VOCAL Vovida Open Communication Application Library System Administration Guide Software Version 1.3.
Copyright Copyright © 2001, Cisco Systems, Inc. Guide Versions The following table matches the software versions with the guide versions: Software Version Guide Version Date Comments 1.0.0 Internal Trials Only 1.1.0 Internal Trials Only 1.2.0 1.2 1.2.0 1.2 A 1.3.0 1.3 March 26, 2001 Open Release to Public April 11, 2001 Copy edit errors corrected. December 21, Support new open release to public 2001 Version This manual is written to support VOCAL Version 1.3.0.
UHIDFH Introduction This chapter is a general introduction to the System Administration manual, and provides information about the intentions and organization of the manual. It also provides information about additional resources available from http://www.vovida.org. Objectives This guide provides Information about adding users and assigning features. Information for installing and provisioning a VOCAL system is provided in the Installation Guide.
Chapter Appendix C Documentation Conventions Title Call Flows Provides illustrations and descriptions of call flows for various call scenarios. The following is a list of conventions used in this guide: Convention Description bold text Names of elements found on the GUI screen, including buttons, and selectable entities such as, servers and server groups. <> Text that appears between angle brackets describes variables such as, .
Table of Contents Preface Chapter 1. Setting Up Users Working With The GUI Environment . . . . . . . . . . . . . . . . . . . . . . . . . Adding, Viewing, Editing, and Deleting Users . . . . . . . . . . . . . . . . . . 1-2 1-9 Chapter 2. Network Management SNMP Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 2-2 Appendix A. Features Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Core System Features . . .
Table of Contents (continued) vi
6HWWLQJ 8S 8VHUV This chapter describes how to add users to the system and how to maintain the user data base. Topic See Page Working With The GUI Environment. . . . . . . . . . . . . . . . . . . . . . . . . . Logging In. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Overview of the User Configuration Screen . . . . . . . . . . . . . . . . . . . . 1-2 1-3 1-5 Adding, Viewing, Editing, and Deleting Users. . . . . . . . . . . . . . . . . . Adding New Users . . .
Working With The GUI Environment Working With The GUI Environment Overview This section describes: • the login screen and how to log into the VOCAL system • the user configuration screen and available the buttons, options boxes, and data fields. Before you begin The machine that was used to install the VOCAL system is known as the provisioning host. You can access the Provisioning Server from the provisioning host, or from any other PC that is connected to the network where the VOCAL system resides.
Working With The GUI Environment Logging In Introduction The Provisioning Login screen provides access for Administrators to work with the users, and for Technicians to work with the servers. Definition The login screen is a java-enabled graphical user interface (GUI) that runs in a web browser. The browser can be any type that supports JRE 1.3.1_01. Procedure To log in, follow these steps: Table 1-1.
Working With The GUI Environment Items and Fields Table 1-2 describes the items found on the Login Screen. Table 1-2. Login Screen: Item and Field Description Item Password Administration Description Access Level Administrator As an Administrator, you can add, view, edit or delete user entries. In addition, you can setup feature subscriptions for users. Technician As a Technician, you can edit the VOCAL server provisioning. For more information, see the VOCAL Installation Guide.
Working With The GUI Environment Overview of the User Configuration Screen Introduction This section describes the buttons, option boxes, and data fields on the User Configuration Screen. Screen Capture Figures 1-2 and 1-3 show the User Configuration screen as it appears when you login. Figure 1-2 shows what the screen looks like when you login for the first time and the user records are blank. User records. As it appears before any users have been added. Figure 1-2.
Working With The GUI Environment After Data Entry Figure 1-3 shows what the screen looks like after some users have been added. For more information about adding users, see “Adding New Users” on page 1-10. User records. As it appears after some Fills user record fields with data. Finds users. Option Boxes. Figure 1-3. User Configuration Screen: After Data Entry Buttons Table 1-4 describes buttons on the User Configuration screen. Table 1-4.
Working With The GUI Environment Option Boxes The option boxes filter the fields displayed on the User Configuration screen. If none of the boxes is selected, only the Name, User Group, IP and Marshal fields appear. If all of the boxes are selected, then all of the fields appear on the User Configuration screen. For a complete description of the data fields refer to the “Viewing Users: Data Fields Descriptions” section on page 1-19. Table 1-5 describes the option boxes. Table 1-5.
Working With The GUI Environment Right-Mouse-Click Menu Options Table 1-6 shows the options available from the right-mouse-click menu. Table 1-6. User Configuration Screen: Right-Mouse-Click Menu Options Option Option Boxes 1-8 Description View View lets you view data field information in tabular format for one or more users. For more information, see Figure 1-6 on page 1-17. Edit Edit lets you edit information for the user. Selecting edit will open the Edit user screen.
Adding, Viewing, Editing, and Deleting Users Adding, Viewing, Editing, and Deleting Users Introduction The ”Working With The GUI Environment” section discussed the GUI buttons, option boxes and a right-mouse click menu that enables adding, viewing, editing and deleting users. This section provides information about using those GUI elements to perform tasks.
Adding, Viewing, Editing, and Deleting Users Adding New Users Introduction This section describes how to add new users. Procedure: Adding To add a new user, follow these steps: a New User Table 1-7. Adding New Users Step Action 1 Select the Show admin data option box. 2 Right-mouse click and select New. 3 The Edit user screen appears. 4 Enter the user name in the Name field. Caution You are only allowed to enter and modify the Name field when you add a new user entry.
