Avaya 9600 User Manual

Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Administrator Guide Release 2.0

16-601944
Issue 2
December 2007
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, A vaya 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.
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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
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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
.

Contents

Chapter 1: Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
About This Guide . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 7
Major Differences Between 9600 Series SIP IP and 9600 Series H.323 IP Telephones 8
Features & Functions Supported by H.323 and Not Supported by SIP: . . . . 9
Change History . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 9
What’s New in SIP Software Release 2.0. . . . . . . . . . . . . . . . . . . . . . . 10
Document Organization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
Other Documentation . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 13
Chapter 2: Administration Overview and Requirements . . . . . . . . . 15
9600 Series IP Telephones . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 15
Parameter Data Precedence . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 18
The Administrative Process. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Administrative Checklist . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 19
Telephone Initialization Process . . . . . . . . . . . . . . . . . . . . . . . . . . . 21
Step 1: Telephone to Network . . . . . . . . . . . . . . . . . . . . . . . . . . 21
Step 2: Telephone to LLDP-Enabled Network . . . . . . . . . . . . . . . . . . 21
Step 3: Telephone to DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . 22
Step 4: Telephone and File Server . . . . . . . . . . . . . . . . . . . . . . . . 22
Step 5: Telephone and the SES Server. . . . . . . . . . . . . . . . . . . . . . 22
Error Conditions . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 23
Chapter 3: Network Requirements . . . . . . . . . . . . . . . . . . . . . 25
Network Assessment . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
Hardware Requirements. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 25
Server Requirements . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 26
HTTP/HTTPS Server . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
Network Time Protocol (NTP) Server. . . . . . . . . . . . . . . . . . . . . . . 27
Required Network Information . . . . . . . . . . . . . . . . . . . . . . . . . . . . 27
Other Network Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
SNMP . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 28
Registration and Authentication . . . . . . . . . . . . . . . . . . . . . . . . . 28
Reliability and Performance. . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
IEEE 802.1D and 802.1Q. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 29
Network Audio Quality Display on 9600 Series SIP IP Telephones . . . . . . 30
SIP Station Number Portability . . . . . . . . . . . . . . . . . . . . . . . . . . 30
Issue 2 December 2007 3
Contents
TCP/UDP Port Utilization . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 31
Security. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 34
Registration and Authentication . . . . . . . . . . . . . . . . . . . . . . . 35
Chapter 4: Communication Manager Administration . . . . . . . . . . . 37
Call Server Requirements. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
Switch Compatibility. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 37
Communication Manager Administrative Requirements . . . . . . . . . . . . . . 37
System-Level Preparation Tasks . . . . . . . . . . . . . . . . . . . . . . . . . 38
SIP Trunk Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 38
Call Routing Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
IP Interface and Addresses . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
UDP Port Selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 39
RSVP and RTCP/SRTCP. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
QoS . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
IEEE 802.1D and 802.1Q. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
NAT . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
DIFFSERV . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 40
Voice Mail Integration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
Auto Hold. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
Call Transfer Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 41
Conferencing Call Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . 42
Telephone Administration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 42
CM/SIP IP Telephone Configuration Requirements . . . . . . . . . . . . . . . . . 44
Administering Stations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 48
Administering Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 49
Chapter 5: SIP Enablement Services (SES) Administration . . . . . . . 51
Introduction . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 51
Using the Web Browser to Configure SES. . . . . . . . . . . . . . . . . . . . . . 51
Chapter 6: Server Administration . . . . . . . . . . . . . . . . . . . . . 53
Software Checklist. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
DHCP and File Servers . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 53
DHCP Server Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 54
Configuring DHCP for 9600 Series SIP IP Telephones . . . . . . . . . . . . . 54
DHCP Generic Setup . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 56
4 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0
Windows NT 4.0 DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . . . 59
Verifying the Installation of the DHCP Server . . . . . . . . . . . . . . . . 59
Creating a DHCP Scope for the IP Telephones . . . . . . . . . . . . . . . 60
Editing Custom Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
Adding the DHCP Option . . . . . . . . . . . . . . . . . . . . . . . . . . . 61
Activating the Leases . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 62
Verifying Your Configuration . . . . . . . . . . . . . . . . . . . . . . . . . 62
Windows 2000 DHCP Server . . . . . . . . . . . . . . . . . . . . . . . . . . . 63
Verifying the Installation of the DHCP Server . . . . . . . . . . . . . . . . 63
Adding DHCP Options. . . . . . . . . . . . . . . . . . . . . . . . . . . . . 65
Activating the New Scope. . . . . . . . . . . . . . . . . . . . . . . . . . . 65
HTTP Generic Setup. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 66
Chapter 7: Telephone Software and Binary Files . . . . . . . . . . . . . 67
General Download Process . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
Software . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 67
9600 Series SIP IP Telephone Scripts and Binary Files. . . . . . . . . . . . . . . 68
Choosing the Right Binary File and Upgrade Script File . . . . . . . . . . . . 68
Upgrade Script File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Settings File . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 69
Contents of the Settings File . . . . . . . . . . . . . . . . . . . . . . . . . . . 70
The GROUP System Value . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 72
Contents
Chapter 8: Administering Telephone Options . . . . . . . . . . . . . . . 73
Administering Options for the 9600 Series SIP IP Telephones. . . . . . . . . . . 73
VLAN Considerations . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
VLAN Tagging . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 94
VLAN Detection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 95
VLAN Default Value and Priority Tagging . . . . . . . . . . . . . . . . . . . . 95
VLAN Separation. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 96
DNS Addressing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
IEEE 802.1X . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 98
802.1X Pass-Through and Proxy Logoff . . . . . . . . . . . . . . . . . . . . . 99
802.1X Supplicant Operation . . . . . . . . . . . . . . . . . . . . . . . . . . . 99
Link Layer Discovery Protocol (LLDP) . . . . . . . . . . . . . . . . . . . . . . . . 101
Visiting User Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 105
Emergency Number Administration . . . . . . . . . . . . . . . . . . . . . . . . . 106
Local Administrative (Craft) Options Using the Telephone Dialpad . . . . . . . . 107
Language Selection . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 107
Issue 2 December 2007 5
Contents
Enhanced Local Dialing . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 108
Enhanced Local Dialing Requirements . . . . . . . . . . . . . . . . . . . . . 110
Chapter 9: Administering Applications and Options . . . . . . . . . . . 111
Customizing Telephone Applications and Options . . . . . . . . . . . . . . . . . 111
Avaya “A” Menu Administration . . . . . . . . . . . . . . . . . . . . . . . . . . . 112
Administering Standard Avaya Menu Entries . . . . . . . . . . . . . . . . . . 112
Administering the WML Browser . . . . . . . . . . . . . . . . . . . . . . . . . 112
Appendix A: Glossary of Terms . . . . . . . . . . . . . . . . . . . . . . 115
Appendix B: Related Documentation . . . . . . . . . . . . . . . . . . . 119
IETF Documents . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
ITU Documents. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 119
ISO/IEC, ANSI/IEEE Documents . . . . . . . . . . . . . . . . . . . . . . . . . 119
Appendix C: Sample Station Forms . . . . . . . . . . . . . . . . . . . . 121
Index . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 135
6 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

