network.

Software Checklist

Ensure that you have purchased and/or own licenses to install and use any or all of the DHCP, TFTP, and HTTP server software.

Note:

Note: It is possible to install the DHCP, TFTP, and HTTP server software on the same

machine.

CAUTION:

CAUTION: The circuitry in the 4600 Series IP Telephones reserves IP Addresses of the form

192.168.2.x for internal communications. The telephone(s) will not properly use addresses you specify if they are of that form.

Required Network Information

DHCP is the control point where an enterprise controls its IP telephones. Before administering DHCP and TFTP, HTTP, and TLS, as applicable, complete the information in Table 3: Required

Network Information Before Installation - Per DHCP Server on page 59. Completing the pre-

installation steps ensures that you have the necessary information regarding your network. If you have more than one Gateway, TFTP/HTTP/TLS server, subnet mask, and Gatekeeper in your configuration, you need to complete Table 3

Release 1.5 of the 4600 Series Telephones supported specifying a list of IP Addresses for a gateway/router, TFTP server, and Avaya Media Server Gatekeeper(s). We explain this specification in Overview of Voice over IP (VoIP) total ASCII characters, with IP Addresses separated by commas with no intervening spaces. Note that depending on the specific DHCP application, only 127 characters might be supported.

for each DHCP server.

on page 27. Each list can contain up to 255

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Required Network Information

When specifying IP Addresses for the file server or media server, use either dotted decimal format (“xxx.xxx.xxx.xxx”) or DNS names. If you use DNS, note that the system value DOMAIN is appended to the DNS names you specify. If DOMAIN is null, the DNS names must be fully qualified, in accordance with IETF RFCs 1034 and 1035. For more specific information about DNS, see DHCP Generic Setup

on page 61 and DNS Addressing on page 114.

Table 3: Required Network Information Before Installation - Per DHCP Server

1. Gateway (router) IP Address(es)

2. TFTP server IP Address(es) H.323 only, if applicable.

3. Subnet mask

4. Avaya Media Server

H.323 only.

Gatekeeper IP Address(es)

5. Avaya Media Server Gatekeep er port

H.323 only. Although this can be a value between 0 and 65535, the default value is

1719. Do not change the default value unless that value conflicts with an existing port assignment.

6. TFTP server file path H.323 only, if applicable.

7. Telephone IP Address range

From: To:

8. DNS server address(es) If applicable.

9. HTTP server address(es) If applicable.

10. HTTPS/TLS server

If applicable.

address(es)

The file server file path is the “root” directory the server uses for all transfers. This is the default directory all files will be uploaded to or downloaded from. In configurations where the upgrade script and application files are in the default directory, do not use item 6

in Table 3: Required

Network Information Before Installation - Per DHCP Server on page 59.

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DHCP

This section provides basic information on DHCP servers and generic information on DHCP server administration.

Choosing a DHCP Configuration

A discussion on how to best set up your network to work with the 4600 Series IP Telephones is beyond the scope of this document. See Network Assessment concentrates on the simplest case of the single LAN segment. Information provided here can be extrapolated for more complex LAN configurations.

CAUTION:

CAUTION: Before you start, it is important that you understand your current network

configuration. An improper installation can cause network failures or reduce the reliability and performance of your network.

on page 33. This document

DHCP Software Alternatives

Three DHCP software alternatives are common to Windows operating systems:

● Windows NT

● Windows 2000

● Windows 2003

Any other DHCP application might work. It is the customer’s responsibility to install and configure the DHCP server correctly. This document is limited to describing a generic administration that works with the 4600 Series IP Telephones.

4.0 DHCP Server

DHCP Server

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DHCP Generic Setup

This document is limited to describing a generic administration that works with the 4600 Series IP Telephones. Three DHCP software alternatives are common to Windows operating systems:

● Windows NT

● Windows 2003

● Windows 2000

Any other DHCP application might work. It is the responsibility of the customer to install and configure the DHCP server correctly.

DHCP server setup involves:

1. Installing the DHCP server software according to vendor instructions.

2. Configuring the DHCP server with:

● IP Addresses available for the 4600 Series IP Telephones.

● The following DHCP options:

4.0 DHCP Server

DHCP Server

DHCP

- Option 1 - Subnet mask. As described in Table 3

, item 3.

- Option 3 - Gateway (router) IP Address(es). As described in Table 3

, item 1. If using more than one address, the total list can contain up to 255 total ASCII characters. You must separate IP Addresses with commas with no intervening spaces.

- Option 6 - DNS server(s) address list. If using more than one address, the total list can contain up to 127 total ASCII characters. Y ou must separate IP Addresses with commas with no intervening sp aces. At least one address in Option 6 must be a valid, non zero, dotted decimal address. Otherwise, DNS will fail.

- Option 15 - DNS Domain Name. This string contains the domain name to be used when DNS names in system parameters are resolved into IP Addresses. This domain name is appended to the DNS name before the 4600 IP Telephone attempts to resolve the DNS address. Option 15 is necessary if you want to use a DNS name for the TFTP server. Otherwise, you can specify a DOMAIN as part of customizing TFTP as indicated in

DNS Addressing

on page 114.

- Option 51 - DHCP lease time. Optional. Avaya recommends a lease time of six weeks or greater. Expired leases cause Avaya IP Telephones to reboot.Avaya recommends providing enough leases so an IP Address for an IP telephone does not change if it is briefly taken offline.

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Note:

Note: The DHCP standard states that when a DHCP lease expires, the device should

immediately cease using its assigned IP Address. If the network has problems and the only DHCP server is centralized, the server is not accessible to th e given telephone. In this case the telephone is not usable until the server can be reached.

Avaya recommends that, once assigned an IP Add ress, the telephone continues using that address after the DHCP lease expires, until a conflict with another device is detected. As Table 10: 4600 Series IP Telephone Customizable

System Parameters indicates, the system parameter DHCPSTD allows an

administrator to specify that the telephone will either:

- Comply with the DHCP standard by setting DHCPSTD to “1”, or

- Continue to use its IP Address after the DHCP lease expires by setting DHCPSTD

The latter case is the default. If the default is invoked, after the DHCP lease expires the telephone sends an ARP Request for its own IP Address every five seconds. The request continues either forever, or until the telephone receives an ARP Reply. After receiving an ARP Reply, the telephone displays an error message, sets its IP Address to 0.0.0.0, and attempts to contact the DHCP server again.

to “0.”

- Option 52 - Overload Option, if desired. If this option is received in a message, the telephone interprets the sname and file fields in accordance with RFC 2132, Section 9.3.

- Option 58 - DHCP lease renew time, if desired. If not received or if this value is greater than that for Option 51, the d efault value of T1 (renewal timer) is used as per RFC 2131, Section 4.5.

- Option 59 - DHCP lease rebind time, if desired. If not received or if this value is greater than that for Option 51, the d efault value of T2 (rebinding timer) is used as per RFC 2131, Section 4.5.

- Option 66 - TFTP Server Name. Applicable to H.323 only.

Note:

Note: Microsoft DHCP servers support only dotted-decimal format for file server

addresses, not symbolic names. Option 66 need not be used if the TFTP server is identified in the Site Specific Option Number string (Option 176), or if HTTP is to be used exclusively instead of TFTP. However , to simplify configuration, A vaya recommends that you use Option 66 if you are using TFTP. If you use both Option 66 and Option 176 to identify TFTP servers, the value(s) in Option 176 overrides the value(s) in Option 66.

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DHCP

- A 4600 Series IP Telephone-specific DHCP option specifying information such as: TFTP server and Avaya Media Server Gatekeeper IP Addresses, or SIP-specific parameters, such as SIPPROXYSRVR, as defined in 4600 Series IP Telephone

Customizable System Parameters. Use the Site-Specific Option Number (SSON) at

#176. You can set this option value, for example, to either of the following strings:

MCIPADD=xxx.xxx.xxx.xxx,MCPORT=yyyy,TFTPDIR=<path>,TFTPSRVR= zzz.zzz.zzz.zzz

MCIPADD={list of DNS names}, MCPORT=yyyy, TFTPDIR=<path>, TFTPSRVR= {list of DNS names}

As of Release 2.4, also use Option 176 to identify the VLAN IDs used for the Ethernet line interface and the secondary Ethernet interface for VLAN separation. In this instance, if your voice VLAN ID is xxx and your data VLAN ID is yyy, at a minimum, add the following string to Option 176:

L2Q=1,L2QVLAN=xxx,PHY2VLAN=yyy The following is the recommended format for VLAN separation on dynamically

programmed IP telephones. In this example, L2Q=1,L2QVLAN=xxx,VLANSEP=1,PHY2VLAN=yyy,PHY2PRIO=z To enable VLAN separation, also set related para meters in the 46xxsettings.txt file, as

described in 4600 Series IP Telephone Scripts and Application Files

on page 78.

Note:

Note: The total length of the DHCP packet cannot exceed 1024 bytes, except for 4690 phones.

The total length of the DHCP packet for 4690 phones cannot exceed 576 bytes.

List the TFTPDIR value before the TFTPSRVR value, if the latter is specified in the SSON.

Some DHCP applications limit the length of Option 176 to 247 characters. You can have SIP-specific, H.323-specific, or common system parameters in the

same SSON Option. All such parameters are downloaded to all IP telephones, but only the relevant parameters for a given telephone are acted upon.

The 4600 Series IP Telephones ignore any vendor proprietary options such as Microsoft-specific Option 250, used by a Microsoft DHCP server when any option value exceeds 255 bytes.

The 4600 Series IP Telephones do not support Regular Expression Matching, and therefore, do not use wildcards. See Administering Options for the 4600 Series IP Telephones

on page 101.

In configurations where the upgrade script and application files are in the default directory, do not use the TFTPDIR=<path>.

