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 PRODUCT ARE SET 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 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 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 without Cisco's written authorization may result in the equipment no longer complying with FCC requirements for Class A or Class B digital devices. In that event, your right to use the equipment may be limited by FCC regulations, and you may be required to correct any 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 interference stops, 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 interference stops. · Move the equipment to one side or the other of the television or radio. · Move the equipment farther away from the television or radio. · Plug the equipment into an outlet that is on a different circuit from the television or radio. (That is, make certain the equipment and the television or radio are on circuits controlled by different circuit breakers or fuses.) Modifications to this product not authorized by Cisco Systems, Inc. could void the FCC approval and negate your authority to operate the product. The Cisco implementation of TCP header compression is an adaptation of a program developed 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 trademarks mentioned in this document or Web site are the property of their respective owners. The use of the word partner does not imply a partnership relationship between Cisco and any other company. (0010R) Cisco SIP IP Phone 7960 Administrator Guide Copyright © 2000, Cisco Systems, Inc. All rights reserved. CONTENTS About This Guide ix Overview ix Who Should Use This Guide ix Objectives x Organization x Related Documentation xi Document Conventions xi Obtaining Documentation xv World Wide Web xv Documentation CD-ROM xv Ordering Documentation xv Obtaining Technical Assistance xv Cisco Connection Online xvi Technical Assistance Center xvi Documentation Feedback xvii 1 CHAPTER Product Overview 1-1 What is Session Initiation Protocol? 1-1 Components of SIP 1-3 SIP Clients 1-4 SIP Servers 1-5 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 iii Contents What is the Cisco SIP IP Phone 7960? 1-5 Supported Features 1-7 Supported Protocols 1-10 Prerequisites 1-12 Cisco SIP IP Phone Connections 1-13 Connecting to the Network 1-13 Connecting to Power 1-14 Using a Headset 1-15 The Cisco SIP IP Phone with a Catalyst Switch 1-16 2 CHAPTER Getting Started with Your Cisco SIP IP Phone 2-1 Initialization Process Overview 2-1 Installing the Cisco SIP IP Phone 2-3 Installation Task Summary 2-3 Downloading Files to Your TFTP Server 2-4 Configuring SIP Parameters 2-5 Configuring SIP Parameters via a TFTP Server 2-6 Manually Configuring the SIP Parameters 2-11 Configuring Network Parameters 2-13 Configuring Network Parameters via a DHCP Server 2-14 Manually Configuring the Network Parameters 2-14 Connecting the Phone 2-16 Adjusting the Placement of the Cisco SIP Phone 2-18 Verifying Startup 2-20 Cisco SIP IP Phone 7960 Administrator Guide iv 78-10497-02 Contents Using the Cisco SIP IP Phone Menu Interface 2-21 Reading the Cisco SIP IP Phone Icons 2-22 Customizing the Cisco SIP IP Phone Ring Types 2-24 Creating Dial Plans 2-24 3 CHAPTER Managing Cisco SIP IP Phones 3-1 Entering Configuration Mode 3-1 Unlocking Configuration Mode 3-2 Locking Configuration Mode 3-2 Modifying the Phone's Network Settings 3-2 Modifying the Phone's SIP Settings 3-5 Modifying SIP Parameters via a TFTP Server 3-8 Modifying the Default SIP Configuration File 3-8 Modifying the Phone-Specific SIP Configuration File 3-15 Modifying the SIP Parameters Manually 3-18 Setting the Date, Time, and Daylight Savings Time 3-22 Erasing the Locally-Defined Settings 3-28 Erasing the Locally-Defined Network Settings 3-28 Erasing the Locally-Defined SIP Settings 3-29 Accessing Status Information 3-30 Viewing Status Messages 3-31 Viewing Network Statistics 3-31 Viewing the Firmware Version 3-33 Upgrading the Cisco SIP IP Phone Firmware 3-33 Performing an Image Upgrade and Remote Reboot 3-35 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 v Contents APPENDIX A SIP Compliance with RFC-2543 Information A-1 SIP Functions A-2 SIP Methods A-2 SIP Responses A-3 1xx Response--Information Responses A-4 2xx Response--Successful Responses A-4 3xx Response--Redirection Responses A-5 4xx Response--Request Failure Responses A-5 5xx Response--Server Failure Responses A-10 6xx Response--Global Responses A-10 SIP Header Fields A-10 SIP Session Description Protocol (SDP) Usage A-12 APPENDIX B SIP Call Flows B-1 Call Flow Scenarios for Successful Calls B-2 Gateway-to Cisco SIP IP Phone--Successful Call Setup and Disconnect B-3 Gateway-to-Cisco SIP IP Phone--Successful Call Setup and Call Hold B-7 Gateway to-Cisco SIP IP Phone--Successful Call Setup and Call Transfer B-11 Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold B-16 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Hold with Consultation B-20 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting