Copyright 2002, Avaya Inc.
All Rights Reserved, Printed in U.S.A.
Notice
Every effort was made to ensure that the information in this book
was complete and accurate at the time of printing. However,
information is subject to change.
Avaya Web Page
The world wide web home page for Avaya is:
http://www.avaya.com
variety of losses to your company, including but not limited to,
human/data privacy, intellectual property, material assets, financial resources, labor costs, and/or legal costs.
Federal Communications Commission Statement
Part 68: Network Registration Number. This equipment is reg-
istered with the FCC in accordance with Part 68 of the FCC Rules.
It is identified by FCC registration number AV1USA-43058-MFE.
Preventing Toll Fraud
Toll fraud is the unauthorized use of your telecommunications system by an unauthorized party (for example, a person who is not a
corporate employee, agent, subcontractor, or working on your
company’s behalf). Be aware that there is a risk of toll fraud associated with your system and that, if toll fraud occurs, it can result
in substantial additional charges for your telecommunications services.
Avaya Fraud Intervention
If you suspect that you are being victimized by toll fraud and you
need technical support or assistance and are in within the United
States, call the Technical Service Center Toll Fraud Intervention
Hotline at 1.800.643.2353. If you need technical support or assistance and are outside of the United States, contact the equipment
vendor from whom you purchased your equipment service maintenance contract. If you need to report toll fraud issues regarding a
public telephone, contact the in-country telephone service provider.
Providing Telecommunications Security
Telecommunications security of voice, data, and/or video communications is the prevention of any type of intrusion to, that is,
either unauthorized or malicious access to or use of, your company’s telecommunications equipment by some party.
Your company’s “telecommunications equipment” includes both
this Avaya product and any other voice/data/video equipment that
could be accessed via this Avaya product (that is, “networked
equipment”).
An “outside party” is anyone who is not a corporate employee,
agent, subcontractor, or working on your company’s behalf.
Whereas, a “malicious party” is anyone, including someone who
may be otherwise authorized, who accesses your telecommunications equipment with either malicious or mischievous intent.
Such intrusions may be either to/through synchronous (time-multiplexed and/or circuit-based) or asynchronous (character-, message-, or packet-based) equipment or interfaces for reasons of:
• Utilization (of capabilities special to the accessed equipment)
• Theft (such as, of intellectual property, financial assets, or
toll-facility access)
• Eavesdropping (privacy invasions to humans)
• Mischief (troubling, but apparently innocuous, tampering)
• Harm (such as harmful tampering, data loss or alteration,
regardless of motive or intent)
Be aware that there may be a risk of unauthorized intrusions associated with your system and/or its networked equipment. Also
realize that, if such an intrusion should occur, it could result in a
Part 68: Answer-Supervision Signaling. Allowing this equipment to be operated in a manner that does not provide proper
answer-supervision signaling is in violation of Part 68 Rules. This
equipment returns answer-supervision signals to the public
switched network when:
• Answered by the called station
• Answered by the attendant
• Routed to a recorded announcement that can be administered by the CPE user
This equipment returns answer-supervision signals on all DID
calls forwarded back to the public switched telephone network.
Permissible exceptions are:
• A call is unanswered
• A busy tone is received
• A reorder tone is received
Industry Canada (IC) Interference Information
NOTICE: This equipment meets the applicable Industry Canada
Terminal Equipment Technical Specifications. This is confirmed
by the registration number. The abbreviation, IC, before the
registration number signifies that registration was performed
based on a Declaration of Conformity indicating that Industry
Canada technical specifications were met. It does not imply that
Industry Canada approved the equipment.
Le Présent Appareil Nomérique n’émet pas de bruits radioélectriques dépassant les limites applicables aux appareils numériques
de la class A préscrites dans le reglement sur le brouillage
radioélectrique édicté par le Industrie Canada.
Trademarks
DEFINITY is a registered trademark of Avaya, Inc.
MultiVantage is a trademark of Avaya, Inc.
Ordering Information
Call:Avaya Publications Center
U.S. and Canada Voice 1 800 457 1235
Outside U.S. and Canada Voice +1 410 568 3680
U.S. and Canada Fax 1 800 457 1764
Outside U.S. and Canada Fax +1 410 891 0207
Write:GlobalWare Solutions
200 Ward Hill Avenue
Haverhill, MA 01835 USA
Attention: Avaya Account Manager
Email:totalware@gwsmail.com
For additional documents, refer to the section in About This Guide
titled Related Documents. An online copy of this and other related
Avaya product documentation can be found at: http://
www.avaya.com/support.
Obtaining Products
To learn more about Avaya products and to order products,
access the Avaya web site at http://www.avaya.com. Or call
the following numbers: customers 1 800 451 2100, account
executives 1 888 778 1880 (voice) or 1 888 778 1881 (fax).
European Union Declaration of Conformity
The “CE” mark affixed to the equipment means that it conforms to the referenced European Union (EU) Directives
listed below:
EMC Directive 89/336/EEC
Low-Voltage Directive 73/23/EEC
For more information on standards compliance, contact your
local distributor.