Chapter 1: Introduction

About This Guide

This guide is for personnel who administer Avaya Communication Manager, DHCP, HTTP/ HTTPS servers for 9600 Series SIP IP Telephones, a Local Area Network (LAN), SIP Enablement Services (SES) or a Network Time server.
The 9600 Series IP Telephones use Internet Protocol (IP) technology with Ethernet line interfaces and support both SIP and H.323 protocols. The 9600 Series IP Telephones provide support for DHCP, HTTP, and HTTPS over IPv4/UDP, which enhance the administration and servicing of the telephones. These telephones use DHCP to ob t ain dynamic IP Add resses, and HTTPS or HTTP to download new versions of software or customized settings for the telephones.
!
Important:
Important: This document covers administration for 9600 Series SIP IP Telephones only . For
administration for 9600 Series IP Telephones using the H.323 protocol, see the
Avaya one-X™ Deskphone Edition for 9600 Series IP Telephones Administrator Guide (Document Number 16-300698), available at: www.avaya.com/support
.
This document does not cover administration for Avaya Distributed Office. Full documentation for Avaya Distributed Office is available on the Avaya support Web site, www.avaya.com/support
Avaya does not provide product support for many of the products mentioned in this document. Take care to ensure that there is adequate technical support available for servers used with any 9600 Series IP and/or SIP IP Telephone system. If the servers are not functioning correctly, the 9600 Series IP Telephones might not operate correctly.
.
Issue 2 December 2007 7
Introduction

Major Differences Between 9600 Series SIP IP and 9600 Series H.323 IP Telephones

Review this section if your administrative environment includes both SIP and H.323 signaling protocols for 9600 Series IP Telephones.
General IP Telephony - Two major protocols handle 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 simultaneously over the same network, but in general, no endpoint supports both p rotocols at the same time. Neither 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.
Signaling - 96xx Series IP Telephones ship from the factory with H.323 signaling. To use the SIP protocol, applicable H.323 96xx Series IP Telephones must be appropriately converted and configured. See the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide for detailed conversion/configuration information.
Avaya Communication Manager Release - 9600 Series SIP IP Telephones are supported only by Communication Manager Release 4.0 and greater. SIP telephones use Avaya OPS (Outbound SIP Proxy) features on the “trunk” side of Avaya Communication Manager whereas the H.323 (IP) telephones are supported on the “line” side of the Communication Manager. When a 9600 Series SIP IP Telephone is running under Communication Manager Release 5.0, an additional feature, Extend Call, is available.
Required Servers - SIP 9600 Series IP Telephones use two [additional] servers that H.323 telephones do not:
- SIP Proxy server - provided by SIP Enablement Services (SES) software, and
- Network Time server - which controls time-related parameters.
These servers are not necessarily separate hardware units.
Features & Functions supported by H.323 9600 Series IP Telephones, Not SIP - Button modules are not currently supported by 9600 SIP IP Telephones.
Backup/Restore - 9600 Series (H.323) IP Telephones use HTTP to store backup files. 9600
Series SIP IP Telephones use the Personal Profile Manager (PPM) functionality within SIP Enablement Services (SES) for backup and restore functions.
Settings File & System Parameters - Both SIP and H.323 9600 Series IP Telephones (and 4600 Series IP Telephones) use the same settings file. Some of the same system parameters are used, however, numerous SIP-specific parameters support SIP operation only. In H.323 9600 Series IP Telephones, the parameters OPSTAT and APPSTAT control all user interface functions, whereas 9600 Series SIP IP Telephones use a separate parameter (for example ENABLE_CONTACTS, ENABLE_CALLLOG) for each user interface function.
Supported by
8 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