B-25 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer without 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) B-41 Cisco SIP IP Phone 7960 Administrator Guide vi 78-10497-02 Contents 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 Cisco SIP IP Phone-to Cisco SIP IP Phone 3-Way Calling B-52 Call Flow Scenarios for Failed Calls B-58 Gateway-to-Cisco SIP IP Phone--Called User is Busy B-58 Gateway-to-Cisco SIP IP Phone--Called User Does Not Answer B-60 Gateway-to-Cisco SIP IP Phone--Client, Server, or Global Error B-63 Cisco SIP IP Phone-to-Cisco SIP IP Phone--Called User is Busy B-66 Cisco SIP IP Phone-to-Cisco SIP IP Phone--Called User Does Not Answer B-68 Cisco SIP IP Phone-to-Cisco SIP IP Phone--Authentication Error B-70 C APPENDIX Technical Specifications C-1 Physical and Operating Environment Specifications C-1 Cable Specifications C-3 Connections Specifications C-3 APPENDIX D Translated Safety Warnings D-1 Installation Warning D-1 Product Disposal Warning D-2 Lightning Activity Warning D-3 SELV Circuit Warning (other versions available) D-4 Circuit Breaker (15A) Warning D-6 Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 vii Contents GLOSSARY INDEX Cisco SIP IP Phone 7960 Administrator Guide viii 78-10497-02 About This Guide Overview The Cisco Session Initiation Protocol (SIP) IP Phone 7960 Administrator Guide provides information about how to setup, connect cables to, and configure a Cisco SIP IP phone 7960 (hereafter referred to as a Cisco SIP IP phone). The administrator guide 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 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 telecommunication engineers should use this guide to learn the steps required to properly set up the Cisco SIP IP phone on the network. 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 network and require an understanding of IP networking and telephony concepts. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 ix About This Guide Objectives Objectives The Cisco SIP IP Phone 7960 Administrator 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 chapters and appendixes: · · · 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 configure the Cisco SIP IP phone. Chapter 3, "Managing Cisco SIP IP Phones" 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-2543 Information" provides reference information about the SIP IP phone compliance to RFC 2543. Appendix B, "SIP Call Flows" provides reference information about the SIP IP phone call flows. Appendix C, "Technical Specifications" lists the physical and operating 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. · · · · Cisco SIP IP Phone 7960 Administrator Guide x 78-10497-02 About This Guide Related Documentation Related Documentation The following is a list of related Cisco SIP VoIP publications. For more information about implementing a SIP VoIP network refer to the following publications: · · · · Session Initiation Protocol Gateway Call Flows Session Initiation for VoIP on Cisco Access Platforms Getting Started with the Cisco IP Phone 7960 Installing the Wall Mount Kit for the Cisco IP Phone The following is a list of Cisco VoIP publications that provide information about implementing a VoIP network: · · · · · Service Provider Features for Voice over IP (introduced in Cisco IOS Release 12.0(3)T) Cisco IOS IP and IP Routing Configuration Guide Cisco IOS Release 12.1 Multiservice Applications Configuration Guide Voice over IP for the Cisco 2600 and Cisco 3600 Series Routers Voice over IP for the Cisco AS5300 Documents Document Conventions This document uses the following conventions: · · · · · · · Commands and keywords are in boldface font. Arguments for which you supply values are in italic font. Elements in square brackets ([ ]) are optional. 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 information the system displays are in Information you must enter is in boldface screen screen font. font. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 xi About This Guide Document Conventions Notes use the following conventions: Note Means reader take note. Notes contain helpful suggestions or references to material not covered in the publication. Cautions use the following conventions: Caution Means reader be careful. In this situation, you might do something that could result in equipment damage or loss of data. Warnings use the following conventions: Warning This warning symbol means danger. You are in a situation that could cause bodily injury. Before you 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.) Waarschuwing Cisco SIP IP Phone 7960 Administrator Guide xii 78-10497-02 About This Guide Document Conventions Varoitus 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. 