4600 Series IP Telephone LAN Administrator’s Guide
This guide provides a description of Voice over IP, describes how to administer the DHCP and
TFTP servers and covers how to troubleshoot operational problems with the 4600 Series IP
Telephones and the servers.
The 4600 Series IP Telephone product line is a supplement to Avaya’s IP Solutions platform.
1
Unless otherwise indicated, references in this document to "the DEFINITY® server" also
refer to the MultiVantage
TM
media servers.
Intended Audience1
This document is intended for personnel administering the DHCP and TFTP servers to support the
4600 Series IP Telephones and those administering the Local Area Network (LAN) itself.
CAUTION:
Many of the products mentioned in this document are not supported by Avaya. Care should
be taken to ensure there is adequate technical support available for the TFTP and DHCP
servers. If the TFTP or DHCP servers are not functioning correctly, the 4600 Series IP
Telephones may not operate correctly.
About This Guide
1-1
4600 Series IP Telephone LAN Administrator’s Guide
Document Organization1
The guide contains the following sections:
Chapter 1, Introduction
Provides an overview of the 4600 Series IP Telephone
Administrator’s document.
Chapter 2, Overview of Voice
over IP (VoIP)
Chapter 3, Requirements
Describes VoIP and factors influencing its performance that
must be considered when implementing this feature.
Describes the hardware and software requirements for Avaya’s
VoIP offering.
Chapter 4, Server
Administration
Chapter 5, Troubleshooting
Guidelines
Appendix A, Avaya - 46xx IP
Telephone MIB
Appendix B, Creating
Websites for the 4630 IP
Telephone
Appendix C, Creating
Websites for the 4620 IP
Telephone
Describes the administration of DHCP and TFTP for the 4600
Series IP Telephones.
Describes messages that may occur during the operation of the
4600 Series IP Telephones.
Provides the MIB specification for the 46xx IP Telephones
(4606, 4612, 4624, and 4630).
Provides information on creating and customizing websites for
viewing on the 4630 IP Telephone. Also describes the current
capabilities and limitations of the 4630’s web browser.
Provides information on creating and customizing websites for
viewing on the 4620 IP Telephone. Also describes the current
capabilities and limitations of the 4620’s web browser.
Change History1
Issue 1.0This document was issued for the first time in November 200 0.
Issue 1.1This version of the document, revised and issued in Ap ril 2001,
supports through DEFINITY
®
Release 9.
Issue 1.5 This version of the document was revised in June, 2001 to
support DEFINITY
®
Release 9.5.
Issue 1.6This version of the document was revised to support DEFINITY
Release 10 and the 4630 IP Telephone.
Issue 1.7This is the current version of this document, revised to support
MultiVantage
TM
Release 11 and the 4602 and 4620 IP
Telephones.
Document Organization
1-2
®
Terms Used in This Guide1
Introduction
802.1p
802.1Q
ARPAddress Resolution Protocol, used to verify that the IP address
CELPCode-excited linear-predictive; voice compression requiring only
CLANControl LAN, type of TN799 circuit pack.
DHCPDynamic Host Configuration Protocol, an IETF protocol used to
DiffServDifferentiated Services, an IP-based QoS mechanism.
DNSDomain Name System, an IETF standard for ASCII strings to
IETFInternet Engineering Task Force, the organization that produces
LANLocal Area Network.
LDAPLightweight Directory Access Protocol, an IETF standard for
802.1Q defines a layer 2 frame structure that supports VLAN
identification and a QoS mechanism usually referred to as
802.1p, but the content of 802.1p is now incorporated in 802.1D.
provided by the DHCP server is not in use by another IP
Telephone.
16 kbps of bandwidth.
automate IP Address allocation and management.
represent IP addresses.
standards for communications on the internet.
database organization and query exchange.
MACMedia Access Control, ID of an endpoint.
NAPTNetwork Address Port Translation.
NATNetwork Address Translation.
PSTNPublic Switched Telephone Network, the network used for
traditional telephony.
QoSQuality of Service, used to refer to a number of mechanisms
intended to improve audio quality over packet-based networks.
RRQRead Request packet, a message sent from the 4600 Series IP
Telephone to the TFTP server, requesting to download the
upgrade script and the application file.
RSVPResource ReSerVation Protocol, used by hosts to request
resource reservations throughout a network.
RTCPRTP Control Protocol, monitors quality of the RTP services and
can provide real-time information to users of an RTP service.
RTPReal-time Transport Protocol, provides end-to-end services for
real-time data (such as voice over IP).
TCP/IPTransmission Control Protocol/Internet Protocol, a network-layer
protocol used on LANs and internets.
Document Organization
1-3
4600 Series IP Telephone LAN Administrator’s Guide
TFTPTrivial File Transfer Protocol, used to provide downloading of
upgrade scripts and application files to the IP Telephones.
UDPUser Datagram Protocol, a connectionless transport-layer
protocol.
VLANVirtual LAN.
Conventions Used in This Guide1
This guide uses the following textual, symbolic, and typographic conventions to help you interpret
information.
Symbolic Conventions1
This symbol precedes additional information about a topic. This information is not
required to run your system.
CAUTION:
This symbol is used to emphasize possible harm to software, possible loss of data, or
possible service interruptions.
