Cisco IP phone 7960 User Manual

Cisco SIP IP Phone 7960 Administrator Guide

Version 2.0

Corporate Headquarters

Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA http://www.cisco.com Tel:

408 526-4000 800 553-NETS (6387)

Fax: 408 526-4100

Customer Order Number: DOC-7810497= Text Part Number: 78-10497-02

THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS.

THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PR ODUCT ARE S ET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY.

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 accordance 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 w hich case users will be required to correct t he interference at their own expense.

The following information is for FCC complia nce of Cl ass B devices: The equi pment desc ribed in thi s manual generates and may rad iate radio-frequency energy. 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 wit hou t Cisc o’s written authori zatio n may res ult in the e quipm ent no lon ger comply ing with FCC re quirem ents for Class A or Class B digital devices. In that event, your right to use the equipm ent may be lim ited by FCC regulations, and you may be requi red to correct any interference to radio or television communicati ons at you r own expense.

You can determine whether your equipmen t is causing interf erence by turni ng it off. If the in terference s tops, it was probably caused by the Cisco equipment 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 following measures:

• Turn the television or radio antenna until the interf erenc e stops.

• Move the equipment to one side or the other of the te levision or r adio.

• Move the equipment farther away from the television or radi o.

• Plug the equipment into an outlet that is on a different circuit from the tel evision or radio. (That is, make certain the equ ipment and the television or radio are on circuits controlled by different circuit br eakers or fuses.)

Modifications to this product not authori zed by Cisco Sys tems, Inc. could void the FCC app roval and negate your authori ty to operate the product. The Cisco implementation of TCP header compression is an adaptation of a pr ogr am d eveloped by the University of California, Berkeley (UCB) as

part of UCB’s public domain version of the UNIX operating system. All rights reserved. Copyright © 1981, Regents of the University of California. NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE

PROVIDED “AS IS” WITH ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE.

IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES.

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All other brands, names, or trademark s mentioned in this do cument or Web site ar e the property of their re spective owner s. The use of t he word partner does not imply a partnership relat ionsh ip between Cis co and any oth er com pany. (001 0R)

Cisco SIP IP Phone 7960 Administrator Guide

About This Guide ix

CONTENTS

CHAPTER

Overview Who Should Use This Guide Objectives Organization Related Documentation Document Conventions Obtaining Documentation

Obtaining Technical Assistance

Product Overview

What is Session Initiation Protocol?

World Wide Web

Documentation CD-ROM Ordering Documenta tion

Cisco Connection O nline

xvi

Technical Assistance Center Documentation Feedback

1-1

xvi

xvii

1-1

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Components of SIP

SIP Clients SIP Servers

1-3

1-4

1-5

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Contents

CHAPTER

What is the Cisco SIP IP Phone 7960?

Supported Features Supported Protocols

Prerequisites

1-12

1-7

1-10

Cisco SIP IP Phone Con nections

Connecting to the Network Connecting to Power Using a Headset

1-14

1-15

The Cisco SIP IP Phon e with a Catalyst Switch

Getting Started with Your CiscoSIP IP Phone

Initialization Process Overview Installing the Cisco SIP IP Phone

Installation Task Summary Downloading Files to Your TFTP Server Configuring SI P Parameters

Configuring SI P Parameters via a TFTP Server

1-5

1-13 1-13

1-16

2-1

2-3

2-4

2-5

2-6

Manually Configuring the SIP Parameters

Configuring Network Parameters

Configuring Network Parameters via a DHCP Server Manually Configuring the Network Parameters

Connecting the Phone

Adjusting the Pl acement of the Cisco SIP Phone

Verifying Startup

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2-11

2-13

2-14

2-16

2-18

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CHAPTER

Using the Cisco SIP IP Phone Menu Interface Reading the Cisco SIP IP Phone Icons

2-22

Customizing the Cisco SIP IP Phone Ring Types Creating Dial Plans

Managing Cisco SIP IP Phones

Entering Configuration Mode

Unlocking Configuration Mode Locking Configu ration Mode

2-24

3-1

3-2

Modifying the Phone’s Network Settings Modifying the Phone’s SIP Settings

3-5

Modifying SIP Parameters via a TFTP Server

Modifying the Default SIP Configuration File Modifying the Phone-Specific SIP Confi guration File