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). Attention Warnung Avvertenza Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 xiii About This Guide Document Conventions Advarsel 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].) Aviso Advertencia Varning! Cisco SIP IP Phone 7960 Administrator Guide xiv 78-10497-02 About This Guide Obtaining Documentation 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. Ordering Documentation Registered CCO users can order the Documentation CD-ROM and other Cisco Product documentation through our online Subscription Services at http://www.cisco.com/cgi-bin/subcat/kaojump.cgi. Nonregistered CCO users can order documentation through a local account representative by calling Cisco's corporate headquarters (California, USA) at 408 526-4000 or, in North America, call 800 553-NETS (6387). Obtaining Technical Assistance Cisco provides Cisco Connection Online (CCO) as a starting point 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-addressed stamped response card included in many printed docs, or by sending mail to Cisco. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 xv About This Guide 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. This highly integrated Internet application is a powerful, easy-to-use tool for doing business with Cisco. CCO's broad range of features and services helps customers and partners to streamline business processes and improve productivity. Through CCO, you will find information about Cisco and our networking solutions, services, and programs. In addition, you can resolve technical issues with 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 services. Registered users may order products, check on the status of an order and view benefits specific to their relationships with Cisco. You can access CCO in the following ways: · · · WWW: www.cisco.com Telnet: cco.cisco.com Modem using standard connection rates and the following terminal settings: VT100 emulation; 8 data bits; no parity; and 1 stop bit. ­ From North America, call 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 Center (TAC) is available to warranty or maintenance contract customers who need technical assistance with a Cisco product that is under warranty or covered by a maintenance contract. Cisco SIP IP Phone 7960 Administrator Guide xvi 78-10497-02 About This Guide Obtaining Technical Assistance To display the TAC web site that includes links to technical support information and software upgrades and for requesting TAC support, use www.cisco.com/techsupport. To contact by e-mail, use one of the following: Language English Hanzi (Chinese) Kanji (Japanese) Hangul (Korean) Spanish Thai E-mail Address tac@cisco.com chinese-tac@cisco.com japan-tac@cisco.com korea-tac@cisco.com tac@cisco.com thai-tac@cisco.com In North America, TAC can be reached at 800 553-2447 or 408 526-7209. For other telephone numbers and TAC e-mail addresses worldwide, consult 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 Wide 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@cisco.com. To 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 Connection 170 West Tasman Drive San Jose, CA 95134-9883 We appreciate and value your comments. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 xvii About This Guide Obtaining Technical Assistance Cisco SIP IP Phone 7960 Administrator Guide xviii 78-10497-02 CHAPTER 1 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 Phone 7960?, page 1-5 Prerequisites, page 1-12 Cisco SIP IP Phone Connections, 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 (defined in RFC 2543) that can be used to establish, maintain, and terminate calls between two or more end points. Like other VoIP protocols, SIP is designed to address the functions of signaling and session management within a packet telephony network. Signaling allows call information to be carried across network boundaries. Session management provides the ability to control the attributes of an end-to-end call. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 1-1 Chapter 1 What is Session Initiation Protocol? Product Overview 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. Conferences are established 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 already on the phone or did not answer 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 or the changing of a media characteristic or codec. Handle the transfer and termination of calls--SIP supports the transfer of calls from one end point to another. During a call transfer, 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. · · · Conferences can consist of two or more users and can be established using multicast or multiple unicast sessions. Note The term conference means an established session (or call) between two or more end points. In this document, the terms conference and call are used interchangeably. Cisco SIP IP Phone 7960 Administrator Guide 1-2 78-10497-02 Chapter 1 Product Overview What is Session Initiation Protocol? 