You do not have to use Option 176. If you do not use this option, you must ensure that the key information, especially TFTPSRVR, MCIPADD, and MCPORT, is administered appropriately

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elsewhere. For example, when you specify the DNS server in Option 6, and the Domain Name in Option 15, you can use the configured names “A vayaTFTPServer” and “AvayaCallServer” fo r TFTPSRVR and MCIPADD, respectively. Upgra ding from I P teleph one Releases prior to R1.60 requires Option 176 to be minimally administered with MCIPADD.

Administer DHCP servers to deliver only the options specified in this document. Administering additional, unexpected options might have unexpected consequences, including possibly causing the IP telephone to ignore the DHCP server.

The media server name and TFTP server name must each be no more than 32 characters in length.

Note:

Note: Examples of good DNS administration include:

- Option 6: “aaa.aaa.aaa.aaa”

- Option 15: “yourco.com”

- Option 66: “yourTFTPserver,zzz.zzz.zzz.zzz”

- Option 176: “MCIPADD=xxx.xxx.xxx.xxx” Depending on the DHCP application you choose, be aware that the application most

likely will not immediately recycle expired DHCP leases. An expired lease might remain reserved for the original client for a day or more. For example, Windows NT DHCP reserves expired leases for about one day. This reservation period protects a client’s lease for a short time. If the client and the DHCP server are in two different time zones, the computers’ clocks are not in synch, or the client is not on the network when the lease expires, there is time to correct the situation.

The following example shows the implication of having a reservation period: Assume two IP Addresses, therefore two possible DHCP leases. Assume three IP telephones, two of which are using the two available IP Addresses. When the lease of the first two telephones expires, the third telephone cannot get a lease until the reservation period expires. Even if you remove the other two telephones from the network, the third telephone remains without a lease until the reservation period expires.

In Table 4

, the 4600 Series IP Telephone sets the system values to the DHCPACK message

field values shown.

Table 4: DHCPACK Setting of System Values

System Value Set to

DNSSRVR Option #6 (if received). DOMAIN Option #15 (if received). GIPADD The first four octets of Option #3 (if received). IPADD The yiaddr field.

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Table 4: DHCPACK Setting of System Values (continued)

System Value Set to

NETMASK Option #1 (if received). SNTPSRVR Option #42 (if received). TFTPSRVR First set to the value of the siaddr field (only if

non-zero), then set to the value of Option #66 (if received), then set to the value of TFTPSRVR in Option #176 (if any).

The rest of this section describes some common DHCP servers.

Windows NT 4.0 DHCP Server

DHCP

This section contains details on how to verify and configure the DHCP server included in the

Windows NT Use V erifying the Inst allation of the DHCP Server

4.0 server operating system. to verify whether the DHCP server is installed.

If it is not, install the DHCP server. If it is installed, go to the section Initial Configuration page 65 and the subsequent sections.

Verifying the Installation of the DHCP Server

Use the following procedure to verify whether the DHCP server is installed.

1. Select Start-->Settings-->Control Panel.

2. Double-click the Network icon.

3. Verify that Microsoft DHCP Server is listed as one of the Network Services on the Services tab.

4. If it is listed, go to the following section, Initial Configuration the DHCP server.

Initial Configuration

The Windows NT® 4.0 DHCP server configuration involves setting up a scope for the IP telephone. A DHCP scope is essentially a grouping of IP devices, in this case IP telephones, running the DHCP client service in a subnet. The scope defines parameters for each subnet. Each scope has the following properties:

. If it is not listed, then install

● A unique subnet mask used to determine the subnet related to a given IP Address.

● A scope name assigned by the administrator when the scope is created.

● Lease duration values to be assigned to DHCP clients with dynamic addresses.

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In addition, the DHCP server can assign configuration parameters to a client, and these can be specified for each DHCP scope.

Setting up the Windows NT

4.0 DHCP server requires the following steps:

1. Creating a DHCP scope for the IP telephones.

2. Editing custom options.

3. Adding the DHCP options.

4. Activating the new scope.

5. Verifying your configuration.

Each step is detailed in the next five sub-sections.

Creating a DHCP Scope for the IP Telephones

Use the following procedure to create a DHCP scope for the IP telephones.

1. Select Start-->Programs-->Admin Tools-->DHCP Manager.

2. Expand Local Machine in the DHCP Servers window by double clicking it until the + sign changes to a - sign.

3. Select Scope-->Create.

4. Define the range of IP Addresses used by the IP telephones listed in Table 3: Required

Network Information Before Installation - Per DHCP Server.

The Start Address is the first IP Address to be used for the IP telephones. The End Address is the last IP Address to be used for the IP telephones. Set the Subnet Mask to the value recorded in Table 3: Required Network Information

Before Installation - Per DHCP Server.

To exclude any IP Addresses you do not want assigned to IP telephones within the Start and End Addresses range:

a. In the Exclusion Range Start Address field, enter the first IP Address in the range that

you want to exclude.

b. In the Exclusion Range End Address field, enter the last IP Address in the range that

you want to exclude. c. Click the Add button. d. Repeat steps a. through c. for each IP Address range to be excluded.

Example: Suppose the range of IP Addresses available for your IP telephone network are:

● 135.254.76.7 to 135.254.76.80

● 135.254.76.90 to 135.254.76.200

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● 135.254.76.225 to 135.254.76.230

Your start address and end address are 135.254.76.7 and 135.254.76.230 respectively.

Exclude the ranges 135.254.76.81 to 135.254.76.89 and 135.254.76.201 to

135.254.76.224.

Note:

Note: We recommend that you provision the 4600 Series IP Telephones with

sequential IP Addresses. We recommend not mixing 4600 Series IP Telephones and PCs in the

same scope.

5. Under Lease Duration, select the Limited To option and set the lease duration to the maximum.

6. Enter a sensible name for the Name field, such as “DEFINITY IP Telephones.”

7. Click OK.

A dialog box prompts you: Activate the new scope now?

DHCP

8. Click No. Note:

Note: Activate the scope only after setting all options.

Editing Custom Options

Use the following procedure to edit custom options.

1. Highlight the newly created scope.

2. Select DHCP Options-->Defaults in the menu.

3. Click the New button.

4. In the Add Option T ype dialog box, enter an appropriate cu stom option name, for example, “46XXOPTION.”

5. Change the Data Type Byte value to String.

6. Enter 176 in the Identifier field.

7. Click the OK button.

The DHCP Options menu displays.

8. Select the Option Name for 176 and set the value string.

9. Click the OK button.

10. For the Option Name field, select 003 Router from the drop-down list.

11. Click Edit Array.

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12. Enter the Gateway IP Address recorded in Table 3: Required Network Information Before

Installation - Per DHCP Server for the New IP Address field.

13. Select Add and then OK.

Adding the DHCP Option

Use the following procedure to add the DHCP option.

1. Highlight the scope you just created.

2. Select Scope under DHCP Options.

3. Select the 176 option that you created from the Unused Options list.

4. Click the Add button.

5. Select option 003 from the Unused Options list.

6. Click the Add button.

7. Click the OK button.

8. Select the Global parameter under DHCP Options.

9. Select the 176 option that you created from the Unused Options list.

10. Click the Add button.

11. Click the OK button.

Activating the Leases

Use the following procedure to activate the leases.

1. Click Activate under the Scope menu. The light-bulb icon for the scope lights.

Verifying Your Configuration

This section describes how to verify that the 46XXOPTIONs are correctly configured for the Windows NT

Verify the Default Option, 176 46XXOPTION

Use the following procedure to verify the default option.

1. Select Start-->Programs-->Admin Tools-->DHCP Manager.

2. Expand Local Machine in the DHCP servers window by double clicking until the + sign

changes to a - sign.

4.0 DHCP server.

3. In the DHCP servers frame, click the scope for the IP telephone.

4. Select Defaults from the DHCP_Options menu.

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5. In the Option Name pull-down list, select 176 46XXOPTION.

6. Verify that the Value String box contains the correct string from DHCP Software

Alternatives on page 60.

If not, update the string and click the OK button twice.

Verify the Scope Option, 176 46XXOPTION

Use the following procedure to verify the scope option:

1. Select Scope under DHCP OPTIONS.

2. In the Active Options: scroll list, click 176 46XXOPTION.

3. Click the Value button.

DHCP

4. Verify that the Value String box contains the correct string from DHCP Generic Setup

page 61.

If not, update the string and click the OK button.

Verify the Global Option, 176 46XXOPTION

Use the following procedure to verify the global option:

1. Select Global under DHCP OPTIONS.

2. In the Active Options: scroll list, click 176 46XXOPTION.

3. Click the Value button.

4. Verify that the Value String box contains the correct value from DHCP Generic Setup

page 61. If not, update the string and click the OK button.

Windows 2000 DHCP Server

This section describes the configuration of the DHCP server in Windows 2000®.

Verifying the Installation of the DHCP Server

Use the following procedure to verify whether the DHCP server is installed.

1. Select Start-->Program-->Administrative Tools-->Computer Management.

2. Under Services and Applications in the Computer Management tree, find DHCP.

3. If DHCP is not installed, install the DHCP server. Otherwise, proceed directly to

Creating and Configuring a DHCP Scope

for instructions on server configuration.

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Creating and Configuring a DHCP Scope

Use the following procedure to create and configure a DHCP scope.

1. Select Start-->Programs-->Administrative Tools-->DHCP.

2. In the console tree, click the DHCP server to which you want to add the DHCP scope for the

IP telephones. This is usually the name of your DHCP server machine.

3. Select Action-->New Scope from the menu. Windows displays the New Scope Wizard to guide you through rest of the setup.