Typographic Conventions1
This guide uses the following typographic conventions:
commandWords printed in this type are commands that you enter into your
system.
deviceWords printed in this type indicate parameters associated with a
command for which you must substitute the appropriate value. For
example, when entering the mount command, device must be
replaced with the name of the drive that contains the installation disk.
AdministrativeWords printed in bold type are menu or screen titles and labels, or items
on menus and screens that you select to perform a task.
italicsItalic type indicates a document that contains additional information
about a topic.
<Enter>Words enclosed in angle brackets represent a single key that should be
pressed. These include <Ctrl>, <Enter>, <Esc>, <Insert>, and
<Delete>.
Document Organization
1-4
Online Documentation1
The online documentation for the 4600 Series IP Telephones is located at the following URL:
http://www.avaya.com/support
Related Documents1
■DEFINITY
This CD contains documentation that describes, among other things, how to administer a
DEFINITY switch with Release 8.4 software.
This document is provided with the DEFINITY Release 8.4 product.
■DEFINITY
This CD contains documentation that describes, among other things, how to administer a
DEFINITY switch with Release 9 software.
This document is provided with the DEFINITY Release 9 product.
■DEFINITY
®
Documentation Release 8.4
®
Documentation Release 9
®
Documentation Release 10
Introduction
This CD contains documentation that describes, among other things, how to administer a
DEFINITY switch with Release 10 software.
This document is provided with the DEFINITY Release 10 product.
■ Avaya MultiVantage
This document describes how to administer a switch with Avaya MultiVantage
This document is provided with the MultiVantage
TM
Software Documentation Release 11
TM
Release 11 product.
TM
software.
The following documents are available on the web site listed above under Online
Documentation.
■4600 Series IP Telephones Safety Instructions (for 4602/4606/4612/4620/4624/4630 IP
Telephones), Issue 1, July 2002 (555-233-779)
This document contains important user safety instructions for the 4600 Series IP Telephones.
■30A Switched Hub Set Up Quick Reference, Issue 2, July 2002 (Comcode 700234750;
Document Number 555-236-700)
This document contains important safety and installation information for the 30A Switched
Hub.
■4600 Series IP Telephone Installation Guide (555-233-128)
This document describes how to install 4600 Series IP Telephones. It also provides
troubleshooting guidelines for the 4600 Series IP Telephones.
■4602 IP Telephone User Guide (555-233-780)
This document provides detailed information about using the 4602 IP Telephone.
Online Documentation
1-5
4600 Series IP Telephone LAN Administrator’s Guide
■4606 IP Telephone User Guide (555-233-765) (Int’l 555-233-769)
This document provides detailed information about using the 4606 IP Telephone.
■4612 IP Telephone User Guide (555-233-766) (Int’l 555-233-770)
This document provides detailed information about using the 4612 IP Telephone.
■4620 IP Telephone User Guide (555-233-781)
This document provides detailed information about using the 4620 IP Telephone.
■4624 IP Telephone User Guide (555-233-768) (Int’l 555-233-771)
This document provides detailed information about using the 4624 IP Telephone.
■4630 IP Telephone User Guide (555-233-764)
This document provides detailed information about using the 4630 IP Telephone.
The following documents provide standards relevant to IP Telephony.
IETF Documents1
The following documents are available for free from the IETF web site: http://www.ietf.org/
rfc.html.
Requirements for Internet Hosts - Communication Layers, October 1989, by R. Braden (STD
3: RFC 1122)
Requirements for Internet Hosts - Application and Support, October 1989, by R. Braden (STD
3: RFC 1123)
Internet Protocol (IP), September 1981, by Information Sciences Institute (STD 5: RFC 791),
as amended by Internet Standard Subnetting Procedure, August 1985, by J. Mogul and J.
Postel (STD 5: RFC 950)
Broadcasting Internet Datagrams, October 1984, by J. Mogul (STD 5: RFC 919)
Broadcasting Internet Datagrams in the Presence of Subnets, October 1984, by J. Mogul
(STD 5: RFC 922)
User Datagram Protocol (UDP), August 28, 1980, by J. Postel (STD 6: RFC 768)
Transmission Control Protocol (TCP), September 1981, by Information Sciences Institute
(STD 7: RFC 793)
Domain Names - Concepts and Facilities (DNS), November, 1987, by P. Mockapetris (STD 13:
RFC 1034)
Domain Names - Implementation and Specification (DNS), November 1987, by P. Mockapetris
(STD 13: RFC 1035)
The TFTP Protocol (Revision 2), (TFTP), July 1992, by K. Sollins, (STD 33: RFC 1350:) as
updated by TFTP Option Extension, May 1998, by G. Malkin and A. Harkin (RFC 2347)
An Ethernet Address Resolution Protocol (ARP), November 1982, by David C. Plummer (STD
37: RFC 826)
Dynamic Host Configuration Protocol (DHCP), March 1997, by R. Droms (RFC 2131)
Related Documents
1-6
DHCP Options and BOOTP Vendor Extensions, March 1997, by S. Alexander and R. Droms
(RFC 2132)
RTP: A Transport Protocol for Real-Time Applications (RTP/RTCP), January 1996, by H.