Adding, Viewing, Editing, and Deleting Users Adding Users: Administrator’s Edit User Screen Edit User Screen Figure 1-5 illustrates the edit user screen that appears when the show administrator data option box is checked. Features that can be enabled for the user by the administrator Figure 1-5. Edit User Screen: Show Admin Data Name Specify the name of the user in alphanumeric characters. A unique name must be specified for each user.
Adding, Viewing, Editing, and Deleting Users Group This field is a text identifier to help you classify your users. Marshal Group Allows you to select a User Agent Marshal server group from the pull down menu. The list of marshal server groups in the pull down menu corresponds to the marshal server groups provisioned under servers/marshalServer/ serverType UserAgent. You can load balance users among different User Agent Marshal server groups.
Adding, Viewing, Editing, and Deleting Users JTAPI Check the Enabled option box to enable the JTAPI feature. With this feature enabled the user can place calls using a JTAPI User Agent. Note You must load a Java application, such as the JTAPI User Agent application that is bundled with VOCAL, to work with this JTAPI feature. ForwardAllCalls Option Box Check the Enabled option box to enable the ForwardAllCalls feature for the user.
Adding, Viewing, Editing, and Deleting Users Call Screening Option Box Check the Enabled option box to enable the Call Screening feature for the user. Note For version 1.3.0 of VOCAL, phone numbers entered for call screening must include the area code, regardless if they are local or long-distance phone numbers. Call Processing Language does not provide a pattern matching method that differentiates seven digit (local) phone numbers from ten digit (long-distance) numbers.
Adding, Viewing, Editing, and Deleting Users Call Return Option Box Check the Call Return option box to enable the Call Return feature for the user. Pull Down Menu The pull down menu allows you to select a Feature server group from the pull down menu. The list of Feature server groups in the pull down menu corresponds to the Feature server groups provisioned under servers/ featureServer/serverType CallReturn. You can load balance users among different Feature Server groups.
Adding, Viewing, Editing, and Deleting Users Viewing Users: Individually Introduction This section describes how to view records for individual users. If you have thousands of users loaded into your system, you will find that it is faster to load the data for individual users, or small groups of users, as required, rather than loading the data all users every time you login as an Administrator. Viewing individual records requires using the right-mouse-click menu, which is described below.
Adding, Viewing, Editing, and Deleting Users Screen Capture: Viewing A Single User Figure 1-6 illustrates selecting the data for a single user. 1. Select a record 2. Right mouse click, select View The data is displayed for the 3. Select display options Figure 1-6.
Adding, Viewing, Editing, and Deleting Users Screen Capture: Viewing Small Groups of Users Figure 1-7 illustrates selecting the data for a small group of users. 1. Select several records 2. Right mouse click, select View The data is displayed for the selected records. 3. Select display options Figure 1-7. Displaying Data for Small Groups of Users Load all Users 1-18 To view information for all users, select the Load all users button as shown in Figure 1-8 on page 1-25.
Adding, Viewing, Editing, and Deleting Users Viewing Users: Data Fields Descriptions Introduction Different data fields appear in the user configuration screen depending on the option boxes selected. Default Data Fields When none of the option boxes are checked, the user configuration screen displays these default data fields: Table 1-9. Default Data Field Field Admin Data Field Description Name Specifies the unique name of the user. User Group Specifies the group name that the user is grouped in.
Adding, Viewing, Editing, and Deleting Users Table 1-10. Admin Data Fields (Continued) Field 1-20 Description Forward Busy/No Ans. Group Indicates the name of the ForwardBusyNoAnswer Feature server group. Failure Case Indicates the number or address to forward calls to there is a problem with contacting the destination called party. Call Screen Enabled Indicates whether the Call Screen feature is enabled for the user: • Deselected: indicates that this feature is disabled for the user.
Adding, Viewing, Editing, and Deleting Users Table 1-10. Admin Data Fields (Continued) Field Description Call Return Enabled Indicates whether the Call Return feature is enabled for the user: • Deselected: indicates that this feature is disabled for the user. • Selected: indicates that this feature is enabled for the user. Call Return Group Indicates the name of the CallReturn Feature server group.
Adding, Viewing, Editing, and Deleting Users User Data Field When the Show user data option box is checked, these data field appear in addition to the default data fields: Table 1-11. User Data Fields Field 1-22 Description Forward All Set Indicates whether Forward All feature is set by the user: • OFF: indicates that call forwarding off. • ON: indicates that call forwarding is on and all calls are forward to a number specified by the user.
Adding, Viewing, Editing, and Deleting Users Table 1-11. User Data Fields (Continued) Field Description 900 # User Block Indicates whether 900 Number Block feature is set by the user: • Deselected: indicates that 900 Number block is not enabled by the user. 900 numbers are not blocked and user can dial 900 numbers. • Selected: indicates that 900 Number block is enabled by the user. 900 numbers are blocked and users cannot dial 900 numbers.
Adding, Viewing, Editing, and Deleting Users Viewing Users: All Users Introduction This section explains how to use the Load all users button and the option boxes to view user data. Overview For situations where you need to compare the data between users, you can click the Load all users button. This button activates a program that reads a flat file on the Provisioning Server, containing all user data, and displays the data in the GUI.
Adding, Viewing, Editing, and Deleting Users Load All Users Figure 1-8 shows the use of the Load all users button. 1. Click Load all users The data is displayed. 2. Select options 3. Use the horizontal scroll bar to see the other fields. Figure 1-8.
Adding, Viewing, Editing, and Deleting Users Finding Users Introduction You can highlight any of the users by clicking their record with the mouse. If you have thousands of users, the Find User utility will make your search easier. Procedure: Finding The Find button activates a program that automatically searches the Name column for the first match of your criteria as you type it in. For example, if you Users type a 6, the first name that starts with 6 will be highlighted.