4. Click the Next button. The Scope Name dialog box displays.

5. In the Name field, enter a name for the scope such as “DEFINITY IP Telephones,” then

enter a brief comment in the Description field.

6. When you finish Steps 1 - 5, click the Next button. The IP Address Range dialog box displays.

7. Define the range of IP Addresses used by the IP telephones listed in Table 3: Required

Network Information Before Installation - Per DHCP Server. The Start IP Address is the

first IP Address available to the IP telephones. The End IP Address is the last IP Address available to the IP telephones.

Note:

Note: We recommend not mixing 4600 Series IP Telephones and PCs in the

same scope.

8. Define the subnet mask in one of two ways:

● The number of bits of an IP Address to use for the network/subnet IDs.

● The subnet mask IP Address.

Enter only one of these values. When you finish, click the Next button. The Add Exclusions dialog box displays.

9. Exclude any IP Addresses in the range specified in the previous step that you do not want

assigned to an IP telephone. a. In the Start Address field under Exclusion Range, enter the first IP Address in the

range you want to exclude.

b. In the End Address field under Exclusion Range, enter the last IP Address in the

range you want to exclude.

c. Click the Add button. d. Repeat steps a. through c. for each IP Address range that you want to exclude.

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Note:

Note: You can add additional exclusion ranges later by right clicking the Address Pool

under the newly created scope and selecting the New Exclusion Range option. Example: Suppose the ranges of IP Addresses available for your IP telephone network are:

● 135.254.76.7 to 135.254.76.80

● 135.254.76.90 to 135.254.76.200

● 135.254.76.225 to 135.254.76.230

The Start IP Address and End IP Address you enter in the IP Address Range dialog box are 135.254.76.7 and 135.254.76.230 respectively.

In the Add Exclusions dialog box, exclude the following ranges:

● 135.254.76.81 to 135.254.76.89

● 135.254.76.201 to 135.254.76.224

Click the Next button after you enter all the exclusions. The Lease Duration dialog box displays.

10. For all telephones that obtain their IP Addresses from the server, enter 30 days in the Lease Duration field. This is the duration after which a device’s IP Address expires and which the device needs to renew.

DHCP

11. Click the Next button. The Configure DHCP Options dialog box displays.

12. Click the No, I will activate this scope later button. The Router (Default Gateway) dialog box displays.

13. For each router or default gateway, enter the IP Address and click the Add button. When you are done, click the Next button.

The Completing the New Scope Wizard dialog box displays.

14. Click the Finish button. The new scope appears under your server in the DHCP tree. The scope is not yet active

and will not assign IP Addresses.

15. Highlight the newly created scope and select Action-->Properties from the menu.

16. Under Lease duration for DHCP clients, select 2 weeks (minimum) and click the OK

button.

CAUTION:

CAUTION: IP Address leases are kept active for varying periods of time. To avoid having

calls terminated suddenly, ensure that the lease duration is not too short, for example, not set to less than two weeks. A lease duration of two to four weeks is reasonable.

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Adding DHCP Options

Use the following procedure to add DHCP options to the scope you created in the previous procedure.

1. On the DHCP window, right-click the Scope Options fold er under the scope you created in the last procedure.

A drop-down menu displays.

2. Click the Configure Options... option.

The Scope Options dialog box displays.

3. In the General tab page, under the Available Options, check the 066’Boot Server Host Name’ Options checkbox.

The String Value dialog box displays.

4. Enter the TFTP Server address(es) in the String Value. Use the same TFTPSRVR value format as discussed in TFTP Generic Setup server at IP Address “zzz.zzz.zzz.zzz” and a second TFTP server at address “tftpserver.yourco.com,” in the string value enter:

on page 74. For example, if you had a TFTP

“zzz.zzz.zzz.zzz,tftpserver.yourco.com”

5. In the left pane of the DHCP, right click the DHCP Server name, then click

Set Predefined Options....

6. Under Predefined Options and Values, click Add.

7. In the Option Type Name field, enter any appropriate name, for example, “Avaya IP Telephones.”

8. Change the Data Type to String.

9. In the Code field, enter 176, then click the OK button twice.

The Predefined Options and Values dialog box closes, leaving the DHCP dialog box enabled.

10. Expand the newly created scope to reveal its Scope Options.

11. Click Scope Options and select Action-->Configure Options from the menu.

12. In the General tab page, under the Available Options, check the Option 176 checkbox.

13. In the Data Entry box, enter the DHCP IP telephone option string as described in

DHCP Generic Setup

on page 61.

Note:

Note: You can enter the text string directly on the right side of the Data Entry box under

the ASCII label.

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14. From the list in Available Options, check option 003 Rou ter.

15. Enter the gateway (router) IP Address from the IP Address field of Table 3: Required

Network Information Before Installation - Per DHCP Server.

16. Click the Add button.

17. Click the OK button.

Activating the New Scope

Use the following procedure to activate the new scope.

1. In the DHCP console tree, click the IP Telephone Scope you just created.

2. From the Action menu, select Activate. The small red down arrow over the scope icon disappears, indicating tha t the scope was

activated.

TFTP (H.323 Only)

This section describes how to set up a TFTP server for downloading software updates to the 4600 Series IP Telephones.

CAUTION:

CAUTION: The files defined by the TFTP server configuration have to be a ccessible from all

IP telephones. Ensure that the filenames match the names in the upgrade script, including case, since UNIX systems are case-sensitive.

Note:

Note: SIP IP telephones download upgrade script files, and hence, firmware and

settings files, from HTTP servers only. Y ou can use any TFTP application you w ant. However , Avaya recommends using

the TFTP server capability on the S8300 media serve r or the Ava ya IP Telephone File Server Application.

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TFTP Generic Setup

The following phases are involved in setting up a TFTP server.

● Install the TFTP server software.

● Configure the file path parameter to the directory where the files are to be stored. This is

the file path in Table 3: Required Network Information Before Installation - Per DHCP

Server on page 59. For increased security , we also recommend that you disable the ability

to upload to the server. Note that this option might not be available to all TFTP servers.

● Download the upgrade script file and application file from the Avaya Web site

(http://www.avaya.com/support

● Table 10, Table 11, and Table 12 list the parameters you can administer when manually

creating the TFTP script file. Manual administration is discussed in 4600 Series IP

Telephone Scripts and Application Files.

Note:

Note: Many LINUX servers distinguish between uppercase and lowercase names.

Ensure that you accurately specify the 46xxsettings filename, and the names and values of the data therein.

) to the directory as specified by the file path.

TFTP Server on S8300 Media Server

The S8300 Media Server provides all the TFTP support required for the 4600 Series IP Telephones. In addition, the media server has an easy to use, PC-based interface for creating script files. Thus, you do not need to manually create the text files discussed in 4600 Series IP

Telephone Scripts and Application Files. The media server creates the files for you. For more

information about the media server , see Downloading Avaya 46xx IP Telephone Software Using Avaya Media Servers, mentioned in Related Documents

Avaya File Server Application

The Avaya IP Telephone File Server Application provides a Windows or Linux TFTP or HTTP Server you can install on your own server. For more information, see Avaya IP Telephone File Server Application Reference Guide (Document # 16-601433), mentioned in Related

Documents and available from the Avaya support Web site.

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HTTP

This section gives general guidance to set up an HTTP server for downloading software updates to 4600 Series IP Telephones.

CAUTION:

CAUTION: The files defined by HTTP server configuration must be accessible from all IP

telephones invoking those files. Ensure that the file names match the names in the upgrade script, including case, since UNIX systems are case-sensitive.

Note:

Note: Use any HTTP application you want. In addition to the HTTP application on the

Avaya S8300 Media Server, other commonly used HTTP applications include Apache and Microsoft IIS.

HTTP Generic Setup

HTTP

These are the phases involved in setting up an HTTP server:

● Install the HTTP server application.

● Administer the system parameter HTTPSRVR to the address(es) of the HTTP server.

Include this parameter in DHCP Option 176, or the appropriate SSON Option.

● Download the upgrade script file and application file(s) from the Avaya Web site

http://www.avaya.com/support

to the HTTP server.

Note:

Note: Many LINUX servers distinguish between upper and lower case names. Ensure

that you specify the 46xxsettings filename accurately, as well as the names and values of the data within the file.

If you choose to enhance the security of your HTTP environment by using Transport Layer Security (TLS), you also need to:

● Install the TLS server application.

● Administer the system parameter TLSSRVR to the address(es) of the Avaya HTTP server.

Note:

Note: TLS is supported only on an Avaya server.

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HTTP Configuration for Backup/Restore

In addition to the procedures in this section, you can use the Avaya File Server Application for configuration, firmware file download, and backup/restore. You can download this application from http://www.avaya.com/support

In general, if you are migrating from FTP to HTTP, all you need to do is administer the 46XX settings file to point to the correct file server by using BRURI follow the steps below.

For IIS Web Servers

For IIS 4.0 (WinNT4.0), IIS 5.0 (Win2000), IIS 5.1 (WinXP), IIS 6.0 (Win2003):

1. Create a “backup” folder under the root directory of your Web server. All backup files will be

stored in that directory. For example, if your backup folder is C:/Inetpub/wwwroot/backup the 46xxsettings.txt file

should have a line similar to:

, as described in Table 10, and then

[SET BRURI http://www.website.com/backup/]

If your backup folder is the root directory, the 46xxsettings.txt file should have a line similar to:

[SET BRURI http://www.website.com/

2. Use Internet Information Services Manager or Internet Information Services depending

on your OS. Go to Start --> Settings --> Control Panel --> Administrative Tools.

3. Right click on the folder created for backup, or right click on Default Web Site if there is no

specific backup directory.

4. Select Properties.

5. In the Directory tab, make sure the Write box is checked.

Additional step for IIS 6.0 (Win2003):

1. Use Internet Information Services. Go to Start --> Settings --> Control Panel -->

Administrative Tools.