Change History

Language Support - SIP telephones support the same languages as H.323 telephones, with
the exception of Hebrew. SIP does not support Hebrew or the English Large Text Font for any language. Further, all SIP language files have .xml file extensions whereas H.323 language files have .txt file extensions.
SNMP & MIBs - Although both SIP and H.323 telephones support SNMP v2c and have custom Management Information Bases (MIBs), the MIBs are formatted somewhat differently.
RSVP & VMON (VMM) - 9600 Series SIP IP Telephones do not use RSVP (Resource ReSerVation Protocol) software to provide real-time monitoring and historical data of audio quality for VoIP calls. 9600 Series SIP IP Telephones do support Avaya Voice over IP (VoIP) Monitoring Manager (VMON, now called VMM). 9600 Series IP T elephones use both RSVP and VMON.
QoS - Unlike H.323 telephones, 9600 Series SIP IP Telephones do not use Avaya Communication Manager to set Quality of Service (QoS). The SIP IP telephones use the parameters L2QAUD, L2QSIG, DSCP AUD, and DSCPSIG (described in Table 11:
9600 Series
SIP IP Telephones Customizeable System Parameters).
NAT - 9600 Series SIP IP Telephones do not support Network Address T ranslation (NAT); 9600 Series IP (H.323) Telephones do support NAT.

Features & Functions Supported by H.323 and Not Supported by SIP:

SIP Software Release 2.0 SIP Software Release 1.0
Calltype Digit Conversion
RSVP
Remote Ping & Trace Route
SBM24 Button Modules
Push (Top Line, web page, and/or audio)
Link Layer Discovery Protocol (LLDP)
GigE (Gigabit Ethernet)
Calltype Digit Conversion
IEEE 802.1X
RSVP
VMON
Remote Ping & Trace Route
Web browser
SBM24 Button Modules
Push (Top Line, web page, and/or audio)
Autodial feature buttons
Change History
Issue 1 This document was issued for the first time in May 2007 to support the first release of
9600 Series SIP IP Telephones.
Issue 2 This is the current version of the document, revised and issued in December, 2007 to
support SIP IP Software Release 2.0. This release provides the 9600 SIP IP Telephones with similar functionality to their H323 9600 IP Telephone counterparts, despite their signaling protocol differences. Release 2.0 introduces several new functions, new configuration parameters, and adds telephone models 9630G and 9640G. What’s New in
SIP Software Release 2.0 describes this release in more detail.
Issue 2 December 2007 9
Introduction

What’s New in SIP Software Release 2.0

New material in this issue to support SIP Release 2.0 software includes: New GigE Models Support SIP - This release extends SIP capability to two additional
telephones, the 9630G and 9640G. Both models provide built-in Gigabyte Ethernet (GigE) support, but are otherwise identical to their 9630 and 9640 SIP IP telephone counterparts.
Language Support - 9600 Series SIP IP Telephones now support 13 languages. See
Language Selection
Emergency Button - Administrators can now program an “Emergency” number using the new
PHNEMERGNUM
logged into the telephone from which they are calling for assistance. For more information, see
Emergency Number Administration
Administration Enhancements - SIP Software Release 2.0 supports functionality introduced on Avaya Communication Manager Release 5.0 and SIP Enablement Services (SES) Release
5.0.
Visiting User Support - Visiting use r su pport allows users to e asily move be tween g eograp hic locations while retaining their telephone extension and settings. 9600 Series SIP IP Tele phones can be provisioned through the settings file VU_MODE modes:
on page 107 for more information.
parameter. Users can dial the Emergency Number whether or not they are
.
configuration parameter to one of three
No Visiting User - the telephone operates “normally” and has no user interface impact for
normal operation. The telephone can be forced to a “registered Inactive” state when a visiting user registers elsewhere.
Optional Visiting User - the telephone prompts the user at registration time if they are
visiting or not.
Forced Visiting User - the telephone allows only visiting user registrations.
For more information, see Visiting User Administration
.
Link Layer Discovery Protocol (LLDP) - 9600 Series SIP IP Telephones now support link layer discovery protocol. See Link Layer Discovery Protocol (LLDP)
for information.
802.1X - 9600 Series SIP IP Telephones now support IEEE standard 802.1X for increased
security. The new configuration parameter DOT1X defines the 802.1X operational mode. The new parameter DOT1XSTAT enables/disables 802.1X. The new parameter DOT1XEAPS specifies the authentication method to use with 802.1X. These parameters can be set through the settings file or on a per-phone basis using a local Craft procedure.
Support for Non-Avaya (Third Party) Environments - Several parameters, most notably ENABLE_AVAYA_ENVIRONMENT, have been added to cover operation for either:
an Avaya environment, which provisions SIP/AST features and uses Personal Profile
Manager (PPM) for download and backup/restore, or
a non-Avaya mode, which complies with 3rd party standard SIP proxy with provision for
SIPPING 19 feature.
10 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0
What’s New in SIP Software Release 2.0
WML Applications/Browser - 9600 Series SIP IP Telephones now provide access to WML
applications via a WML Browser, as described in Chapter 9:
Administering Applications and
Options.
New, Revised, and Deleted Configuration Parameters - The following configuration parameters have been added for this release and are linked to the table that describes them in detail:
CALL_TRANSFER_ MODE
CALLFWDADDR
CALLFWDDELAY
CALLFWDSTAT
CNAPORT
CNASRVR
CONFIG_SERVER_ SECURE_MODE
COVERAGEADDR
DIALPLAN
DOT1X
DOT1XEAPS
DOT1XSTAT
ENABLE_AVAYA_ ENVIRONMENT
INTER_DIGIT_TIMEOUT (replaces INTER_DIGIT_DIALING_TIMEOUT_DURATION)
LAST_LOGIN_STATUS (system-set only)
LLDP_ENABLED
MWISRVR
NO_DIGITS_TIMEOUT
PHNEMERGNUM
PHNNUMOFSA
POE_CONS_SUPPORT
PRESENCE_SERVER
PROVIDE_EDITED_ DIALING
PROVIDE_EXCHANGE_CALENDAR
PROVIDE_EXCHANGE_CONTACTS
QKLOGINSTAT
RTCPCONT
Issue 2 December 2007 11
Introduction
RTCPMON
RTCPMONPORT
SIP_MODE
SIPCONFERENCECONTINUE
TLSSRVRID
VU_MODE
VU_TIMER
WMLEXCEPT
WMLHOME
WMLIDLETIME
WMLIDLEURI
WMLPORT
WMLPROXY
The following parameters have been modified or renamed:
Parameters PHYxDUPLEX and PHYxSPEED were combined. PHY1SPEED has been
renamed to PHY1_OPERATIONAL_MODE. This parameter now includes the current duplex mode. PHY2SPEED has been renamed to PHY2_OPERATIONAL_MODE. This parameter now includes the current duplex mode.
The OUTBOUND_SUBSCRIPTION_REQUEST_DURATION default value has been
changed from 17280000 to 86400 seconds. This parameter can now be set through the settings file.
The dimensions for SNTP_SYNC_INTERVAL and
SNTP_SYNC_RANDOMIZATION_INTERVAL have changed from seconds to minutes.
EXCHANGE_CONTACTS_ENABLED has been renamed to
USE_EXCHANGE_CONTACTS.
EXCHANGE_CALENDAR_ENABLED has been renamed to
USE_EXCHANGE_CALENDAR.
The default value definition of ENABLE_G726 has changed from 0 to 1.
The default values and sidetone definitions of the audio parameters AUDIOSTHD and
AUDIOSTHS
WAIT_FOR_REGISTRATION_TIMER can now be set through the settings file.
have been modified.
The following configuration parameters are no longer valid and have been removed:
PHY1DUPLEX
PHY2DUPLEX
INTER_DIGIT_DIALING_TIMEOUT_DURATION
12 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

