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.
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.
C O N T E N T S 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 CHAPTER 1 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
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 CHAPTER 2 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 Parame
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 CHAPTER 3 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 P
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 Gate
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
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.
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.
About This Guide Related Documentation Related Documentation The following is a list of related Cisco SIP VoIP publications.
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.
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).) Attention Ce symbole d’avertissement indique un danger.
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].
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.
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.
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 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.
About This Guide Obtaining Technical Assistance Cisco SIP IP Phone 7960 Administrator Guide xviii 78-10497-02
C H A P T E R 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 ove
Chapter 1 Product Overview 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. Conferences are established using only the media capabilities that can be supported by all end points.
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.
Chapter 1 Product Overview What is Session Initiation Protocol? Figure 1-1 SIP Architecture SIP Proxy and Redirect Servers SIP SIP SIP SIP User Agents (UA) SIP Gateway RTP PSTN Legacy PBX 42870 IP 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.
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.
Chapter 1 Product Overview 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 Line or speed dial buttons Footstand adjustment Soft keys "i " button On-screen mode buttons Volume buttons Dialing pad Scroll key Function toggles 38007 Handset • 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
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.
Chapter 1 Product Overview What is the Cisco SIP IP Phone 7960? • 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 signal
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.
Chapter 1 Product Overview What is the Cisco SIP IP Phone 7960? – 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.
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.
Chapter 1 Product Overview Prerequisites • 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.
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.
Chapter 1 Product Overview Cisco SIP IP Phone Connections 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.
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.
Chapter 1 Product Overview The Cisco SIP IP Phone with a Catalyst Switch 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.
C H A P T E R 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.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Initialization Process Overview 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.
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.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the 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 Description OS79XX.TXT (Required) Enables the phone to automatically determine and initialize for the VoIP environment in which it is being installed.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Configuring SIP Parameters Note This section describes how to configure the basic SIP parameters that are required for the phone to operate in a SIP VoIP environment. For a complete list of the SIP parameters that you can configure, see the “Modifying the Phone’s SIP Settings” section on page 3-5. The SIP parameters are those parameters that a Cisco SIP IP phone needs to operate in a SIP VoIP environment.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Configuring SIP Parameters via a TFTP Server If you are configuring SIP parameters via a TFTP server, you must use configuration files. There are two configuration files that you can use to define the SIP parameters; the default configuration file (optional) and the phone-specific configuration file (required).
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone • The default configuration file must be stored in the root directory of the TFTP server. The phone-specific configuration file can be stored in the root directory or in a subdirectory in which all phone-specific configuration files are located. • Each line in the configuration files must use the following format: variable-name : value ; optional comments • Use colons to separate variable names and values.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone By maintaining these parameters in the default configuration file, you can perform global changes, such as upgrading the image version, without having to modify the phone-specific configuration file for each phone. Before You Begin • Ensure that you have downloaded the SIPDefault.cnf file from CCO to the root directory of your TFTP server.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone The following is an example of a SIP default configuration file: ; sip default configuration file #Image Version image_version:P0S3xxyy ; #Proxy server address proxy1_address: 192.168.1.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Procedure Step 1 Step 2 Using an ASCII editor, create a phone-specific configuration file for each phone that you plan to install. In the phone-specific configuration file, define values for the following SIP parameters (where x is a number 1 through 6): • linex_name—(Required) Number or e-mail address used when registering. When entering a number, enter the number without any dashes.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Manually Configuring the SIP Parameters If you did not configure the SIP parameters via a TFTP server, you must manually configure them after you have connected the phone as described in the “Connecting the Phone” section on page 2-16. Before You Begin • Connect your phone as described in the “Connecting the Phone” section on page 2-16.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Step 5 Highlight and press the Select soft key to configure the following parameters: • Name—(Required) Number or e-mail address used when registering. When entering a number, enter the number without any dashes. For example, enter 555-1212 as 5551212. When entering an e-mail address, enter the e-mail ID without the host name.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Configuring Network Parameters Note This section describes how to configure the basic network parameters that are required for the phone to operate on the network. For a complete list of the network parameters that you can configure, see the “Modifying the Phone’s Network Settings” section on page 3-2.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Note By default, DHCP is enabled on your phone. Before you can manually configure the network parameters, you must disable DHCP after connecting your phone to a power supply.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone • When configuring a domain name: – Press the Number soft key if entering a numerical ID or press the Alpha soft key to enter a name. – If entering letters, use the numbers on the dial pad associated with a particular letter. For example, the 2 key has the letters A, B, and C. For a lower case “a”, press the 2 key once. To scroll through the available letters and numbers, press the key repeatedly.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Step 7 Caution • Domain Name—Name of the DNS domain in which the phone resides. • DNS Servers 1 through 5—IP address of the DNS server used by the phone to result names to IP addresses. The phone will attempt to use DNS Servers 2 through 5 if DNS Server 1 is unavailable. When done, press the Save soft key. The phone programs the new information into Flash memory and resets.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Figure 2-1 Cisco SIP IP Phone Cable Connections Cisco IP Phone 7960 (rear view) Power outlet AC adapter port (DC48V) Headset port (optional power cable) RJ-11 port Network port (10/100 SW) Access port (10/100 PC) 38006 Handset port Procedure Step 1 Connect a Category 3 or 5 straight-through Ethernet cable from the switch or hub to the network port on the phone.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Step 3 Connect a Category 3 or 5 straight-through Ethernet cable from another network device, such as a desktop computer, to the access port on the phone (optional). See “Connecting to the Network” section on page 1-13 for more information on the access port. Step 4 Connect the power plug to the Cisco AC Adapter port (optional). See “Connecting to Power” section on page 1-14 for more information.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Installing the Cisco SIP IP Phone Before You Begin • Mounting the Cisco SIP IP phone on the wall requires some tools and equipment that are not provided as standard equipment. Following are the tools and parts required for a typical Cisco SIP IP phone installation: – Screwdriver – Screws to secure the Cisco SIP IP phone to the wall • Refer to Figure 2-1 for a graphical overview of these procedures.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Verifying Startup Figure 2-2 Adjusting the Footstand Cisco IP Phone 7960 (rear view) Adjustment plate raises and lowers phone vertically 38036 Footstand adjustment Adjustment plate installation button raises and screws holes (2) lowers adjustment plate Verifying Startup After the phone has power connected to it, the phone begins its startup process by cycling through these steps: 1.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Using the Cisco SIP IP Phone Menu Interface 3. These messages display as phone starts up: – Configuring VLAN—The phone is configuring the Ethernet connection. – Configuring IP—The phone is contacting the DHCP server to obtain network parameters and the IP address of the TFTP server. – Requesting Configuration—The phone is contacting the TFTP server to request its configuration files and compare firmware images.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Reading the Cisco SIP IP Phone Icons – If entering letters, use the numbers on the dial pad associated with a particular letter. For example, the 2 key has the letters A, B, and C. For a lower case “a”, press the 2 key once. To scroll through the available letters and numbers, press the key repeatedly. – Press the << soft key to delete any mistakes.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Reading the Cisco SIP IP Phone Icons Table 1 Icon Cisco SIP IP Phone User Interface Icon Meanings (continued) Meaning The line is configured for E.164 number dialing and ready for you to place the call. When a line is configured for E.164 number dialing, you can enter only numbers when placing the call. You can change to URL dialing at any time while dialing on a line by pressing the more soft key and then the URL soft key.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Customizing the Cisco SIP IP Phone Ring Types Customizing the Cisco SIP IP Phone Ring Types The Cisco SIP IP phone ships with two ring types: Chirp1 and Chirp2. By default, your ring type options will be those two choices. However, using the RINGLIST.DAT file, you can customize the ring types that are available to the Cisco SIP IP phone users.