Schulzrinne, S. Casner, R. Frederick, V. Jacobson (RFC 1889)
Definition of the Differentiated Services Field (DS Field) in the IPv4 and IPv6 Headers,
(DIFFSRV), December 1998, by K. Nichols, S. Blake, F. Baker and D. Black (RFC 2474)
Introduction to version 2 of the Internet-standard Network Management Framework
(SNMPv2), April 1993, by J. Case, K. McCloghrie, M. Rose, and S. Waldbusser (RFC 1441)
Management Information Base for Network Management of TCP/IP Internets: MIB-II, March
1991, edited by K. McCloghrie and M. Rose (RFC 1213)
SNMPv2 Management Information Base for the Internet Protocol using SMIv2, November
1996, edited by K. McCloghrie (RFC 2011)
Structure of Management Information Version 2 (SMIv2), April 1999, edited by K. McCloghrie,
D. Perkins, and J. Schoenwaelder (RFC 2578)
Resource ReSerVation Protocol VI, September 1997, by R. Braden, L. Zhang, S. Berson, S.
Herzog, and S. Jamin (RFC 2205)
ITU Documents1
Introduction
The following documents are available for a fee from the ITU web site: http://www.itu.int.
Recommendation G.711, Pulse Code Modulation (PCM) of Voice Frequencies, November
1988.
Recommendation G.729, Coding of speech at 8 kbit/s using Conjugate-Structure AlgebraicCode-Excited Linear-Prediction (CS-ACELP), March 1996.
Annex A to Recommendation G.729: Reduced complexity 8 kbit/s CS-ACELP speech codec,
November 1996.
Annex B to Recommendation G.729: A silence compression scheme for G.729 optimized for
terminals conforming to Recommendation V.70, November 1996.
Recommendation H.225.0, Call signalling protocols and media stream packetization for
packet-based multimedia communications systems, February 1998.
Recommendation H.245, Control protocol for multimedia communication, February 1998.
Recommendation H.323, Packet-based multimedia communications systems, February 1998.
Related Documents
1-7
4600 Series IP Telephone LAN Administrator’s Guide
ISO/IEC, ANSI/IEEE Documents1
The following documents are available for a fee from the ISO/IEC standards web site: http:/
/www.iec.ch.
International Standard ISO/IEC 8802-2:1998 ANSI/IEEE Std 802.2, 1998 Edition, Information
technology - Telecommunications and information exchange between systems - Local and
metropolitan area networks- Specific requirements- Part 2: Logical Link Control.
ISO/IEC 15802-3: 1998 ANSI/IEEE Std 802.1D, 1998 Edition, Information technologyTelecommunications and information exchange between systems- Local and metropolitan area
networks- Common specifications- Part 3: Media Access Control (MAC) Bridges.
IEEE Std 802.1Q-1998, IEEE Standards for Local and Metropolitan Area Networks: Virtual
Bridged Local Area Networks.
Customer Support1
For support for your 4600 Series IP Telephones, call the Avaya support number provided to you by
your Avaya representative or Avaya reseller.
Information about Avaya products can be obtained at the following URL:
http://www.avaya.com/support
Customer Support
1-8
Overview of Voice over IP (VoIP)2
PIn
Introduction2
This chapter describes the differences between data and voice networks, and the factors that
influence the performance of VoIP. The installation and administration of 4600 Series IP
Telephones on DEFINITY
addressed.
Overview of Voice over IP2
®
servers, and the installation and configuration of DHCP and TFTP are
2
The 4600 Series IP Telephones allow enterprises to use Voice over IP (that is, packet-switched
networks) instead of telephony over the Public Switched Telephone Network (PSTN). However,
the use of data networks for transmitting voice packets poses the problem that data networks were
not designed for the specific qualities required by voice traffic.
Data and Voice Network Similarities2
Data and voice networks share similar functions due to the nature of networking.
■Signaling is used to establish a connection between two endpoints.
In a voice network, signaling is used to identify who the calling party is trying to call and where
the called party is on the network. Traditional telephony uses terminals with fixed addresses
and establishes a fixed connection for the communication session between two such
terminals, allocating fixed bandwidth resources for the duration of the call.
IP communications constitute a connectionless network, having neither fixed addresses nor
fixed connections.
■Addressing. Each terminal on a network must be identified by a unique address.
In a voice network the unique address is a permanent attribute, based on international and
national numbering plans, as well as those based on local telephone company practices and
internal customer-specific codes.
In IP communications, dial plans track extension numbers assigned to terminals. No fixed
connection path is needed.
Introduction
2-1
4600 Series IP Telephone LAN Administrator’s Guide
■Routing is related to addressing and allows connections to be established between
endpoints.
Though these functions are common to data and voice networks, the implementations differ.
Delay and Jitter2
Data traffic is generally short and comes in bursts. Data networks like the Internet were designed
to manage these bursts of traffic from many sources on a first-come, first-served basis. Data
packets are sent to multiple destinations, often without any attempt to keep them in a particular
order.
Voice networks are designed for continuous transmission during a call. The traffic is not bursty,
and the conversation uses a specific amount of bandwidth between the two ends for the duration
of the call.