Document Organization

The guide contains the following sections:
Document Organization
Chapter 1: Chapter 2: Administration
Overview and Requirements Chapter 3:
Requirements Chapter 4: Communication
Manager Administration Chapter 5: SIP Enablement
Services (SES) Administration Chapter 6: Server Administration Describes DHCP and HTTP/HTTPS administration for the 9600
Chapter 7: and Binary Files
Chapter 8: Administering Telephone Options
Chapter 9: Applications and Options
Appendix A: Glossary of Terms Provides a glossary of terms used in this document or which can be
Introduction Provides an overview of this document.
Provides an overview of the administrative process and describes general hardware, software, and operational requirements.
Network
Telephone Software
Administering
Describes administrative requirements for your Local Area Network.
Describes how to administer Avaya Communication Manager to operate with 9600 Series SIP IP Telephones.
Covers SIP Enablement Services (SES) configuration for 9600 Series SIP IP Telephones.
Series IP Telephones. Describes telephone software, covers software downloads, and
provides information about the configuration file. Describes how to use file parameters and options to administer
9600 Series SIP IP Telephones. Covers backup and restoration of telephone data. Also describes how to use local procedures to customize a single telephone from the dialpad.
Describes customizeable application-specific parameters, to provide administrative control of telephone functions and options.
applicable to 9600 Series SIP IP Telephones.
Appendix B: Related Documentation
Appendix C: Sample Station Forms
Provides references to Web sites with external documents that relate to telephony in general, and can provide additional information about specific aspects of the telephones.
Provides examples of Avaya Communication Manager forms related to system wide and individual telephone administration.

Other Documentation

See the Avaya support site at http://www.avaya.com/support for 9600 Series SIP IP Telephone technical and end user documentation.
See Appendix B: Related Documentation such as those published by the Internet Engineering Task Force (IETF) and the International Telecommunication Union (ITU).
for Web sites that list related, non-Avaya documents,
Issue 2 December 2007 13
Introduction
14 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0
Chapter 2: Administration Overview and
Requirements