Modifying the SIP Parameters Manually Setting the Date, Time, and Daylight Savings Time Erasing the Locally-Defined Settings

3-28

2-21

2-24

3-2

3-8

3-15

3-18

3-22

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Erasing the Local ly-Defined Networ k Settings

Erasing the Local ly-Defined SIP Settings Accessing Status Information

Viewing Status Messages

Viewing Networ k S tatistics

Viewing the Firmware Version

3-30

3-31

3-33

Upgrading the Cisco SIP IP Phone Firmware

3-29

3-33

Performing an Image Upgrade and Remote Reboot

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3-35

Contents

APPENDIX

SIP Compliance with RFC-2543 Information

SIP Functions SIP Methods SIP Responses

A-2

A-3

1xxResponse—Information Responses

2xxResponse—Successful Responses

3xxResponse—Redirection Responses

4xxResponse—Request Failure Responses

5xx Response—Server Failure R es po n se s

6xxResponse—Global Responses SIP Header Fields

A-10

SIP Session Description Protocol (SDP) Usage

SIP Call Flows

B-1

Call Flow Scenarios for Successful Calls

Gateway-to Cisco SIP IP Phone—Successful Call Setup and Disconnect

Gateway-to-Cisco SIP IP Phone—Succes sful Call Setup and Call Hold

A-1

A-4

A-5

A-10

A-12

B-2

B-3

B-7

Gateway to-Cisco SIP IP Phone—Successf ul Call Setup and Call

Transfer

B-11

Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold

Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Hold with Consultation

Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting

Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer witho ut

Consultation

B-31

Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer with

Consultation

B-35

Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding

(Unconditional)

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B-20

B-25

B-41

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Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding

(Busy)

B-44

Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (No

Answer)

B-48

APPENDIX

Cisco SIP IP Phone-to Cisco SIP IP Phone 3-Way Calling Call Flow Scenarios for Failed Calls

B-58

Gateway-to-Cisco SIP IP Phone—Called User is Busy

Gateway-to-Cisco SIP IP Phone—Called User Does Not Answer

Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error

Cisco SIP IP Phone-to-Cisco SIP IP Phone— Called User is Busy

B-52

B-58

B-60

B-63

B-66

Cisco SIP IP Phone-to-Cisco SIP IP Phone— Called User Does Not

Answer

Cisco SIP IP Phone-to-Cisco SIP IP Phone— Authentication Error

Technical Specifications

Physical and Operating Environment Specifications Cable Specifications Connections Specifications

Translated Safety Warnings

Installation Warning

B-68

B-70

C-1

C-3

D-1

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Product Disposal Warning

Lightning Activity Warning

D-2

D-3

SELV Circuit Warning (other versions available)

Circuit Breaker (15A) Warning

D-6

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vii

Contents

GLOSSARY

INDEX

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About This Guide

Overview

The Cisco Session Initiation Protocol (SIP) IP Phone 7960 Admini strator Guide provides information about how to setup, connect cables to, and configure a Cisco SIP IP phone 7960 ( here after refe rr ed to a s a C isco SIP IP phone) . Th e administrator guide also p rovides info rmation on how to con figure the network and SIP settings and change the settings and options of the Cisco SIP IP phone. The administrator guide also includes reference information such as Cisco SIP IP phone call flows and compliance information.

Who Should Use This Guide

Network engineers, system administrators, or telecommunic ation engineers should use this guide to learn the steps required to properly set up the Cisco SIP IP phone on the networ k.

The tasks described are considered to be administration-level tasks and are not intended for end-users of the phones. Many of the tasks involve configuring network settings which could affect the phone’s ability to function in the netw ork and require an un de rstanding of IP networ king and tele phony con cepts.

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Cisco SIP IP Phone 7960 Administrator Guide

Objectives

The Cisco SIP I P Pho ne 79 60 Adm inistrator Guide provides necessary information to get the Cisco SIP IP phone operational 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 implementing a SIP VoIP network, refer to the documents listed in the “Related Documentation” section on page xi.

Organization

This administrator guide is divided into the following cha pters a nd a ppend ixes:

About This Guide

•

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, and con figure th e Cisco SI P IP ph one.

•

Chapter 3, “Managing Cisco SIP IP Phone s” de scribe s how to mo dify the Cisco SIP IP phone’s network and SIP settings, how to a ccess network and call status information, and how to u pgrad e the firmware.