Adding, Viewing, Editing, and Deleting Users Deleting Users Deleting User To delete a user or multiple users, follow these steps: Table 1-14. Procedure for Deleting Users Step Action 1 To select a user, left mouse click on a row in the table. To select multiple users, hold down the shift key while left mouse clicking the rows in the table. 2 Using the right mouse click, select Delete.
Adding, Viewing, Editing, and Deleting Users Editing Users: Administrator Controlled Introduction This section describes how to edit users. Procedure: Editing To edit a user, follow these steps: a User Table 1-15. Procedure for Editing User Step Action 1 Select the Show admin data option box. 2 Use the Left mouse click to select a user entry. 3 Right mouse click and select Edit. The Edit user screen appears. 4 Edit the fields and option boxes as required.
Adding, Viewing, Editing, and Deleting Users Editing User: Show Alias Introduction The aliases names associated with each users can be displayed using the show alias option box. User names with aliases appear in italics. What’s an Alias? An alias is another address or phone number by which a user can be reached. A telephone call directed to the alias will terminate at the user’s telephone.
Adding, Viewing, Editing, and Deleting Users Editing User Features: User Controlled Introduction The VOCAL system provides a web page for users to maintain some of their features. These features are call User Controlled Features and they include: • JTAPI • Forward all calls • Call blocking • Call screening • Caller ID blocking • Forward unanswered • Forward busy Procedure: Editing To edit a user, follow these steps: a User Table 1-17.
Adding, Viewing, Editing, and Deleting Users Editing User Feature: Edit User Screen Show User Data View Figure 1-10 illustrates the edit user screen that appears when Show user data option box checked and the edit right mouse option is used. This screen displays features that can be enabled by the user. This screen is provided for the administrator to view the user’s settings. If required, the administrator can modify the user’s setting. Figure 1-10.
Adding, Viewing, Editing, and Deleting Users Aliases This field displays aliases associated with this user. To add aliases for the user: 1) Right mouse click over the Alias area and select add. 2) Type the alias name for the user. To remove aliases, right mouse click the alias name and select remove. JTAPI The user can use a JTAPI dial pad to place calls if the JTAPI set option box is checked.
Adding, Viewing, Editing, and Deleting Users Call Screening The user can screen a call by name and number. To add numbers to screen: 1) Right mouse click near the name and number box. Select Add. 2) A Screen Calls From dialog box appears. Enter the name and number to screen. Click OK. The format is the user ID, for example, 7000. 3) If you enter “6” in the number field, then all numbers beginning with 6 will be screened.
Adding, Viewing, Editing, and Deleting Users 1-34
1HWZRUN 0DQDJHPHQW This chapter describes network management and statistics for the VOCAL system. Topic See Page SNMP Support . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . MIBs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . VOCAL SNMP GUI. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . VOCAL SNMP GUI Screen Elements . . . . . . . . . . . . . . . . . . . . . .
SNMP Support SNMP Support Overview VOCAL supports Simple Network Management Protocol (SNMP) monitoring from: • the VOCAL SNMP GUI - this supports monitoring of VOCAL server status. • a third party SNMP network manager. SNMP Version VOCAL supports SNMP version 2 (RFCs 1441 to 1452).
SNMP Support MIBs Introduction In a TCP/IP-based network, each device maintains a set of variables describing its state. In Simple Network Management Protocol (SNMP), these variables are known as objects, but these objects do not hold the same meaning as those within an object-oriented programming architecture. SNMP objects contain information about their state without any methods, other than the ability to read and write their values.
SNMP Support VOCAL Enterprise MIB 2-4 For more information refer to the /usr/local/vocal/proxies/netMgnt directory: • VOVIDA-LOCAL-GRP-MIB.txt • VOVIDA-NOTIFICATIONS-MIB.txt • VOVIDA-SERVERGRP-MIB.txt • VOVIDA-SOFTSWITCHSTATS-MIB.txt • VOVIDA-SUBSCRIBERSTATS-MIB.
SNMP Support VOCAL SNMP GUI Server Status Monitoring Each VOCAL system server sends (via multicast) heartbeat packets to its peers at a predefined interval. The Heartbeat Server monitors the exchange of heartbeat packets between VOCAL servers and sends server status trap messages to the network management system. The network management system displays server status on the VOCAL SNMP GUI. VOCAL SNMP GUI Figure 2-1 illustrates an example of the VOCAL SNMP GUI.
SNMP Support VOCAL SNMP GUI Screen Elements Hosts & Processes This frame displays the host server and indicates whether they are active (blue) or inactive (red). If a host server contains several processes, it will display a red ball if one or more of the processes is inactive. SNMP Controller The process controller allows you to start or stop the SNMP control process. To start or stop the process controller, follow these steps: 1) From the Host View, select a hostname and select a process.
)HDWXUHV This chapter describes features supported by the VOCAL system. Topic See Page Features. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A-2 Core System Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . A-3 Set-Based Features . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
Features Features Introduction This section describes the types of feature supported by the VOCAL system. Overview The VOCAL system supports two types of features—core system features and set-based features. Core System Features Core system features are those that involve rerouting calls under certain conditions, such as the called party is busy, or the phone number being called is forbidden.
Core System Features Core System Features Types of Core System Features There are two types of system features—calling features and called features. The calling features are assigned to the call originator. The called features are assigned to the calling destination.