9600 Series IP Telephones

The 9600 Series IP Telephones currently support the H.323 signaling protocol and the SIP signaling protocol.
The H.323 standard provides for real time audio, vide o, 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 Secure Real Time Transfer Protocol (SRTP)
Real Time Control Protocol (RTCP) and Secure Real Time Control Protocol (SRTCP)
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)
The 9600 Series SIP IP Telephones support Media Encryption (SRTP) and use built-in Avaya SIP Certificates for trust management. Trust management involves downloadin g certificates for additional trusted Certificate Authorities (CA) and the policy management of those CAs. Identity management is handled by Simple Certificate Enrollment Protocol (SCEP) with phone certificates and private keys.
The 9600 Series IP Telephones are loaded with either H.323 or SIP software as part of initial script file administration and initialization during installation. Post-installation, sof tware upgrades automatically download using the proper signaling protocol.
Issue 2 December 2007 15
Administration Overview and Requirements
The parameters under which the 9600 Series SIP IP Telephones need to operate are summarized as follows:
Telephone Administration on the Communication Manager (CM) call server, as covered in
Chapter 4:
Administration on SIP Enablement Services (SES), as covered in Chapter 5: SIP
Communication Manager Administration.
Enablement Services (SES) Administration.
IP Address management for the telephone, as covered in Chapter 6: Server
Administration for dynamic addressing. For static addressing, see the Avaya one-X™
Deskphone Edition for 9600 Series SIP IP Telephones Installation and Maintenance Guide.
Tagging Control and VLAN administration for the telephone, if appropriate, as covered in
Chapter 8:
Quality of Service (QoS) administration for the telephone, if appropriate. QoS is covered in
QoS
Protocol administration, for example, Simple Network Management Control (SNMP) and
Administering Telephone Options.
on page 29 and QoS on page 40.
Link Layer Discovery Protocol (LLDP).
Interface administration for the telephone, as appropriate. Administer the telephone to
LAN interface using the PHY1 parameter described in Chapter 3: Administer the telephone to PC interface using the PHY2 parameter described in “Interface Control” in the Avaya one-X™ Deskphone Edition for 9600 Series SIP IP
Telephones Installation and Maintenance Guide.
Network Requirements.
Application-specific telephone administration, if appropriate, as described in Chapter
8: Administering Telephone Options. An example of application-specific data is
Web-specific information required for the optional Web browser application.
Table 1
indicates that you can administer system configuration parameters in a variety of ways
and use a variety of administrative mechanisms like:
Maintaining the information on the call server.
Manually entering the information by means of the telephone dialpad.
Administering the DHCP server.
Editing the configuration file on the applicable HTTP or HTTPS file server.
User modification of certain parameters, when given administrative permission to do so.
Note:
Note: Not all parameters can be administered on all administrative mechanisms.
16 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0
Table 1: Administration Alternatives and Options for 9600 Series SIP IP Telephones
Administrative
Parameter(s)
Mechanisms
For More Information See:
9600 Series IP Telephones
Telephone Administration
Avaya Communication Manager and SES
IP Addresses DHCP
(strongly
Chapter 4: Chapter 6: B: Related Documentation.
DHCP and File Servers on page 53, and especially DHCP Server Administration
Communication Manager Administration, Server Administration, and Appendix
on page 54.
recommended) Settings file Chapter 7: Telephone Software and Binary Files and
Manual administration at the telephone
LLDP
Tagging and VLAN LLDP
DHCP
Chapter 8:
“St atic Addressing Inst allation” in the Avaya one-X™
Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.
Link Layer Discovery Protocol (LLDP) Link Layer Discovery Protocol (LLDP)
DHCP Server Administration
Administering Telephone Options.
on page 101. on page 101.
on page 54, and
Chapter 8: Administering Telephone Options.
Settings file DHCP and File Servers
on page 53 and
Chapter 8: Administering Telephone Options.
Manual administration at the telephone
Network Time Server (NTS)
DHCP Settings file
Quality of Service Settings file Chapter 8:
“St atic Addressing Inst allation” in the Avaya one-X™
Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.
DHCP Server Administration on page 54 and Network Time Protocol (NTP) Server on page 27.
Administering Telephone Options.
Interface DHCP DHCP and File Servers on page 53, and Chapter
7: Telephone Software and Binary Files.
Settings file (strongly
DHCP and File Servers on page 53, and Chapter 7: Telephone Software and Binary Files.
recommended) LLDP
Link Layer Discovery Protocol (LLDP) on page 101.
Application ­specific parameters
Manual administration at the telephone
“Secondary Ethernet Interface Enable/Disable” in the
Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.
DHCP DHCP and File Servers on page 53, and especially
DHCP Server Administration
on page 54. Also,
Chapter 8: Administering Telephone Options.
Settings file (strongly recommended)
DHCP and File Servers
on page 53, and especially
HTTP Generic Setup on page 66. Also, Chapter 8:
Administering Telephone Options.
Issue 2 December 2007 17
Administration Overview and Requirements
General information about administering DHCP servers is covered in DHCP and File
Servers on page 53, and more specifically, DHCP Server Administration on page 54. General
information about administering HTTP servers is covered in DHCP and File Servers specifically, HTTP Generic Setup
. Once you are familiar with that material, you can administer
telephone options as described in Chapter 8:

Parameter Data Precedence

As shown in Table 1: Administration Alternatives and Options for 9600 Series SIP IP
Telephones, you can administer a given parameter in a number of ways. If a given parameter is
administered through different mechanisms, the last server to provide the parameter has precedence. The precedence, from lowest to highest, is:
1. LLDP
2. DHCP
3. Settings file
, and more
Administering Telephone Options.
4. Personal Profile Manager (PPM),
Note:
Note: Exception: In the case of the parameter SIPDOMAIN, the settings file has a
higher precedence than PPM.
5. Manual administration, unless the system parameter USE_DHCP is set to 1 (Get IP Address automatically by DHCP), or backup file data obtained through PPM.
For example, if the SIP outbound proxy server address is defined to have the precedence information so that the value retrieved from DHCP server has a lower precedence than the value retrieved from the settings file, and the value retrieved from the settings file is higher than the value retrieved from PPM, then the following determination occurs:
If the most recent value the telephone has is from DHCP and new server address
information is retrieved from the settings file, the telephone will use the new value from the settings file.
If later on, the telephone receives a new server address value from PPM, it will not use this
value because PPM’s precedence as a data source for the server address is lower than the current value (which came from the settings file).
If the server to which a specific telephone points is changed manually using the Craft
ADDR procedure, that value now takes precedence over the previous value.
18 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0
Note:
Note: 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. For the L2QVLAN and L2Q system values, LLDP settings of VLAN IDs are the absolute authority only if the LLDP task receives the VLAN IDs before DHCP, and the DHCP client of the telephone is activated at all. If the LLDP task receives the VLAN IDs after DHCP negotiation, several criteria must be successful before the telephone accepts VLAN IDs from LLDP. For more information, see Link Layer Discovery Protocol (LLDP)