•

Appendix A, “SIP Complianc e w ith R FC-25 43 Infor ma tion” p rovides reference information a bou t the SI P IP phone c omplian ce to RFC 25 43.

•

Appendix B, “SIP Call Flows” provides reference information about the SIP IP phone call flows.

•

Appendix C, “Technical Specifications” lists the physical and oper ating environment 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 phone.

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About This Guide

Related Documentation

•

Voice over IP for the Cisco AS5300 Doc umen ts

Document Conventions

This docume nt u ses t he fo ll owing conventions:

•

Commands and keywords are in boldface font.

•

Arguments for which you supply values are in italic font.

•

Elements in square br ackets ([ ]) are optional.

•

Alternative keywords are grouped in braces and separated by vertical bars (for example, { x | y | z }).

•

Optional alternative keywords are grouped in brac kets and sepa ra ted by vertical bars (for examp le, [ x | y | z ] ).

•

Terminal sessions and information the system displays are in

•

Information you must enter is in

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boldface screen

Cisco SIP IP Phone 7960 Administrator Guide

font.

screen

font.

Document Conventions

About This Guide

Notes use the following conventions:

Note

Caution

Warning

Waarschuwing

Means reader take note. Notes contain helpful suggestions or references to material not covered in the publication.

Cautions use the following conventions:

Means read er be careful . In this situation, you m ight do something that could result in equipment damage or loss of data.

Warnings use the following conventions:

This warning symbol means danger. You are in a situation that could cause bodily injury. Before you work on any equipm ent, 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.)

xii

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Document Conventions

Varoitus

Attention

Warnung

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 Warnings" (käännetyt turvallisuutta koskevat varoitukset).)

Ce symbole d’avertissement indique un danger. V ous 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).)

Avvertenza

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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).

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Document Conventions

About This Guide

Advarsel

Aviso

Advertencia

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.”)

xiv

Varning!

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 d enna publikation i appendix "Translated Safety Warnings" [Översatta säkerhetsvarningar].)

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About This Guide

Obtaining Documentation

World Wide Web

You can access the most current Cisco documentation on the World Wide Web at http://www.cisco.com, http://www-china.cisco.com, or http://www-europe.cisco.com.

Documentation CD-ROM

Cisco documentation and additional literature are available in a CD-ROM package, which ships with your product. The Documentation CD-ROM is updated monthly. Therefore, it is probably more current than printed documentation. The CD-ROM package is available as a single unit or as an annual subscription.

Obtaining Do cu m e ntation

Ordering Documentation

Registered CCO users can order the Docu mentation CD-ROM and other Cisc o Product documentation throug h our onlin e Subscrip tion Se rv ices a t http://www.cisco.com/cgi-bin/subcat/kaojump.cgi.

Nonregistered CCO users can o rder do cum entation throug h a lo cal ac count representative by calling Cisco’s corporate headquarters (Califor nia, U SA) at 408 526-4000 or, in North Am erica , call 8 00 553 -NET S (6387).

Obtaining Technical Assistance

Cisco provides Cisco Connection On line (CC O) as a starting p oint for all technical assistance. Warranty or maintenance contract customers can use the Technical Assistance Center. All customers can submit technical feedback on Cisco documentation using the web, e-mail, a self-addr essed stamped response card included in many printed docs, or by sending mail to Cisco.

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Obtaining Technical Assistance

Cisco Connection Online

Cisco continues to revolutionize how business is done on the Internet. Cisco Connection Online is the foundation of a suite of interactive, networked services that provides immediate, open access to Cisco information and resources at anytime, from anywhere in the world. T his highly integrated Internet application is a powerful, easy-to-use tool for doing business with Cisc o.

CCO’s broad range of features a nd serv ice s helps cu stomers a nd partne rs to streamline business processes and improve productivity. Through CCO, you will find information about C isco and our n etworking solu tions, se rvices, and programs. In addition, you can resolve technical issues w ith online support services, download and test software packages, and order Cisco learning materials and merchandise. Valuable online skill assessment, training, and certification programs are also available.

Customers and partners can self-register on CCO to obtain additional personalized information and service s. Registered users may order prod ucts, check on the status o f an orde r and v i ew benefits specific to their rela tionshi ps with Cisco.