Core System Features Call Return Call return allows the user to call back the last caller. The user dials *69 to dial up the last caller. Call Screening Call screening allows the user to block calls from a list of numbers. For example, when an screened number calls the user, the caller will receive a busy signal. When call screening is enabled the user’s phone set will not ring for a screened number. Note For version 1.3.
Set-Based Features Set-Based Features Definition Set based features are features that a user can enable from a phone set. These features are an example of how SIP-based networks are able to transfer much of its intelligence to its end-points. Many SIP IP phone sets have a variety of “smart” features.
Set-Based Features AdHoc Conferencing A-6 The Conference key on a phone set allows the user to set up a conference call with a number of people. To set up a conference call: • Call the first person. Press the conference button to place the first caller on hold. • Call the second person. Press the conference button to add the second caller to the call. • Repeat until all callers are added to the call.
6XSSRUWHG 6,3 0HVVDJHV Topic See Page SIP Request Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . B-2 SIP Response Messages . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
SIP Request Messages SIP Request Messages Supported SIP The VOCAL system supports these SIP request messages: Request Messages Table B-1. SIP Request Messages Descriptions SIP Request Messages B-2 Descriptions INVITE Indicates that the user or service is being invited to participate in a session. ACK Confirms that the client has received a final response to an INVITE request. BYE Indicate the user wishes to terminate the call.
SIP Response Messages SIP Response Messages SIP Response Messages Category The VOCAL system supports all SIP response messages: • 1xx Responses - Information Responses • 2xx Responses - Successful Responses • 3xx Responses - Redirection Responses • 4xx Responses - Request Failures Responses • 5xx Responses - Server Failure Responses • 6xx Responses - Global Failure Responses For More Information Refer to the SIP RFC 2543 for a list of the status codes and their reason codes: http://www.ietf.
SIP Response Messages • • • • • • • • • • • • • • 409 Conflict 410 Gone 411 Length Required 413 Request Entity Too Large 414 Request-URI Too Large 415 Unsupported Media Type 420 Bad Extension 480 Temporarily not available 481 Call Leg/Transaction Does Not Exist 482 Loop Detected 483 Too Many Hops 484 Address Incomplete 485 Ambiguous 486 Busy Here 5xx Responses 5xx SIP response message are server error responses: • 500 Internal Server Error • 501 Not Implemented • 502 Bad Gateway • 503 Service Unavailabl
&DOO )ORZV This chapter provides call flows diagram and IP trace logs for several call scenarios. Topic See Page SIP Phone: Registration. . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . Registration: Access List Authentication . . . . . . . . . . . . . . . . . . . . . . Registration: Digest Authentication . . . . . . . . . . . . . . . . . . . . . . . . . . C-3 C-4 C-6 SIP IP Phone to SIP IP Phone: Call Setup and Disconnect . . . . . . C-8 SIP IP Phone to Analog Phone via Gateway .
Topic (continued) See Page User Agent to User Agent: Consulted Transfer. . . . . . . . . . . . . . . . C-103 User Agent to User Agent: Blind Transfer . . . . . . . . . . . . . . . . . . . . C-122 JTAPI . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . C-140 Ad Hoc Conference Call Between User Agents. . . . . . . . . . . . . . . .
SIP Phone: Registration SIP Phone: Registration Call Scenario Figure C-1 illustrates a SIP phone registering with the Marshal server. Authentication Methods There are three registration methods, no authentication, access list authentication or digest authentication. Table C-1 shows the authentication criteria used by each method. Table C-1.
SIP Phone: Registration Registration: Access List Authentication Call Flow Diagram Figure C-2 shows a SIP IP phone registering with the Redirect server. The User Agent Marshal server is using the Access List authentication method. SIP Phone UA Marshal Redirect Server 1. REGISTER 2. REGISTER 3. 200 4. 200 Figure C-2. Call Flow Diagram: SIP Phone Registration Call Trace The following trace shows a SIP IP phone registering with the Redirect server.
SIP Phone: Registration Header: Expires: 3600 Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.180:5060->192.168.26.10:5060] Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To: Header: Call-ID: c2943000-1e262-14ae-2e323931@192.168.26.
SIP Phone: Registration Registration: Digest Authentication Call Flow Diagram Figure C-3 shows a SIP IP phone registering with the Redirect server. The User Agent Marshal server is using the Digest authentication method. SIP Phone UA Marshal Redirect Server 1. REGISTER 2. 401 3. REGISTER 4. REGISTER 5. 200 6. 200 Figure C-3. Call Flow Diagram: SIP IP Phone Registration — Digest Authentication Call Trace The following call trace shows a SIP IP phone registering with the Redirect server.
SIP Phone: Registration Header: Authorization: Digest username=”6711”,realm=”vovida.com”,uri=”sip:192.168.26.180”,response=”fee2efef60a99b4576c 0437947959deb”,nonce=”966645751”,algorithm=MD5 Header: Contact: Header: Expires: 3600 Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: REGISTER sip:@192.168.26.200:5060 SIP/2.0 [192.168.26.180:5060>192.168.26.
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect SIP IP Phone to SIP IP Phone: Call Setup and Disconnect Call Scenario Figure C-4 illustrates a call between two, on-network SIP IP phones. VOCAL System Redirect Server User Agent Marshal Server SIP Phone Calling Party SIP Phone Called Party Figure C-4. SIP Phone to SIP Phone Call Flow Diagrams Figures C-5 and C-6 show a successful call setup between two, on-network SIP IP phones. In this example, the called party terminates the call.
UA Marshal Redirect Server SIP Phone 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 302 8. ACK 9. INVITE 10. 100 11. 180 12. 180 13. 200 14. 200 15. ACK Figure C-5.