The Administrative Process

The following list depicts administration for a typical 9600 Series SIP IP Telephone network. Your own configuration might differ depending on the servers and system you have in place.
1. Avaya Communication Manager (4.0 or greater) administered for 9600 Series IP Telephones. All 9600 Series SIP IP Telephones must be administered with the 4620SIP station type.
The Administrative Process
on page 101.
2. SES (SIP Enablement Services) administered.
3. LAN and applicable servers (file servers, Network Time server) administered to accept the telephones.
4. Telephone software downloaded from the Avaya support site.
5. 46xxsettings file updated with site-specific and SIP-specific information, as applicable.
6. 9600 Series Telephones installed. For more information, see the Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.
7. Individual 9600 Series IP Telephones updated using Craft procedures, as applicable. For more information, see “Local Administrative Procedures” in the Avaya one-X™ Deskphone Edition for 9600 SIP P Telephones Installation and Maintenance Guide.

Administrative Checklist

Use the following checklist as a guide to system and LAN administrator responsibilities. This high-level list helps ensure that all telephone system prerequisites and requirements are met prior to telephone installation.
Note:
Note: One person might function as both the system administrator and the LAN
administrator in some environments.
Issue 2 December 2007 19
Administration Overview and Requirements
Table 2: Administrative Checklist
Task Description For More Information See:
Network Requirements Assessment
Administer Avaya Communication Manager
Administer the Proxy Server
DHCP server installation
Administer DHCP application
Administer Network Time Server
HTTP/HTTPS server installation
Determine that network hardware is in place and can handle telephone system requirements.
Verify that the call server is licensed and is administered for Voice over IP (VoIP).
Verify the individual telephones are administered as desired.
Administer for SIP Enablement Services (SES).
Install a DHCP application on at least one new or existing PC on the LAN.
Add IP telephone administration to DHCP application.
Set value(s) for Simple Network Time Protocol (SNTP)
Install an HTTP/HTTPS application on at least one new or existing PC on the LAN.
Chapter 3:
Network Requirements.
Chapter 4: Communication Manager Administration.
Chapter 4:
Communication Manager
Administration.
Installing and Administering SIP Enablement Services (03-600768),
available on the Avaya support Web site,
http://www.avaya.com/support.
Vendor-provided instructions.
DHCP Server Administration
in
Chapter 6: Server Administration.
Option 42 under DHCP Generic
Setup.
Vendor-provided instructions.
Binary file(s), script file, and settings file installation on HTTP/ HTTPS server
Modify settings file as needed
Download the files from the Avaya support site.
Edit the settings file as necessary for your environment, using your
http://www.avaya.com/support Chapter 7: Telephone Software and
Binary Files. Chapter 7:
Binary Files.
own tools.
Administer telephones locally as applicable
As a Group: The GROUP System Value on
page 72 and the Avaya one-X™
Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.
Individually: The applicable Craft Local
Procedures in the Avaya one-X™
Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.
20 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0
Telephone Software and
1 of 2

Telephone Initialization Process

Table 2: Administrative Checklist (continued)
Task Description For More Information See:
Installation of telephones in the network
Allow user to modify Options, if applicable
Set the following parameters in the settings file:
ENABLE_CALL_LOG ENABLE_CONTACTS ENABLE_MODIFY_CONTACTS ENABLE_PRESENCE
PROVIDE_OPTIONS_SCREEN PROVIDE_NETWORKINFO_SCR
EEN PROVIDE_LOGOUT
Telephone 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 9600 Series SIP IP Telephones relate to the routers and servers in your network.
Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.
9600 Series SIP IP Telephones Customizeable System Parameters.
2 of 2

Step 1: Telephone to Network

The telephone is appropriately installed and powered. After a short initialization process, the telephone identifies the LAN speed and sends a message out into the network, identifying itself and requesting further information. A router on the network re ceives and relays this message to the appropriate DHCP server.

Step 2: Telephone to LLDP-Enabled Network

An LLDP-enabled network provides information to the telephone, as described in Link Layer
Discovery Protocol (LLDP) on page 101. Among other data passed to the telephone is the IP
Address of the HTTP or HTTPS server.
Issue 2 December 2007 21
Administration Overview and Requirements

Step 3: Telephone to DHCP Server

The DHCP server provides information to the telephone, as described in DHCP and File
Servers on page 53. Among other data passed to the telephone is the IP Address of the HTTP
or HTTPS server.