Chapter 2 Getting Started with Your Cisco SIP IP Phone Creating Dial Plans Note We recommend that you define the dial_template parameter in the default configuration file for maintenance and control purposes. Specify the dial_template parameter in a phone-specific configuration file only if that phone needs to use a different dial plan than is being used by the other phones in the same system. When creating a dial plan, remember the following: • Dial plans must be in an .
Chapter 2 Getting Started with Your Cisco SIP IP Phone Creating Dial Plans • User=”type” is the either IP or Phone. Enter User=phone or User=IP to have the tag automatically added to the dialed number. • Rewrite=”altstrng” is the alternate string to be dialed instead of what the user enters. Step 4 If desired, specify at the end of each string where comment defines the type of plan (for example, Long Distance or Corporate Dial Plan).
C H A P T E R 3 Managing Cisco SIP IP Phones This chapter provides information on the following: • Entering Configuration Mode, page 3-1 • Modifying the Phone’s Network Settings, page 3-2 • Modifying the Phone’s SIP Settings, page 3-5 • Setting the Date, Time, and Daylight Savings Time, page 3-22 • Erasing the Locally-Defined Settings, page 3-28 • Accessing Status Information, page 3-30 • Upgrading the Cisco SIP IP Phone Firmware, page 3-33 Entering Configuration Mode When you access the net
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s Network Settings Unlocking Configuration Mode To unlock the Cisco SIP IP phone, press **#. Note You have activated the configuration mode for your phone. There is no indication an action has taken place. If the Network Configuration or SIP Configuration panel is displayed, the lock icon in the upper right corner of your LCD will change to an unlocked state.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s Network Settings Before You Begin When configuring network settings, remember the following: • Unlock configuration mode as described in the “Unlocking Configuration Mode” section on page 3-2. By default, the network parameters are locked to ensure that end-users cannot modify settings that might affect their network connectivity.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s Network Settings • TFTP Server—IP address of the TFTP server from which the phone downloads its configuration files and firmware images. To edit this field, DHCP must be disabled. • Default Routers 1 through 5—IP address of the default gateway used by the phone. Default Routers 2 through 5 are the IP addresses of the gateways that the phone will attempt to use as an alternate gateway if the primary gateway is NA.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings network before the phone attempts to acquire another IP address from the DHCP server. When moving the phone to a new network segment, you should first release the DHCP address. Step 4 Caution • Alternate TFTP—Whether to use an alternate TFTP server. This field enables an administrator to specify the remote TFTP server rather than the local one. Possible values for this parameter are Yes and No. The default is No.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings Table 3-1 lists each of the SIP parameters that you can configure. In the Configuration column, the name of a parameter as you would specify it in a configuration file is listed. In the menu column (SIP Configuration, Network Configuration, and Services), the name of the same parameter as it would appear on the user interface is listed. If NA appears for a parameter name in a menu column, it can cannot be defined via that menu.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings Table 3-1 SIP Parameters Summary (continued) Configuration File SIP Configuration Menu Network Configuration Menu Services Menu dtmf_outofband Out of Band DTMF NA NA image_version NA NA NA linex_authname (line1 to line6) Authentication Name NA NA linex_displayname (line1 to line6) Display Name NA NA linex_name (line1 to line6) Name NA NA linex_password (line1 to line6) Authentication Password NA NA linex
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings Table 3-1 SIP Parameters Summary (continued) Configuration File SIP Configuration Menu Network Configuration Menu Services Menu timer_register_expires Register Expires NA NA timer_t1 NA NA NA timer_t2 NA NA NA tos_media NA NA NA Modifying SIP Parameters via a TFTP Server If you have set up your phones to retrieve their SIP parameters via a TFTP server as described in the “Configuring SIP Parameters via a TFTP S
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings Before You Begin • Ensure that you have downloaded the SIPDefault.cnf file from CCO to the root directory of your TFTP server. • Review the guidelines and restrictions documented in the “Configuration File Guidelines” section on page 2-6. Procedure Step 1 Using an ASCII editor, open the SIPDefault.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings • dtmf_inband—(Optional) Whether to detect and generate in-band signaling format. Valid values are 1 (generate DTMF digits in-band) and 0 (do not generate DTMF digits in-band). The default is 1. • dtmf_db_level—(Optional) In-band DTMF digit tone level. Valid values are: – 1 (6 db below nominal) – 2 (3 db below nominal) – 3 (nominal) – 4 (3 db above nominal) – 5 (6 db above nominal) The default is 3.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings • sip_retx—(Optional) Maximum number of times a SIP message other than an INVITE request will be retransmitted. The valid value is any positive integer. The default is 10. • sip_invite_retx—(Optional) Maximum number of times an INVITE request will be retransmitted. The valid value is any positive integer. The default is 6. • proxy_register—(Optional) Whether the phone must register with a proxy server during initialization.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings – dst_offset—(Optional) Offset from the phone’s time when DST is in effect. – dst_start_month—(Optional) Month in which DST starts. – dst_start_day—(Optional) Day of the month on which DST begins. – dst_start_day_of_week—(Optional) Day of the week on which DST begins. – dst_start_week_of_month—(Optional) Week of month in which DST begins. – dst_start_time—(Optional) Time of day on which DST begins.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings – 2—The Do Not Disturb feature is off permanently and cannot be turned on and off locally via the phone’s user interface. If specifying this value, specify this parameter in the phone-specific configuration file. – 3—The Do Not Disturb feature is on permanently and cannot be turned on and off locally via the phone’s user interface. This setting sets the phone to be a “call out” phone only.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings – 2—The Anonymous Call Blocking feature is disabled permanently and cannot be turned on and off locally via the phone’s user interface. If specifying this value, specify this parameter in the phone-specific configuration file. – 3—The Anonymous Call Blocking feature is enabled permanently and cannot be turned on and off locally via the phone’s user interface.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings Step 2 Save the file with the same file name, SIPDefault.cnf, to the root directory of your TFTP server. The following is an example of a SIP default configuration file: ; sip default configuration file #Image Version image_version:P0S3xxyy ; #Default Codec preferred_codec :g711ulaw #Enable Registration proxy_register :1 ; #Registration expiration timer_register_expires :3600 ; #Proxy address proxy1_address: 192.168.1.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings Procedure Step 1 Using an ASCII editor, create a phone-specific configuration file for each phone that you plan to install. In the phone-specific configuration file, define values for the following SIP parameters (where x a number 1 through 6): • linex_name—(Required) Number or e-mail address used when registering. When entering a number, enter the number without any dashes. For example, enter 555-1212 as 5551212.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings • linex_displayname—(Optional) Identification as it should appear for caller identification purposes. For example, instead of jdoe@company.com displaying on phones that have caller ID, you can specify John Doe in this parameter to have John Doe display on the callee end instead. If a value is not specified for this parameter, nothing is used.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings Step 2 Save the file to your TFTP server (in the root directory or a subdirectory containing all the phone-specific configuration files). Name the file “SIPXXXXYYYYZZZZ.cnf” where XXXXYYYYZZZZ is the MAC address of the phone. The MAC address must be in uppercase and the extension, cnf, must be in lower case (for example, SIP00503EFFD842.cnf).
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings • When configuring the Preferred Codec and Out of Band DTMF parameters, press the Change soft key until the option you desire is displayed and then press the Save soft key. • After making your changes, relock configuration mode as described in the “Locking Configuration Mode” section on page 3-2. Procedure Step 1 Press the settings key. The Settings menu is displayed. Step 2 Highlight SIP Configuration.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings Authentication Password parameter when registration is enabled, the default logical password is used. The default logical password is SIPmacaddress where macaddress is the MAC address of the phone. • Display Name—(Optional) Identification as it should appear for caller-identification purposes. For example, instead of jdoe@company.
