Cisco SIP IP Phone Administrator's Manual
Cisco SIP IP Phone Administrator Guide
Version 4.0
August 2002
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STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT
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The following information is for FCC compliance of Class A devices: This equipment has been tested and found to comply with the limits for a Class A digital device, pursuant
to part 15 of the FCC rules. These limits are designed to provide reasonable protection against harmful interference when the equipment is operated in a commercial
environment. This equipment generates, uses, and can radiate radio-frequency energy and, if not installed and used in accor dance with the instruction manual, may cause
harmful interference to radio communications. Operation of this equipment in a residential area is likely to cause harmful interference, in which case users will be required
to correct the interference at their own expense.
The following information is for FCC compliance of Class B devices: The equipment described in this manual generates and may radiate radio-frequency ener gy. If it is not
installed in accordance with Cisco’s installation instructions, it may cause interference with radio and television reception. This equipment has been tested and found to
comply with the limits for a Class B digital device in accordance with the specifications in part 15 of the FCC rules. These specifications are designed to provide reasonable
protection against such interference in a residential installation. However, there is no guarantee that interference will not occur in a particular installation.
Modifying the equipment without Cisc o’s writ ten author ization m ay resul t in the equi pment no lo nger comp lyi ng with FCC requi rements for Class A or Class B digital
devices. In that event, your right to use the equ ipment may be limit ed by FCC regul ations , and you may be requir ed to correct a ny interference to radio or television
communications at your own expense.
You can determine whether your equipment is causing interference by turning it off. If the interferen ce stops, it was probably caused by the Cis co equipm ent or one of its
peripheral devices. If the equipment causes interference to radio or television reception, try to correct the interference by using one or more of the followi ng measures:
• Turn the television or radio antenna unt il the int erference st ops.
• Move the equipment to one side or the other of the televisio n or radi o.
• Move the equipment farther away from the te levision or radio.
• Plug the equipment into an outlet that is on a di fferent cir cuit from the televi sion o r radio. (That is, make certain th e equipment and the te levision or radio are on circuit s
controlled by different circuit breaker s or fuses.)
Modifications to this product no t author ized by Cis co Syst ems, Inc. coul d voi d the FCC appro val and ne gate your authorit y to op erate the pr odu ct.
The Cisco implementation of TCP head er compressi on is an adap tation of a program developed by the Universi ty of Ca lifornia, Berk eley (UCB) as part of UCB ’s public
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Cisco SIP IP Phone Administrator Guide
Copyright © 2001-2002, Cis co Sys tems, In c.
All rights reserved.
Preface vii
Overview vii
Who Should Use This Guide vii
Objectives viii
Document Organization viii
Related Documentation viii
Document Conventions ix
Obtaining Documentation xi
World Wide Web xi
Documentation CD-ROM xi
Ordering Documentation xi
Documentat ion Feedback xi
CONTENTS
CHAPTER
Obtaining Technical Assistance xii
Cisco.com xii
Technical Assistance Center xii
Cisco TAC Web Site xiii
Cisco TAC Escalation Center xiii
1 Product Overview 1-1
What Is Session Initiation Protocol? 1-1
Components of SIP 1-2
SIP Clients 1-3
SIP Servers 1-3
What Is the Cisco SIP IP Phone? 1-3
BTXML Support 1-5
Cisco CallManager XML Support 1-5
Supported Features 1-6
Physical Features 1-6
Network Features 1-6
Configuration Features 1-7
Codec and Protocol Support 1-7
Dialing and Messaging Features 1-7
Call Options 1-8
Routing and Proxy Features 1-8
Cisco SIP IP Phone Administrator Guide
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Contents
Character Support 1-10
Supported Protocols 1-11
Prerequisites 1-12
Cisco SIP IP Phone Con nections 1-12
Connecting to the Network 1-13
Network Port (10/100 SW) 1-13
Access Port (10/100 PC) 1-13
Connecting to Power 1-13
Using a Headset 1-14
The Cisco SIP IP Phone with a Catalyst Switch 1-14
CHAPTER
2 Getting Started with Your CiscoSIPIP Phone 2-1
Initializatio n Pr oc e ss Ov er v ie w 2-1
Installing the Cisco SIP IP Phone 2-2
Installation Task Summary 2-2
Downloading File s to Your TFTP Server 2-3
Configuring SI P Parameters 2-3
Configuring SI P Parameters via a TFTP Server 2-4
Manually Configuring the SIP Parameters 2-7
Configuring Network Parameters 2-9
Configuring Net w ork Parameters via a DHCP Server 2-10
Manually Configuring the Network Parameters 2-10
Connecting the Phone 2-11
Adjusting the Pl acement of the Cisco SIP Pho ne 2-12
Verifying Startup 2-14
Using the CiscoSIP IPPhone Menu I nterface 2-15
Reading the Cisco SIP IP Phone Icons 2-15
Customizing the Cisco SIP IP Phone Ring Types 2-17
Creating Dial Plans 2-17
CHAPTER
3 Managing Cisco SIP IP Phones 3-1
Changing Your Configuration 3-1
Modifying the Ph one’s Network Se tt in gs 3-2
Entering Configuration Mode 3-2
Unlocking Configuration Mode 3-2
Locking Configuration Mode 3-2
Changing the Network Settings 3-2
Modifying the Ph one’s SIP Settin gs 3-5
Cisco SIP IP Ph one Administrator Guide
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Modifying SIP Parameters via a TFTP Server 3-8
Modifying the Default SIP Configuration File 3-8
Modifying the Phone-Specific SIP Configuration File 3-23
Modifying the S IP Par am e te rs Directly on Your Ph o ne 3-26
Using the Command-Line Interface 3-30
Setting the Date, Time, and Daylight Saving Ti me 3-36
Erasing the Locally Defined Settings 3-41
Erasing the Local ly Defined Network Settings 3-41
Erasing the Locally Defined SIP Settin gs 3-42
Accessing Status Information 3-42
Viewing Status Messages 3-43
Viewing Netw or k S ta tis tics 3-43
Viewing the Firm w a re Ve rs io n 3-44
Upgrading the Cisco SIP IP Phone Firmware 3-44
Upgrading from Release2.2 or Later Releases to Release4.0 3-45
Upgrading from Release2.1 or Earlier Releases to Release4.0 3-45
Dual Booting from SCC P or MGCP to Release 4.0 3-46
Contents
APPENDIX
Performing an Image Upgr ade and Remote Reboot 3-46
A SIP Compliance with RFC 3261 Information A-1
SIP Functions A-1
SIP Methods A-2
SIP Responses A-2
1xxResponse—Information Responses A-2
2xxResponse—Successful Respo nses A-3
3xx Response—Redirection Responses A-3
4xxResponse—Request Failure Responses A-4
5xx Response—Server Failure Responses A-6
6xxResponse—Global Responses A-7
SIP Header Fields A-7
SIP Session Description Protocol (SDP) Usage A-8
Transport Layer Protocols A-9
SIP Security A-9
Authentication A-9
SIP DNS Records Usage A-9
SIP DTMF Digit Transport A-9
Cisco SIP IP Phone Administrator Guide
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Contents
APPENDIX
B SIP Call Flows B-1
Call Flow Scenarios for Successful Calls B-1
Gateway-to Cisco SIP IP Phone—Successful Call Setup and Disconnect B-2
Gateway-to- C isco SIP IP Phone—S u cc essful Call Setu p an d C all Hold B-4
Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold B-6
Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Hold with Consultation B-9
Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting B-13
Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer Without Consultation B-17
Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer Without Consultation Using Failo ver to
Bye/Also
Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer with Consult ation B-25
Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer with Consult ation Using Failover to
Bye/Also
Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Unco nditional) B-35
Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Bus y) B-37
Cisco SIP IP Phone-to-Cisco SIP IP Phone Networ k Call Forwarding (No Answer) B-39
Cisco SIP IP Phone-to Cisco SIP IP Phone Three-Way Calling B-42
Call Flow Scenarios for Failed Calls B-46
Gateway-to-Cisco SIP IP Phone—Called User Is Busy B-46
Gateway-to-Cisco SIP IP Phone—Called User Does Not Answer B-48
Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Err or B-50
Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User Is Busy B-51
Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User Does Not Answer B-52
Cisco SIP IP Phone-to-Cisco SIP IP Phone—Aut hentication Error B-53
Call from a Cisco SIP IP Phone to a Gateway Acting as a Backup Proxy B-54
Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone via a Backup Proxy B-56
Call from a Cisco SIP IP Phone to a Gateway Acting as an Emergency Proxy B-59
Call from a Cisco SIP IP Phone to a Cisco SIP IP Phone via Emergency Proxy B-60
B-21
B-30
APPENDIX
C Technical Specifications C-1
Physical and Ope r ating Environm en t Specificati on s C-1
Cable Specifications C-2
Regulatory Safety Compliance C-2
Connection s S pe c ifications C-3
APPENDIX
D Translated Safety Warnings D-1
Installation Warning D-1
Product Disposal Warning D-1
Lightning Activity Warning D-2
Cisco SIP IP Ph one Administrator Guide
iv
G
LOSSARY
I
NDEX
Contents
SELV Circuit Warning D-2
Circuit Breaker (15A) Warning D-3
Cisco SIP IP Phone Administrator Guide
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Contents
Cisco SIP IP Ph one Administrator Guide
vi
Preface
This document describes the Cisco SIP IP phone. This chapter describes the objectives and organization
of the document and explains how to find additio nal inform ation on relat ed product s and servi ces.
This chapter contains the following sections:
• Overview, page vii
• Who Should Use This Guide, pa ge vii
• Objectives, page viii
• Document Organization, pag e viii
• Related Document ation , page vii i
• Document Conventions, page ix
• Obtaining Docu ment ati on , pa ge xi
• Obtaining Technical Assistance, page xii
Overview
The Cisco SIP IP Phone Administrator Guide provides information about how to set up, connect cab les
to, and configure a Cisco Se ssion Initiation Protocol (SIP) IP phone 7940 or 7960 (hereafte r referred to
as a Cisco SIP IP phone). It also provides information on how to configure the network and SIP settings
and change the settings and options of the Cisco SIP IP phone. The administra tor guide al so include s
reference info rmat ion su ch as Cisc o SIP IP phone ca ll flows a nd comp lia nc e i nfo rm ation .
Who Should Use This Guide
Network engineers, syste m administ rators, or teleco mmunic ations engineers shou ld use this guide to
learn the steps required to properly set up t he Cisco SIP IP phone on the network.
The tasks described are considered to be administration-level tasks and are not intended for the end users
of the phones. M any of th e t asks i nvolve configuring network se ttin gs t hat coul d affect t he pho ne ’s
ability to function in the network and require an understanding of IP networking and telephony concepts.
Cisco SIP IP Phone Administrator Guide
vii
Preface
Objectives
Objectives
The Cisco SIP IP Phone A dministrator Guide provides nec essary infor mation to get th e Cisco SIP IP
phone operation al in a Voice-over-IP (VoIP) network.
It is not the intent of this administrator guide to provide information on how to implement a SIP VoIP
network. For information on imple menting a SIP VoIP network, refer to the documen ts list ed in the
“Related Documentation” section on page viii.
Document Organization
Table 1 lists the chapters and appendixes in this document:
Table 1 Document Organization
Section Title Description
Chapter 1 Product Overview Describes SIP and the Cisco SIP IP phone.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Describes how to install, connect, an d configure th e
Cisco SIP IP phone.
Chapter 3 Managing Cisco SIP I P Ph ones Describes how to modify the Cisco SIP IP phone’s
network and SIP settings, how to access network and
call status information, and how to upgrade the
firmware.
Appendix A SIP Compliance with RFC 3261 Information Provides reference information about the SIP IP phone
compliance to RFC 32 61.
Appendix B SIP C all Flows Provides reference information about the SIP IP phone
call flows.
Appendix C Technical Specifications Lists the physical an d ope ra ting e nvironment
specifications, cable specifications, and connection
specifications
Appendix D Translated Safety Warnings Lists translated safety warnings that should be
followed when installing an electrical device such as
the SIP IP phon e.
Related Documentation
The following is a list of rel ated Cisco SIP VoIP publications. For m ore information a bout implementin g
a SIP VoIP network, refer to the following publications:
• Session Initiation Protocol Gateway Call Flows
• Cisco IP Phone 7960 an d 794 0 Serie s A t a Gl ance
• Regulatory Compliance and Safety Information for the Cisco IP Phone 7960, 7940, and 7910 Series
• Installing the Wall Mount Kit for the Cisco IP Phone
Cisco SIP IP Ph one Administrator Guide
viii
Preface
The following is a list of Cisco VoIP publications that prov ide info rmation ab out impl ement ing a VoIP
network:
• Cisco IOS Voice, Video, and Fax Configuration Guide , Release 12.2
• Cisco IOS Voice, Video, and Fax Command Reference, Release 12.2
• Cisco IOS IP Configuration Guide, Release 12.2
• Cisco IOS IP Comman d R eference, Volume 1 of 3: Addressing and Services, Release 12.2
• Cisco IOS IP Comman d R efe rence, Volume 2 of 3: Ro uting Protocols, Release 12.2
• Cisco IOS IP Command Reference, Volume 3 of 3: Multicast, Release 12.2
Document Conventions
This docume nt u s es the f ol lowing conventions:
• Commands and keywords are in boldface font.
• Arguments for which you supply values are in italic font.
• Elements in sq uare br acket s ([ ]) are optional .
Document Conventions
• Alternative keywords are grouped in braces and separated by vertical bars (for example, { x | y | z }).
• Optional alternative keywords are grouped in brackets and separated by vertical bars (for example,
[ x | y | z ]).
• Terminal sessions and informa tion the sys tem displays ar e in screen font.
• Information you must enter is in boldface screen font.
Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the
publication.
Caution Means reader b e c areful. In this situation, you might do something that could result in equipment
damage or loss of data.
Warning
Waarschuwing
This warning symbol means danger. You are in a situation that could cause bodily injury. Before yo u
work on any equipment, be aware of the hazards involved with electrical circuitry and be familiar
with standard practices for preventing accidents. (To see translations of the warnings that appear
in this publication, refer to the appendix, “Translated Safety Warnings.”)