You can access CCO in the following ways:

About This Guide

•

WWW: ww w.cisco.com

•

Telnet: cco.cisco.com

•

Modem using standard connec tion rates and the fo llowing terminal settings: VT100 emulation; 8 data bits; no parity; and 1 stop bit.

–

From North America , c all 408 526 -8070

–

From Europe, call 33 1 64 46 40 82

You can e-mail questions about using CCO to cco-team@ cisco.com.

Technical Assistance Center

The Cisco Technical Assistance Ce nter (TAC) is available to warranty or maintenance co ntract custo mers w ho need te chnica l assi stan ce with a Cisco product that is under warranty or covered by a maintenance co ntract.

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About This Guide

Obtaining Technical Assistance

To display the TAC web site that includes links to technical support information and software upgr ades an d for req ues ting TAC support, use www.cisco.com/techsupport.

To contact by e-mail, use one of the following:

Language E-mail Address

English tac@cisco.com Hanzi (Chinese) chinese-tac@cisco.com Kanji (Japanese) japan-tac@cisco.com Hangul (Korean) korea-tac@cisco.com Spanish tac@cisco.com Thai thai-tac@cisco.com

In North America, TAC can be reached at 800 553-2447 or 408 526- 7209. For other telephone numbe rs a nd TAC e-mail addresses world wide, consu lt the following web site: http://www.cisco.com/warp/public/687/Directory/DirTAC.shtml.

Documentation Feedback

If you are reading Cisco product documentation on the World Wi de Web, you can submit technical comments electronically. Click Feedback in the toolbar and select Documentation. After you complete the form, click Submit to send it to Cisco.

You can e-mail your comments to bug-doc@c isco.com. T o submit your comments by mail, for your convenience many documents contain

a response card behind the front cover. Otherwise, you can mail your comments to the following address:

Cisco Systems, Inc. Document Resource C onnectio n 170 West Tasman Drive San Jose, CA 95134-9883

We appreciate and valu e your co mmen ts.

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Obtaining Technical Assistance

About This Guide

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CHAPTER

Product Overview

This chapter contains the following information about the Cisco SIP IP phone:

•

What is Session Initiation Protocol?, page 1-1

•

What is the Cisco SIP IP Ph one 7 960 ?, pa ge 1-5

•

Prerequisites, page 1-12

•

Cisco SIP IP Phone Connec tions, page 1-13

•

The Cisco SIP IP Phone with a Catalyst Switch, page 1-16

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-layer control protocol (de fined in RFC 2543) that can be used to establish, maintain, and terminate calls between two or more end points.

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Like other VoI P protocols, SIP is designed to address the func tions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundar ies. Session management provides the ability to control the attributes of an end-to-end call.

Cisco SIP IP Phone 7960 Administrator Guide

1-1

What is Session Initiation Protocol?

SIP provides the capabilities to:

•

Determine the location of the target end point—SIP supports address resolution, name mapping, and call redirection.

•

Determine the media capabilities of the target end point—Via Session Description Protocol (SDP), SIP determines the “lowest level” of common services between the end points. Confer ences are establishe d using only the media capabilities that can be supported by all end points.

•

Determine the availability of the target end point—If a call cannot be completed because the target end point is unavailable, SIP determines whether the called party is al read y on the ph one o r d id n ot answ er in the allotted number of rings. It then returns a message indicating why the target end point was unavailable.

•

Establish a session between the originating and target end point—If the call can be completed, SIP establishes a session between the end points. SIP also supports mid-call changes, such as the addition of another end point to the conference o r t h e ch an ging of a m ed ia ch arac t er isti c o r co de c.

•

Handle the transfer and termination of calls—SIP supports the transfer of calls from one end poin t to an other. During a call transfe r, SIP simply establishes a session between the transferee and a new end point (specified by the transferring party) and terminates the session between the transferee and the transferring party. At the end of a call, SIP terminates the sessions between all parties.

Chapter1 Product Overview

1-2

Conferences can consist of two or more users and can be establish ed using multicast or multiple unicast sessions.

Note

The term conference means an established session (or call) between two or more end points. In this documen t, the terms conference and call are used interchangeably.

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Chapter 1 Product Overvi ew

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 serve r (UAS)—A server application t ha t c ontacts the user when a SIP request is received and that returns a response on behalf of the user.

