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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
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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
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License Type(s):
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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
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and search for the document number in
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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
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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
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 20077
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 2.0SIP 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 2This 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 20079
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:
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 TermsProvides a glossary of terms used in this document or which can be
IntroductionProvides 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 200713
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 200715
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 AddressesDHCP
(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 fileChapter 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.
InterfaceDHCPDHCP 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 200717
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 200719
Administration Overview and Requirements
Table 2: Administrative Checklist
TaskDescriptionFor 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)
TaskDescriptionFor More Information See:
Installation of
telephones in the
network
Allow user to modify
Options, if applicable
Set the following parameters in the
settings file:
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 200721
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 200723
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 200725
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 200727
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 200729
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
ParameterPossible Values
Received Audio Coding G.711, G.722, G.726A, or G.729.
Packet LossNo data or a percentage. Late and out-of-sequence packet s are
Packetization DelayNo 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 DelayNo 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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