Chapter 3 Managing Cisco SIP IP Phones Modifying the Phone’s SIP Settings • Register with Proxy—(Optional) Whether the phone must register with a proxy server during initialization. Valid values are Yes and No. Select the No soft key to disable registration during initialization. Select the Yes soft key to enable registration during initialization. The default is No.
Chapter 3 Managing Cisco SIP IP Phones Setting the Date, Time, and Daylight Savings Time Setting the Date, Time, and Daylight Savings Time The current date and time is supported on the Cisco SIP IP phone via SNTP and is displayed on the phone’s LCD. In addition to supporting the current date and time, daylight savings time (DST) and time zone settings are also supported.
Chapter 3 Managing Cisco SIP IP Phones Setting the Date, Time, and Daylight Savings Time Table 3-2 Actions Based on sntp_mode When the sntp_server Parameter is Set to a Null Value sntp_server sntp_mode= =0.0.0.0 unicast sntp_mode= multicast sntp_mode= anycast sntp mode= directedbroadcast Sends Nothing. Nothing. No known server with which to communicate. SNTP packet to the local network address. When in multicast mode, SNTP requests are not sent.
Chapter 3 Managing Cisco SIP IP Phones Setting the Date, Time, and Daylight Savings Time Table 3-3 lists the actions that take place when a valid IP address is specified in the sntp_server parameter. Table 3-3 Actions Based on sntp_mode When the sntp_server Parameter is Set to an IP Address sntp_server sntp_mode= = 0.0.0.0 unicast Sends Receives sntp_mode= multicast sntp_mode= anycast SNTP request to the SNTP request to the Nothing. SNTP server. SNTP server.
Chapter 3 Managing Cisco SIP IP Phones Setting the Date, Time, and Daylight Savings Time Procedure Step 1 Using an ASCII editor, open the SIPDefault.cnf file and define or modify values for the following SNTP-specific SIP parameters as necessary: • sntp_mode—(Required) Mode in which the phone will listen for the SNTP server. Valid values are unicast, multicast, anycast, or directedbroadcast.
Chapter 3 Managing Cisco SIP IP Phones Setting the Date, Time, and Daylight Savings Time Step 3 • dst_start_time—Time of day on which DST begins. Valid values are hour/minute (02/00) or hour (14:30). • dst_stop_time—Time of day on which DST ends. Valid values are hour/minute (02/00) or hour (14:30). To configure absolute DST, specify values for the following parameters or to configure relative DST, proceed to Step 4: • dst_start_day—Day of the month on which DST begins.
Chapter 3 Managing Cisco SIP IP Phones Setting the Date, Time, and Daylight Savings Time • dst_stop_day_of_week—Day of the week on which DST ends. Valid values are Sunday or Sun, Monday or Mon, Tuesday or Tue, Wednesday or Wed, Thursday or Thu, Friday or Fri, Saturday or Sat, or Sunday or Sun or 1 through 7 with 1 being Sunday and 7 being Saturday. When specifying the name of the day, the value is case-sensitive and should be typed as cited in this description.
Chapter 3 Managing Cisco SIP IP Phones Erasing the Locally-Defined Settings The following is an example of the configuration for a relative DST configuration: ; sip default configuration file (additional configuration text omitted) time_zone : PST dst_offset : 01/00 dst_start_month : April dst_start_day : 0 dst_start_day_of_week : Sunday dst_start_week_of_month : 1 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 02/
Chapter 3 Managing Cisco SIP IP Phones Erasing the Locally-Defined Settings Procedure Step 1 Press the settings key. The Settings menu is displayed. Step 2 Highlight Network Configuration. Step 3 Press the Select soft key. The Network Configuration settings are displayed. Step 4 Highlight Erase Configuration. Step 5 Press the Yes soft key. Step 6 Press the Save soft key. The phone programs the new information into Flash memory and resets.
Chapter 3 Managing Cisco SIP IP Phones Accessing Status Information Step 4 Highlight the parameter for which you wish to erase the setting. Step 5 Press the Edit soft key. Step 6 Press the << soft key to delete the current value. Step 7 Press the Validate soft key to save your change and exit the Edit panel. Step 8 If modifying a line parameter, press the Back soft key to exit the Line Configuration panel. Step 9 Press the Save soft key.
Chapter 3 Managing Cisco SIP IP Phones Accessing Status Information Viewing Status Messages To view status messages that you can use to diagnose network problems, complete the following steps: Step 1 Press the Settings key. The Settings menu is displayed. Step 2 Highlight Status. Step 3 Press the Select soft key. The Setting Status menu is displayed. Step 4 Highlight Status Messages. Step 5 Press the Select soft key. The Status Messages panel is displayed.
Chapter 3 Managing Cisco SIP IP Phones Accessing Status Information • Phone State Message—TCP messages indicating the state of the phone. Possible messages are: – Phone Initialized—TCP connection has not gone down since the phone was powered on. – Phone Closed TCP—TCP connection was closed by the phone. – TCP Timeout—TCP connection was closed because of a retry timeout. – Error Code—Error messages indicating unusual reasons the TCP connection was closed.
Chapter 3 Managing Cisco SIP IP Phones Upgrading the Cisco SIP IP Phone Firmware Viewing the Firmware Version To view network statistics, complete the following steps: Step 1 Press the settings key. The Settings menu is displayed. Step 2 Highlight Status. Step 3 Press the Select soft key. The Setting Status menu is displayed. Step 4 Highlight Firmware Versions. Step 5 Press the Select soft key. The Firmware Versions panel is displayed.
Chapter 3 Managing Cisco SIP IP Phones Upgrading the Cisco SIP IP Phone Firmware Procedure Step 1 Copy the binary file P0S3xxyy.bin (where xx is the version number and yy is the subversion number) from CCO to the root directory of the TFTP server. Step 2 Using a text editor, open the configuration file and update the image version specified in the image_version variable. The version name in image_version variable should match the version name (without the .
Chapter 3 Managing Cisco SIP IP Phones Performing an Image Upgrade and Remote Reboot Performing an Image Upgrade and Remote Reboot With Version 2.0 of the Cisco SIP IP Phone 7960, you can perform an image upgrade and remote reboot using Notify messages and the synchinfo.xml file. Note To perform an image grade and remote reboot, a SIP proxy server and a TFTP server must exist in the phone network. To upgrade the firmware image and perform a remote reboot, complete the following tasks: 1.
Chapter 3 Managing Cisco SIP IP Phones Performing an Image Upgrade and Remote Reboot The following is an example of a Notify message: NOTIFY sip:lineX_name@ipaddress:5060 SIP/2.0 Via: SIP/2.0/UDP ipaddress:5060;branch=1 Via: SIP/2.
A P P E N D I X A SIP Compliance with RFC-2543 Information This section describes how the Cisco SIP IP phone complies with the IETF definition of SIP as described in RFC 2543.
Appendix A SIP Compliance with RFC-2543 Information SIP Functions SIP Functions Function Supported? User Agent Client (UAC) Yes User Agent Server (UAS) Yes Proxy Server Third-party only Redirect Server Third-party only SIP Methods Five of the six methods used by the SIP gateway are supported: Method Supported? Comments INVITE Yes The Cisco SIP IP phone supports mid-call changes such as putting a call on hold as signaled by a new INVITE that contains an existing Call-ID.
Appendix A SIP Compliance with RFC-2543 Information SIP Responses SIP Responses Release 1.
Appendix A SIP Compliance with RFC-2543 Information SIP Responses 1xx Response—Information Responses 1xx Response Supported? Comments 100 Trying Yes The Cisco SIP IP phone generates this response for an incoming INVITE. Upon receiving this response, the phone waits for a 180 Ringing, 183 Session progress, or 200 OK response. 180 Ringing Yes None See comments The Cisco SIP IP phone does not generate these responses, however, the phone does receive them.