Dit waarschuwingssymbool betekent gevaar. U verkeert in een situatie die lichamelijk letsel
kan veroorzaken. Voordat u aan enige apparatuur gaat werken, dient u zich bewust te zijn van
de bij elektrische schakelingen betrokken risico’s en dient u op de hoogte te zijn van standaard
maatregelen om ongelukken te voorkomen. (Voor vertalingen van de waarschuwingen die in
deze publicatie verschijnen, kunt u het aanhangsel “Translated Safety Warnings” (Vertalingen
van veiligheidsvoorschriften) raadplegen.)
Cisco SIP IP Phone Administrator Guide
ix
Document Conventions
Preface
Varoitus
Attention
Warnung
Avvertenza
Tämä varoitusmerkki merkitsee vaaraa. Olet tilanteessa, joka voi johtaa ruumiinvammaan.
Ennen kuin työskentelet minkään laitteiston parissa, ota selvää sähkökytkentöihin liittyvistä
vaaroista ja tavanomaisista onnettomuuksien ehkäisykeinoista. (Tässä julkaisussa esiintyvien
varoitusten käännökset löydät liitteestä "Translated Safety W arnings" (käännetyt turvallisuutta
koskevat varoitukset).)
Ce symbole d’avertissement indique un danger. Vous vous trouvez dans une situation pouvant
entraîner des blessures. Avant d’accéder à cet équipement, soyez conscient des dangers posés
par les circuits électriques et familiarisez-vous avec les procédures courantes de prévention
des accidents. Pour obtenir les traductions des mises en garde figurant dans cette publication,
veuillez consulter l’annexe intitulée « Translated Safety Warnings » (Traduction des avis de
sécurité).
Dieses Warnsymbol bedeutet Gefahr. Sie befinden sich in einer Situation, die zu einer
Körperverletzung führen könnte. Bevor Sie mit der Arbeit an irgendeinem Gerät beginnen, seien
Sie sich der mit elektrischen Stromkreisen verbundenen Gefahren und der Standardpraktiken
zur Vermeidung von Unfällen bewußt. (Übersetzungen der in dieser Veröffentlichung
enthaltenen Warnhinweise finden Sie im Anhang mit dem Titel “Translated Safety Warnings”
(Übersetzung der Warnhinweise).)
Questo simbolo di avvertenza indica un pericolo. Si è in una situazione che può causare
infortuni. Prima di lavorare su qualsiasi apparecchiatura, occorre conoscere i pericoli relativi
ai circuiti elettrici ed essere al corrente delle pratiche standard per la prevenzione di incidenti.
La traduzione delle avvertenze riportate in questa pubblicazione si trova nell’appendice,
“Translated Safety Warnings” (Traduzione delle avvertenze di sicurezza).
Advarsel
Aviso
Advertencia
Varning!
Dette varselsymbolet betyr fare. Du befinner deg i en situasjon som kan føre til personskade.
Før du utfører arbeid på utstyr, må du være oppmerksom på de faremomentene som elektriske
kretser innebærer, samt gjøre deg kjent med vanlig praksis når det gjelder å unngå ulykker.
(Hvis du vil se oversettelser av de advarslene som finnes i denne publikasjonen, kan du se i
vedlegget "Translated Safety Warnings" [Oversatte sikkerhetsadvarsler].)
Este símbolo de aviso indica perigo. Encontra-se numa situação que lhe poderá causar danos
fisicos. Antes de começar a trabalhar com qualquer equipamento, familiarize-se com os
perigos relacionados com circuitos eléctricos, e com quaisquer práticas comuns que possam
prevenir possíveis acidentes. (Para ver as traduções dos avisos que constam desta publicação,
consulte o apêndice “Translated Safety Warnings” - “Traduções dos Avisos de Segurança”).
Este símbolo de aviso significa peligro. Existe riesgo para su integridad física. Antes de
manipular cualquier equipo, considerar los riesgos que entraña la corriente eléctrica y
familiarizarse con los procedimientos estándar de prevención de accidentes. (Para ver
traducciones de las advertencias que aparecen en esta publicación, consultar el apéndice
titulado “Translated Safety Warnings.”)
Denna varningssymbol signalerar fara. Du befinner dig i en situation som kan leda till
personskada. Innan du utför arbete på någon utrustning måste du vara medveten om farorna
med elkretsar och känna till vanligt förfarande för att förebygga skador. (Se förklaringar av de
varningar som förekommer i denna publikation i appendix "Translated Safety Warnings"
[Översatta säkerhetsvarningar].)
Cisco SIP IP Ph one Administrator Guide
x
Preface
Obtaining Documentation
The following sections explain how to obtain docum entati on from Cisc o Systems.
World Wide Web
You can access the most current Cisco docume ntation on the World Wide We b at the fo llowing URL:
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Ordering Documentation
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Cisco SIP IP Phone Administrator Guide
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Obtaining Technical As sistance
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Cisco.com
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Cisco.com is the foundation of a suite of interactive, networked services that provides immediate, open
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Cisco SIP IP Ph one Administrator Guide
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Preface
• Priority leve l 1 (P1)—Your production network is down, and a critical impact to business operations
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Cisco TAC Web Site
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registered user, you can open a case online by using the TAC Case Open tool at the following URL:
Obtaining Technical Assistance
will occur if se rv ice is n ot r esto re d qui ck ly. No workaround i s available.
http://www.cisco.com/tac/caseopen
If you have Internet access, it is recomme nded that you open P3 and P4 ca ses throug h the Cisco TAC
We b Site.
Cisco TAC Escalation Center
The Cisco TAC Escalation Center addresses issues that are classified as priority level 1 or priority
level 2 ; these cla ssifications ar e assig ned when severe network degrada tion signi ficantly impa cts
business operations. When you contact the TAC Escalation Center with a P1 or P2 problem, a Cisco T A C
engineer will automatically open a case.
To obtain a dir ect ory o f t oll-fr ee C i sco TAC telephone n umb er s f or yo ur co untr y, go to the fol lowing
URL:
http://www.cisco.com/warp/public/687/Directory/DirTAC.shtml
Before calling, please check with your network operations center to determine the level of Cisco support
services to which your company is entitled; for example, SMARTnet, SMARTnet Onsite, or Network
Supported Accounts (NSA). In addition, please have available your service agreement number and your
product serial numb er.
Cisco SIP IP Phone Administrator Guide
xiii
Obtaining Technical As sistance
Preface
Cisco SIP IP Ph one Administrator Guide
xiv
Product Overview
This chapter contains the following information about the Cisco SIP IP phone:
• What Is Session Initia tion Prot ocol? , pa ge 1 -1
• What Is the Cisco SIP IP Phon e? , pa ge 1 -3
• Prerequisites, page 1-12
• Cisco SIP IP Phone Conne ctio ns, pa ge 1 -12
• The Cisco SIP IP Phone with a Catalyst Switch, page 1-14
What Is Session Initiation Protocol?