UA Marshal Redirect Server SIP Phone 16. ACK 17. BYE 18. BYE 19. 200 20. 200 Figure C-6.
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect Call Trace The following call trace shows a successful call setup between two, onnetwork IP phones. In this example, the called party terminates the call. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5120@192.168.36.180 SIP/2.0 [192.168.6.21:50623>192.168.36.180:5060] Header: Via: SIP/2.0/UDP 192.168.6.
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.36.200:5060->192.168.36.180:5060] Header: Via: SIP/2.0/UDP 192.168.36.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.6.21:5060 Header: From: Header: To:
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect ----------------------------------------------------------------sip-req: ACK sip:5120@192.168.36.200:5060;user=phone SIP/2.0 [192.168.36.180:5060>192.168.36.200:5060] Header: Via: SIP/2.0/UDP 192.168.36.180:5060;branch=3 Header: From: Header: To: Header: Call-ID: c2943000-e0563-2a1ce-2e323931@192.168.6.
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect Header: To: ;tag=c29430002e0620-0 Header: Call-ID: c2943000-e0563-2a1ce-2e323931@192.168.6.21 Header: CSeq: 100 INVITE Header: Server: Cisco IP Phone/ Rev. 1/ SIP enabled Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.6.20:50753->192.168.36.180:5060] Header: Via: SIP/2.
SIP IP Phone to SIP IP Phone: Call Setup and Disconnect Header: Call-ID: c2943000-e0563-2a1ce-2e323931@192.168.6.21 Header: Route: , Header: CSeq: 100 ACK Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: ACK sip:5120@192.168.6.20:5060 SIP/2.0 [192.168.36.180:5060>192.168.6.20:5060] Header: Via: SIP/2.
SIP IP Phone to Analog Phone via Gateway SIP IP Phone to Analog Phone via Gateway Call Scenario Figure C-7 illustrates a SIP phone to analog phone call made over an IP network via a gateway. VOCAL System Redirect Server PSTN User Agent Marshal Server Gateway Marshal Server Gateway SIP Phone Calling Party Analog Phone Called Party Figure C-7.
UA Marshal Redirect Server 5300 Marshal Cisco 5300 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 100 8. INVITE 9. 100 10. 180 11. 180 12. 180 13. 200 14. 200 15. 200 Figure C-8.
UA Marshal Redirect Server 5300 Marshal Cisco 5300 16. ACK 17. ACK 18. ACK 19. BYE 20. BYE 21. BYE 22. 200 23. 200 24. 200 Figure C-9.
SIP IP Phone to Analog Phone via Gateway Call Trace The following trace shows a call originating from an on-network SIP phone and being routed through a gateway to the PSTN. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:93831073@192.168.36.180 SIP/2.0 [192.168.6.20:50753>192.168.36.180:5060] Header: Via: SIP/2.0/UDP 192.168.6.20:5060 Header: From: sip:5120@192.168.6.
SIP IP Phone to Analog Phone via Gateway Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.36.200:5060>192.168.36.180:5060] Header: Via: SIP/2.0/UDP 192.168.36.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.6.20:5060 Header: From: Header: To:
SIP IP Phone to Analog Phone via Gateway sip-req: INVITE sip:93831073@192.168.16.210:5060;user=phone SIP/2.0 [192.168.36.110:5060->192.168.16.210:5060] Header: Via: SIP/2.0/UDP 192.168.36.110:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.36.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.6.20:5060 Header: From: Header: To: Header: Call-ID: c2943000-23e062-2e278-2e323931@192.168.6.
SIP IP Phone to Analog Phone via Gateway Header: Via: SIP/2.0/UDP 192.168.6.20:5060 Header: From: Header: To: Header: Call-ID: c2943000-23e062-2e278-2e323931@192.168.6.20 Header: CSeq: 100 INVITE Header: Server: Cisco VoIP Gateway/ IOS 12.
SIP IP Phone to Analog Phone via Gateway Header: Via: SIP/2.0/UDP 192.168.36.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.6.20:5060 Header: From: Header: To: ;tag=1AF49448-1D50 Header: Call-ID: c2943000-23e062-2e278-2e323931@192.168.6.20 Header: CSeq: 100 INVITE Header: Contact: Header: Record-Route: ,
SIP IP Phone to Analog Phone via Gateway Header: CSeq: 100 ACK Header: Route: Header: Proxy-Authorization: Basic VovidaClassXSwitch Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: ACK sip:93831073@192.168.16.210:5060 SIP/2.0 [192.168.36.110:5060>192.168.16.210:5060] Header: Via: SIP/2.0/UDP 192.168.36.110:5060;branch=4 Header: Via: SIP/2.
SIP IP Phone to Analog Phone via Gateway Header: Via: SIP/2.0/UDP 192.168.36.180:5060;branch=4,SIP/2.0/UDP 192.168.36.110:5060;branch=2,SIP/2.0/UDP 192.168.16.210:50110 Header: From: ;tag=1AF49448-1D50 Header: To: Header: Call-ID: c2943000-23e062-2e278-2e323931@192.168.6.20 Header: Server: Cisco IP Phone/ Rev.
SIP Phone to Phone via Gateway: Called Party is Busy SIP Phone to Phone via Gateway: Called Party is Busy Call Scenario Figure C-10 illustrates User A initiating a call to User B while User B is busy. VOCAL System Redirect Server PSTN User Agent Marshal Server Gateway Marshal Server Gateway User A Calling Party User B (Busy) Figure C-10.