2. Below Default Web Site select Web Services Extension.

3. Make sure the WebDAV option is set to Allowed.

]

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For Apache Web Servers

1. Create a “backup” folder under the root directory of your Web server, and make the folder

writable by everyone. All backup files will be stored in that directory. If your backup folder is for instance C:/Program Files/Apache Group/Apache2/htdocs/

backup, the 46xxsettings.txt file should have a line similar to:

HTTP Configuration for Backup/Restore

[SET BRURI http://www.website.com/backup/

]

If your backup folder is the root directory, the 46xxsettings.txt file should have a line similar to:

[SET BRURI http://www.website.com/]

2. Edit your Web server configuration file httpd.conf.

3. Uncomment the two LoadModule lines associated with DAV:

LoadModule dav_module modules/mod_dav.so LoadModule dav_fs_module modules/mod_dav_fs.so

Note:

Note: If these modules are not available on your system, typically the case on some

Unix/Linux Apache servers, you have to recompile these two modules (mod_dav & mod_dav_fs) into the server. Other ways to load these modules might be available. Check your Apache documentation at http://httpd.apache.org/docs/ for more details.

4. Add the following lines in the httpd.conf file:

# # WebDAV configuration # DavLockDB "C:/Program Files/Apache Group/Apache2/var/DAVLock" <Location /> Dav On </Location>

For Unix/Linux Web servers the fourth line might look more like:

DavLockDB/usr/local/apache2/var/DAVLock

Create the var directory and make it writable by everyone. Right click Properties-->Security-->Add-->Everyone-->Full Control.

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4600 Series IP Telephone Scripts and Application Files

Important:

Important: You can convert a 4602, 4602SW, 4602SW+, 4610SW, 4620SW, 4621SW, and

4625SW IP Telephone from H.323 to SIP software, or from SIP to H.323 software. However, depending on the telephone model and the software version you start from, additional steps may be required from those mentioned in this section. When converting from one protocol type to another on a given telephone, please see “Converting Software on Avaya 4600 Series IP Telephones” in the 4600 Series IP Telephone Installation Guide (Document Number 555-233-128).

The files necessary to operate the 4600 Series IP Telephones are available on the Avaya Web site at: http://www.avaya.com/support

Two f iles on the file se rver are essential. Other files are needed when the Avaya IP Telephones need an upgrade. The essential files are:

● An upgrade script file, which tells the IP telephone whether the telephone needs to

upgrade software. The Avaya IP Telephones attempt to read this file whenever they reset. The upgrade script file is also used to point to the settings file. There are separate upgrade script files for the 4630 Telephones.

Note:

Note: The 4630 IP Telephones have a different upgrade process than the other

telephones. This is because the 4630 touch screen operation is significantly more complex than any of the other Avaya IP Telephones. There are some common elements between the 4630 and other IP telephones. Any differences are highlighted as appropriate in this section.

● The settings file contains the option settings that enable many of the options you will need

to customize the Avaya IP Telephones for your enterprise. You can use one settings file for all your Avaya IP Telephones.

In addition to the upgrade script and settings files you need the latest binary code used in the Avaya IP Telephones.

The upgrade script file and settings file are available from the Avaya Web site. The files allow you to upgrade to new software releases and new functionality without having to replace IP telephones. These two files, plus other useful information such as a ReadMe file, information about infrared capabilities, and a settings file template, are contained in a self-extracting executable file you download to your file server. Application files for all current 4600 Series IP Telephones except the 4630/4630SW, and an upgrade script file, are bundled together in that self-extracting executable file. The self-extracting executable file comes in both zipped and unzipped format. See Choosing the Right Application File and Upgrade Script File

on page 80

for more information.

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The Avaya-provided upgrade script files, and the binaries included in the zip files, upgrade the Avaya IP Telephones. You should not need to modify them. It is essential that all the binary files be together on the file server. When downloading a new release onto a file server with an existing release already on it, we recommend that you:

● Stop the file server.

● Back up all the current file server directories as applicable.

● Copy your 46xxsettings.txt file to a backup location.

● Remove all the files in the download directory. This ensures that you do not have an

inappropriate binary or configuration file on the server.

● Download the self-extracting executable file, or the corresponding zip file.

● Extract all the files. When extracting the 4630 files, ensure that you allow the directories to

be created.

● Copy your 46xxsettings.txt file back into the download directory.

● Check the Readme files for release-specific information.

● Modify the 46xxsettings.txt file as desired.

● Restart the TFTP/HTTP Server.

● Reset your Avaya IP Telephones.

You can download a default upgrade script file, sometimes called merely the “script file,” from

http://www.avaya.com/support

. This file allows the telephone to use default settings for customer-definable options. Of course, these settings can also be changed with DHCP or in some cases, from the telephone’s dialpad itself. However, you might want to open the default file and administer the options to add useful functionality to your Avaya IP Telephones. This file must reside in the same directory as the upgrade script file, and must be called

46xxsettings.scr or 46xxsettings.txt. The Avaya IP Telephones can operate without this file.

Note:

Note: Most Windows systems interpret the file extension *.scr as a screen saver. The

4600 IP Telephones originally used *.scr to indicate a script file. Starting with Release 1.7, the settings file can also have the extension *.txt.

The settings file can include any of the five types of statements, one per line:

● Comments, which are statements with a “#” character in the first column.

● Tags, which are comments that have exactly one space character after the initial #,

followed by a text string with no spaces.

● Goto commands, of the form GOTO tag. Goto commands cause the telephone to

continue interpreting the settings file at the next line after a # tag statement. If no such statement exists, the rest of the settings file is ignored.

● Conditionals, of the form IF $name SEQ string GOTO tag. Conditionals cause the Goto

command to be processed if the value of name is a case-insensitive equivalent to string. If no such name exists, the entire conditional is ignored.

● SET commands, of the form SET parameter_name value. Invalid values cause the

specified value to be ignored for the associated parameter_name so the default or previously administered value is retained. All values must be text strings, even if the value itself is numeric, a dotted decimal IP Address, etc.

Note:

Note: Enclose all data in quotation marks for proper interpretation.

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The Avaya-provided upgrade script file includes lines that tell the telephone to GET

46xxsettings.scr and 46xxsettings.txt. These lines cause the telephone to use TFTP/HTTP to attempt to download the file specified in the GET command. If the file is obtained, its contents are interpreted as an additional script file. That is how your settings are changed from the default settings. If the file cannot be obtained, the telephone continues processing the upgrade script file. The upgrade script file is processed so that if there is no 46xxsettings.scr file, the telephone looks for a 46xxsettings.txt file. If the settings file is successfully obtained but does not include any setting changes the telephone stops using TFTP or HTTP. This happens when you initially download the script file template from the Avaya support W eb site, before you make any changes. When the settings file contains no setting changes, the telephone does not go back to the upgrade script file. You can change the settings file name, if desired, as long as you also edit the corresponding GET command in the upgrade script file. However, we encourage you not to alter the Avaya-provided upgrade script file. If Avaya changes the upgrade script file in the future, any changes you have made will be lost. We strongly encourag e you to use the 46xxsettings file to customize your settings instead. For more details on customizing your settings file, see Contents of the Settings File

Choosing the Right Application File and Upgrade Script File

The 4600 IP Telephone software Releases are bundled together in *exe and *zip files on the Avaya support Web site. See 4600 Series IP Telephone Scripts and Application Files

for a detailed description. As of Release 2.7, you have fou r “bundles” from which to choose. Only one bundle is likely to be optimal for any one environment.

Which bundle to choose depends on the answer to two questions:

● Which version of 4610SW/4620SW software do you need in that environment?

● Are the majority of your 4602/4602SW/4602SW+, 4610SW, 4620SW, 4621SW, and

4625SW Telephones in that environment H.323-based or SIP-based? The 4610SW, 4620SW, 4621SW, 4622SW, and 4625SW IP Telephones support multi-byte characters, so the software bundles come in one of four versions:

● a default version which only supports single-byte characters like those used in English,

French, Japanese Katakana, etc.

● a multi-byte version for 4610SW, 4620SW, 4621SW, 4622SW, and 4625SW IP

Telephones that support Chinese, Russian, and Hebrew.

● a separate multi-byte version for 4610SW, 4620SW, 4621SW, 4622SW and 4625SW IP

Telephones that support Japanese, Russian, and Hebrew. Note that the 4625SW does not

currently support Japanese (Kanji).

● a separate multi-byte version for 4610SW, 4620SW, 4621SW, 4622SW and 4625SW IP

Telephones that support Korean, Russian, and Hebrew. If multi-byte support is not relevant to you, select the default bundle, even if you do not have any 4610SW, 4620 SW , 4621SW , 4622SW, and 4625SW telephones. Otherwise, select the software bundle that includes Chinese, Japanese, or Korean as appropriate.

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Note:

Note: All bundles include the complete software for the other, non-4610SW/4620SW/

4621SW/4622SW/4625SW Telephones. The software includes the 4620 IP Telephone, but not the 4630/4630SW, which remains separate. The only differences between the four bundles are the software for the 4610SW, 4620SW, 4621SW, 4622SW, and 4625SW, and a slight change in the associated upgrade script file.

The 4602/4602SW/4602SW+, 4610SW, 4620SW, 4621SW and 4625SW IP Telephones can support either H.323 or SIP signaling protocols. If a majority of your 4600 Series IP Telephones are H.323-based, which is the most common situation, you can use any or all of the software bundles identified in this section. If a majority are SIP-based, select the fourth software bundle, identified as the “SIP” software bundle on the Web site. The application files in this SIP software bundle are the same as in the default bundle. The difference is a modified upgrade script file that assumes SIP is the default protocol for 4602/4602SW/4602SW+, 4610SW, 4620SW, 4621SW, and 4625SW IP Telephones, and that H.323 is the exception.