Appendix A SIP Compliance with RFC-2543 Information SIP Responses 3xx Response—Redirection Responses 3xx Response Supported Comments 300 Multiple Choices Yes 301 Moved Permanently Yes 302 Moved Temporarily Yes The Cisco SIP IP phone does not generate this response at this time. Upon receiving this response, the phone sends an INVITE containing the contact information received in the 302 Moved temporarily response.
Appendix A SIP Compliance with RFC-2543 Information SIP Responses 4xx Response Supported? Comments 401 Unauthorized Yes This response is only received in this release. If a 401 Unauthorized response is received during registration, the phone accepts the response and sends a new request that contains the user’s authentication information in the format of the HTTP digest as modified by RFC 2543. 402 Payment Required Yes The phone does not generate the 402 Payment Required response.
Appendix A SIP Compliance with RFC-2543 Information SIP Responses 4xx Response Supported? Comments 407 Proxy Authentication Required See comments This response is only received in this release. 408 Request Timeout See comments The SIP phone does not generate a 408 Request Timeout response. For an incoming response, the gateway initiates a graceful call disconnect (during which the caller hears a busy or fast busy tone) before clearing the call request.
Appendix A SIP Compliance with RFC-2543 Information SIP Responses 4xx Response Supported? Comments 411 Length Required See comments This response is only received by the phone in this release. This response indicates that the user refuses to accept the request without a defined content length. If received the phone resends the INVITE request if it can add a valid Content-Length header field. 413 Request Entity Too Large See comments This response is only received by the phone in this release.
Appendix A SIP Compliance with RFC-2543 Information SIP Responses 4xx Response Supported? Comments 480 Temporarily Unavailable See comments This response is only received by the phone in this release. The user is notified if this response is received. If this response is received, the user is notified that the callee is temporarily unavailable (perhaps not logged on) and any retry information is displayed.
Appendix A SIP Compliance with RFC-2543 Information SIP Header Fields 5xx Response—Server Failure Responses 5xx Response Comments 500 Internal Server Error 501 Not Implemented 502 Bad Gateway 503 Service Unavailable 504 Gateway Timeout For an incoming response, the Cisco SIP IP phone sends a new request if an additional contact address is present. If an additional contact address is not present, the gateway initiates a graceful call disconnect.
Appendix A SIP Compliance with RFC-2543 Information SIP Header Fields Header Field Supported? Authorization Yes Call-ID Yes Contact Yes Content-Encoding Yes Content-Length Yes Content-Type Yes Cseq Yes Date Yes Encryption No Expires Yes From Yes Hide No Max-Forwards Yes Organization No Priority No Proxy-Authenticate Yes Proxy-Authorization Yes Proxy-Require Yes ReBy Yes Record-Route Yes Require Yes Response-Key No Retry-After Yes Route Yes Cisco SIP IP Pho
Appendix A SIP Compliance with RFC-2543 Information SIP Session Description Protocol (SDP) Usage Header Field Supported? Server No Subject No Timestamp Yes To Yes Unsupported Yes User-Agent Yes Via Yes Warning Yes WWW-Authenticate Yes SIP Session Description Protocol (SDP) Usage SDP Headers Supported? v—Protocol version Yes o—Owner/creator and session identifier Yes a—Session name Yes c—Connection information Yes m—Media name and transport address Yes Cisco SIP IP Phone 79
A P P E N D I X B SIP Call Flows SIP uses six request methods: • INVITE—Indicates a user or service is being invited to participate in a call session. • ACK—Confirms that the client has received a final response to an INVITE request. • BYE—Terminates a call and can be sent by either the caller or the callee. • CANCEL—Cancels any pending searches but does not terminate a call that has already been accepted. • OPTIONS—Queries the capabilities of servers.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Call Flow Scenarios for Successful Calls This section describes call flows for the following scenarios, which illustrate successful calls: • Gateway-to Cisco SIP IP Phone—Successful Call Setup and Disconnect, page B-3 • Gateway-to-Cisco SIP IP Phone—Successful Call Setup and Call Hold, page B-7 • Gateway to-Cisco SIP IP Phone—Successful Call Setup and Call Transfer, page B-11 • Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call H
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Gateway-to Cisco SIP IP Phone—Successful Call Setup and Disconnect Figure B-1 illustrates a successful gateway-to-Cisco SIP IP phone call setup and disconnect. In this scenario, the two end users are User A and User B. User A is located at PBX A. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. User B is located at a Cisco SIP IP phone. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-1 User A Gateway-to-Cisco SIP IP Phone—Successful Setup and Disconnect PBX A GW1 IP Network SIP IP Phone User B IP 1. Setup 2. INVITE 3. Call Proceeding 4. 100 Trying 5. 180 Ringing 6. Alerting 7. 200 OK 8. Connect 9. Connect ACK 10. ACK 2-way voice path 2-way RTP channel 11. BYE 12. Disconnect 13. Release 15. Release Complete 41724 14.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 Setup—PBX A to Gateway 1 Call Setup is initiated between PBX A and Gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call User B. 2 INVITE—Gateway 1 to Cisco SIP IP phone Gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer includes the IP address and the port number of the SIP enabled entity to contact.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 6 Alerting—Gateway 1 to PBX A Gateway 1 sends an Alert message to User A. The Alert message indicates that Gateway 1 has received a 180 Ringing response from the Cisco SIP IP phone. User A hears the ringback tone that indicates that User B is being alerted. 7 200 OK—Cisco SIP IP phone to The Cisco SIP IP phone sends a SIP 200 OK response to Gateway 1 Gateway 1.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Gateway-to-Cisco SIP IP Phone—Successful Call Setup and Call Hold Figure B-2 illustrates a successful gateway-to-Cisco SIP IP phone call setup and call hold. In this scenario, the two end users are User A and User B. User A is located at PBX A. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. User B is located at a Cisco SIP IP phone. Gateway 1 is connected to the Cisco SIP IP phone over an IP network.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-2 User A Gateway-to-Cisco SIP IP Phone Call—Successful Call Setup and Call Hold PBX A GW1 IP Network SIP IP Phone User B IP 1. Setup 2. INVITE 3. Call Proceeding 4. 100 Trying 5. 180 Ringing 6. Alerting 7. 200 OK 8. Connect 9. ACK 10. Connect ACK 2-way voice path 2-way RTP channel 11. INVITE (c=IN IP4 0.0.0.0) 12. 200 OK 13. ACK No RTP packets being sent 14. INVITE (c=IN IP4 IP-User B) 15.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 Setup—PBX A to Gateway 1 Call Setup is initiated between PBX A and Gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call User B. 2 INVITE—Gateway 1 to Cisco SIP IP phone Gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer includes the IP address and the port number of the SIP enabled entity to contact.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 6 Alerting—Gateway 1 to PBX A Gateway 1 sends an Alert message to User A. The Alert message indicates that Gateway 1 has received a 180 Ringing response from the Cisco SIP IP phone. User A hears the ringback tone that indicates that User B is being alerted. 7 200 OK—Cisco SIP IP phone to The Cisco SIP IP phone sends a SIP 200 OK response to Gateway 1 Gateway 1.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Gateway to-Cisco SIP IP Phone—Successful Call Setup and Call Transfer Figure B-3 illustrates a successful gateway-to-Cisco SIP IP phone PC call setup and call transfer without consultation. In this scenario, there are three end users: User A, User B, and User C. User A is located at PBX A. PBX A is connected to Gateway 1 (SIP Gateway) via a T1/E1. User B is located at a Cisco SIP IP phone and is directly connected to the IP network.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-3 Gateway-to-Cisco SIP IP Phone Call—Successful Call Setup and Call Transfer SIP IP Phone User B User A PBX A 1. Setup GW1 IP Network IP GW2 PBX B User C 2. INVITE 3. Call Proceeding 4. 100 Trying 5. 180 Ringing 6. Alerting 7. 200 OK 8. Connect 9. Connect ACK 10. ACK 2-way voice path 2-way RTP channel 11. BYE (Also: C) 12. 200 OK 13. INVITE (Requested-By: B) 15. 100 Trying 18. 180 Ringing 14. Setup 16.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 Setup—PBX A to Gateway 1 Call Setup is initiated between PBX A and Gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call User B. 2 INVITE—Gateway 1 to Cisco SIP IP phone Gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer includes the IP address and the port number of the SIP enabled entity to contact.