UA Marshal Redirect Server 5300 Marshal Cisco 5300 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 100 8. INVITE 9. 100 10. 183 11. 183 12. 183 13. 200 14. 200 15. 200 Figure C-11.
UA Marshal Redirect Server 5300 Marshal Cisco 5300 16. ACK 17. ACK 18. ACK Figure C-12.
SIP Phone to Phone via Gateway: Called Party is Busy Call Trace The following call trace shows a call originating from an on-network SIP phone, being routed through a gateway to the PSTN, and returning a busy signal. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:93831069@192.168.26.180 SIP/2.0 [192.168.26.10:50373>192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.
SIP Phone to Phone via Gateway: Called Party is Busy Header: c=IN IP4 192.168.26.10 Header: t=0 0 Header: m=audio 26268 RTP/AVP 0 101 Header: a=rtpmap:0 pcmu/8000 Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.26.200:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.
SIP Phone to Phone via Gateway: Called Party is Busy Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:93831069@192.168.16.210:5060;user=phone SIP/2.0 [192.168.26.110:5060->192.168.16.210:5060] Header: Via: SIP/2.0/UDP 192.168.26.110:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From:
SIP Phone to Phone via Gateway: Called Party is Busy ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 183 Session Progress [192.168.26.110:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To:
SIP Phone to Phone via Gateway: Called Party is Busy ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.110:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To:
SIP Phone to Phone via Gateway: Called Party is Busy Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To: ;tag=25A5AD44-1FE9 Header: Call-ID: c2943000-3e262-e4dc-2e323931@192.168.26.10 Header: CSeq: 100 ACK Header: Route:
SIP IP Phone to SIP IP Phone: Forward All Calls SIP IP Phone to SIP IP Phone: Forward All Calls Call Scenario Figure C-13 illustrates the following call scenario: • User A initiates a call to User B • User B has call forwarding enabled • The call is forwarded to User C Note In this example, all SIP phones are connected to the same Marshal server.
UA Marshal Redirect Server Feature Server: FAC SIP Phone 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 100 8. 302 9. ACK 10. 302 11. ACK 12. INVITE 13. 100 14. INVITE 15. 302 Figure C-14.
UA Marshal Redirect Server Feature Server: FAC SIP Phone 16. ACK 17. INVITE 18. 100 19. 180 20. 180 21. 200 22. 200 23. ACK 24. ACK 25. BYE 26. BYE 27. 200 28. 200 Figure C-15.
SIP IP Phone to SIP IP Phone: Forward All Calls Call Trace The following call trace shows a call originating from an on-network SIP IP phone being forwarded to a call forward destination. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:6715@192.168.26.180 SIP/2.0 [192.168.26.10:50373>192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: sip:6711@192.168.26.
SIP IP Phone to SIP IP Phone: Forward All Calls Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.26.200:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To:
SIP IP Phone to SIP IP Phone: Forward All Calls Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To: Header: Call-ID: c2943000-ce262-1b5c2-2e323931@192.168.26.10 Header: CSeq: 100 INVITE Header: Contact: Header: Content-Length: 0 Header: CC-Redirect:
SIP IP Phone to SIP IP Phone: Forward All Calls Header: m=audio 30224 RTP/AVP 0 101 Header: a=rtpmap:0 pcmu/8000 Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 100 Trying [192.168.26.180:5060->192.168.26.10:5060] Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To:
SIP IP Phone to SIP IP Phone: Forward All Calls sip-req: INVITE sip:6716@192.168.26.12:5060 SIP/2.0 [192.168.26.180:5060>192.168.26.12:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To: Header: Call-ID: c2943000-de262-1b626-2e323931@192.168.26.
SIP IP Phone to SIP IP Phone: Forward All Calls Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4,SIP/2.0/UDP 192.168.26.180:5060;branch=2,SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To: ;tag=c2943000312610-0 Header: Call-ID: c2943000-de262-1b626-2e323931@192.168.26.10 Header: Server: Cisco IP Phone/ Rev. 1/ SIP enabled Header: Contact: sip:6716@192.168.26.12:5060 Header: Record-Route:
SIP IP Phone to SIP IP Phone: Forward All Calls sip-req: ACK sip:6716@192.168.26.12:5060 SIP/2.0 [192.168.26.180:5060>192.168.26.12:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To: ;tag=c2943000312610-0 Header: Call-ID: c2943000-de262-1b626-2e323931@192.168.26.
Phone to SIP Phone via Gateway: Call Screening Phone to SIP Phone via Gateway: Call Screening Call Scenario Figure C-16 illustrates the following call scenario: • User A initiates a call to User B • User B has call screening enabled • The feature server screene the call and returns a forbidden call message back to the gateway.
UA Marshal Redirect Server 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 100 8. INVITE 9. 302 10. ACK 11. 403 12. ACK 13. 403 14. ACK Figure C-17.
Phone to SIP Phone via Gateway: Call Screening Call Trace The following call trace shows a call, originating from an on-network SIP IP phone, being screened by the feature server. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:6711@192.168.26.180 SIP/2.0 [192.168.26.11:50783>192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.11:5060 Header: From: sip:6715@192.168.26.
Phone to SIP Phone via Gateway: Call Screening Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.26.200:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.26.11:5060 Header: From: Header: To:
Phone to SIP Phone via Gateway: Call Screening Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.11:5060 Header: From: Header: To: Header: Call-ID: c3943000-6978b-2995c-2e323931@192.168.26.11 Header: CSeq: 100 INVITE Header: Proxy-Authorization: Basic 123 Header: Expires: 180 Header: Record-Route: Header: Contact:
Phone to SIP Phone via Gateway: Call Screening Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 403 Forbidden [192.168.26.180:5060->192.168.26.11:5060] Header: Via: SIP/2.0/UDP 192.168.26.11:5060 Header: From: Header: To: Header: Call-ID: c3943000-6978b-2995c-2e323931@192.168.26.