When you have a mixture of H.323 and SIP telephones, use the SIG system value to ensure that each telephone type has appropriate software downloaded. The SIG system value has three legal values:

● the default value “0” which indicates “use the default protocol,”

● “1” meaning “use H.323,” and

● “2” meaning “use SIP.”

You decide the meaning of “the default protocol.” If the majority of your IP telephones are H.323-based, that should be the default. Otherwise, SIP is the default.

The SIG system value cannot be set in the 46xxsettings file or in the upgrade script file. SIG can only be set on a telephone-by-telephone basis. Instead of manually setting SIG yourself, first instruct the installers of the non-default phones to perform the SIGnaling Protocol Identifier procedure in Chapter 3 of the 4600 Series IP Telephone Installation Guide. For example, if yours is a largely H.323 environment, when SIP phones are installed the SIG system value should be set to “2.” If yours is a largely SIP environment, when H.323 phones are installe d the SIG system value should be set to “1.”

Detailed information about SIP is available in the SIP-related documentation, provided elsewhere on the Avaya support Web site.

Note:

Note: As indicated above, although the SIG system value is a Release 2.0 feature, the

4601 IP Telephone supports SIG functionality, even though the 4601 currently supports only Release 1.8 software.

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Contents of the Upgrade Script

See Appendix F: Sample Upgrade Script File for a typical upgrade script file.

Contents of the Settings File

Check the last lines of the Upgrade Script file example in the previous section. They show that after checking the application software, the 4600 Series IP Telephone looks for a 46xx settings file. This optional file is under your control and is where you can identify non-default option settings, application-specific parameters, etc. The Avaya support Web site has a template for this file for downloading. An example of what the file could look like follows.

Note:

Note: The following is intended only as an example. Your settings will vary from the

settings shown. This sample assumes specification of a DNS Server , par ameters for the 4630/4630SW Directory application, and a 4620 Web Browser. See

Administering Options for the 4600 Series IP Telephones

about specific values. You need only specify settings that vary from defaults, although specifying defaults is harmless.

on page 101, for details

DNSSRVR=”dnsexample.yourco.com” DIRSRVR=”123.123.123.123”

DIRTOPDN=”yourco” WMLHOME=”http://support.avaya.com/elmodocs2/avayaip/4620/home.wml”

WEBPROXY=”11.11.11.11”

As of Release 2.4, VLAN separation provides for tagged frames to be received by a secondary Ethernet interface, typically a PC. Add commands to the 46xxsettings.txt file to enable VLAN separation, provide the VLAN ID for tagged frames received on the secondary Ethernet interface, and set the Layer 2 priority for those tagged frames. The following example assumes the data VLAN ID is “yyy” and the data traffic priority is “z”:

SET VLANSEP 1 SET PHY2VLAN yyy SET PHY2PRIO z

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The GROUP System Value

You might have different communities of end users, all of which have the same model telephone, but which require different administered settings. For example, you might want to restrict Call Center agents from being able to Logoff, which might be an essential capability for “hot-desking” associates. Or you might want to assign your SIP telephone users to different messaging systems or registration/proxy servers. We provide examples of the group settings for each of these situations later in this section.

As of Release 2.8, Communication Manager Release 3.1.3 and above supports the downloading of the GROUP settings upon registration of the individual phone.

As of Release 2.0, the simplest way to separate groups of users is to associate each of them with a number. You then edit the 46xxsettings file so each group is assigned the appropriate settings. Use the GROUP system value for this purpose. The GROUP system value cannot be set in the 46xxsettings file. The GROUP System value can only be set on a telephone-by-telephone basis. To do so, first identify which phones are associated with which group, and designate a number for each group. The number can be any integer from 0 to 999, with 0 as the default, meaning your largest group would be assigned as Group 0.

The GROUP System Value

Then, at each non-default telephone, instruct the installer or end-user to invoke the GROUP Local (dialpad) Administrative procedure as specified in the 4600 Series IP Telephone Installation Guide and specify which GROUP number to use. Once the GROUP assignments are in place, edit the settings file to allow each telephone of the appropriate group to download its proper settings.

Here is an example of the settings file for the Call Center agent:

IF $GROUP SEQ 1 goto CALLCENTER IF $GROUP SEQ 2 goto HOTDESK

{specify settings unique to Group 0}

goto END # CALLCENTER

{specify settings unique to Group 1}

goto END # HOTDESK

{specify settings unique to Group 2}

# END

{specify settings common to all Groups}

Here is an example of the settings file for the SIP telephone users. Note that there are two messaging systems and two registration/proxy servers, for a total of four possible combinations. These sample GROUP assignments are unique to the SIP users. Other GROUP assignments could be in the same file for other purposes.

IF $GROUP SEQ 10 goto GROUP10 IF $GROUP SEQ 20 goto GROUP20

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IF $GROUP SEQ 30 goto GROUP30 IF $GROUP SEQ 40 goto GROUP40

{specify settings unique to Group 0}

goto END # GROUP10

{specify settings unique to Group 10, for example}

SET COVERAGEADDR "22000" SET SIPPROXYSRVR "{IP Address of server1}" SET SIPREGISTRAR "{IP Address of server1}" goto END

# GROUP20

{specify settings unique to Group 20, for example}

SET COVERAGEADDR "23000" SET SIPPROXYSRVR "{IP Address of server1}" SET SIPREGISTRAR "{IP Address of server1}" goto END

# GROUP30

{specify settings unique to Group 30, for example}

SET COVERAGEADDR "22000" SET SIPPROXYSRVR "{IP Address of server2}" SET SIPREGISTRAR "{IP Address of server2}" goto END

QoS

# GROUP40

{specify settings unique to Group 40, for example}

SET COVERAGEADDR "23000" SET SIPPROXYSRVR "{IP Address of server2}" SET SIPREGISTRAR "{IP Address of server2}" goto END

# END

{specify settings common to all Groups}

The 4600 Series IP Telephones support both IEEE 802.1D/Q and DiffServ. Other network-based QoS initiatives such as UDP port selection do not require support by the telephones. Those initiatives nonetheless can contribute to improved QoS for the entire network.

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IEEE 802.1D and 802.1Q

IEEE’s 802.1Q standard defines a tag that can be added to voice and dat a p acket s. Most of the information associated with this tag deals with Virtual LAN (VLAN) management, but 3 bits are reserved for identifying packet priority. These 3 bits allow any one of 8 priorities to be assigned to a specific packet. As defined in the standard, the 8 priorities are, from highest to lowest:

● 7: Network management traffic

● 6: Voice traffic with less than 10ms latency

● 5: Voice traffic with less than 100ms latency

● 4: “Controlled-load” traffic for mission-critical data applications

● 3: Traffic meriting “extra-effort” by the network for prompt delivery, for example, executives’

e-mail

● 2: Reserved for future use

● 0: Traffic meriting the network’s “best-effort” for prompt delivery. This is the default priority.

● 1: Background traffic such as bulk data transfers and backups

QoS

Note:

Note: Priority 0 is a higher priority than Priority 1.

To support IEEE 802.1D/Q, the 4600 Series IP Telephones can be administered either of two ways:

● from the network by appropriate administration of the DHCP or TFTP/HTTP servers, as

covered in 4600 Series IP Telephone Scripts and Application Files

● at the telephone itself using dialpad input, as covered under local administrative

on page 78, or

procedures in the 4600 Series IP Telephone Installation Guide. Eight IEEE 802.1D/Q QoS parameters in the telephones can be administered, as follows. The

first five parameters are for standard 802.1Q tagging and apply to any 4600 Series IP Telephone. The last three parameters apply to all 4600 Series IP Telephones with a secondary Ethernet interface.

● L2Q: 802.1Q framing parameter (1=On, 2=Off, or 0=AUTO). The default is 0, but the

preferred setting is 1 (ON). You can manually set the L2Q value of a specific 4600 IP

Telephone to any value, for example, AUTO, ON, or OFF. However , a ny subsequent value

administered via DHCP or TFTP/HTTP settings file will override the manual value. To use

the QoS Local Administrative Option to set L2Q manually see the 4600 Series IP

Telephone Installation Guide.

● L2QVLAN: the VLAN ID on which the telephone should operate. For example, what VLAN

ID to use for DHCP Discovery, etc. (up to 4 digits, from 0 to 4094, default is 0).

● VLANTEST: the number of seconds to wait for a DHCPOFFER when using a non--zero

VLAN ID (up to 3 digits, from 0 to 999, default is 60).

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● L2QAUD: 802.1Q VoIP (voice) user RTP traffic priority value (between 0 and 7, default

is 6).

● L2QSIG: 802.1Q VoIP (voice) user Call Control Signaling traffic priority value (between 0

and 7, default is 6).

● VLANSEP: enables VLAN separation when set to 1 (the default); disables VLAN

separation when set to 0. If the value is zero, broadcasts are transmitted. Can only be

administered using DHCP/TFTP, not by a manual procedure.

● PHY2VLAN: VLAN ID to be used for tagged (data) frames received on the secondary

Ethernet interface when VLAN separation is enabled (up to 4 digits, from 0 to 4094, default

is 0). Can only be administered using DHCP/TFTP, not by a manual procedure.

● PHY2PRIO: Layer 2 (data) priority value to be used for tagged frames received on

the secondary Ethernet interface when VLAN separation is enabled (1 digit, 0 (zero)

through 7, default is 0). Can only be administered using DHCP/TFTP, not by a manual

procedure. In the 4600 Series IP Telephone Installation Guide, the Local Administrative Option ADDR also

allows you to specify VLAN IDs and VLANTEST values. The Local Administrative Option QoS allows you to specify values for L2Q, L2QAUD, and L2QSIG.