Several features of data networks are unsuitable for voice telephony:
■Data networks are designed to deliver data at the destination, but not necessarily within a
certain time. This produces delay (latency). In data networks, delay tends to be variable. For
voice messages, variable delay results in jitter, an audible chopiness in conversations.
■Variable routing also can result in loss of timing synchronization, so that packets are not
received at the destination in the proper order.
■Data networks have a strong emphasis on error correction, resulting in repeated
transmissions.
While data network concepts include prioritization of traffic types to give some forms of traffic
greater reliability (for example, for interactive transactions), data requirements tend to be not as
strict as most voice requirements.
Release 1.1 of the 4600 Series IP Telephones includes a dynamic jitter buffer. This feature
automatically smooths jitter to improve audio quality.
Tandem Coding2
Tandem coding (also called transcoding) refers to the conversion of a voice signal from analog to
digital and back again. When calls are routed over multiple IP facilities, they may be subject to
multiple transcodings. The multiple conversions between analog and digital coding result in a
deterioration in the voice quality. Tandem coding should be avoided wherever possible in any
compressed voice system (for example, minimizing analog trunking on the PBX).
Overview of Voice over IP
2-2
Overview of Voice over IP (VoIP)
Voice Coding Standards2
There are a number of voice coding standards. The Avaya 4600 Series IP Telephones offer the
options described below.
G.711, which describes the 64 kbps PCM voice coding technique. G.711-encoded voice is
already in the correct format for digital voice delivery in the public phone network or through
PBXs.
G.729A and G.729B, which describe adaptive code-excited linear-predictive (CELP)
compression that enables voice to be coded into 8 kbps streams.
Release 1.6 of the 4600 Series IP Telephones provides support for G.711 silence suppression and
custom packet loss concealment, which can improve audio quality significantly.
H.323 Standard2
Internal signaling provides connection control and call progress (status) information. The H.323
standard is used for internal signaling for IP packet voice networks. H.323 defines more than
simply voice. It defines a complete multimedia network (voice, video, data), with everything from
devices to protocols. The H.245 standard links all the entities within H.323 by negotiating facilities
among participants and H.323 network elements.
The H.323 standard makes G.711 PCM compression the default form of compression. All other
compression formats are optional.
DHCP2
Dynamic Host Configuration Protocol (DHCP) allows a server to assign IP addresses and other
parameters to devices such as the 4600 Series IP Telephones on an as-needed basis. This
eliminates the need to configure each end station with a static IP address. The DHCP application
also passes information to the 4600 Series IP Telephone, identifying the IP Addresses of the PBX
and the TFTP server and the path to the upgrade script and the application file on the TFTP server.
For further information, refer to DHCP and TFTP Servers
on page 2-7 and DHCP on page 4-6.
TFTP2
During the installation and, if necessary, during the reset of the 4600 Series IP Telephones, the
upgrade script and potentially, the application file, are downloaded from the Trivial File Transfer
Protocol (TFTP) server to each IP Telephone, simplifying the software upgrade process. For
further information, refer to DHCP and TFTP Servers
on page 2-7 and TFTP on page 4-17.
Overview of Voice over IP
2-3
4600 Series IP Telephone LAN Administrator’s Guide
NAT2
A Network Address Translator is an application that can be administered between your network
and the Internet. The NAT translates network layer IP addresses so your local intranet IP
addresses can duplicate global, Internet addresses. A detailed discussion of NAT is beyond the
scope of this document, but it should be noted that use of NAT can lead to problems affecting the
consistency of addressing throughout your network. In Release 1.6 and earlier Releases of the
4600 Series IP Telephones, NAT is not recommended for networks handling IP-based telephony
traffic. As of Release 1.7, all 4600 Series IP Telephones support NAT interworking; hence, there
are no problems with NAT and Release 1.7 of the 4600 Series IP Telephones. Note, however, that
support for NAT does not imply support for Network Address Port Translation (NAPT). Specifically,
the 4600 Series IP Telephones do not support communication to the PBX through any NAPT
device.
QoS2
Quality of Service (QoS) is a term covering several initiatives to maximize the quality of the voice
heard at both ends of a call that originates, terminates, or both, on an IP-based telephone. These
initiatives include various prioritization schemes to offer voice packets a larger or prioritized share
of network resources. These schemes include standards such as IEEE’s 802.1D and 802.1Q, the
Internet Engineering Task Force’s (IETF’s) “Differentiated Services”, RTP Control Protocol (RTCP)
and Resource ReSerVation Protocol (RSVP), and port-based priority schemes such as UDP port
selection. Documentation for your LAN equipment will elaborate on the extent your network can
support any or all of these initiatives. See Chapter 4, Server Administration
of QoS for the 4600 Series IP Telephones.
for some implications
As of Release 1.7, both the 4620 and 4630 IP Telephones provide information to the end user
about network audio quality that may be of use to the LAN Administrator. For specific information,
see QoS with 4620 and 4630 IP Telephones
on page 4-22.