Session Initiation Protocol (SIP) is the Internet Engineering Task Force’s (IETF’s) standard for
multimedia conferencing over IP. SIP is an ASCII-based, application-l ayer control protocol (defined in
RFC 3261) that can be used to establish, maintain, and terminate calls between two or more endpoints.
CHAPTER
1
Like other V oIP protocols, SIP is designed to address the functions of signaling and session management
within a packet te lep hony net work. Si gnali ng allows call information to be carried across network
boundaries. Session m anagement provides the ability to control the attributes of an end-to-end call.
SIP provides the capabilities to:
• Determine the location of the target endpoint—SIP supports address resolution, name mapping, and
call redirection.
• Determine the media capabilities of the target endpoint—Via Session Description Protocol (SDP),
SIP determines the “lowest level” of common services between the endpoints. Conferences are
established using only the media capab ilities that can be s uppo rted by all e ndpoints.
• Determine the availability of the target endpoint—If a call canno t be compl eted beca use th e target
endpoint is unavailable, SIP determin es whe ther the cal led part y is alrea dy on the phon e or did no t
answer in the allotted number of rings. It t hen retur ns a messag e indi cat ing wh y the target endpoint
was unavailable.
• Establish a session between the ori ginat ing and target endpoi nt—If the call can be completed, SIP
establishes a session between the endpoints. SIP also supports mid-call changes, such as the addition
of another endpoint to the confere nce or the changing o f a media cha racteri stic or code c.
• Handle the transfer and termination of calls—SIP suppor ts t he t rans fe r of ca lls from o ne endp oint
to another . During a call tra nsfer, SIP simply estab lishes a s ession betwee n the tran sfer ee and a n e w
endpoint (specified by the trans f er ring p ar ty) and terminates the sess io n be twe en th e tr an sfe re e a nd
the transferring party. At the end of a call, SIP terminates the sessions between all parties.
Cisco SIP IP Phone Administrator Guide
1-1
What Is Session Initiation Protocol?
Conferences can consist of two or more users and can be established using multicast or multiple unicast
sessions.
Note The term conference m eans an es tablished se ssion (o r call) between two or more endpoints. In this
document, the terms co nferenc e and call are used inte rchang eably.
Components of SIP
SIP is a peer-to-peer protocol. The peers in a session are called user agents (UAs). A user agent can
function in one of the following roles:
• User agent client (UAC)—A client application that initiates the SIP request.
• User agent server (UAS)—A server application that cont acts the us er when a SIP r equest is r ecei ved
and that return s a r esponse o n beha lf o f th e use r.
T ypically, a SIP endpoint is capable of functioning as both a UAC and a UAS, but functions only as one
or the other per transaction. Whether the endpoint functions as a UA C or a UAS depends on the UA that
initiated the request.
Chapter 1 Product Overview
From an architecture standpoint, the physical components of a SIP network can also be grouped into two
categories: clie nts an d ser vers. Figure 1-1 illustrates the architecture of a SIP network.
Note In addition, the SIP s ervers c a n inte rac t w ith oth er a ppli cat ion serv ice s, suc h as Li ght weght Direc tor y
Access Protocol (LDAP) servers, a da tabase app lica tion , or a n eX ten sible M arkup Lan gua ge ( XML )
application. These application services provide back-end services such as directory, authentication, and
billing services.
Figure 1-1 SIP Architecture
SIP Proxy and
Redirect Servers
SIP
SIP SIP
SIP User
Agents (UA)
SIP Gateway
Cisco SIP IP Ph one Administrator Guide
1-2
IP
RTP
PSTN
Legacy PBX
42870
Chapter 1 Product Overview
SIP Client s
SIP Server s
What Is the Cisco SIP IP Phone?
SIP clients include:
• Phones—Can ac t a s e ither a UAS or UAC. Softphones (PCs that have phone c apa bil ities i nsta lled)
and Cisco SIP IP phones can initiate SIP requests and respond to requests.
• Gateways—Provide call control. Gateways provide many services, the most common being a
translation function betwe en SI P conf e renc ing end point s and othe r term i nal t ypes. Thi s func ti on
includes translation between transmission formats and between communications procedures. In
addition, the gat eway also tra nsla tes bet wee n aud io a nd vid eo code cs a nd perf orm s c all setu p and
clearing on both the LA N si de and the switc hed-c irc uit n etwor k sid e.
SIP servers includ e:
• Proxy server—The pr oxy server is a n in term ed iate device tha t r ece ives SIP reques ts f rom a c li ent
and then forwards the re quests on the clie nt’s behalf. Basically , proxy servers receive SIP messages
and forward them to the next SIP server in the network. Proxy servers can provide functi o ns such as
authentication, authorization, network access control, routing, reliable request retransmission, and
security.
• Redirect server—Receives SIP requests, strips out the address in the request, checks its address
tables for any other addresses tha t may be mapp ed to the one in the reque st, and th en retur ns the
results of the address mapping to the client. Basically, redirect servers provide the client with
information about t he next hop or hops that a message should take an d then the client cont acts th e
next hop server or UAS directl y.
• Registra r se rv er —Processes requests from UACs for registration of their current location. Registrar
servers are often co-located with a redirect or proxy server.
What Is the Cisco SIP IP Phone?
Cisco SIP IP phones are full-featured telephones that can be plugged directly into an IP network and can
be used very much like a standard private branch exchange (PBX) telephone. The Cisco SIP IP phone is
an IP telephony inst ru ment t hat ca n b e use d in VoIP networks.
The Cisco SIP IP ph one mo del t erm i nals c an at tach t o t he exi sti ng data ne twork i nfras tr uct ure, via
10BASE-T/100BASE-T interfaces on a n E the rnet s witc h. Wh en use d with a voice- capa ble E the rnet
switch (one that understands type of service [ToS] bits and can prioritize VoIP traffic), the phones
eliminate the need for a traditional proprietary telephone set and key system and PBX.
The Cisco SIP IP phone complies with RFC 3261, as listed in Appendix A, “SIP Compliance with RFC
3261 Information”.
Figure 1-2 illustrates physical features of the Cisco SIP IP phone.
Cisco SIP IP Phone Administrator Guide
1-3
What Is the Cisco SIP IP Phone?
Figure 1-2 Cisco SIP IP Phone Physical Features
Chapter 1 Product Overview
LCD
Handset
•
LCD screen—Desktop, which di sp lays inf orm ation a bout yo ur Ci sco SIP IP phone , suc h as the
Dialing
pad
Scroll
Line or speed dial
key
buttons
Function
toggles
Footstand
adjustment
Soft keys
i button
On-screen
mode buttons
Volume
buttons
38007
time, date, yo ur pho ne nu mb er, caller ID, li ne and ca ll sta tus a nd the soft key ta bs.
• Line or speed-dial buttons—Opens a new line or speed dials the number on the L CD screen .
• Footstand adjustment—Adjusts the angl e of t he p hon e base.
• Soft keys—Activate the feature described by the text message directly above on the LCD screen.
• Information (i) button—Provides online he lp for selec ted keys or features and ne twork statisti cs
about the active call. Displays a descriptor of the key directly after pressing the i button. For
example, pressing t he i button, then up or d own displa ys a scr een inst ru cting yo u h ow to scrol l u p
and down on the LCD.