Step 4: Telephone and File Server

The 9600 Series IP Telephones can download script files, binary files, certificates, language files, and settings files from either an HTTP or HTTPS 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 binary file the telephone must use. The binary file is the software that has the telephony functionality.
The telephone uses the script file to determine if it has the proper binary file. If the telephone determines the proper binary file is missing, the telephone requests an binary file download from the file server. The file server the n 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 as described in the next paragraph without downloading the binary file again.
The telephone checks and loads the binary file, then uses the script file to look for a settin gs file, if appropriate. The optional settings file can contain settings you have administered for any or all of the 9600 Series SIP IP Telephones in your network. For more information about this download process and settings file, see Chapter 7:

Step 5: Telephone and the SES Server

In this step, the telephone might prompt the user for an extension and password. The telephone uses that information to exchange a series of messages with SES, which in turn communicates with Avaya Communication Manager (CM). For a new installation and for full service, the user can enter the telephone extension and the SES password. For a restart of an existing installation, this information is already stored on the telephone, but the user might have to confirm the information. The telephone and SES and SES and CM exchange more messaging. The expected result is that the telephone is appropriately registered and CM call server data such as feature button assignments are downloaded.
For more information about the installation process, see the Ava ya one-X™ Deskphone Editio n for 9600 SIP IP Telephones Installation and Maintenance Guide.
Telephone Software and Binary Files.
22 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

Error Conditions

Assuming proper administration, most of the problems reported by telephone users are likely to be LAN-based. Quality of Service, server administration, and other issues can impact user perception of IP telephone performance.
The Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide covers possible operational problems that might be encountered after successful 9600 Series SIP IP Telephone installation. The User Guides for a specific tele phone model also contain guidance for users having problems with specific IP telephone applications.
Error Conditions
Issue 2 December 2007 23
Administration Overview and Requirements
24 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

Chapter 3: Network Requirements

Network Assessment

Perform a network assessment to ensure that the network will have the capacity for the expected data and voice traffic, and that it can support for all applications:
SIP,
DHCP,
HTTP/HTTPS, and
Jitter buffers
Also, QoS support is required to run VoIP on your configuration. For more information, see
Appendix B: Related Documentation
and DSCPSIG in Table 11:
9600 Series SIP IP Telephones Customizeable System
Parameters.
and the QoS parameters L2QAUD, L2QSIG, DSCPAUD,

Hardware Requirements

To operate properly, you need:
Category 5e cables designed to the IEEE 802.3af-2003 standard, for LAN powering,
TN2602 IP Media Processor circuit pack. Sites with a TN2302 IP Media Processor circuit
pack are strongly encouraged to install a TN2602 circuit p ack to benefit from the increased capacity.
TN799C or D Control-LAN (C-LAN) circuit pack.
!
Important:
Important: IP telephone firmware Release 1.0 or greater requires TN799C V3 or greater
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
To ensure that the appropriate circuit pack(s) are administered on your Communication Manager call server, see Chapter 4: information about hardware requirements in general, see the Avaya one-X™ Deskphone Edition for 9600 SIP IP Telephones Installation and Maintenance Guide.
.
Communication Manager Administration. For more
Issue 2 December 2007 25
Network Requirements

Server Requirements

Four server types can be configured for the 9600 Series IP Telephones:
DHCP server
HTTP or HTTPS server
SIP Proxy or Registration server
Network Time Protocol server for SNTP
Note:
Note: 9600 Series SIP IP Telephones need SIP Enablement Services (SES) to work
properly. The SIP Proxy and Registration servers reside on the SES server. Avaya Communication Manager (CM) is considered a “feature server” behind SES that provides Outboard Proxy SIP (OPS) features. SIP software Release 2.0 supports both SES 4.X and 5.X, but when the corresponding server is running SES 4.X, the telephones assume only those features compatible with SES 4.X.
While the servers listed provide different functions that relate to the 9600 Se ries IP Telephones, they are not necessarily different boxes. For example, DHCP provides network information whereas HTTP provides configuration and application file management, yet both functions can co-exist on one hardware unit. Any standards-based server is recommended.
For parameters related to Avaya Communication Manager information, see Chapter
4: Communication Manager Administration. For parameters related to DHCP and file servers,
see Chapter 6:
!
Important:
Important: The telephones obtain important information from the script files on the server(s)
and depend on the binary file for software upgrades. If these servers are unavailable when the telephones reset, the telephones will not operate properly. Some features might not be available. To restore them you need to reset the telephone(s) when the file server is available.

DHCP Server

Avaya recommends that a DHCP server and application be installed and that static addressing be avoided. Install the DHCP server and application as described in DHCP and File Servers page 53.
Server Administration.
on
26 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

HTTP/HTTPS Server

Administer the HTTP or HTTPS file server and application as described in HTTP Generic
Setup on page 66.

Network Time Protocol (NTP) Server

SIP IP telephones require NTP server support to set the time and date, used in system log time stamps and other time/date functions. The NTP server is typically needed by one or more servers within the enterprise. Administration of the NTP server is beyond the scope of this document.

Required Network Information

Before you administer DHCP and HTTP/HTTPS, as applicable, complete the information in
Table 3
configuration, complete Table 3
. If you have more than one router, HTTP/TLS server and subnetwork mask in your
for each DHCP server.
Required Network Information
The 9600 Series SIP IP Telephones support specifying a list of IP Addresses for a gateway/ router and the HTTP/HTTPS server . Each list can contain up to 255 tot al ASCII characters, with IP Addresses separated by commas with no intervening spaces. Depending on the specific DHCP application, only 127 characters might be supported.
When specifying IP Addresses for the file server, use either dotted decimal format (“xxx.xxx.xxx.xxx”) or DNS names. If you use DNS, the system value DOMAIN is appended to the IP Addresses you specify. If DOMAIN is null, the DNS names must be fully qualified, in accordance with IETF RFCs 1034 and 1035. For more information about DNS, see DHCP
Generic Setup on page 56 and DNS Addressing on page 98.
Table 3: Required Network Information Before Installation - Per DHCP Server
1. Gateway (router) IP Address(es)
2. HTTP server IP Address(es)
3. Subnetwork mask
4. HTTP server file path (HTTPDIR)
5. Telephone IP Address range
From: To:
6. DNS server address(es) If applicable.
7. HTTPS server address(es) If applicable.
Issue 2 December 2007 27
Network Requirements
The default file server file path is the “root” directory used for al l tra nsfe r s by the server. All files are uploaded to or downloaded from this default directory. In configurations where the upgrade script and binary files are in the default directory, do not use item 4 in Table 3
As the LAN or System Administrator, you are also responsible for:
Administering the DHCP server as described in Chapter 6: Server Administration.
Editing the configuration file on the applicable HTTP or HTTPS file server, as covered in
9600 Series SIP IP Telephone Scripts and Binary Files