SNMP2
Simple Network Management Protocol (SNMP) is a family of standards-based protocols and
procedures to allow vendor-independent management of data networks. Using a simple set of
protocol commands, an SNMP-compliant device will store information in standard format in one or
more Management Information Bases (MIBs). In general, devices will support the standardsspecific MIB termed MIB-II. In addition, devices may define one or more "custom MIBs" that
contain information about the specifics of the device.
Release 1.1 of the 4600 Series IP Telephones is fully compatible with SNMPv2c (a later version of
SNMP) and with Structure of Management Information Version 2 (SMIv2), although the telephones
will respond correctly to queries from entities that comply with earlier versions of SNMP, such as
SNMPv1. "Fully compatible" means that the telephones will respond to queries directed either at
the MIB-II or the Custom MIB. The 4600 Series IP Telephone Custom MIB is read-only (values
therein cannot be changed externally via network management tools). Similarly, although the 4600
Series IP Telephone’s MIB-II has read/write permissions in accordance with the standard, to
improve security any writes to MIB-II are saved but otherwise ignored.
SNMP
2-4
Overview of Voice over IP (VoIP)
More information about SNMP and MIBs can be found in the IETF references listed in Chapter 1,
Related Documents
also available for download in *.txt format on the Avaya support website.
. Appendix A of this LAN Administration Guide lists the Custom MIB, which is
Network Assessment2
The current technology allows optimum network configurations to deliver VoIP with perceived
voice quality close to that of the Public Switched Telephone Network (PSTN). Not every network is
able to take advantage of packet voice transmissions. Some data networks have insufficient
residual capacity for even compressed voice traffic. In addition, the usual approach to developing
data networks by integrating products from many vendors makes it necessary to test the
components for compatibility with Voice over IP traffic.
It is assumed that your organization has performed a network assessment (with or without the
assistance of Avaya) before attempting to install Voice over IP, in order to have a high degree of
confidence that the existing data network has the capacity to carry voice packet traffic and is
compatible with the required technology.
A network assessment would include a determination of the following:
■A network audit to review existing equipment and evaluate its capabilities, including its ability
to meet planned voice and data needs.
■A determination of network objectives, including the dominant traffic type, choice of
technologies, and setting voice quality objectives.
The assessment should leave you confident that the implemented network will have the capacity
for the foreseen data and voice traffic, and can support H.323, DHCP, TFTP, and jitter buffers in
H.323 applications.
It is important to distinguish between compliance with the minimal VoIP standards and support for
QoS which is needed to run VoIP on your configuration.
Suggestions for Installation and Configuration2
Reliability and Performance2
There is a cost/performance trade-off associated with Voice over IP. Greater reliability and
improved performance can be obtained through server redundancy and components with higher
bandwidth capabilities.
The reliability and performance of the traditional PBX systems have been very high. Although
much of the LAN is outside of the control of the PBX, there are several points to consider which
enhance the reliability and performance of the IP Telephone network.
Network Assessment
2-5
4600 Series IP Telephone LAN Administrator’s Guide
All 4600 Series IP Telephones support the tools "ping" and "traceroute." These are standard LAN/
WAN tools for identifying whether two points on a network can communicate with each other, and
what path a sample communication takes as it traverses the network from one point to the other.
All 4600 Series IP Telephones will respond appropriately to a ping or a traceroute message sent
from the DEFINITY
®
or MultiVantageTM switch or any other source on your network, although these
telephones will not, in general, initiate a ping or traceroute. With Release 1.6 of the 4600 Series IP
Telephones comes support of "remote ping" and "remote traceroute." The switch can instruct such
a 4600 Series IP Telephone to initiate a ping, or a traceroute, to a specified IP address. The
telephone will carry out that instruction and send a message to the switch informing it of the
results. See your DEFINITY
®
or MultiVantageTM Administration documentation for more details.
IP Address Lists and Station Number Portability2
With Release 1.5 of the 4600 Series Telephones comes the capability to specify lists of IP
addresses (either dotted decimal or DNS format) for key elements of the network, rather than
merely one address for each. Specifically, you can specify up to 127 total characters in each list of
the following: router/gateways, TFTP servers, and the call server. When the 4600 telephone is
powered up or is rebooted, it attempts to establish communication with these various network
elements in turn, starting with the first address on the respective list. If the communication is
denied, or times out, the telephone proceeds to the next address on the appropriate list and tries
that one. The telephone does not report failure unless all the addresses on a given list have failed.
Obviously, this capability can significantly improve the reliability of IP telephony by maximizing the
likelihood the telephone can communicate with backup equipment if the primary equipment is
down or inaccessible (say, perhaps due to a limited network outage).
However, this capability also has the advantage of making station number portability easier.
Assume a situation where the company has multiple locations (for example, London and New
York), all sharing a corporate IP network. Users want to take their telephones from their offices in
London and bring them to New York. When users power up their telephones in the new location,
the local DHCP server will generally route them to the local switch, which denies service because
it knows nothing about these new users. However, with proper administration of the local DHCP
server, the telephone knows to try a second call server IP address, this one in London. The user
can then be automatically registered with the London switch.
Chapter 4
router/gateways, and TFTP servers. For specific information, see DNS Addressing
contains details on administration of DCHP servers for lists of alternate call servers,
in Chapter 4.