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 contacts the user when a SIP request is received and that returns a response on behalf of the user. Typically, a SIP end point 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 UAC or a UAS depends on the UA that initiated the request. From an architecture standpoint, the physical components of a SIP network can also be grouped into two categories: clients and servers. Figure 1-1 illustrates the architecture of a SIP network. Note In addition, the SIP servers can interact with other application services, such as Lightweght Directory Access Protocol (LDAP) servers, a database application, or an extensible markup language (XML) application. These application services provide back-end services such as directory, authentication, and billing services. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 1-3 Chapter 1 What is Session Initiation Protocol? Product Overview Figure 1-1 SIP Architecture SIP Proxy and Redirect Servers SIP SIP SIP User Agents (UA) SIP SIP Gateway RTP PSTN IP Legacy PBX SIP Clients 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 requests. Gateways--Provide call control. Gateways provide many services, the most common being a translation function between SIP conferencing endpoints and other terminal types. This function includes translation between transmission formats and between communications procedures. In addition, the gateway also translates between audio and video codecs and performs call setup and clearing on both the LAN side and the switched-circuit network side. · Cisco SIP IP Phone 7960 Administrator Guide 1-4 42870 78-10497-02 Chapter 1 Product Overview What is the Cisco SIP IP Phone 7960? SIP Servers SIP servers include: · Proxy server--The proxy server is an intermediate device that receives SIP requests from a client and then forwards the requests on the client's behalf. Basically, proxy servers receive SIP messages and forward them 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 addresses that may be mapped 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 client contacts the next hop server or UAS directly. Registrar server--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 7960? Cisco SIP IP phones 7960s (hereafter referred to as Cisco SIP IP phones) 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 10BaseT/100BaseT interfaces on an Ethernet switch. When used with a voice-capable Ethernet 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/PBX. The Cisco SIP IP phone complies with RFC 2543. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 1-5 Chapter 1 What is the Cisco SIP IP Phone 7960? Product Overview Figure 1-2 illustrates physical features of the Cisco SIP IP phone: Figure 1-2 Cisco SIP IP Phone Physical Features LCD Line or speed dial buttons Footstand adjustment Soft keys "i " button On-screen mode buttons Volume buttons Handset 38007 Dialing pad · Scroll key Function toggles LCD screen--Desktop which displays information 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 angle of the phone base. Soft keys--Activates the feature described by the text message directly above on the LCD screen. Information (i) button--Provides online help for selected keys or features and network statistics about the active call. This feature will be available in a future release. · · · · Cisco SIP IP Phone 7960 Administrator Guide 1-6 78-10497-02 Chapter 1 Product Overview What is the Cisco SIP IP Phone 7960? · · · 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 options: ­ Headset and speaker--Toggles these functions enabling you to answer the phone using 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 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 In addition to the physical features illustrated in Figure 1-2, the Cisco SIP IP phone also 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 10BaseT or 100BaseT Ethernet (RJ-45) network (half- or full-duplex connections are supported) A large (4.25 x 3 in.) display with adjustable contrast G.711 (u-law and a-law) and G.729a audio compression IP address assignment--Dynamic Host Configuration Protocol (DHCP) client or manually configured via a local setup menu Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 1-7 Chapter 1 What is the Cisco SIP IP Phone 7960? Product Overview · Ability to: ­ Configure Ethernet port mode and speed ­ Register with or unregister from a proxy server ­ Specify a TFTP boot directory ­ Configure a label for phone identification display purposes ­ Configure a name for caller identification purposes for each active line on a phone ­ Configure a 12- or 24-hour user interface time display · · · · · · · · · In-band dual-tone multifrequency (DTMF) support for touch-tone dialing Out-of-band DTMF signaling for codecs that do not transport the DTMF signaling correctly (for example, G.729 or G.729A) Local or remote (using the SIP 183 Ringing message) call progress tone AVT payload type negotiation Network startup via DHCP and Trivial File Transfer Protocol (TFTP) Dial plan support that enables automatic dialing and automatic generation of a secondary dial tone Current date and time support via Simple Network Time Protocol (SNTP) and time zone and daylight savings time support Call redirection information support via the CC-Diversion