T ypically, a SIP end point is capable of functioning as both a UAC and a UAS, b ut functions only as one or the other per transaction. Whether the endpoint functions as a UAC or a UAS depends on the UA that initiated the request.

From an architecture s tandpoint, the p hysic al com ponen ts of a SIP n etwork can also be grouped into two categories: clients and servers. Figure 1-1 illustrates the architecture of a SIP ne twork.

What is Session Initiation Protocol?

Note

In addition, the SIP servers can interact with other application services, such as Lightw eght Direc tory Acce ss Protoc ol (LDAP) servers, a database application, o r an extensible marku p lang uage (XML) application. These application services provide back-end services such as directory, authentication, and billing services.

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What is Session Initiation Protocol?

Figure 1-1 SIP Architecture

SIP User

gents (UA)

Chapter1 Product Overview

SIP Proxy and

Redirect Servers

SIP

SIP SIP

SIP Gateway

SIP Client s

PSTN

42870

RTP

Legacy PBX

SIP clients include:

•

Phones—Can act as either a UAS or UAC. Softphones (PCs that have phone capabilities installed) and Cisco SIP IP phones can initiate SIP requests and respond to reques ts .

•

Gateways—Provide call control. Gateways provide many services, the most common being a tra nslation f unction be twee n SIP c onfere ncing e ndpoin ts and other terminal type s. T his func tion include s transla tion be twee n transmission formats a nd be twee n com mu nications pr oced ure s. In addition, the gateway also translates between audio and video codecs and performs call setup and clearing on both the LAN side a nd the switched- circuit ne twork side.

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SIP Server s

What is the Cisco SIP IP Phone 7960?

SIP servers includ e:

•

Proxy server—The proxy server is an intermediate device that receives SIP requests from a client and then forwards the re quests on the client’s behalf. Basically, proxy servers receive SIP messages and for ward th em to the next SIP server in the network. Proxy servers can provide functions 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 addresse s that may be mapp ed to the one in the request, and then returns the results of the address mapping to the client. Basically, redirect servers provide the client with information about the next hop or hops that a message should take and then the clien t contacts the next hop server or UAS directly.

•

Registrar server—Processes requ ests 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 7960?

Cisco SIP IP phones 7960 s (he reafter r eferred to a s C isco SIP IP phon es ) are full-featured telephones that can be plugged directly into an IP network and used very much like a standard private branch exchange (PBX) telephone. The Cisco SIP IP phone is an IP telephony instrument that can be used in VoIP networks.

The Cisco SIP IP phone model terminals can attach to the existing in place data network infrastructure, via 10B aseT /100B ase T interfa ces on an Ethe rnet sw itch. When used with a voice-capable Ethernet switch (one that understa nds Type of Service [ToS] bits and can prioritize VoIP traffic), the phones eliminate the need for a traditional proprietary tele phone set and key system/PBX.

The Cisco SIP IP phone com plies with RFC 25 43.

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What is the Cisco SIP IP Phone 7960?

Figure 1-2 illustrates physical features of the Cisco SIP IP phone:

Figure 1-2 Cisco SIP IP Phone Physical Features

LCD

Chapter1 Product Overview

Line or speed dial

buttons

Footstand adjustment

Soft keys

" button

" On-screen

mode buttons Volume

buttons

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Handset

Dialing

pad

•

LCD screen—Deskto p which displays infor mation about your Cisco SIP IP phone, such as the time, date, your phone number, caller ID, line/call status and the soft key tabs.

•

Line or speed dial buttons—Opens a new line or speed dials the number on the LCD screen.

•

Footstand adjustment—Adjusts the an gle of the p hone ba se.

•

Soft keys—Acti v ates the feature d escribed b y the text me ssage directl y abo ve on the LCD screen.

•

Information (i) button—Pr ovide s online hel p for sele cted ke ys or fea tures and network statistics about the active call. This feature will be available in a future rele ase .

Cisco SIP IP Phone 7960 Administrator Guide

Scroll

key

Function

toggles

38007

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•

On-screen mode buttons—Retrieves information about current settings, recent calls, available services, and voice mail messages.

•

Volume buttons—Adjusts the volume of the handset, headset, speaker, ringer and adjusts the brightness contrast settings on the LCD screen.

•

Function toggles—Includes these optio ns:

–

•

Scroll key—Enables you to move among different soft key options displayed on LCD screen.