Security2
In VoIP, physical wire is replaced with an IP connection. The connection is more mobile.
Unauthorized relocation of the IP telephone allows unauthorized users to send and receive calls
as the valid owner. For further details on toll fraud, refer to the DEFINITY
documents in Chapter 1, Related Documents
.
®
or MultiVantageTM
Suggestions for Installation and Configuration
2-6
4600 Series IP Telephones2
Dual Connection Architecture2
Overview of Voice over IP (VoIP)
Releases 1.0 and 1.1 of the 4600 Series IP Telephones use dual connection architecture to
communicate with the DEFINITY
two station extensions must be administered for each telephone.
®
or MultiVantageTM switch. In the dual connection architecture,
Single Connection Architecture2
Release 1.5 and subsequent releases of the 4600 Series IP Telephones use single connection
architecture to communicate with the DEFINITY
connection architecture, only one station extension must be administered for each telephone.
®
or MultiVantageTM switch. In the single
Registration and Authentication2
The DEFINITY
Series IP Telephones using the extension and password. For further information, see Related
Documents on page 1-5.
®
or MultiVantageTM switch supports the registering and authentication of 4600
Software2
As shipped from the factory, the 4600 Series IP Telephones may not contain sufficient software for
registration and operation. When the phone is first plugged in, a software download from a TFTP
server is initiated. This gives the phone its proper functionality.
For downloads of software upgrades, the PBX provides the capability for a remote restart of the
4600 Series IP Telephone. As a consequence of restarting, the phone automatically restarts
reboot procedures. If new software is available, a download will result.
WAN Considerations2
QoS is harder on a WAN than a LAN. A LAN assumes no bandwidth concerns. A WAN assumes a
finite amount of bandwidth. Therefore, QoS considerations are more significant when the IP
telephony environment includes a WAN. In addition, there are administrative and hardware
compatibility issues unique to WANs.
DHCP and TFTP Servers2
The DHCP server provides the following information to the 4600 Series IP Telephone:
■IP address of the 4600 Series IP Telephone
■IP Address and port number of the TN799 board on the DEFINITY
server. On the call server, the standard port number is 1719.
®
or MultiVantageTM Call
4600 Series IP Telephones
2-7
4600 Series IP Telephone LAN Administrator’s Guide
■IP Address of the TFTP server
■The subnet mask
■IP Address of the router
You should administer the LAN so that every IP Telephone can access a DHCP server with the
above information.
The IP Telephone will not function without an IP address. The failure of a DHCP server at boot
time will leave all the affected voice terminals unusable. (Although it is possible for the user to
manually assign an IP address to an IP Telephone, when the DHCP server finally returns, the
telephone will never look for a DHCP server unless the static IP data is unassigned manually. In
addition, manual entry of IP data is an error-prone process.) It is therefore strongly recommended that a DHCP server be available when the IP Telephone reboots.
A minimum of two DHCP servers is recommended for reliability.
The TFTP server provides the 4600 Series IP Telephone with a script file and, if appropriate, new
or updated application software (see Step 3, Telephone and TFTP Server under Initialization
Process below). In addition, you can edit the script file to customize telephone parameters for your
specific environment (see Administering Options for the 4600 Series IP Telephones
in Chapter 4,
Server Administration).
Initialization Process2
The following is a high-level description of the information exchanged when the telephone is
initializing and registering. This description, which assumes all equipment is properly administered
ahead of time, may be helpful in explaining how the 4600 Series IP Telephones relate to the
routers and servers in your network.
Step 1: Telephone to Network2
The telephone is appropriately installed and powered, and 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 in the network receives this message, and relays it to
the appropriate DHCP server.
Step 2: DHCP Server to Telephone2
The DHCP server provides information to the telephone, as described in DHCP and TFTP Servers
on page 2-7. Among other data passed to the telephone is the IP address of the TFTP server,
which is crucial for the next step.
Initialization Process
2-8
Overview of Voice over IP (VoIP)
Step 3: Telephone and TFTP Server2
The telephone queries the TFTP server, which transmits a script file to the telephone. This script
file, at a minimum, tells the telephone which application file the telephone should be using (the
application file is the software that has the telephony functionality, and can be easily updated for
future enhancements).
The telephone uses the script file to determine if it has the proper application file. A newly-installed
telephone will have no application file, and hence does not have the proper one. A previouslyinstalled telephone may or may not have the proper application file. In any event, if the telephone
determines it does not have the application file the script file says the telephone should have, the
telephone requests a download of the proper application file from the TFTP server. The TFTP
server then downloads the file and conducts some checks to ensure the file was downloaded
properly. If the telephone determines it already has the proper file, it proceeds to the next step
without downloading the application file again.
Step 4: Telephone and the DEFINITY
®
/MultiVantage
TM
Call Server2
In this step, the telephone and the PBX exchange a series of messages which cause the display
on the telephone to prompt the user. For a new installation, the user must enter the telephone’s
extension and the call server password. For a restart of an existing installation, this information is
already stored on the telephone, but the user must confirm the information. In either case, manual
intervention is required. The telephone and the switch exchange more messaging, with the
expected result that the telephone is appropriately registered on the switch.