Other Network Considerations

SNMP

The 9600 Series SIP IP Telephones are fully compatible with SNMPv2c 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 read-only Custom MIB. 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. For more information, see Chapter 6:
Series SIP IP Telephones Customizeable System Parameters.
Note:
Note: SNMP is disabled by default. Administrators must initiate SNMP by setting the
SNMPADD and SNMPSTRING system values appropriately.
For more information about SNMP and MIBs, see the IETF Web site listed in
Appendix B: Related Documentation
. The Avaya Custom MIB for the 9600 Series SIP IP Telephones is available for download in *.txt format on the Avaya support Web site at
http://www.avaya.com/support
.

Registration and Authentication

A 9600 Series SIP IP Telephone requires an outboard proxy SIP (OPS) extension on Avaya Communication Manager and a login and password on the SES Server to register and authenticate it. Registration is described in the Initialization process, in Step
the SES Server on page 22. For further information, see Installing and Administering SIP
Enablement Services R 4.0 (03-600766), available on the Avaya support Web site,
http://www.avaya.com/support
.
Server Administration and Table 11: 9600
5: Telephone and
28 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0

Reliability and Performance

All 9600 Series SIP IP Telephones respond to a ping or traceroute message sent from Avaya Communication Manager or any other network source. The telephones do not originate a ping or traceroute. The 9600 Series SIP IP Telephones offer and support “remote ping” and “remote traceroute.” The switch can instruct the telephone to originate 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. For more information, see your switch administration documentation.
If applicable, the telephones test whether the network Ethernet switch port supports IEEE
802.1D/q tagged frames by ARPing the router with a tagged frame. For more information, see
VLAN Considerations
on page 94. If your LAN environment includes Virtual LANs (VLANs),
your router must respond to ARPs for VLAN tagging to work properly.
QoS
Other Network Considerations
For more information about the extent to which your network can support any or all of the QoS initiatives, see your LAN equipment documentation. See QoS for the 9600 Series SIP IP Telephones.
All 9600 Series SIP IP Telephones provide some detail about network audio quality. For more information see, Network Audio Quality Display on 9600 Series SIP IP Telephones

IEEE 802.1D and 802.1Q

For more information about IEEE 802.1D and IEEE 802.1Q and the 9600 Series SIP IP Telephones, see IEEE 802.1D and 802.1Q Three bits of the 802.1Q tag are reserved for ide ntifying p acket priority to allow any one of eight priorities to be assigned to a specific packet.
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 critical data applications
3: Traffic meriting “extra-effort” by the network for prompt delivery, for example, executive
e-mail
2: Reserved for future use
0: The default priority for traffic meriting the “best-effort” for prompt delivery of the network.
1: Background traffic such as bulk data transfers and backups
on page 40 for QoS implications
on page 30.
on page 40 and VLAN Considerations on page 94.
Note:
Note: Priority 0 is a higher priority than Priority 1.
Issue 2 December 2007 29
Network Requirements

Network Audio Quality Display on 9600 Series SIP IP Telephones

All 9600 Series SIP IP Telephones give the user an opportunity to monitor network audio performance while on a call. For more information, see the telephone user guide.
While on a call, the telephones display network audio quality parameters in real-time, as shown in Table 4
Table 4: Parameters in Real-Time
Parameter Possible Values
Received Audio Coding G.711, G.722, G.726A, or G.729. Packet Loss No data or a percentage. Late and out-of-sequence packet s are
Packetization Delay No data or an integer number of milliseconds. The number
:
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.
reflects the amount of audio data in each RTP packet.
One-way Network Delay No data or an integer number of milliseconds. The number is
one-half the value RTCP or SR TCP computes for the round-trip delay.
Network Jitter Compensation Delay
The implication for LAN administration depends on the values the user reports and the specific nature of your LAN, like topology, loading, and QoS administration. This information gives the user an idea of how network conditions affect the audio quality of the current call. Avaya assumes you have more detailed tools available for LAN troubleshooting.
No data or an integer number of milliseconds reporting the average delay introduced by the jitter buffer of the telephone.

SIP Station Number Portability

The 9600 Series SIP IP Telephones provide station number portability. On startup or a reboot, the telephone attempts to establish communication with its home Personal Profile Manager (PPM)/SIP Enablement Services (SES) server based on the User Name and Password.
Assume a situation where the company has multiple locations in London and New York, all sharing a corporate IP network. Users want to take their telephone functionality from their offices in London to their New York office. When users start up their telephones in the new location and enter their credentials, the local SES/PPM server usually routes them to the local call server. With proper administration of the local SES/PPM server, the telephone knows to try its home SES/PPM server, the one in London. The user can then be automatically registered with the London SES/PPM server.
30 9600 Series SIP IP Telephones Administrator Guide SIP Release 2.0
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