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 6 Alerting—Gateway 1 to PBX A Gateway 1 sends an Alert message to User A. The Alert message indicates that Gateway 1 has received a 180 Ringing response from the Cisco SIP IP phone. User A hears the ringback tone that indicates that User B is being alerted. 7 200 OK—Cisco SIP IP phone to The Cisco SIP IP phone sends a SIP 200 OK response to Gateway 1 Gateway 1.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 13 INVITE—Gateway 1 to Gateway 2 Gateway 1 sends a SIP INVITE request to Gateway 2. In the INVITE request, a unique Call-ID is generated and the Requested-By field indicates that User B requested the call. 14 Setup—Gateway 2 to PBX B Gateway 2 receives the INVITE request from Gateway 1 and initiates a Call Setup with User C via PBX B.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold Figure B-4 illustrates a successful call between Cisco SIP IP phones in which one of the participants places the other on hold and then returns to the call. In this call flow scenario, the two end users are User A and User B. User A and User B are both using Cisco SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: 1. User A calls User B.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-4 Cisco SIP IP Phone-to-Cisco SIP IP Phone Simple Call Hold SIP IP Phone User A IP Network SIP IP Phone User B IP IP 1. INVITE B 2. 180 RINGING 3. 200 OK 4. ACK 2-way RTP channel 5. INVITE (c=IN IP4 0.0.0.0) 6. 200 OK 7. ACK A is on hold. The RTP channel between A and B is torn down. 8. INVITE (c=IN IP4 IP-User B) 9. 200 OK 10. ACK 41465 A is taken off hold. The RTP channel between A and B is reestablished.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: 2 • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 3 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to to Cisco SIP IP phone A Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 8 INVITE—Cisco SIP IP phone B Cisco SIP IP phone B sends a mid-call INVITE to Cisco to Cisco SIP IP phone A SIP IP phone A with the same call ID as the previous INVITE and new SDP session parameters (IP address), which are used to reestablish the call. Call_ID=1 SDP: c=IN IP4 181.23.250.2 To reestablish the call between phone A and phone B, the IP address of phone B is inserted into the c= SDP field.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-5 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Hold with Consultation SIP IP Phone User A SIP IP Phone User B IP Network IP IP SIP IP Phone User C IP 1. INVITE B 2. 180 Ringing 3. 200 OK 4. ACK 2-way RTP channel 5. INVITE (c=IN IP4 0.0.0.0) 6. 200 OK 7. ACK A is put on hold. The RTP channel between A and B is torn down. 8. INVITE C 9. 180 Ringing 10. 200 OK 11. ACK 2-way RTP channel 12. BYE 13. 200 OK 14.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: 2 • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 3 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to to Cisco SIP IP phone A Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 8 INVITE—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP INVITE request to Cisco to Cisco SIP IP phone C SIP IP phone C. The INVITE request is an invitation to User C to participate in a call session. 9 180 Ringing—Cisco SIP IP Cisco SIP IP phone C sends a SIP 180 Ringing response to phone C to Cisco SIP IP phone B Cisco SIP IP phone B.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 14 INVITE—Cisco SIP IP phone B Cisco SIP IP phone B sends a mid-call INVITE to Cisco to Cisco SIP IP phone A SIP IP phone A with the same call ID as the previous INVITE and new SDP session parameters (IP address), which are used to reestablish the call. Call_ID=1 SDP: c=IN IP4 181.23.250.2 To reestablish the call between phone A and phone B, the IP address of phone B is inserted into the c= SDP field.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-6 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Waiting SIP IP Phone User A IP Network SIP IP Phone User B IP SIP IP Phone User C IP IP 1. INVITE B 2. 180 Ringing 3. 200 OK 4. ACK 2-way RTP channel 5. INVITE C 6. 180 Ringing 7. INVITE (c=IN IP4 0.0.0.0) 8. 200 OK 9. ACK A is put on hold. The RTP channel between A and B is torn down. 10. 200 OK 11. ACK 2-way RTP channel 12. INVITE (c=IN IP4 0.0.0.0) 13. 200 OK 14.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: 2 • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 3 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to to Cisco SIP IP phone A Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 9 ACK—Cisco SIP IP phone B to Cisco SIP IP phone A Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone A. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A. The RTP channel between Cisco SIP IP phone A and Cisco SIP IP phone B is torn down.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 15 INVITE—Cisco SIP IP phone B Cisco SIP IP phone B sends a mid-call INVITE to Cisco to Cisco SIP IP phone A SIP IP phone A with the same call ID as the previous INVITE (sent to Cisco SIP IP phone A) and new SDP session parameters (IP address), which are used to reestablish the call. Call_ID=1 SDP: c=IN IP4 181.23.250.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 16 ACK—Cisco SIP IP phone B to Cisco SIP IP phone C Cisco SIP IP phone B sends a SIP ACK to Cisco SIP IP phone C. The ACK confirms that Cisco SIP IP phone B has received the 200 OK response from Cisco SIP IP phone A. A two-way RTP channel is reestablished between Cisco SIP IP phone B and Cisco SIP IP phone C.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-7 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer without Consultation SIP IP Phone User A IP Network IP SIP IP Phone User B IP SIP IP Phone User C IP 1. INVITE B 2. 180 Ringing 3. 200 OK 4. ACK 2-way RTP channel 5. BYE (Also: C) 6. 200 OK A and B are disconnected. 7. INVITE C (Requested by B) 8. 180 Ringing 9. 200 OK 10.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: 2 • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 3 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to to Cisco SIP IP phone A Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 7 INVITE—Cisco SIP IP phone A to Cisco SIP IP phone C (Requested by Cisco SIP IP phone B) At the request of Cisco SIP IP phone B, Cisco SIP IP phone A sends a SIP INVITE request to Cisco SIP IP phone C. The INVITE request is an invitation to User C to participate in a call session.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-8 Cisco SIP IP Phone-to-Cisco SIP IP Phone Call Transfer with Consultation SIP IP Phone User A IP Network IP SIP IP Phone User B SIP IP Phone User C IP IP 1. INVITE B 2. 180 Ringing 3. 200 OK 4. ACK 2-way RTP channel 5. INVITE (c=IN IP4 0.0.0.0) 6. 200 OK 7. ACK A is put on hold. The RTP channel between A and B is torn down. 8. INVITE C 9. 180 Ringing 10. 200 OK 11. ACK 2-way RTP channel 12. BYE 13. 200 OK 14.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: 2 • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 3 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to to Cisco SIP IP phone A Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 9 180 Ringing—Cisco SIP IP Cisco SIP IP phone C sends a SIP 180 Ringing response to phone C to Cisco SIP IP phone B Cisco SIP IP phone B. 10 200 OK—Cisco SIP IP phone C Cisco SIP IP phone C sends a SIP 200 OK response to to Cisco SIP IP phone B Cisco SIP IP phone B. The 200 OK response notifies Cisco SIP IP phone B that the connection has been made.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 14 BYE—Cisco SIP IP phone B to Cisco SIP IP phone A The call continues and then User B hangs up. Cisco SIP IP phone B sends a SIP BYE request to Cisco SIP IP phone A. The SIP BYE request includes the Also header. The Also header indicates that User C needs to be brought into the call while User B hangs up.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Unconditional) Figure B-9 illustrates successful call forwarding between Cisco SIP IP phones in which User B has requested unconditional call forwarding from the network. When User A calls User B, the call is immediately transferred to Cisco SIP IP phone C. In this call flow scenario, the end users are User A, User B, and User C.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-9 Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Unconditional) IP Network IP SIP IP Phone User A Proxy Server Redirect Server IP IP SIP IP Phone User B SIP IP Phone User C 1. INVITE B 2. INVITE B 3. 302 Moved Temporarily 4. INVITE C 5. 180 Ringing 6. 200 OK 7. 200 OK 8. ACK 2-way RTP channel 41471 9.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to the to SIP proxy server SIP proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 6 200 OK—Cisco SIP IP phone C Cisco SIP IP phone C sends a SIP 200 OK response to the to SIP proxy server SIP proxy server. 7 200 OK—SIP proxy server to Cisco SIP IP phone A SIP proxy server forwards the SIP 200 OK response to Cisco SIP IP phone A. 8 ACK—Cisco SIP IP phone A to SIP proxy server Cisco SIP IP phone A sends a SIP ACK to the SIP proxy server.