•

Dialing pad—Press the dia l pa d buttons to d ial a p hon e num be r. Dial pad buttons work exactly like those on your existing telephone.

•

Handset—Lift the handset and press the dial pad numbers to place a call, review voice mail messages, answer a call, and so on.

Supported Features

What is the Cisco SIP IP Phone 7960?

Headset and speaker—Toggles these functions enabling you to answer

the phone using a headset or speakerphone.

Mute—Stops or resumes voice transmission.

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In addition to the physical features illustrated in Figure 1-2, the Cisco SIP IP ph one a lso provides the following:

•

An adjustable ring tone

•

A hearing-aid compatible handset

•

Headset compatibility

•

An integrated two-port Ethernet switch that allows the telephone and a computer to share a single Ethernet jack

•

A direct connection to a 10Bas eT or 10 0Ba seT Ethe rnet (RJ- 45) n etwor k (half- or full-duplex conn ections a re sup ported )

•

A large (4.25 x 3 in.) display with adjustable contrast

•

G.711 (u-law and a-law) and G.729a audio comp ression

•

IP address assignment—Dynamic Ho st Configuration Protocol (DHCP) client or manually configured via a local setup menu

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What is the Cisco SIP IP Phone 7960?

•

Ability to:

– – – – –

–

•

In-band dual-tone multifreque ncy (DTMF) sup port for touc h-tone dialing

•

Out-of-band DTMF signaling for codecs that do not tran sport the DTMF signaling correctly (for example, G.72 9 or G.729A)

•

Local or remote (using the SI P 183 R i nging m essage ) ca ll p rogre ss tone

•

AVT payload type negotiation

•

Network startup via DHC P a nd Trivial File Transfer Protocol (TFTP)

•

Dial plan support that enables automatic dialing and automatic generation of a secondary dial ton e

•

Current date and time support via Simple Network Time Protocol (SNTP) and time zone and daylight savings time supp ort

Chapter1 Product Overview

Configure Ethernet port mode and speed

Specify a TFTP boot dire ctory

Configure a label for phone ide ntification display p urpose s

Configure a name for caller identification purposes for each active line

on a phone

Configure a 12- or 24-hour user interfac e time disp lay

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•

Call redirection information supp ort via the CC-Diversion head er

•

Third-party call control via d elayed media negotia tion. A de layed m edia negotiation is one where the Session Description Protocol ( SDP) informati on is not completely advertised in the initial call setup.

•

Support for endpoints specified as Fully Qualified Domain Names (FQDNs) in the SDP

•

Local directory configuration (save and rec all) a nd auto matic dial completion—Each time a call i s successfully made or recei v ed, the number is stored in a local dir ector y th at is mainta ined o n the pho ne . The maximu m number of entries is 32. Entries are aged-out based on their usage and age. 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.

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•

What is the Cisco SIP IP Phone 7960?

Message Waiting Indication (via unsolicited NOTIFY)—Lights to indicate that a new voice m essage 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 off. The message waiting indicator is controlled by the voicemail server.

Speed dial to voicemail via the messages button Remote reset support (via the E vent header in NOTIFY messa ges) The following call options:

–

Call forward (network)—A llows the Cisco SIP IP phon e user to re qu est

forwarding service from the network (v ia a third party tool that en ables

this feature to be c onfigured). When a call is placed to the user ’s phone,

it is redirected to the appropriate forward destination by the SIP proxy

server.

–

Call hold—Allows the Cisco SI P IP phone u ser (use r A) to pla ce a ca ll

(from user B) on hold. When user A places user B on hold, the 2-way RTP

voice path between user A and user B is temporarily disconnected but the

call session is still connected. When user A takes user B off hold, the

2-way RTP voice path is reestablished.

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–

Call transfer—Allows the Cisco SIP IP phone user (user A) to transfer a

call from one user (u ser B ) to a nothe r use r (user C) . U ser A p lac es user

B on hold and calls use r C. If user C a ccepts th e transfe r, a session is

established between u ser B and use r C a nd the s ession betw een user A

and user B is terminated.

–

Three-way calling—Allows a “bridged” 3-way call. When a 3-way call

is established, the Cisco SIP IP phone through which the call is

established acts as a bridge, mixing the audio media for the other parties.

–

Do not disturb—Allows the user to instruct the system to intercept

incoming calls during specified periods of time when the user does not

want to be disturbed.