SIP Phone to PSTN: Call Blocking SIP Phone to PSTN: Call Blocking Call Scenario Figure C-16 illustrates the following call scenario: • User A initiates a long distance or 1-900 number call • The VOCAL System blocks the call VOCAL System Redirect Server PSTN User Agent Marshal Server Call Blocking Feature Server Gateway Marshal Server Gateway 1-900 Called Party User A Calling Party Figure C-18.
UA Marshal Redirect Server Feature Server: Call Blocking 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 100 8. 403 9. ACK 10. 403 Figure C-19.
SIP Phone to PSTN: Call Blocking Call Trace The following call trace shows a call, originating from an on-network SiP IP phone, being blocked by the feature server. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:914083831059@192.168.26.180 SIP/2.0 [192.168.26.12:50420>192.168.26.180:6060] Header: Via: SIP/2.0/UDP 192.168.26.12:5060 Header: From: sip:6715@192.168.26.
SIP Phone to PSTN: Call Blocking Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.26.200:6060>192.168.26.180:6060] Header: Via: SIP/2.0/UDP 192.168.26.180:6060;branch=1 Header: Via: SIP/2.0/UDP 192.168.26.12:5060 Header: From: Header: To:
SIP Phone to PSTN: Call Blocking sip-res: SIP/2.0 403 Forbidden [192.168.26.220:6072->192.168.26.180:6060] Header: Via: SIP/2.0/UDP 192.168.26.180:6060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.12:5060 Header: From: Header: To: Header: Call-ID: c2943000-d061261-221d950-2e323931@192.168.26.
SIP IP Phone to SIP IP Phone: Call Return SIP IP Phone to SIP IP Phone: Call Return Call Scenario Figure C-20 illustrates the following call scenario: • User A dials *69 to determine the last number that was called, User B • User A calls User B VOCAL System Redirect Server User Agent Marshal Server Call Return Feature Server User A Calling Party User B Called Party Figure C-20.
UA Marshal Redirect Server Feature Server: Call Return 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 100 8. INVITE 9. 302 10. ACK 11. INVITE 12. 100 13. INVITE 14. 302 15. ACK Figure C-21.
UA Marshal Redirect Server Feature Server: Call Return 16. INVITE 17. 100 18. 180 19. 180 20. 180 21. 180 22. CANCEL 23. 200 24. CANCEL 25. 200 26. CANCEL 27. 200 28. CANCEL 29. 200 30. INVITE 31. 100 Figure C-22.
UA Marshal Redirect Server Feature Server: Call Return 32. INVITE 33. 302 34. ACK 35. INVITE 36. 100 37. 302 38. ACK 39. 302 40. ACK 41. INVITE 42. 100 43. INVITE 44. 302 45. ACK 46. INVITE 47. 100 Figure C-23.
UA Marshal Redirect Server Feature Server: Call Return 48. 180 49. 180 50. 200 51. 200 52. ACK 53. ACK 54. BYE 55. BYE 56. 200 57. 200 Figure C-24.
SIP IP Phone to SIP IP Phone: Call Return Call Trace The following call trace shows a call return request leading to an established call between two on-network SIP IP phones. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:6711@192.168.26.180 SIP/2.0 [192.168.26.11:50783>192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.11:5060 Header: From: sip:6715@192.168.26.
SIP IP Phone to SIP IP Phone: Call Return Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.26.200:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.26.11:5060 Header: From: Header: To:
SIP IP Phone to SIP IP Phone: Call Return Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.11:5060 Header: From: Header: To: Header: Call-ID: c3943000-2978b-aa0e-2e323931@192.168.26.11 Header: CSeq: 100 INVITE Header: Proxy-Authorization: Basic 123 Header: Expires: 180 Header: Record-Route: Header: Contact:
SIP IP Phone to SIP IP Phone: Call Return ----------------------------------------------------------------SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=CiscoSystemsSIP-IPPhone-UserAgent 25077 6500 IN IP4 192.168.26.11 Header: s=SIP Call Header: c=IN IP4 192.168.26.
SIP IP Phone to SIP IP Phone: Call Return Header: Contact: Header: Content-Length: 0 Header: CC-Redirect: ;redirreason=unconditional;redir-counter=0;redir-limit=99 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: ACK sip:6711@192.168.26.200:5060;user=phone SIP/2.0 [192.168.26.180:5060>192.168.26.200:5060] Header: Via: SIP/2.0/UDP 192.168.26.
SIP IP Phone to SIP IP Phone: Call Return Header: To: ;tag=c29430001e2620-0 Header: Call-ID: c3943000-2978b-aa0e-2e323931@192.168.26.11 Header: Server: Cisco IP Phone/ Rev. 1/ SIP enabled Header: CSeq: 100 INVITE Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 180 Ringing [192.168.26.180:5060->192.168.26.220:6074] Header: Via: SIP/2.0/UDP 192.
SIP IP Phone to SIP IP Phone: Call Return Header: To: Header: Call-ID: c3943000-2978b-aa0e-2e323931@192.168.26.11 Header: CSeq: 100 CANCEL Header: Proxy-Authorization: Basic 123 Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.220:6074->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.