header Third-party call control via delayed media negotiation. A delayed media negotiation is one where the Session Description Protocol (SDP) information 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 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 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. · · Cisco SIP IP Phone 7960 Administrator Guide 1-8 78-10497-02 Chapter 1 Product Overview What is the Cisco SIP IP Phone 7960? · Message Waiting Indication (via unsolicited NOTIFY)--Lights to indicate that a new voice 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 off. The message waiting indicator is controlled by the voicemail server. Speed dial to voicemail via the messages button Remote reset support (via the Event header in NOTIFY messages) The following call options: ­ Call forward (network)--Allows the Cisco SIP IP phone user to request · · · forwarding service from the network (via a third party tool that enables this feature to be configured). 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 SIP IP phone user (user A) to place a call (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. ­ Call transfer--Allows the Cisco SIP IP phone user (user A) to transfer a call from one user (user B) to another user (user C). User A places user B on hold and calls user C. If user C accepts the transfer, a session is established between user B and user C and the session between 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 alternate between the two calls. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 1-9 Chapter 1 What is the Cisco SIP IP Phone 7960? 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 email address from phones that have caller identification capabilities. ­ Anonymous call blocking--Allows the user to instruct the system 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 and Quick Reference Cisco IP Phone 7960 documents that shipped with the phone. Supported Protocols The Cisco SIP IP phone supports the following standard protocols: · Domain Name System (DNS) DNS is used in the Internet for translating names of network nodes 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. Cisco SIP IP Phone 7960 Administrator Guide 1-10 78-10497-02 Chapter 1 Product Overview What is the Cisco SIP IP Phone 7960? · Internet Control Message Protocol (ICMP) ICMP is a network layer Internet protocol that enables hosts to send error or control messages to other hosts. ICMP also provides other information relevant to IP packet processing. The Cisco SIP supports ICMP as it is documented 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 supports IP as it is defined in RFC 791. · 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 supports RTP as a media channel. · Session Description Protocol (SDP) SDP is an ASCII-based protocol that describes multimedia sessions and their related scheduling information. The Cisco SIP IP phone uses SDP for session description. · 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. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 1-11 Chapter 1 Prerequisites 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 uses TFTP to download configuration files and software updates. · User Datagram Protocol (UDP) UDP is a simple protocol that exchanges data packets without acknowledgments or guaranteed delivery. SIP can use UDP as the underlying transport protocol. If UDP is used, retransmissions are used to ensure reliability. The Cisco SIP IP phone supports UDP as it is defined in RFC 768 for SIP signaling. 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. For more information about configuring IP, refer to Cisco IOS IP and IP Routing Configuration Guide. · VoIP is configured on your Cisco routers. For more information about configuring VoIP, refer to the Cisco IOS Release 12.1 Multiservice Applications Configuration Guide for the appropriate access platform. For more information about configuring SIP VoIP, refer to the Enhancements to SIP for VoIP on Cisco Access Platforms. · · · VoIP gateways are configured for 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 configured to receive and forward SIP messages. Cisco SIP IP Phone 7960 Administrator Guide 1-12 78-10497-02 Chapter 1 Product Overview Cisco SIP IP Phone Connections Cisco SIP IP Phone Connections The Cisco SIP IP phone has connections for connecting 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 7960 (rear view) Power outlet AC adapter port (DC48V) (optional power cable) RJ-11 port Headset port Handset port Network port (10/100 SW) Access port (10/100 PC) 38006 Connecting to the Network The Cisco SIP IP phone has two RJ-45 ports that each support 10/100 Mbps halfor full-duplex Ethernet connections to external devices--network port (labeled 10/100 SW) and access port (labeled 10/100 PC). You can use either Category 3 or 5 cabling for 10 Mpbs connections, but use Category 5 for 100 Mbps connections. On both the network port and access port, use full-duplex mode to avoid collisions. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 1-13 Chapter 1 Cisco SIP IP Phone Connections Product Overview 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 Cisco Catalyst switch over this connection. See the "Connecting to Power" section on page 1-14 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 on this port. Connecting to Power The Cisco SIP IP phone can be powered by the following sources: · · External power source--Optional Cisco AC adaptor and power cord for connecting to a standard wall receptacle. WS-X6348-RJ45V 10/100 switching module--Provides inline power to the Cisco SIP IP phone when connected to a Catalyst 3500, 4000, or 6000 family 10/100BaseTX switching module. This module sends power on pins 1 & 2 and 3 & 6. · WS-PWR-PANEL--Power patch panel provides power to the Cisco SIP IP phone which allows the Cisco SIP IP phone to be connected to existing Catalyst 4000, 5000, and 6000 family 10/100BaseTX switching modules. This module sends power on pins 4, 5, 7, and 8. · · · WS-X4148-RJ45V--48 port 10/100 Ethernet with inline power module for the Catalyst 4006. WS-X4095-PEM--VoIP DC Power Entry module for the Catalyst 4006. WS-X4608-2PSU and WS-X4608--External -48V DC power shelf common equipment for the Catalyst 4006 with two AC-to-DC PSUs and one empty bay for redundant option and the 110V 15A AC-to48V DC PSU redundant option for the power shelf WS-C3524-PWR-XL-EN--Catalyst 3524-PWR XL switch · Cisco SIP IP Phone 7960 Administrator Guide 1-14 78-10497-02 Chapter 1 Product Overview Cisco SIP IP Phone Connections Note Only the network port (labeled 10/100 SW) supports inline power from the Cisco Catalyst switches. For redundancy, you can use the Cisco AC adapter even if you are using inline power from the Cisco 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. To use this redundancy feature you must set the inline power mode to auto on the Cisco Catalyst switch. Next, connect the un-powered 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 phone supports a four or six-wire headset jack. Specifically, the Cisco SIP IP phone supports the following Plantronics headset models: · · · Tristar Monaural Encore Monaural H91 Encore Binaural H101 The Volume and Mute controls will 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 Phone 7960. For information on ordering compatible headsets and coil cords for the Cisco IP phone 7960, see http://cisco.getheadsets.com. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 1-15 Chapter 1 The Cisco SIP IP Phone with a Catalyst Switch Product Overview The Cisco SIP IP Phone with a Catalyst Switch To function in the IP telephony network, the Cisco SIP IP phone must be connected 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 the 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 the same physical link to the switch and the same port on the switch. This configuration has these implications for the VLAN configuration on the network: · The current VLANs might be configured on an IP subnet basis, and additional IP addresses might not be available to assign the phone to a port so that it belongs to the same subnet as other devices (PC) connected to the same port. Data traffic present on the VLAN supporting phones might reduce the quality 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 phone. The switch port configured for connecting a phone would have separate VLANs configured for carrying: · · Voice traffic to and from the Cisco SIP IP phone (auxiliary 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 network where there are not enough IP addresses. For more information, refer to the documentation included with the Cisco Catalyst switch. Cisco SIP IP Phone 7960 Administrator Guide 1-16 78-10497-02 CHAPTER 2 Getting Started with Your Cisco SIP IP Phone This chapter explains the Cisco SIP IP phone initialization and the process that you should follow to install and connect the Cisco SIP IP phone. This chapter provides the following major sections: · · · · · · · Initialization Process Overview, page 2-1 Installing the Cisco SIP IP Phone, page 2-3 Verifying Startup, page 2-20 Using the Cisco SIP IP Phone Menu Interface, page 2-21 Reading the Cisco SIP IP Phone Icons, page 2-22 Customizing the Cisco SIP IP Phone Ring Types, page 2-24 Creating Dial Plans, page 2-24 Initialization Process Overview The initialization process of the Cisco SIP IP phone is responsible for establishing network connectivity and for making the phone operational in your IP network. Once you connect your phone to the network and to an electrical supply, the phone begins its initialization process. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 2-1 Chapter 2 Initialization Process Overview Getting Started with Your Cisco SIP IP Phone During the initialization process, the following events take place: 1. The stored image is loaded. The Cisco SIP IP phone has non-volatile Flash memory in which it stores the firmware images, user-defined preferences, and permanent factory information about the phone. During initialization, the phone runs a bootstrap loader that loads and executes the phone image stored in Flash memory. 