• On-screen mode buttons—Retrieves information about current settings, recent calls, available
services, and voice-mail messages.
• Volume buttons—Adjust the volume of the handset, headset, speaker, and ringer and adjust the
brightness contrast settings on the LCD screen.
• Function toggles—In cl u des t he se op tio n s:
–
Headset and speaker—Toggles these functions ena bli ng yo u to answ er the pho ne u sing a
headset or speakerphone.
–
Mute—Stops or resumes voice transmission.
• Scroll key—Enables you to move among different soft key options displayed on LCD screen .
• Dialing pad—Press the dial-pad buttons to dial a phone number. Dial-pad buttons work exactly like
those on your existi ng t elepho ne .
• Handset—Lift the handset and press the dial-pad numbers to place a call, review voice-mail
messages, and answer a call.
Cisco SIP IP Ph one Administrator Guide
1-4
Chapter 1 Product Overview
BTXML Support
Basic Telephony eXtensible Markup Language (BTXML) is supported on t he Cisco SIP IP phon e.
BTXML defines XML elements for controlling the user interface of an IP telephone. BTXML describes
what information i s d ispl ayed on the s cree n and h ow the use r provid es inpu t usi ng soft keys and ha rd
keys. User interface control is internal to the phone and there is no external BTXML user interface
control.
Cisco CallManager XML Support
The Cisco SIP IP phon e su p por ts cus to mer-written Cisco CallManager XML cards that ca n be acc es s ed
using buttons or softkeys on the phone. These cards can provid e data such as st ock quotes , calenda rs,
and directory lookups. The XML ca rds can be ac cessed by the fol lowing methods:
• From the Services soft key, configured using the services_url parameter.
• By pressing the i button.
• By pressing the dire ctory button and s electing External Direct ory , configur ed using the director y_url
parameter.
What Is the Cisco SIP IP Phone?
• By specifying a bitmap to be used as the phone's lo go (brandi ng), configured using th e logo_ur l
parameter.
See Chapter 3, “Managi ng Ci sco SI P IP Pho nes ” for information ab out configurin g these p aramete rs.
The Cisco SIP IP phone supports Cisco CallManger XML up to version 3.0. It does not support the XML
objects added in Cisc o Ca llMa nage r XM L version 3. 1:
• CiscoIPPhoneIconMenu
• CiscoIPPhoneExecute
• CiscoIPPhoneError
• CiscoIPPhoneResponse
• SoftKeyItem
The following exceptions apply to the Cisco SIP IP phone:
• External directories cannot be appended to the main list of directories under the directory button. If
external directori es are provisi on ed fo r the Ci sco SIP I P p hone , t he n th ey can be ac cesse d by
pressing the direct ory button and select ing the Ext ernal Di rector y option .
• The Cisco SI P IP ph one rem oves w hi te sp ac e whe n th e Ci sc o C all M an ag er XM L card s ar e
displayed. Multiple spaces are consolidated to a single space.
• Setting x and y coor dina tes fo r th e Cisco IPPho neIm age obje ct is not sup port ed. T he i m age always
appears at location 0,0 . Center ing of the image is not supp orted if x and y are set to -1.
• The Cisco SIP IP phone displays a ny v a lid titl e it re cei v es. Th is dif fe rs from the Cisco CallMan ager
phones in that the CiscoIPPhoneGraphicMenu object does not display a title even if it receives one
and the CiscoIPPhoneImag e objec t displays the pr evious menu item or “Services” rather than
received titles.
• Cisco CallManager phones allow embedded carriage returns and line feeds in menu items. In the
Cisco SIP IP phone, car riage re tur ns and li ne f eeds ar e discar de d.
• The Cisco SIP IP phone always displays the full set of di rectory sof tkeys. For Cisco CallManage r
phones, the softkeys can change depending on what type of object it receives. This is due to support
for Cisco CallManager 3.0 software.
Cisco SIP IP Phone Administrator Guide
1-5
What Is the Cisco SIP IP Phone?
• A parameter is sent along with the initial request for a Services or Directory URL which
differentiates the Cisco SIP IP phone from othe r types of pho nes.
For more information a bout usin g XM L on your Ci sco SI P IP ph one, see t he fo llowing li nk s o r
documents:
• Cisco IP Phone Se rv ice fo rum at th e fo llowing U RL:
http://www.hotdispatch.com/cisco-ip-telephony
• Cisco CallManager Services Developer Kit at the following URL:
http://www.cisco.com/warp/public/570/avvid/voice_ip/cm_xml/cm_xmldown.shtml
• Developing Cisco IP Phone Services by Darrick Deel, Mark Nelson, and Anne Smith, ISBN
1-58705-060-9
Supported Features
In addition to the features illustrated in Figure 1-2 on page 1-4, the Cisco SIP IP phone also provides the
following features.
Chapter 1 Product Overview
Physical Features
Network Features
• Adjustable ring tone
• Hearing-aid compatible handset
• Headset compatibility
• Integrated two-port Ethernet switch that allows the telephone and a computer to share a single
Ethernet jack
• Direct connection to a 10BASE-T or 100 BASE-T Ethernet (RJ- 45) networ k (half- or full-du plex
connections are supp orte d)
• Large (4.25 x 3 in. or 10.79 cm. x 7.62 cm .) display w ith adjusta ble contra st
• IP address assignment—Dyna mic Host Configurati on Protoco l (DHCP) client or ma nuall y
configured via a loca l se tup menu .
• Network startup using D HCP a nd Trivial File Transfer Protocol (TFTP).
• Te lnet support—Allows the user to use Te lnet to connect directly to the Cisco SIP IP phone to debug
and troubleshoot the phone. See the “Managing C i s co S I P IP P ho n e s ” section on page 3-1 for more
information on configura tion p aram eters.
• Ping support—Allows the user to use ping to see if a Cisco SIP IP phone is operational and ho w long
the response time is from the phone.
• Traceroute support—Allows the user t o use trac erou te to se e the p ath tha t a C isco SIP IP phon e
traverses in the route to its desired destination.
Cisco SIP IP Ph one Administrator Guide
1-6
Chapter 1 Product Overview
Configuration Features
The Cisco SIP IP phone provides the ability to:
• Configure Ethernet port mode and spe ed
• Register with or unregister from a proxy server
• Specify a TFTP bo ot d ire ctor y
• Configure a label for phone ide nti fication displ ay p urp oses
• Configure a name for caller identification purposes for each active line on a phone
• Configure a 12- or 2 4-hour use r i nte rface ti me d ispl ay
Codec and Protocol Support
• G.711 (u-law and a-law) and G.729a audio c ompre ssion.
• In-band dual tone mult ifrequency (D TMF) sup port for touc h-to ne dialin g.
• Out-of-band DT MF si gnal ing for cod ec s th at do not tr anspor t the DT MF sign ali ng corr ec tly ( fo r
example, G.729 or G .729 a) .
What Is the Cisco SIP IP Phone?
• Local (180 Ringi ng) o r r emot e (1 83 Sessio n Prog re ss) call progr es s ton e.