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-10 Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (Busy) IP Network IP SIP IP Phone User A Proxy Server Redirect Server IP IP SIP IP Phone User B SIP IP Phone User C 1. INVITE B 2. INVITE B 3. 300 Multiple Choices 4. INVITE B 5. 486 Busy Here 6. ACK 7. INVITE C 8. 180 Ringing 9. 200 OK 10. 200 OK 11. ACK 12.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to the to SIP proxy server SIP proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 5 486 Busy Here—Cisco SIP IP phone B to SIP proxy server Cisco SIP IP phone B sends a 486 Busy here message to the SIP proxy server. The message indicates that Cisco SIP IP phone B is in use and the user is not willing or able to take additional calls. 6 ACK—SIP proxy server to Cisco SIP proxy server forwards the SIP ACK to the Cisco SIP IP SIP IP phone B phone B.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (No Answer) Figure B-11 illustrates successful call forwarding between Cisco SIP IP phones in which User B has requested call forwarding from the network in the event there is no answer. When User A calls User B, the proxy server tries to place the call to Cisco SIP IP phone B and, if there is no answer, the call is transferred to Cisco SIP IP phone C.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-11 Cisco SIP IP Phone-to-Cisco SIP IP Phone Network Call Forwarding (No Answer) IP Network IP SIP IP Phone User A Proxy Server Redirect Server IP IP SIP IP Phone User B SIP IP Phone User C 1. INVITE B 2. INVITE B 3. 300 Multiple Choices 4. INVITE B 5. 180 Ringing 6. 180 Ringing 7. CANCEL 8. 200 OK 9. INVITE C 10. 180 Ringing 11. 200 OK 12. 200 OK 13. ACK 14.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to the to SIP proxy server SIP proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 6 180 Ringing—SIP proxy server to Cisco SIP IP phone A SIP proxy server sends a SIP 180 Ringing response to Cisco SIP IP phone A. The timeout expires before the phone is answered. 7 CANCEL (Ring Timeout)—SIP proxy server to Cisco SIP IP phone B 8 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to the to SIP proxy server SIP proxy server.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Cisco SIP IP Phone-to Cisco SIP IP Phone 3-Way Calling Figure B-11 illustrates successful 3-way calling between Cisco SIP IP phones in which User B mixes two RTP channels and therefore establishes a conference bridge between User A and User C. In this call flow scenario, the end users are User A, User B, and User C. They are all using Cisco SIP IP phones, which are connected via an IP network. The call flow scenario is as follows: 5.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Figure B-12 Cisco SIP IP Phone-to Cisco SIP IP Phone 3-Way Calling IP Network IP IP SIP IP Phone User A Proxy Server SIP IP Phone User B Redirect Server IP SIP IP Phone User C 1. INVITE B (Call-ID=1) 2. 180 Ringing 3. 200 OK 4. ACK 2-way RTP channel 1 between Users A and B established 5. INVITE (Call-ID=1, c=IN IP40.0.0.0) 6. 200 OK 7. ACK User A is on hold. The RTP channel 1 between User A and B is torn down. 8.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: 2 • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 3 200 OK—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP 200 OK response to to Cisco SIP IP phone A Cisco SIP IP phone A. The 200 OK response notifies Cisco SIP IP phone A that the connection has been made.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 8 INVITE—Cisco SIP IP phone B Cisco SIP IP phone B sends a SIP INVITE request to Cisco to Cisco SIP IP phone C SIP IP phone C. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: 9 • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Successful Calls Step Action Description 10 200 OK—Cisco SIP IP phone C Cisco SIP IP phone C sends a SIP 200 OK response to to Cisco SIP IP phone B Cisco SIP IP phone B. The 200 OK response notifies Cisco SIP IP phone B that the connection has been made.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Call Flow Scenarios for Failed Calls This section describes call flows for the following scenarios, which illustrate unsuccessful calls: • Gateway-to-Cisco SIP IP Phone—Called User is Busy, page B-58 • Gateway-to-Cisco SIP IP Phone—Called User Does Not Answer, page B-60 • Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error, page B-63 • Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User is Busy, page B-66 • Cisco SIP IP Ph
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 1 Setup—PBX A to Gateway 1 Call Setup is initiated between PBX A and Gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call User B. 2 INVITE—Gateway 1 to Cisco SIP IP phone Gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer includes the IP address and the port number of the SIP enabled entity to contact.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 6 Disconnect (Busy)—Gateway 1 to PBX A Gateway 1 sends a Disconnect message to PBX A. 7 Release—PBX A to Gateway 1 PBX A sends a Release message to Gateway 1. 8 ACK—Gateway 1 to Cisco SIP IP phone Gateway 1 sends a SIP ACK to the Cisco SIP IP phone. The ACK confirms that User A has received the 486 Busy Here response. The call session attempt is now being terminated.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 1 Setup—PBX A to Gateway 1 Call Setup is initiated between PBX A and Gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call User B. 2 INVITE—Gateway 1 to Cisco SIP IP phone Gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer includes the IP address and the port number of the SIP enabled entity to contact.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 7 CANCEL (Ring Timeout)—Gateway 1 to Cisco SIP IP phone Because Gateway 1 did not return an appropriate response within the time allocated in the INVITE request, Gateway 1 sends a SIP CANCEL request to Gateway 2. A CANCEL request cancels a pending request with the same Call-ID, To, From, and CSeq header field values. 8 Disconnect—Gateway 1 to PBX Gateway 1 sends a Disconnect message to PBX A.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error Figure B-15 illustrates an unsuccessful call in which User A initiates a call to User B and receives a class 4xx, 5xx, or 6xx response. Figure B-15 Gateway-to-Cisco SIP IP Phone—Client, Server, or Global Error User A PBX A GW1 IP Network SIP IP Phone User B IP 1. Setup 2. INVITE 3. Call Proceeding 4. 100 Trying 5. 4xx/5xx/6xx Failure 6. Disconnect 7. Release 41727 8. ACK 9.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 1 Setup—PBX A to Gateway 1 Call Setup is initiated between PBX A and Gateway 1. The Call Setup includes the standard transactions that take place as User A attempts to call User B. 2 INVITE—Gateway 1 to Cisco SIP IP phone Gateway 1 maps the SIP URL phone number to a dial-peer. The dial-peer includes the IP address and the port number of the SIP enabled entity to contact.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 5 4xx/5xx/6xx Failure—Cisco SIP The Cisco SIP IP phone sends a class 4xx, 5xx, or class 6xx IP phone to Gateway 1 failure response to Gateway 1. Depending on which class the failure response is, the call actions differ. If the Cisco SIP IP phone sends a class 4xx failure response (a definite failure response that is a client error), the request will not be retried without modification.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User is Busy Figure B-16 illustrates an unsuccessful call in which User A initiates a call to User B but User B is on the phone and is unable or unwilling to take another call. Figure B-16 Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User is Busy SIP IP Phone User A IP Network IP SIP IP Phone User B IP 1. INVITE B 2. 486 Busy Here 41475 3.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User Does Not Answer Figure B-17 illustrates an unsuccessful call in which User A initiates a call to User B but User B does not answer. Figure B-17 Cisco SIP IP Phone-to-Cisco SIP IP Phone—Called User Does Not Answer SIP IP Phone User A IP Network IP SIP IP Phone User B IP 1. INVITE B 2. 180 Ringing 3. CANCEL 41476 4.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to Cisco to Cisco SIP IP phone B SIP IP phone B. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Cisco SIP IP Phone-to-Cisco SIP IP Phone—Authentication Error Figure B-18 illustrates an unsuccessful call in which User A initiates a call to User B but is prompted for authentication credentials by the proxy server. User A’s SIP IP phone then reinitiates the call with an SIP INVITE request that includes it’s authentication credentials.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Step Action Description 1 INVITE—Cisco SIP IP phone A Cisco SIP IP phone A sends a SIP INVITE request to the to SIP proxy server SIP proxy server. The INVITE request is an invitation to User B to participate in a call session. In the INVITE request: • The phone number of User B is inserted in the Request-URI field in the form of a SIP URL.