2. The VLAN is configured. If the Cisco SIP IP phone is connected to a Catalyst switch, the switch notifies the phone of the voice VLAN defined on the switch. The phone needs to know its VLAN membership before it can proceed with the DHCP request for its IP settings (if using DHCP). 3. An IP address is acquired. If the Cisco SIP IP phone is using DHCP to obtain the IP settings, the phone queries the DHCP server. If the phone is not using DHCP, then the phone will use IP settings that are stored in Flash memory. 4. The TFTP server is contacted. On the TFTP server is the latest Cisco SIP IP phone firmware image and the dual boot file (OS79XX.TXT) that enables the phone to automatically determine and initialize for the VoIP environment in which it is being installed. If the phone is using the TFTP server to obtain its SIP parameters, there should also be a configuration file or files on the TFTP server that the phone will request and download. In the configuration file or files, SIP parameters that are required by the phone to operate in a SIP VoIP environment are defined. If the phone is not obtaining its SIP parameters via the TFTP server, the phone will use SIP settings that are stored in Flash memory. 5. The firmware version is verified. If the phone is obtaining its SIP parameters via a TFTP server, the configuration files are requested. If the phone determines that the image defined in a configuration file differs from the image it has stored in Flash memory, it performs a firmware upgrade. When performing a firmware upgrade, the phone downloads the firmware image from the TFTP server, programs the image into Flash memory, and reboots. Cisco SIP IP Phone 7960 Administrator Guide 2-2 78-10497-02 Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Installing the Cisco SIP IP Phone This section contains information on how to install Cisco SIP IP phones in your IP network. Before getting started, read over the information in this section carefully. Installation Task Summary To successfully install the Cisco SIP IP phone, you must complete the following tasks: 1. 2. Download the required files from CCO to the TFTP server as described in the the "Downloading Files to Your TFTP Server" section on page 2-4. If you are configuring SIP parameters via a TFTP server, create and store the configuration files as described in the "Configuring SIP Parameters via a TFTP Server" section on page 2-6. If you are using DCHP to configure the phones' network settings, configure the required network parameters on your DHCP server as described in the "Configuring Network Parameters via a DHCP Server" section on page 2-14. Connect the phone to the network and to a power supply as described in the "Connecting the Phone" section on page 2-16. If you are not using DHCP to configure network parameters, manually configure the required network parameters as described in the "Manually Configuring the Network Parameters" section on page 2-14. If you are not configuring the SIP parameters via a TFTP server, manually configure the required parameters as described in the "Manually Configuring the SIP Parameters" section on page 2-11. 3. 4. 5. 6. Cisco SIP IP Phone 7960 Administrator Guide 78-10497-02 2-3 Chapter 2 Installing the Cisco SIP IP Phone Getting Started with Your Cisco SIP IP Phone Downloading Files to Your TFTP Server Before installing the Cisco SIP IP phones, copy the following files from CCO to the root directory of your TFTP server. File OS79XX.TXT Description (Required) Enables the phone to automatically determine and initialize for the VoIP environment in which it is being installed. After downloading this file, you will need to use an ASCII editor to open it and specify the file name (without the file extension) of the image version that you plan to run on your phones. SIPDefaultGeneric.cnf (Optional) File in which to configure SIP parameters intended for all phones. For more information on using the SIPDefault.cnf file, see the "Creating the Default SIP Configuration File" section on page 2-7. SIPConfigGeneric.cnf (Required) File which can be used as a template to configure SIP parameters specific to a phone. When customized for a phone, this file must be renamed to the MAC address of the phone. (Optional) Lists audio files that are the custom ring type options for the phones. The audio files listed in the RINGLIST.DAT file must also be in the root directory of the TFTP server. For more information on custom ring types, see the "Customizing the Cisco SIP IP Phone Ring Types" section on page 2-24. (Required) The Cisco SIP IP phone firmware image. P0S3xxyy.bin (where xx is the version number and yy is the subversion number) dialplan.xml syncinfo.xml (Optional) North American example dial plan. (Optional) Controls the image version and associated sync value to be used for remote reboots. RINGLIST.DAT Cisco SIP IP Phone 7960 Administrator Guide 2-4 78-10497-02 Chapter 2