The 4600 Series IP Telephones can simultaneously support receipt of packets using, or not using, 802.1Q parameters.

For additional information on VLAN administration, see VLAN Considerations

DIFFSERV

IETF RFCs 2474 and 2475 define “services” basically as different ways to treat a network’s different traffic subsets at the Internet Protocol (IP) layer, Layer 3. For example, some packets might be routed to expedite delivery and minimize delay, with other packets routed to minimize loss or cost. Redefining an octet in the Layer 3 headers for IP versions 4, or IPv4 and 6, or IPv6 provides the differentiation between these services (Differentiated Services). IPv4 calls this octet a Type of Service (TOS) octet while IPv6 calls this octet a Traffic Class. In both cases, the octet is interpreted differently than it was originally defined. With Differentiated Services, bits 0 through 5 of the octet identify a Differentiated Services Code Point (DSCP). The DSCP identifies a procedure to be used to handle that packet on a per-hop basis. Bits 6 and 7 of the octet are currently unused, and DSCP-compliant routers ignore them.

With DiffServ, the default DSCP is all zeroes, and represents “no special handling.” RFC 2474 also defines eight “Class Selector Codepoints,” which are the eight DSCP encodings that can be represented by xxx000, where “x” represents one bit. These Code Selector Codepoints are considered prioritized, with the larger numeric values having a higher relative order. DSCP-compliant routers should give the associated packets of larger-valued DSCPs a “probability of timely forwarding” greater than a packet with a lower-valued DSCP. In addition to the eight Class Selector Codepoints, a network can define its own DSCPs by defining

on page 91.

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encodings that do not terminate in 000. The specific treatment intended by these custom DSCPs will not necessarily be carried out by routers outside the customer’s own network.

The 4600 Series IP Telephone Installation Guide describes the Local Administrative Option for QoS. This option allows you to specify Diffserv values for Layer 3 audio (“DSCPAUD”) and signaling traffic (“DSCPSIG”) on a telephone-by-telephone basis.

The Avaya H.323 IP Telephones’ DiffServ values change to the values administered on the media server as soon as the telephone registers. For more information, see the document titled Administration for Network Connectivity (555-233-504kk). Unless there is a specific need in your enterprise LAN, we do not recommend you change the default values.

UDP Port Selection (H.323 Only)

Some data networks include equipment that can perform UDP port selection. This is a mechanism that gives packets with port numbers in a given ran ge priority over p acket s with port numbers outside that range.

To support UDP port selection, the 4600 Series IP Telephones can be administered from the Avaya Communication Manager Network Region form. Locate specific implementation details for local administration of MCPORT in the 4600 Series IP Telephone Installation Guide. For Avaya Communication Manager administration, find implementation det ails in Administration for Network Connectivity for Avaya Communication Manager Software. In summary, the system value MCPORT represents the port on the TN2302AP circuit pack. Use this port number to administer routers, etc. supporting UDP port selection, to maximize priority of voice packets being exchanged between the PBX and the telephone.

QoS

The default value for MCPORT is 1719. Administer the switch to use a port within the proper range for the specific LAN, and the IP telephone(s) will copy that port. A related parameter is PORTAUD, which is the RTP port used by the switch. If no UDP port range is administered on the switch, the IP telephone uses an even-numbered port, randomly selected from the interval 4000 to 10000.

Network Audio Quality Display on 4600 Series IP Telephones

With the exceptions of the 4601, 4601+, 4606, 4612, 4624, and 4690 IP Telephones, all Series 4600 IP Telephones are by default administered to allow the end user an opportunity to monitor network audio performance while on a call. The user guides for each telephone provide specific detail on getting to the appropriate screen, what the end user sees, and what the information means.

For 4610SW/4620/4620SW/4621SW/4622SW/4625SW/4630/4630SW IP Telephones, these parameters display in real-time to users on the appropriate screens, while on a call:

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Table 5: Parameters in Real-Time

Parameter Possible Values

Audio Connection Present?

Received Audio

Yes if a receive RTP stream was established. No if a receive RTP stream was not established.

G.711, G.726A, or G.729.

Coding Silence Suppression Yes if the telephone knows the far-end has silence suppression

Enabled. No if the telephone knows the far-end has silence suppression

Disabled, or the telephone does not know either way.

Packet Loss No data or a decimal percentage. Late and out-of-sequence packets

are counted as lost if they are discarded. Packets are not counted as lost until a subsequent packet is received and the loss confirmed by the RTP sequence number.

Packetization Delay No data or an integer number of milliseconds. The number reflects the

amount of delay in received audio packets, and includes any look-ahead delay associated with the codec.

One-way Network Delay

Network Jitter Compensation Delay

No data or an integer number of milliseconds. The number is one-half the value RTCP computes for the round-trip delay.

No data or an integer number of milliseconds reporting the average delay introduced by the telephone’s jitter buffer.

For 4602/4602SW/4602SW+ IP Telephones, the Network Audio Quality Screen gives the user a qualitative assessment of the current overall audio quality. This assessment is based on separate evaluations of:

● the Packet Loss, and

● the total Network Delay , which is th e sum of Packetization Delay, One-way Network Delay,

and Network Jitter Compensation Delay, and

● consideration of the codec in use.

This information’s implication for LAN administration depends, of co urse, on the values the user reports and the specific nature of your LAN, like topology, loading, QoS administration, etc. This information’s major use is to give the user an idea of how network conditions affect the current call’s audio quality. It is assumed you have more detailed tools available for troubleshoo ting the LAN.

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RSVP and RTCP

Avaya IP Telephones implement the Resource ReSerVation Protocol (RSVP) to support WAN bandwidth management. RSVP is administered from the media server. Avaya IP Telephones implement the RTP Control Protocol (RTCP) so Avaya’s Voice over IP (VoIP) Monitoring Manager (VMON) software can provide real-time monitoring and historical data of audio quality for VoIP calls.

Resource ReSerVation Protocol (RSVP) is an IETF-standard protocol hosts use to request resource reservations throughout a network. RSVP-compliant hosts send messages through a network to receivers. Receivers respond with messages requesting a type of service and an amount of resources, for example, bandwidth, to carry out that service. The host is responsible for admitting (approving) or rejecting (denying) the request. In a QoS context, RSVP tries to reserve bandwidth on the network for voice calls on a call-by-call basis. If insufficient bandwid th is available for the target voice quality, a request to use network bandwidth for a voice call is rejected.

RTP Control Protocol (RTCP), as its name implies, is a protocol that provides control functions for Real-time Transport Protocol (RTP). R TP provides end-to-end network services for real-time data such as Voice over IP. But RTP does not provide a reservation function, nor does it guarantee any level of QoS. RTCP supplements RTP by monitoring the quality of the RTP services and can provide real-time information to users of an RTP service. In a QoS context, RTCP is valuable to identify information such as:

QoS

● packet loss,

● 1-way delay or how long a packet has to go from source A to destination B,

● jitter, etc.

RTCP itself does not improve QoS, but provides information to help identify where problem areas might be.

You cannot change the telephone’s RSVP or RTCP parameters directly on the telephone or by TFTP or DHCP administration. The only way to change these parameters is on the H.323 telephones, and such a change requires appropriate switch administration. See your Avaya Media Server administration documentation for more detail. You cannot change these parameters on a SIP IP telephone.

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Internal Audio Parameters

The AUDIOENV variable provides control of some internal audio parameters in the 4600 Series IP Telephones. Avaya does not recommend that customers set these values. In certain situations, particularly noisy environments, Avaya SSE may recommend a change in the AUDIOENV setting to reduce/eliminate the effects environmental noise can have during phone use. This variable applies to R2.4 and later software releases, except R2.5.

AUDIOENV is an index into a table that impacts four internal variables:

Table 6: Internal Audio Variables

Variable Description Possible Values

AGC_Dyn_Range AGC dynamic range. 0 for a typical office environment (+/-9dB), 1

for +/-12dB, 2 for +/-15dB, and 3 for +/-18 AGC Dynamic range variation.

NR_thresh_Hd The noise reduction

threshold for the headset.

NR_thresh_Hs The noise reduction

threshold for the handset.

HD_Tx_Gain Headset transmit

gain.

AUDIOENV= a range of 0 to 107 for pre-2.8 releases., a range of 0 to 191 for Release 2.8, and a range of 0 to 299 beginning with Release 2.8.3.

Set AUDIOENV 0 is the nominal setting (0,0,0,0). Please see Audio Quality Tuning for IP Telephones, Issue 2 on support.avaya.com.

The noise reduction threshold for the headset has a default value of 0 for a typical office environment, 1 for call center applications, 2 and 4 for increasingly noisy audio environments, and 3 where noise reductio n is disabled.

The noise reduction threshold for the handset has a default value of 0 for a typical office environment, 1 for call center applications, 2 and 4 for increasingly noisy audio environments, and 3 where noise reductio n is disabled.

Headset transmit gain has a default value of 0 for normal transmit gain, 1 for +6dB of gain, and 2 for -6dB of gain.

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VLAN Considerations

This section contains information on how to administer 4600 Series IP Telephones to minimize registration time and maximize performance in a Virtual LAN (VLAN) environment. If your LAN environment does not include VLANs, set the system parameter L2Q to 2 (of f) to ensure correct operation.

VLAN Tagging

IEEE 802.1Q tagging (VLAN) is a useful method of managing VoIP traffic in your LAN. Avaya recommends that you establish a voice VLAN, set L2QVLAN to that VLAN and provide voice traffic with priority over other traffic. If LLDP was used set the telephones’ VLAN, that setting has absolute authority. Otherwise, you can set VLAN tagging manually, by DHCP, or in the 46xxsettings.txt file.