–

Multiple directory numbers—Allows the Cisco SIP IP phone to have up

to six directory numbers or lines.

–

Call waiting—Plays an audible tone to indicate that an incoming call is

waiting. The user can then put the existing call on-hold and accept the

other call. The user can alternat e betwee n the two calls .

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What is the Cisco SIP IP Phone 7960?

–

Chapter1 Product Overview

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 email

address instead of a phone number.

Caller ID blocking—Allows the user to instruct the system to block their

phone number or e mail a ddr ess f rom p hone s th at have caller

identification capabilities.

Anonymous call blocking—Allows the user to ins truct the sy stem to

block any calls for which the identification is blocked.

Note

For information on how to use the standard telephony features and URL dialing, refer to the Getting Started

Cisco IP Phone 7960 a nd Quick Reference Cisco IP Phone 7960 documents that shippe d w ith the p hon e.

Supported Protocols

The Cisco SIP IP phone s upports the fo llowing standard pr otoc ols:

•

Domain Nam e Syst em (D NS) DNS is used in the Internet for translating names of network node s into

addresses. SIP uses DNS to resolve the host names of end points to IP addresses.

•

Dynamic Host Control Protocol (DHCP) DHCP is used to dynamically allocate and assign IP addresses. DHCP allows

you to move network devices from one subnet to another without administrative attention. If using DHCP , you can connect Cisco SIP IP phones to the network and become operational without having to manually assign an IP address and additional network parameters.

The Cisco SIP IP phone complies with the DHCP specifications documented in RFC 2131. By default, Cisco SIP IP phones are DHCP-enabled.

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•

What is the Cisco SIP IP Phone 7960?

Internet Control Message Protocol (ICMP) ICMP is a network layer Interne t protocol that enables hosts to send error or

control messages to other hosts. ICMP also provides other inform ation relevant to IP packet processing.

The Cisco SIP supports I CMP as it is docum en ted in RFC 792 . Internet Protocol (IP) IP is a network layer protocol that sends datagram packets between nodes on

the Internet. IP also provides features for addressing, type-of-service (ToS) specification, fragmentation and reassembly, and security.

The Cisco SIP IP phone s upports IP a s it is de fined in RFC 79 1. Real-Time Transport Protocol (RTP) RTP transports real-time data (such as voice data) over data networks. RTP

also the ability to obtain Quality of Service (QoS) information. The Cisco SIP IP phone s upports RTP as a media channel . Session Description Protoco l (SDP ) SDP is an ASCII-based protocol that describes multimedia sessions and their

related scheduling information.

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The Cisco SIP IP phone us es SD P for se ssion de scr iption.

•

Simple Network Time Protocol (SNTP) SNTP sychronizes computer clocks on an IP network. The Cisco SIP IP

phones use SNTP for their date and time support.

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Prerequisites

•

Prerequisites

For the Cisco SIP IP phone to su cce ssfully ope rate as a SIP e ndpoint in y our network, your network must meet the following requirements:

•

Chapter1 Product Overview

Trivial File Transfer Protocol (TFTP) TFTP allows files to be transferred from one computer to another over a

network. The Cisco SIP IP phone us es T FTP to d ownload configuration files and

software updates. User Datagram Pr otocol (UD P) UDP is a simple protocol that exchanges data packets witho ut

acknowledgments or guaranteed delivery . SIP can use UDP as the underlying transport protocol. If UDP is use d, r etran smissions are u sed to en sure reliability.

The Cisco SIP IP phone s upports UD P as it is d efined in RFC 76 8 for SIP signaling.

A working IP network is established.

1-12

For more informati on abou t configuri ng IP, refer to Cisco IOS IP and IP Routing Configuration Guide.

•

VoI P is con figured on you r Cisco r outers. For more information about configuring VoIP, refer to the Cisco IOS

Release 12.1 Multiservice Applications Configuration Guide for the appropriate access platfor m. For more informa tion about configurin g SIP VoIP, refer to the Enhancements to SIP for VoIP on Cisco Access Platforms.

•

VoI P ga teways are co nfigured fo r SIP.

•

A TFTP server is active and contains the latest Cisco SIP IP phone firmware image in its root directory.

•

A proxy server is active and conf ig ured to r eceive and forw ar d SIP messages.

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