Appendix B SIP Call Flows Call Flow Scenarios for Failed Calls Cisco SIP IP Phone 7960 Administrator Guide B-72 78-10497-02
A P P E N D I X C Technical Specifications This appendix provides physical and operating environment and cable technical specifications for the Cisco SIP IP phones. This appendix also provides the connection specifications of your Cisco SIP IP phone. Physical and Operating Environment Specifications The following table lists the physical and operating specifications of the Cisco SIP IP phone.
Appendix C Technical Specifications Physical and Operating Environment Specifications Table C-1 Cisco SIP IP Phone Operational and Physical Specifications (continued) Specification Value or Range Power 48 VDC, supplied locally at the desktop using an optional AC-to-DC poser supply Regulatory CE Marking IC CS-03 FCC Part 15 Class A FCC (CFR47) Part 68 FCC Part 68 EN 55022 Class A UL 1459 CSA-C22.2 No. 225-M90 EN 60950: 1992 IEC 950 AS/NZS 3260 TS001 Safety UL-1950 EN 60950 CSA-C22.2 No.
Appendix C Technical Specifications Cable Specifications Table C-1 Cisco SIP IP Phone Operational and Physical Specifications (continued) Specification Value or Range Cables Two (2) pair of Category 3 for 10 Mbps cables Two (2) pair of Category 5 for 100 Mbps cables Distance Requirements As supported by the Ethernet Specification, it is assumed that most sets that are deployed in the field will be within 100 m (330 ft.) of a phone closet.
Appendix C Technical Specifications Connections Specifications Cisco SIP IP Phone 7960 Administrator Guide C-4 78-10497-02
A P P E N D I X D Translated Safety Warnings This appendix repeats in multiple languages the warnings that appear in the “Getting Started with Your Cisco SIP IP Phone” chapter of this guide. Installation Warning Warning Read the installation instructions before you connect the system to its power source. Waarschuwing Raadpleeg de installatie-aanwijzingen voordat u het systeem met de voeding verbindt. Varoitus Lue asennusohjeet ennen järjestelmän yhdistämistä virtalähteeseen.
Appendix D Translated Safety Warnings ¡Advertencia! Ver las instrucciones de instalación antes de conectar el sistema a la red de alimentación. Varning! Läs installationsanvisningarna innan du kopplar systemet till dess strömförsörjningsenhet. Product Disposal Warning Warning Ultimate disposal of this product should be handled according to all national laws and regulations. Waarschuwing Dit produkt dient volgens alle landelijke wetten en voorschriften te worden afgedankt.
Appendix D Translated Safety Warnings Lightning Activity Warning Warning Do not work on the system or connect or disconnect cables during periods of lightning activity. Waarschuwing Tijdens onweer dat gepaard gaat met bliksem, dient u niet aan het systeem te werken of kabels aan te sluiten of teontkoppelen. Varoitus Älä työskentele järjestelmän parissa äläkä yhdistä tai irrota kaapeleita ukkosilmalla.
Appendix D Translated Safety Warnings SELV Circuit Warning (other versions available) Warning To avoid electric shock, do not connect safety extra-low voltage (SELV) circuits to telephone-network voltage (TNV) circuits. LAN ports contain SELV circuits, and WAN ports contain TNV circuits. Some LAN and WAN ports both use RJ-45 connectors. Use caution when connecting cables.
Appendix D Translated Safety Warnings Avvertenza Per evitare scosse elettriche, non collegare circuiti di sicurezza a tensione molto bassa (SELV) ai circuiti a tensione di rete telefonica (TNV). Le porte LAN contengono circuiti SELV e le porte WAN contengono circuiti TNV. Alcune porte LAN e WAN fanno uso di connettori RJ-45. Fare attenzione quando si collegano cavi. Advarsel Unngå å koble lavspenningskretser (SELV) til kretser for telenettspenning (TNV), slik at du unngår elektrisk støt.
Appendix D Translated Safety Warnings Circuit Breaker (15A) Warning Warning This product relies on the building’s installation for short-circuit (overcurrent) protection. Ensure that a fuse or circuit breaker no larger than 120 VAC, 15A U.S. (240 VAC, 10A international) is used on the phase conductors (all current-carrying conductors). Waarschuwing Dit produkt is afhankelijk van de installatie van het gebouw voor kortsluit- (overstroom)beveiliging.
Appendix D Translated Safety Warnings Aviso Este produto depende das instalações existentes para protecção contra curto-circuito (sobrecarga). Assegure-se de que um fusível ou disjuntor não superior a 240 VAC, 10A é utilizado nos condutores de fase (todos os condutores de transporte de corrente). ¡Advertencia! Este equipo utiliza el sistema de protección contra cortocircuitos (o sobrecorrientes) deló propio edificio.
Appendix D Translated Safety Warnings Cisco SIP IP Phone 7960 Administrator Guide D-8 78-10497-02
G L O S S A R Y A AAA Authentication, Authorization, and Accounting. AAA is a suite of network security services that provides the primary framework through which access control can be set up on your Cisco router or access server. ANI Automatic number identification. C CAS Channel associated signaling. CCAPI Call control applications programming interface. CLI Command line interface. CO Central office.
Glossary CPE Customer premises equipment. Terminating equipment, such as terminals, telephones, and modems, supplied by the telephone company, installed at the customer sites, and connected to the telephone company network. CSM Call switching module. D dial peer An addressable call endpoint. In Voice over IP (V0IP), there are two types of dial peers: POTS and VoIP. DNS Domain name system used to address translation to convert H.323 IDs, URLs, or e-mail IDs to IP addresses.
Glossary E E.164 The international public telecommunications numbering plan. A standard set by ITU-T which addresses telephone numbers. E&M Ear and mouth RBS signaling. endpoint A SIP terminal or gateway. An endpoint can call and be called. It generates and/or terminates the information stream. G gateway A gateway allows SIP or H.323 terminals to communicate with terminals configured to other protocols by converting protocols.