If VLAN tagging is enabled (L2Q= 0 or 1), the 4600 Series IP Telephones set the VLAN ID to L2QVLAN, and VLAN priority for packets from t he telephon e to L2 QAUD for audio packets and L2QSIG for signalling packets. The default value (6) for these parameters is the recommended value for voice traffic in IEEE 802.1D.

VLAN Considerations

Regardless of the tagging setting, a 4600 Series IP Telephone will always transmit packet s from the telephone at absolute priority over packets from secondary Ethernet. The priority settings are useful only if the downstream equipment is administered to give the voice VLAN priority.

VLAN Detection

The Avaya IP Telephones support automatic detection of the condition where the L2QVLAN setting is incorrect. When VLAN tagging is enabled (L2Q= 0 or 1) initially the 4600 Series IP Telephone transmits DHCP messages with IEEE 802.1Q tagging and the VLAN set to L2QVLAN. The telephones will continue to do this for VLANTEST seconds.

● If the VLANTEST timer expires and L2Q=1, the telephone sets L2QVLAN=0 and transmits

DHCP messages with the default VLAN (0).

● If the VLANTEST timer expires and L2Q=0, the telephone sets L2QVLAN=0 and transmits

DHCP messages without tagging.

● If VLANTEST is 0, the timer will never expire.

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Note:

Note: Regardless of the setting of L2Q, VLANTEST, or L2QVLAN, you must have

DHCP administered so that the telephone will get a response to a DHCPDISCOVER when it makes that request on the default (0) VLAN.

After VLANTEST expires, if an Avaya IP Telephone running R2.6 receives a non-zero L2QVLAN value, the telephone will release the IP Address and send DHCPDISCOVER on that VLAN. Any other release will require a manual reset before the telephone will attempt to use a VLAN on which VLANTEST has expired. See the Reset procedure in Chapter 3 of the 4600 Series IP Telephone Installation Guide.

The telephone ignores any VLAN ID administered on the media server if a non-zero VLAN ID is administered either:

- by LLDP

- manually,

- through DHCP, and/or

- through TFTP or HTTP.

VLAN Separation

In Releases 2.4 and 2.6+, VLAN separation is available to control priority tagging from the device on the secondary Ethernet, typically PC data. The following system parameters control VLAN separation:

● VLANSEP - enables (1) or disables (0) VLAN separation.

● PHY2VLAN - provides the VLAN ID for tagged frames received on the secondary Ethernet

interface.

● PHY2PRIO - the layer 2 priority value to be used for tagged frames received on the

secondary Ethernet interface.

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Table 7 provides several VLAN separation guidelines.

Table 7: VLAN Separation Rules

If Then

VLAN Considerations

VLANSEP is “0” (Disabled),

OR the telephone is not tagging frames,

Frames received on the secondary Ethernet interface will not be changed before forwarding. For example, tagging is not added or removed

OR the telephone is tagging frames with a VLAN ID equal to PHY2VLAN.

and the VLAN ID and tagged frames priority are not changed. The Ethernet switch forwarding logic determines that frames received on the Ethernet line interface are forwarded to the secondary Ethernet interface or to the telephone without regard to specific VLAN IDs or the existence of tags.

VLANSEP is “1” (Enabled) All tagged frames received on the secondary

Ethernet interface are changed before forwarding to make the VLAN ID equal to the PHY2VLAN value and the priority value equal to the PHY2PRIO value.

Untagged frames received on the secondary Ethernet interface are not changed before forwarding.

VLANSEP is “1” (Enabled)

AND the telephone is not tagging frames,

The Ethernet switch forwarding logic determines that frames received on the Ethernet line interface are forwarded to the secondary

OR if the telephone is tagging frames with a VLAN ID equal to

Ethernet interface or to the telephone without regard to specific VLAN IDs or the existence of tags.

PHY2VLAN,

VLANSEP is “1” (Enabled)

OR if the PHY2VLAN value is zero.

AND the telephone is tagging frames with a VLAN ID not equal to PHY2VLAN,

AND the PHY2VLAN value is not zero.

Tagged frames received on the Ethernet line interface will only be forwarded to the secondary Ethernet interface if the VLAN ID equals PHY2VLAN.

Tagged frames received on the Ethernet line interface will only be forwarded to the telephone if the VLAN ID equals the VLAN ID used by the telephone.

Untagged frames will continue to be forwarded or not forwarded as determined by the Ethernet switch forwarding logic.

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Unnamed Registration

As of Release 2.4, 4600 Series IP Telephones support unnamed registration. A telephone can register with the call server and receive limited service without requiring an extension and password entry. Unless otherwise disabled using the system value UNNAMEDSTAT telephone automatically attempts to register unnamed if no action is taken on the telephone Extension entry screen.

A telephone registered without the extension and password has the following characteristics:

● only one call appearance, preventing conferences or call transfers,

● no administrable feature buttons,

● on-hook dialing cannot be invoked,

● limited to the calling capability administered for PSA (Personal Station Access) on the call

server, for example, only outgoing calls permitted subject to call server Class of

Restriction/Class of Service limitations, and

● can be converted to normal, named registration by a valid extension and password entry.

, the

IEEE 802.1X

As of Release 2.6, certain IP telephones support the IEEE 802.1X standard for pass-through and supplicant operation. The system parameter DOT1X telephones handle 802.1X multicast packets and proxy logoff, as follows:

● When DOT1X = 0, the telephone forwards 802.1X multicast packets from the

Authenticator to the PC attached to the telephone and forwards multicast p ackets from the

attached PC to the Authenticator (multicast pass-through). Proxy Logoff is not supported.

● When DOT1X = 1, the telephone supports the same multicast pass-through as when

DOT1X=0. Proxy Logoff is supported.

● When DOT1X = 2, the telephone forwards multicast packets from the Authenticator only to

the telephone, ignoring multicast packets from the attached PC (no multicast

pass-through). Proxy Logoff is not supported.

● Regardless of the DOT1X setting, the telephone always properly directs unicast packets

from the Authenticator to the telephone or its attached PC, as dictated by the MAC

address in the packet. The telephones support supplicant operation and parameter values as specified in IEEE

802.1X, but as of software Release 2.9, only if the value of the p ara meter DOT 1XSTAT is “1” or “2”. If DOT1XSTAT has any other value, supplicant operation is not supported.

determines how applicable

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IEEE 802.1X

IP telephones will respond to unicast EAPOL frames (frames with the telephone’s MAC address as the destination MAC address, and a protocol type of 88-8E hex) received on the Ethernet line interface if the value of DOT1XSTAT is g1h or g2h, but will only respond to EAPOL frames that have the PAE group multicast address as the destination MAC address if the value of DOT1XSTAT is g2h. If the value of DOT1XSTAT is changed to g0h from any other value after the supplicant has been authenticated, an EAPOL-Logoff will be transmitted before the supplicant is disabled.

As of software Release 2.9, the system parameter DOT1XSTAT determines how the telephone handles supplicants as follows:

● When DOT1XSTAT = 0, Supplicant operation is completely disabled. This is the default

value.

● When DOT1XSTAT = 1, Supplicant operation is enabled, but responds only to received

unicast EAPOL messages.

● When DOT1XSTAT = 2, Supplicant operation is enabled and responds to received unicast

and multicast EAPOL messages. Note: If the Ethernet line interface link fails, the 802.1X Supplicant, if enabled, enters the

Disconnected state. The 802.1X Supplicant variable userLogoff normally has a value of FALSE. This variable will be set to TRUE before the telephone drops the link on the Ethernet line interface (and back to FALSE after the link has been restored). The userLogoff variable may also be briefly set to TRUE to force the Supplicant into the LOGOFF state when new credentials are entered.

I802.1X Pass-Through and Proxy Logoff

As of Release 2.2.3, IP telephones support pass-through of 802.1x packets to and from an attached PC. This enables an attached PC running 802.1x supplicant software to be authenticated by an Ethernet data switch.

As of Release 2.6, and as of Release 2.7 for the 4625SW, the IP Telephones support two pass-through modes:

● pass-through and

● pass-through with proxy logoff.

The DOT1X parameter setting controls the pass-through mode. In Proxy Logoff mode (DOT1X=1), when the secondary Ethernet interface loses link integ rity, the telephone sends an

802.1X EAPOL-Logoff message to the data switch on behalf of the attached PC. The message alerts the switch that the device is no longer present. For example, a message would be sent when the attached PC is physically disconnected from the IP telephone. When DOT1X = 0 or 2, the Proxy Logoff function is not supported.

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802.1X Supplicant Operation

As of Release 2.7, the 4602SW+ and 4625SW IP Telephones support Supplicant operation. As of Release 2.6, the 4610SW, 4620SW, 4621SW, and 4622SW IP Telephones support Supplicant operation. As of software Release 2.9, supplicant operation can be disabled using the system parameter DOT1X.

IP telephones that support Supplicant operation also support Extensible Authentication Protoco l (EAP), but only with the MD5-Challenge authentication method as specified in IETF RFC 3748 [8.5-33a].

A Supplicant identity (ID) and password of no more than 12 numeric characters are stored in reprogrammable non-volatile memory. The ID and password are not overwritten by telephone software downloads. The default ID is the MAC address of the telephone, converted to ASCII format without colon separators, and the default password is null. Both the ID and password are set to defaults at manufacture. EAP-Response/Identity frames use the ID in the T ype-Dat a field. EAP-Response/MD5-Challenge frames use the password to compute the digest for the Value field, leaving the Name field blank.