More details about the installation process are available in the 4600 Series IP Telephone Installation Manual and in Chapter 3 of this document.
Initialization Process
2-9
4600 Series IP Telephone LAN Administrator’s Guide
Initialization Process
2-10
Requirements3
Program
Introduction3
3
The 4600 Series IP Telephones use Internet Protocol (IP) technology with Ethernet line interfaces.
The IP telephones supplement the existing DEFINITY
feature provides the user with the capability to natively administer and maintain the new 4600
Series IP Telephones.
The 4600 Series IP Telephones provide support for DHCP and TFTP over IPv4/UDP which
enhance the administration and servicing of the phones. These phones use DHCP to obtain
dynamic IP addresses and TFTP to download new versions of software for the phones.
The 4600 Series IP Telephones provide the ability to have one connection on the desktop for both
the telephone set and the PC using the telephone’s built-in hub.
®
/MultiVantageTM IP Solutions platform. This
Hardware Requirements3
Before plugging in the 4600 Series IP Telephone, verify that all of the following requirements have
been met. Failure to do so will prevent the telephone from working and may have a negative
impact on your network.
The following hardware is required for 4600 Series IP Telephones to work properly.
■The DEFINITY
— DEFINITY
— DEFINITY
— DEFINITY
— MultiVantage
®
switch must be installed and administered correctly, with Release 8.4 or later.
®
Release 8.4 supports the 4612 and 4624 IP Telephones.
®
Release 9 and later support the 4606, 4612, and 4624 IP Telephones.
®
Release 10 and later support the 4630 IP Telephone.
TM
Release 11 supports the 4602 and 4620 IP Telephones.
Introduction
3-1
4600 Series IP Telephone LAN Administrator’s Guide
For Release 1.1 of any of the 4600 Series IP Telephones, the DEFINITY® switch must
have Release 9 software installed. For Release 1.5 of any of the 4600 Series
Telephones, the DEFINITY
the 4600 Series IP Telephones, and for support of the 4630 IP Telephone, the
DEFINITY
®
switch must have Release 10 software. For Release 1.7 of the 4600 Series
®
switch must have Release 9.5 software. For Release 1.6 of
IP Telephones, and for support of the 4602 and 4620 IP Telephones, the MultiVantage
switch must have Release 11 software.
■The following two circuit packs must be resident on the PBX server used to support the IP
telephones:
— TN2302 AP Media Processor circuit pack converts the audio levels for the IP telephone to
audio levels for DCP phones when IP phones are used in a call with non-IP telephones.
— TN799 Control-LAN (CLAN) circuit pack for the signaling capability (either the B or C
vintage) on the csi, si, and r platforms.
■A Category 5 LAN. If the telephones are to be powered from the LAN, the LAN must comply
with the IEEE 802.3af standard for LAN powering.
■Electrical power provided to each phone by one of the following two sources.
TM
— A Telephone Power Module (DC power jack) (must be ordered separately).
— IEEE 802.3af, if the LAN supports this powering scheme (although the 4630 cannot be
powered this way).
■Verify that the 4600 Series IP Telephone package includes the following components:
— 1 telephone set.
— 1 AB1C handset.
— 1 H4DU 9-foot long (when extended) 4-conductor coiled handset cord, plugged into the
telephone and the handset.
— 1 Category 5 modular line cord for the connection between the IP Telephone and the
Ethernet wall plug.
— Non-system-specific safety and installation documentation.
— Power brick
— Stylus (4630 IP Telephones only)
■You may need a Category 5 modular line cord for the connection from the 4600 Series IP
Telephone to the PC.
Refer to the 4600 Series IP Telephone Installation Guide.
The IP telephones work the same on all DEFINITY
®
/MultiVantage
TM
platforms.
Hardware Requirements
3-2
Requirements
Software Requirements3
The following software is required for 4600 Series IP Telephones to work properly.
■DEFINITY
station customer options must be turned on.
■The DHCP server and application should be installed and properly administered, as described
in DHCP
■The TFTP server and application must be installed and properly administered, as described in
TFTP
■For 4630 IP Telephone environments, if users are to have access to LDAP directories or
corporate Websites, the appropriate servers must be in place, and the 4630 telephones must
be appropriately administered in accordance with Chapter 4, Server Administration
®
/MultiVantageTM Release 8.4 or later software, as appropriate. The H.323 and IP
on page 4-6.
WARNING:
A DHCP server is not mandatory, but static addressing is necessary when a DHCP
server is unavailable.
Due to the difficulties associated with static addressing, it is very strongly recommended
that a DHCP server be installed and that static addressing be avoided.
on page 4-17.
.
Ensure that all required parameters are configured correctly. For DEFINITY/MultiVantage, see
your administration documentation. For the DHCP and TFTP servers, see Chapter 4, Server
Administration.
Figure 3-1 on the following page illustrates a sample configuration using the 4600 Series IP
Telephones.
Software Requirements
3-3
4600 Series IP Telephone LAN Administrator’s Guide
Figure 3-1. Sample Configuration Using 4600 Series IP Telephones
Software Requirements
3-4
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