Glossary I IVR Integrated voice response. When someone dials in, IVR responds with a prompt to get a personal identification number (PIN), and so on. L LEC Local exchange carrier. Location Server A SIP redirect or proxy server uses a a location service to get information about a caller’s location(s). Location services are offered by location servers. M MF Multi-frequency tones are made of six frequencies that provide 15 two frequency combinations for indication digits 0-9 and KP/ST signals.
Glossary P PDU Protocol data units used by bridges to transfer connectivity information. POTS Plain old telephone service. Basic telephone service supplying standard single line telephones, telephone lines, and access to the PSTN. Proxy Server An intermediary program that acts as both a server and a client for the purpose of making requests on behalf of other clients. Requests are serviced internally or by passing them on, possibly after translation, to other servers.
Glossary S SIP Session Initiation Protocol. This is a protocol developed by the IETF MMUSIC Working Group as an alternative to H.323. SIP features are compliant with IETF RFC 2543, published in March 1999. SIP equips platforms to signal the setup of voice and multimedia calls over IP networks. SPI Service provider interface. T TDM Time division multiplexing. Technique in which information from multiple channels can be allocated bandwidth on a single wire based on preassigned time slots.
Glossary V VoIP Voice over IP. The ability to carry normal telephone-style voice over an IP-based Internet with POTs-like functionality, reliability, and voice quality. VoIP is a blanket term, which generally refers to Cisco’s standards based (for example H.323) approach to IP voice traffic.
Glossary Cisco SIP IP Phone 7960 Administrator Guide 8 78-10497-02
I N D E X Numerics administrative VLAN ID parameter 3-4 Allow header field A-10 10/100 PC port 1-14 Also header field A-10 10/100 SW port 1-14 alternate TFTP server, enabling 3-5 1xx responses A-4 authentication 2xx responses A-4 name, configuring 3-16 3xx responses A-5 services 1-3 4xx responses A-5 Authorization header field A-11 5xx responses A-10 6xx responses A-10 A B billing services 1-3 book Accept-Encoding header field A-10 objectives x Accept header field A-10 organization x A
Index phone-specific 2-6, 2-10 C creating 2-9 cables example 3-18 connecting 2-16 modifying 3-15 specifications C-3 naming convention 2-6 call flows B-1 successful B-2 unsuccessful B-58 SIPDefault.
Index Content-Type header field A-11 DNS conventions, document xi description 1-10 Cseq header field A-11 server parameters 3-4 documentation conventions xi D related xi Date header field A-11 domain name parameter 3-3 default configuration file 2-6, 2-7 Domain Name System (DNS) 1-10 example 2-9, 3-15 do not disturb 1-9 guidelines 2-6 downloading required files 2-4 modifying 3-8 DTMF DB level 3-10 SIPDefault.
Index example footstand adjustment 1-6 default configuration file 3-15 From header field A-11 phone-specific configuration file 3-18 functions Expires header field A-11 proxy server A-2 redirect server A-2 UAC A-2 F UAS A-2 features call forward 1-9 call hold 1-9 G call transfer 1-9 gateways 1-4 call waiting 1-9 guidelines 2-13 do not disturb 1-9 secondary directory number 1-9 URL dialing 1-10 file H handset 1-7 default 3-15 header fields A-10 phone-specific 3-18 headset files suppor
Index initialization process 2-1 installation 2-3 lines, configuring authentication name 3-16 downloading required files 2-4 name 3-16 network parameters 2-13 password 3-16 safety warnings D-1, D-3 short name 3-16 SIP parameters 2-5 linex_authname parameter 3-16 task summary 2-3 linex_name parameter 3-16 Internet Control Message Protocol (ICMP) 1-11 Internet Protocol (IP) 1-11 linex_password parameter 3-16 linex_shortname parameter 3-16 locking, configuration mode 3-2 INVITE retransmission ex
Index modifying port 1-14 statistics 3-31 network parameters 3-2, 3-3 SIP parameters 3-8, 3-15 network connections mute toggle 1-7 access port 1-14 N O name, configuring 3-16 on-screen mode keys 1-7 naming convention, phone-specific configuration file 2-6 operating environment specifications C-1 network Organization header field A-11 connections 1-13 parameters operational VLAN ID parameter 3-4 OS79XX.
Index administrative VLAN ID 3-4 Name 3-19 alternate TFTP 3-5 Out of Band DTMF 3-20 default routers 3-4 Prefered Codec 3-20 DHCP address release 3-4 preferred_codec 3-9 DHCP enable 3-4 proxy_register 3-11 DHCP server 3-3 proxy1_address 3-9 DNS server 3-4 proxy1_port 3-9 domain name 3-3 Proxy Address 3-20 erase configuration 3-5 Proxy Port 3-20 guidelines 2-13 Register Expires 3-21 host name 3-3 Register with Proxy 3-21 IP address 3-3 required 2-8 MAC address 3-3 Short Name 3-19 mo
Index connections 1-13 ICMP 1-11 access port 1-14 IP 1-11 network 1-13 RTP 1-11 network port 1-14 SDP 1-11 features SNTP 1-11 dialing pad 1-7 TFTP 1-12 footstand adjustment 1-6 UDP 1-12 handset 1-7 telephony features headset 1-15 telephony 1-9 headset and speaker toggle 1-7 URL dialing 1-10 information button 1-6 verifying startup 2-20 LCD screen 1-6 phone-specific configuration file line buttons 1-6 creating 2-10 mute toggle 1-7 example 2-9, 3-18 on-screen mode keys 1-7 modify
Index DHCP 1-10 registration DNS 1-10 enabling 3-11 ICMP 1-11 timer 3-11, 3-21 IP 1-11 related documentation xi RTP 1-11 release, DHCP address 3-4 SDP 1-11 request methods B-1 SNTP 1-11 Require header field A-11 TFTP 1-12 resetting UDP 1-12 network statistics 3-32 Proxy-Authenticate header field A-11 Response-Key header field A-11 Proxy-Authorization header field A-11 responses A-3 proxy port specifying 3-9 global (6xx) A-10 information (1xx) A-4 Proxy-Required header field A-11 re
Index clients 1-3, 1-4 S gateways 1-4 safety warnings, translated D-1 phones 1-4 circuit breaker (15A) warning D-6 compliance information A-1 installation warning D-1 components 1-3 lightning activity warning D-3 UAC 1-3 product disposal warning D-2 user agent server 1-3 SELV circuit warning D-4 default configuration file, example 2-9 scroll key 1-7 dtmf_inband 3-10 SDP, description 1-11 end point 1-3 SDP, usage A-12 funtions A-2 secondary directory number 1-9 gateways 1-4 SELV circui
Index Register Expires 3-21 operating environment C-1 Register with proxy 3-21 physical C-1 Short Name 3-19 specifying 3-9 request methods B-1 codec 3-9, 3-20 responses A-3 DTMF level 3-10 global (6xx) A-10 DTMF signaling 3-10 information (1xx) A-4 image version 3-9 redirection (3xx) A-5 proxy port 3-9 request failure (4xx) A-5 proxy server 3-9 server failure (5xx) A-10 retransmission timers 3-10 successful (2xx) A-4 TOS media 3-9 SDP usage A-12 specifying out of bound 3-20 servers
Index timer user registration 3-11, 3-21 retransmission 3-10 timer_t2 3-10 agent server 1-3 User-Agent header field A-12 User Datagram Protocol (UDP) 1-12 timers, retransmission 3-10 Timestamp header field A-12 toggle V headset and speaker 1-7 verifying startup 2-20 mute 1-7 Via header field A-12 To header field A-12 viewing firmware version 3-33 TOS media VLAN specifying 3-9 translated safety warnings D-1 circuit breaker (15A) warning D-6 installation warning D-1 administrative 3-4 operatio