• Audio/Video Transport (AVT) payload type negotiation.
• Current date and time support via Simple Network Time Protocol (SNTP) and time zone and
Daylight Saving Time support.
• Call redirection infor mation supp ort via t he Diversion header.
• Third-party ca ll c ontr ol via d elaye d medi a negotia tio n. A de la yed m edi a negoti ation i s o ne w he re
the Session Description Protocol (SDP) information is not completely advertised in the initial call
setup.
• Support for endpoints specified as fully qual ified domain names (FQ DNs) in the SDP.
• Remote reset and d ial p lan up date suppo rt ( via the Event h eade r in N OTIFY messages).
Note See the “Supported Protocols” section on page 1-11 for additional supported protocols.
Dialing and Messaging Features
• Dial plan suppor t t hat ena bles a utom ati c di al ing and aut om atic g ener ati on of a sec ond ary d i al tone
• Local directory configuration (save and recall) and automatic dial completion—Each time a call is
successfully made or received, the number is stored in a local directory that is maintained on the
phone. The maximum num ber of en tries is 32. Entri es are aged- out bas ed on thei r usage and ag e.
The oldest entry called the least number of times is overwritten first. This feature cannot be
programmed by the user, however, up to 20 entries can be “locked” (via the Locked soft key) so that
they will never be deleted.
• Message W aiting Indication (via u nsolicited NOT IFY)—Lights to indicate that a new v oice message
is in a subscriber’s mailbox. If the subscriber listens to the message but does not save or delete the
message, the light remains on. If a subscriber listens to the new message or messages, and saves or
deletes them, the light goes of f. The message w aiting indi cator is contr olled b y the v oicemai l server.
The indication will be saved over a phone upgrade or reboot.
Cisco SIP IP Phone Administrator Guide
1-7
What Is the Cisco SIP IP Phone?
• Speed dial to voice-mail via the messages button
• Do not disturb—Allows the user to instruct the system to intercept incoming calls during specified
periods of time when t he u ser d oe s no t want to be di sturbe d.
• Multiple directory numbers —Allows the Cisco SIP IP phone to have up to six directory numbers or
lines.
• Call waiting (enabled)—Plays an audible tone to indica te that an incoming call is waiting . Th e us er
can then put the existing call on hold and accept the other call. The user can alternate between the
two calls.
• Call waiting (disabled)—Allows the user to instruct the system to block call waiting calls during a
specified period of time.
• Direct number dialing—Allows users to initiate or receive a call using a standard E.164 number
format in a local, national, or international format.
• Direct URL dialing—Provides the ability to place a call using an e-mail address instead of a phone
number.
• Caller ID blocking—Allows the user to instruct the system to block phone number or e-mail address
from phones that have caller identification capabilities.
Chapter 1 Product Overview
Call Options
• Anonymous call blocking —Allows the user to instruct the system to block any calls for which the
identification is blocked.
• Three-way conferencing—Supports one phone conf eren ci ng w it h two oth er phone s by provid ing
mixing on the initiating phone. To set up a three-way conference call, see the “Making Conference
Calls” section in Chap ter 3 of the C isco IP Pho ne Models 7960 and 794 0 User Guide.
• Call forwa rd (netwo rk)—Allows the Cisco SIP IP phone user to request forwarding service from the
network (via a third-part y tool that enab les this fea ture to be configured) . When a call is pl aced to
the user’s phone, it is redirected to the appropriate forward destina tion by the SIP proxy se rver.
• Call hold—Allows the Cisco SIP IP phone user (user A) to place a call (from user B) on hold. When
user A places user B on hold, the two-way RTP voice path between user A and user B is temporarily
disconnected, but t he call session i s stil l connec ted. When user A tak es user B o ff hold, the two -way
RTP voice path is reestablished.
• Call transfer—Allows the Cisco SIP IP phone user (user A) to transfer a call from one user (user B)
to another user (u ser C). User A pla ces user B on hol d and calls user C. If user C accepts the tran sfer ,
a session is estab lish ed bet wee n user B and use r C an d t he ses s ion betw een u ser A a nd us er B is
terminated.
• Three-way calling—Allows a “bridged” three-way call. When a three-way call is established, the
Cisco SIP IP phone th rou gh whi ch t he c all i s es tablis hed a cts as a br idge , mi xin g th e au dio me dia
for the other parties.
Routing and Proxy Features
• User-defined proxy routing
The Route attribute of the template tag in the dial-plan template file can be used to indicate which
proxy (default, emergency, FQDN) that the call should be init ially route d to. For example, to
configure an emergency proxy, specify value of the Route attribute as “emergency.”
• Backup SIP proxy
Cisco SIP IP Ph one Administrator Guide
1-8
Chapter 1 Product Overview
• Emergency SIP proxy
What Is the Cisco SIP IP Phone?
When the primary proxy does not respo nd to the INV ITE message se nt by the Cisco SIP IP Phone
after the configured number of retries, the Cisco SIP IP Phone sends the INVITE to the backup
proxy. This is independent from the pr oxy defined i n t he Ro ut e a ttri bute in t he d ial -pla n t empl ate
used.
The Cisco SIP IP phone does not have to register with the backup proxy. All interactions with the
backup proxy, such as authentication challenges, are treated the same as the interactions with the
primary proxy.
The backup proxy is used only wit h new INVITE messa ges. Once the back up proxy is used, i t is
active for the duration of the call.
The location of the bac kup SIP p rox y ca n be de fined a s an IP ad dress in the de fault c on figuration
file. See the proxy_backup an d proxy_ backu p_port pa rameter s in the “Modifying the Default SIP
Configuration File” section on page 3-8 .
An optional emergency SIP proxy can be configured with the Route attribute of the template tag in
the dial-plan template file. See “Support of user-defined proxy routi ng.”
When an emergency SIP pro xy is configured and a call is initiated, the p hone gen erates an IN VITE
message to the addre ss spec ified in t he pr ox y_eme rgency para met er. The emergency proxy is used
for the call duration.
The location of the emergency pro xy can be defined as an IP address i n the default configurati on
file. See the proxy_emergency and proxy_ emergen cy_port para meters in “Modifyin g th e D efau lt
SIP Configuration File” section on page 3 -8.
• Support of DNS SRV
The Domain Name Server RR (DNS SRV) is used to locate servers for a given service.
SIP on Cisco’s SIP IP phones uses a DNS SRV query to determine the IP address of th e SI P proxy
or redirect server. The query str ing gen er ate d i s in co mpl ia nce w ith RFC 2782, a nd pr ep en ds th e
protocol label w ith a n und er score _, as in “_protocol._transp ort. ” The addition of the unde rscore
reduces the risk of the same name being used for unre lated pur poses.
In compliance with RFC 2782 and the draft-ietf-sip-srv-01 specification, the system can remember
multiple IP addresses and use them properly. In the draft-ietf-sip-srv-01 specification, it is assumed
that all proxies returned for the SRV record are equivalent such that the p h one can register with any
of the proxies an d i niti ate a ca ll us ing a ny othe r pr oxy.