When a telephone is installed for the first time and 802.1x is in effect, the dynamic address process prompts the installer to enter the Supplicant identity and password. The IP telephone does not accept null value passwords. See “Dynamic Addressing” in the 4600 Series IP Telephone Installation Guide. The IP telephone stores 802.1X credentials when successful authentication is achieved. Post-installation authentication attempts occur using the stored

802.1X credentials, without prompting the user for ID and password entry. An IP telephone can support several different 802.1X authentication scenarios, depending on

the capabilities of the Ethernet data switch to which it is connected. Some switches may authenticate only a single device per switch port. This is known as single-supplicant or port-based operation. These switches typically send multicast 802.1X packets to authenticating devices.

These switches support the following three scenarios:

● Standalone telephone (Telephone Only Authenticates) - When the IP telephone is

configured for Supplicant Mode (DOT1X=2), the telephone can support authentication

from the switch.

● Telephone with attached PC (Telephone Only Authenticates) - When the IP tele phone

is configured for Supplicant Mode (DOT1X=2), the telephone can support authentication

from the switch. The attached PC in this scenario gains access to the network without

being authenticated.

● Telephone with attached PC (PC Only Authenticates) - When the IP telephone is

configured for Pass-Through Mode or Pass-Through Mode with Logoff (DOT1X=0 or 1),

an attached PC running 802.1X supplicant software can be authenticated by the data

switch. The telephone in this scenario gains access to the network without being

authenticated. Some switches support authentication of multiple devices connected through a single switch

port. This is known as multi-supplicant or MAC-based operation. These switches typically send

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unicast 802.1X packets to authenticating devices. These switches support the following two scenarios:

● Standalone telephone (Telephone Only Authenticates) - When the IP telephone is

configured for Supplicant Mode (DOT1X=2), the telephone can support authentication

from the switch.

● Telephone and PC Dual Authentication - Both the IP telephone and the connected PC

can support 802.1X authentication from the switch. The IP telephone may be configured

for Pass-Through Mode or Pass-Through Mode with Logof f (DOT1X=0 or 1). The att ached

PC must be running 802.1X supplicant software.

Link Layer Discovery Protocol (LLDP)

Release 2.6+ 4600 Series IP Telephones support IEEE 802.1AB. Link Layer Discovery Protocol (LLDP) is an open standards layer 2 protocol IP telephones use to advertise their identity and capabilities and to receive administration from an LLDP server. LAN equipment can use LLDP to manage power, administer VLANs, and provide some administration.

Link Layer Discovery Protocol (LLDP)

The transmission and reception of LLDP is specified in IEEE 802.1AB-2005. The 4600 Series IP Telephones use Type-Length-Value (TLV) elements specified in IEEE 802.1AB-2005, TIA TR-41 Committee - Media Endpoint Discovery (LLDP-MED, ANSI/TIA-1057), and Proprietary elements. LLDP Data Units (LLDPDUs) are sent to the LLDP Multicast MAC address (01:80:c2:00:00:0e).

A 4600 Series IP Telephone initiates LLDP after receiving an LLDPDU message from an appropriate system. Once initiated, the telephones send an LLDPDU every 30 seconds with the following contents:

Table 8: LLDPDU Transmitted by the 4600 Series IP Telephones

Category TLV Name (Type) TLV Info String (Value)

Basic Mandatory Chassis ID IPv4 IP Address of telephone. Basic Mandatory Port ID MAC address of the telephone. Basic Mandatory Time-To-Live 120 seconds. Basic Optional System Name The Host Name sent to the DHCP server in

DHCP option 12.

1 of 3

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Table 8: LLDPDU Transmitted by the 4600 Series IP Telephones (continued)

Category TLV Name (Type) TLV Info String (Value)

Basic Optional System Capabilities Bit 2 (Bridge) will be set in the System

Basic Optional Management Address Mgmt IPv4 IP Address of telephone.

Capabilities if the telephone has an internal Ethernet switch. If Bit 2 is set in Enabled Capabilities then the secondary port is enabled.

Bit 5 (Telephone) will be set in the System Capabilities. If Bit 5 is set in the Enabled Capabilities than the telephone is registered.

Interface number subtype = 3 (system port). Interface number = 1.

OID = SNMP MIB-II sysObjectID of the telephone.

IEEE 802.3 Organization

MAC / PHY Configuration / Status

Reports autonegotiation status and speed of the uplink port on the telephone.

Specific TIA LLDP MED LLDP-MED

Capabilities

Media Endpoint Discovery - Class III - IP Telephone.

TIA LLDP MED Network Policy Tagging Yes/No, VLAN ID for voice, L2 Priority,

DSCP Value.

TIA LLDP MED Inventory – Hardware

MODEL - Full Model Name.

Revision

TIA LLDP MED Inventory – Firmware

BOOTNAME.

Revision

TIA LLDP MED Inventory – Software

APPNAME.

Revision

TIA LLDP MED Inventory – Serial

Telephone serial number.

Number

TIA LLDP MED Inventory –

Avaya.

Manufacturer Name

TIA LLDP MED Inventory – Model

MODEL4 - 4 character name.

Name

98 4600 Series IP Telephone LAN Administrator Guide

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Link Layer Discovery Protocol (LLDP)

Table 8: LLDPDU Transmitted by the 4600 Series IP Telephones (continued)

Category TLV Name (Type) TLV Info String (Value)

Avaya Proprietary PoE Conservation

Level Support

Provides Power Conservation abilities/settings, Typical and Maximum Power values.

OUI = 00-40-0D (hex), Subtype = 1.

Avaya Proprietary Call Server IP

Call Server IP Address.

Address

Subtype = 3.

Avaya Proprietary IP Phone Addresses Phone IP Address, Phone Address Mask,

Gateway IP Address. Subtype = 4.

Avaya Proprietary CNA Server IP

Address

CNA Server IP Address = in-use value from CNASRVR.

Subtype = 5.

Avaya Proprietary File Server File Server IP Address.

Subtype = 6.

Avaya Proprietary 802.1Q Framing 802.1Q Framing = 1 if tagging or 2 if not.

Subtype = 7.

Basic Mandatory End-of-LLDPDU Not applicable.

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Server Administration

On receipt of a LLDPDU message the Avaya IP Telephones will act on these Type-Length-Value (TLV) elements.

Table 9: Impact of TLVs on System Parameter Values

System Parameter Name

TLV Name

PHY2VLAN IEEE 802.1 Port

VLAN ID

L2QVLAN and L2Q

IEEE 802.1 VLAN Name

MED Network Policy TLV

Impact

System value changed to the Port VLAN identifier in the TLV.

The system value is changed to the TLV VLAN Identifier. L2Q will be set to 1 (ON).

VLAN Name TLV is only effective if:

● The telephone is not statically

programmed.

● The telephone is not registered with the

Call Server.

● Name begins with VOICE (case does not

matter).

● The VLAN is not zero.

If VLAN Name causes the telephone to change VLAN and the telephone already has an IP Address, the telephone will release the IP Address and send out a new DHCPDISCOVER on the new VLAN ID.

System value will be changed to TLV VLAN Identifier. L2Q will be set to 1(on). Network Policy TLV is only effective if:

● The telephone is not statically

programmed.

● Name begins with VOICE (case does not

matter).

● The new VLAN ID is different than the

original VLAN ID.

● The VLAN is not zero.

If VLAN Name causes the telephone to change VLAN and the telephone already has an IP Address, the telephone will release the IP Address and send out a new DHCPDISCOVER on the new VLAN ID.

MCIPADD Proprietary Call

Server TLV

TLSSRVR, HTTPSRVR and

Proprietary File Server TLV

TFTPSRVR

100 4600 Series IP Telephone LAN Administrator Guide

MCIPADD will be set to this value if it has not already been set.

TLSSRVR, HTTPSRVR and TFTPSRVR will be set to this value if none of them have already been set.

Avaya 4600 Administrator's Guide

Specifications and Main Features

Frequently Asked Questions

User Manual

4600 Series IP Telephone LAN Administrator Guide

Contents

Chapter 1: Introduction

About This Guide

Intended Audience

Document Organization

Change History

What’s New in This Release

Terms Used in This Guide

Conventions Used in This Guide

Symbolic Conventions

Typographic Conventions

Online Documentation

Related Documents

IETF Documents

ITU Documents

ISO/IEC, ANSI/IEEE Documents

Customer Support

Introduction

Overview of Voice over IP (VoIP)

Data and Voice Network Similarities

Delay and Jitter

Tandem Coding

Voice Coding Standards

Telephony Protocols

DHCP

TFTP

HTTP

SNMP

Network Assessment

4600 Series IP Telephones

Software

DHCP and File Servers

H.323

Registration and Authentication

Time-to-Service (TTS)

Registration and Authentication

WAN Considerations

Initialization Process

Step 1: Telephone to Network

Step 2: DHCP Server to Telephone

Step 3: Telephone and File Server

Step 4: Telephone and the Call Server

TCP/UDP Port Utilization

Suggestions for Installation and Configuration

Reliability and Performance

IP Address Lists and Station Number Portability

Security

Chapter 3: Requirements

Introduction

Hardware Requirements

Additional Hardware Requirements

Software Requirements

Chapter 4: Server Administration

Introduction

Parameter Data Precedence

Administering H.323 and SIP IP Telephones on the Same Network

Administering 4600 Series IP Telephones on Avaya Media Servers (H.323 Only)

DEFINITY Releases 9, 9.5, 10, and Avaya Communication Manager Software Release 1.1+

DEFINITY Release 8.4

DHCP and File Servers

Software Checklist

Required Network Information

DHCP

Choosing a DHCP Configuration

DHCP Software Alternatives

DHCP Generic Setup

Windows NT 4.0 DHCP Server

Verifying the Installation of the DHCP Server

Initial Configuration

Creating a DHCP Scope for the IP Telephones

Editing Custom Options

Adding the DHCP Option

Activating the Leases

Verifying Your Configuration

Windows 2000 DHCP Server