• Configurable Voice Activity Detection
Voice Activity Detection (VAD) can be enabled or disabled with enable _v ad par ameter. Use a v alue
of 0 to disable, and a value of 1 to enable. See enable_vad parameter in “Modifying the Default SIP
Configuration File” section on page 3-8.
• Distinctive Alerting
If the INVITE message contains an Alert-Info header, distinctive ringing is invoked. The format of
the header is “Alert-inf o: x ”. Th e value of “x” can be any number. This header is only received by
the phone and is not generate d by the phone.
Distinctive ringing is supported when the phone is idle or during a call. In the idle mode, the phone
rings with a different cadence. The selec t ed ri ng in g t yp e p lay s twice with a short pause in betwe en .
In call-waiting mode, two short beeps are generated instead of one long beep.
• Network Address Translation (NAT) and Outbound Proxy
NAT can be enabled or dis abled with th e nat_e nable p aramete r . You can configur e the address o f the
NAT o r firewall server u sing t he na t_a ddr es s para m eter.
Cisco SIP IP Phone Administrator Guide
1-9
What Is the Cisco SIP IP Phone?
You can configure the IP address and port numbe r of the out bound proxy se rver. When outbound
proxy is enabled, a ll SIP re quests a re sen t t o th e outb ound p roxy se r ver inste ad o f t he
proxyN_address. All respo nses continu e to follow the using the norm al Via processing rules. The
media stream is no t rout ed t hroug h the out boun d p rox y.
NAT and outbound pro xy modes can be indepe ndently en abled or disa bled. Th e received= tag is
added to the Via header of all re spon ses if there is n o received= ta g in the up permos t Via he ader an d
the source IP ad dress is di fferent f rom the IP add ress i n the uppe rm ost Via header. Responses are
sent back to the source under the following conditions:
Note For information on how to use the standard telephony features and URL dialing, refer to the documents
listed in the “Related Documentation” section on page -viii.
Chapter 1 Product Overview
–
If a received= tag is in t he up pe rmo st Via header, the re spon se i s sent back to t he IP a ddr ess
contained in the received= tag.
–
If there is no received= tag and the IP address in the uppermost Via header is different than the
source IP address, the response is sent back to the source IP. Otherwise the response is sent back
to the IP address in the uppermost Via header.
Character Support
Note The XML cards, info text, and menus are all in English. These items are built into the phone image and
The Cisco SIP IP ph one s uppo rts the ISO 8 859 -1 Lat in1 ch ara cters. T he fo l lowing la ngu ag es a re
supported:
French (fr), Spanish (es), Cata lan (ca), Basque (eu) , Portu guese (pt ), Italia n (it), Alba nian (sq) ,
Rhaeto-Romanic (rm), Dutch (nl) , German (de), Danish (da), Swedish (sv), Nor wegian (no), Finnish (f i),
Faroese (fo), Icelandic (is), Irish (ga), Scottish (gd), English (en), Afrikaans (af) and Swahili (sw).
The following languages ar e not supp orted:
Zulu (zu) and other Bantu languages usi ng Latin Ext ended -B letter s, Arabi c in North Afri ca, and
Guarani (gn) missing GEIUY with ~ tilde.
cannot be ch ange d.
ISO 8859-1 Latin1 char acters ca n be used in th e following areas:
• Caller ID information. When a SIP message is received with ISO 8859-1 Latin1 characters in the
caller ID strings, those caller ID strings are displayed on the Cisco SIP IP phone's LCD screen with
the correct ISO 8859-1 Latin1 characters.
• Services menu applications written in CMXML. The customer can develop language-specific
applications for a pa rticular region. F or e x am ple, an application that displayed the current wea th er
in Sweden using Swedish langu age ch aracte rs can be disp layed on the Cisco SIP IP phone. If th e
customer de velo ps the same applicati on for a Sp anish t ow n, they could tr anslat e the applica tion into
Spanish.
• Line key labels. The line keys can be configured to support the Latin1 character. The line key name
can be specified in the c onf ig f ile and it will be display ed corr ectly. The Latin1 characters cannot be
used in the lineX_name, but can be used in the lin eX_shor tname a nd lineX_d isplay name. If the
proxy supports L atin1 characters i n the To/From headers, then they can be used in the lineX_name
parameter as well.
Cisco SIP IP Ph one Administrator Guide
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Chapter 1 Product Overview
Supported Protocols
The Cisco SIP IP ph one s uppo rt s th e fo llowing sta nda rd pr otoc ols:
• Domain Name S yste m (DNS )—Used in the Internet for translating names of network nodes into
addresses. SIP uses D NS t o re solve th e host na mes o f endpoi n ts to IP a dd resses.
• Dynamic Host Control Pr otocol (D HCP)—Used to dynamically allocate and assign IP addresses.
DHCP allows you to move network devices from one subnet to another without admi nistrative
attention. If using DHCP, you can connect Cisco SIP IP phones to the network and become
operational withou t having to man ua lly a ssign an I P addr ess a nd ad di tiona l ne twor k par amet ers.
The Cisco SIP IP ph one com pli es w ith t he DHCP spe cifications do cu ment ed i n RFC 2131. By
default, Cisco SIP IP ph ones ar e DHC P-e nabled .
• Internet Control Message Protocol (ICMP)—A network layer Internet protocol that enables hosts to
send error or con trol mess age s to oth er ho sts. ICMP al so provid es othe r i nfo rmati on rel evant to IP
packet processing.
The Cisco SIP suppo rts I CMP as i t is doc umen t ed in RFC 792 .
• Internet Protocol (IP)—A network layer protocol that sends datagram packets between nodes on the
Internet. IP also provides features for addressing, type-of-service (T oS) specification, fragmentation
and reassembl y, and security.
What Is the Cisco SIP IP Phone?
The Cisco SIP IP ph one s uppo rts IP a s it i s defined i n RFC 79 1.
• Real-Time Transport Protocol (RTP)—Transports re al -ti me d ata ( suc h as voice da ta ) over da ta
networks. RTP also has the ability to obtain quality of service (QoS) information.
The Cisco SIP IP ph one s uppo rts RTP as a media cha nnel.
• Session Description Protocol (SDP)—An ASCII-based protocol that describes multimedia sessions
and their related sche duling info rmat ion.
The Cisco SIP IP ph one u ses SD P for sessi on descr iptio n.
• Simple Network Time Protocol (SNTP)—Sychronizes com pute r clocks on a n IP n etwor k. The
Cisco SIP IP phone s use SNTP for thei r date and tim e support.
• Transmission Control Protocol (TCP)—Provides a reliable byte-stream transfer service betwee n two
endpoints on an inter net . The Cisco SIP IP pho ne suppor ts TCP f or Telnet sessions only.
• Triv ial File Transfer Protocol (TFTP)—Allows f iles to be transferre d from one com puter to an other
over a network. The Cisco SIP IP phone uses TFTP to download configuration files and software
updates.
• User Datagram Pr ot ocol (U DP )—A simple protocol that exchanges data packets without
acknowledgments or guara nteed delivery. SIP can use UD P as the underlying transport protocol. If
UDP is used, retransmissions are used to ensure reliability.
The Cisco SIP IP ph one s uppo rts UDP as it i s defined in RFC 76 8 for SI P signa ling .
• Hypertext Transfer Protocol (HT TP) —The phone contains limi ted suppor t for HTTP 1.1. The
phone uses HTTP to retrieve Cisco CallManager XML files.
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Prerequisites
Prerequisites
For the Cisco SIP IP phone to successfully operate as a SIP endpoint in your network, your network must
meet the following requirements:
• A working IP network is established.
• VoIP is configured on your Cisco routers.
• VoIP gateways are configured for SIP.
• A TFTP server is active and contains the latest Cisco SIP IP phone firmware image in its root
• A proxy server is active and configured to receive and forward SIP messages.
Chapter 1 Product Overview
For more information abou t configuring IP, refer to Cisco IOS IP Configuration Guide,
Release 12.2.
For more information about configuring VoIP, refer to the Cisco IOS Voice, Video, and Fax
Configuration Guide, Release 12.2, for the appropriate access platform. F or more info rmation about
configuring SIP VoIP, refer to the “Configuring SIP for VoIP” chapter.
directory.
Cisco SIP IP Phone Connections
The Cisco SIP IP phone has connections for connect i ng to the data network, for providing power to the
phone, and for connecting a headset to the phone. Figure 1-3 illustrates the connections on the Cisco SIP
IP phone.
Figure 1-3 Cisco SIP IP Phone Cable Connections
Cisco IP Phone (rear view)
Power
outlet
AC adapter
port
(DC48V)
(optional power
cable)
RS-232 port
Headset
port
Handset
port
Cisco SIP IP Ph one Administrator Guide
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Network port
(10/100 SW)
Access port
(10/100 PC)
62472
Chapter 1 Product Overview
Connecting to the Network
The Cisco SIP IP phone has two RJ-45 ports that each support 10/100 Mbps half- or full-duplex Ethernet
connections to ext erna l devices —network port (labeled 10 /100 SW) and acc ess port (labe led
10/100 PC). You can use either Category 3 or 5 cabling for 10 Mpbs connections, but use Category 5 for
100 Mbps connect ions. On bo th t he n etwor k port and acc ess port , use f ull -dupl ex mode to avoid
collisions.
Network Port (10/100 SW)
Use the network port to connect the phone to the network. You must use a straight-through cable on this
port. The phone can also obtain inline power from the Catalyst switch over this connection. See the
“Connecti ng to Power” section on pa ge 1- 13 for details.
Access Port (10/100 PC)
Use the access port to connect a network device, such as a computer, to the phone. You must use a
straight-through cable o n thi s port .
Cisco SIP IP Phone Connections
Connecting to Power
The Cisco SIP IP phone can be powered by the following sources:
• External power source—Optional Cisco AC adapter and power cord for connecting to a standard
wall receptacle.
• WS-X6348-RJ45V 10/100 sw itchin g m odu le—Provides inline power to the C isco SIP IP phon e
when connected to a Ca ta lyst 3500, 4 000 , or 600 0 fa mily 10/10 0BASE-TX sw itchi ng modu le.
This module send s power on pin s 1 and 2, and 3 an d 6 .
• WS-PWR-PANEL—Power patch panel provides power to the Cisco SIP IP phone, which allows the
Cisco SIP IP phone to be con ne cted to existi ng Catal yst 4 000 , 5000, a nd 6000 fa mil y
10/100BASE-TX switching modules.
This module sends p ower on p in s 4 , 5, 7, and 8.
• WS-X4148-RJ45V—48-port 10/100 Eth erne t w ith i nline p ower modu le for t he C ata lyst 4006.
• WS-X4095-PEM—VoIP DC Power Entry module for the Catalyst 4006.
• WS-X4608-2PSU and W S-X 460 8—External -48V DC power shelf common equipment for the
Catalyst 4006 with two AC-to-DC power suppl y units (PSU s) an d one emp ty b ay f or redun da nt
option, and the 110V 15A AC-to-48V DC PSU redu ndant opti on for the power shel f.
• WS-C3524-PWR-XL-EN—Catalyst 3524-PWR XL switch.
Note Only the network port (labeled 10/100 SW) supports inline power from the Catalyst switches.
For redundancy, you can use the Cisco AC adapter even if you are using i n line power from the Catalyst
switches. The Cisco SIP IP phone can share the power load being used from the inline power and external
power source. If either the inline power or the external power goes down, the phone can switch entirely
to the other power source.
Cisco SIP IP Phone Administrator Guide
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The Cisco SIP IP Phone with a Catalyst Switch
T o use this redundancy feature you must set the inline power mode to auto on the Cisco Ca taly st switch .
Next, connect the unpowered Cisco SIP IP phone to the network. After the phone powers up, connect the
external power supply to the phone.
Using a Headset
The Cisco SIP IP ph one su ppo rts a fou r- or six -wire hea dset jack. Spec ifically, the Cisco SIP IP pho ne
supports the following Plantronics headset models:
• Tristar Monaural
• Encore Monaur al H91
• Encore Binaura l H 101
The volume and mute controls also adjust volume to the earpiece and mute the speech path of the
headset. The headset activation key is located on the front of the Cisco SIP IP phone.
Note When using a headset, an amplifier is not required. However, a coil cord is required to connect the
headset to the headset port on the back of your Cisco IP Pho ne 7960/7940. For information on ordering
compatible headset s and coil cord s for the Cisco IP Phone 7960/7460 , go to the fol lowing URL:
Chapter 1 Product Overview
http://cisco.getheadsets.com or http://vxicorp.com/cisco
The Cisco SIP IP Phone with a Ca talyst Switch
To function in the IP telephony network, the Cisco SIP IP phone must be connec ted to a networking
device, such as a Catalyst switch, to obtain network connectivity.
The Cisco SIP IP phone has an internal Ethernet switch, which enables it to switch traffic coming from
the phone, access port, and t he network port .
If a computer is connected to the access port, packets traveling to and from the computer and to and from
the phone share t he sa me physi cal lin k to the swi tch and the same port on t he sw itch .
This configuration has these implications for the VLAN configuration on the network:
• The current VLANs might be configure d on an IP subnet ba sis, and add itional IP add resse s might
not be available to assign the pho ne t o a por t so tha t i t be long s to the same su bnet as o ther devices
(PC) connected to the same port.
• Data traffic present on t he V LAN supp ort ing ph ones mi ght reduc e t he q ua lity of VoIP traffic.
You can resolve these issues by isolating the voice traffic onto a separate VLAN on each of the ports
connected to a p hon e. T he swi tch port configure d f or conne c ting a p hone would have separa te VL ANs
configured for ca rr ying :
• Voice traffic to and from the Cisco SIP IP pho ne (a ux ilia ry VLAN )
• Data traffic to and from the PC connected to the switch through the access port of the Cisco SIP IP
phone (native VLAN)
Isolating the phones on a separate, auxiliary VLAN increases the quality of the voice traffic and allows
a large number of phones to be added to an existing net work whe re there are not enough IP address es.
For more information, refer to the documentation included with the Catalyst switch or available online
at the following URL:
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