SIP IP Phone to SIP IP Phone: Call Return Header: User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled Header: Accept: application/sdp Header: Contact: sip:6711@192.168.26.10:5060 Header: Content-Type: application/sdp Header: Content-Length: 219 ----------------------------------------------------------------SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=CiscoSystemsSIP-IPPhone-UserAgent 6339 9080 IN IP4 192.168.26.10 Header: s=SIP Call Header: c=IN IP4 192.
SIP IP Phone to SIP IP Phone: Call Return ----------------------------------------------------------------sip-req: ACK sip:*69@192.168.26.200:5060;user=phone SIP/2.0 [192.168.26.180:5060>192.168.26.200:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=1 Header: From: Header: To: Header: Call-ID: c2943000-2e262-b27e-2e323931@192.168.26.
SIP IP Phone to SIP IP Phone: Call Return Header: To: Header: Call-ID: c2943000-2e262-b27e-2e323931@192.168.26.10 Header: CSeq: 100 ACK Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.26.180:5060->192.168.26.10:5060] Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From:
SIP IP Phone to SIP IP Phone: Call Return sip-req: INVITE sip:6715@192.168.26.200:5060;user=phone SIP/2.0 [192.168.26.180:5060>192.168.26.200:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To: ;user=phone Header: Call-ID: c2943000-3e262-b4b1-2e323931@192.168.26.
SIP IP Phone to SIP IP Phone: Call Return ----------------------------------------------------------------Header: v=0 Header: o=CiscoSystemsSIP-IPPhone-UserAgent 8962 2811 IN IP4 192.168.26.10 Header: s=SIP Call Header: c=IN IP4 192.168.26.
SIP IP Phone to SIP IP Phone: Call Return Header: m=audio 29956 RTP/AVP 0 101 Header: a=rtpmap:0 pcmu/8000 Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.180:5060->192.168.26.10:5060] Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: Header: To:
SIP IP Phone to SIP IP Phone: Call Return Header: Route: , Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: BYE sip:6711@192.168.26.10:5060 SIP/2.0 [192.168.26.180:5060>192.168.26.10:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.
User Agent to User Agent: Call Waiting User Agent to User Agent: Call Waiting Call Scenario Figure C-25 illustrates the following call scenario: • User A calls User B • While Users A and B are in conversation, User C calls User A • User A is notified that another caller attempting to connect VOCAL System Redirect Server User Agent Marshal Server User A Calling Party Call Waiting Feature Server User B Called Party User C New Calling Party Figure C-25.
UA Marshal Redirect Server VOCAL User Agent 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 302 8. ACK 9. INVITE 10. 180 11. 180 12. 200 13. 200 14. ACK 15. ACK Figure C-26.
UA Marshal Redirect Server VOCAL User Agent 16. INVITE 17. 100 18. INVITE 19. 302 20. ACK 21. INVITE 22. 302 23. ACK 24. INVITE 25. 180 26. INVITE 27. 200 28. 200 29. 200 30. ACK 31. ACK Figure C-27.
UA Marshal Redirect Server VOCAL User Agent 32. INVITE 33. 200 34. INVITE 35. 200 36. ACK 37. ACK 38. BYE 39. 200 40. BYE 41. BYE 42. 200 43. 200 Figure C-28.
User Agent to User Agent: Call Waiting Call Trace The following call trace shows a third party attempting to connect to a phone that is engaged in conversation with another phone. ================================================================= A calls B ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5220@192.168.66.180:5060;user=phone SIP/2.0 [192.168.66.1:5060>192.168.66.
User Agent to User Agent: Call Waiting Header: F ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.66.200:5060->192.168.66.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5220
User Agent to User Agent: Call Waiting ----------------------------------------------------------------sip-req: ACK sip:5220@192.168.66.200:5060;user=phone SIP/2.0 [192.168.66.180:5060>192.168.66.200:5060] Header: Via: SIP/2.0/UDP 192.168.66.180:5060;branch=3 Header: From: UserAgent Header: To: 5220 Header: Call-ID: 4732a6465cfdffbdc0d38708c0728708@192.168.66.
User Agent to User Agent: Call Waiting Header: From: UserAgent Header: To: 5220 Header: Call-ID: 4732a6465cfdffbdc0d38708c0728708@192.168.66.1 Header: CSeq: 1 INVITE Header: Contact: Header: Record-Route: ,
User Agent to User Agent: Call Waiting Header: From: UserAgent Header: To: 5220 Header: Call-ID: 4732a6465cfdffbdc0d38708c0728708@192.168.66.
User Agent to User Agent: Call Waiting Header: m=audio 60335 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 Header: * ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.66.200:5060->192.168.66.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.66.3:5060 Header: From: UserAgent
User Agent to User Agent: Call Waiting Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: ACK sip:5220@192.168.66.200:5060;user=phone SIP/2.0 [192.168.66.180:5060>192.168.66.200:5060] Header: Via: SIP/2.0/UDP 192.168.66.180:5060;branch=3 Header: From: UserAgent Header: To: 5220
User Agent to User Agent: Call Waiting Header: Route: ,, Header: Content-Type: application/sdp Header: Content-Length: 126 ----------------------------------------------------------------SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 944563072 944563072 IN IP4 192.168.66.2 Header: s=VOVIDA Session Header: c=IN IP4 0.0.0.
User Agent to User Agent: Call Waiting sip-res: SIP/2.0 200 OK [192.168.66.180:5060>192.168.66.3:5060] Header: Via: SIP/2.0/UDP 192.168.66.3:5060 Header: From: UserAgent Header: To: 5220 Header: Call-ID: 52f78a2b5cfdffbd60c98708a0688708@192.168.66.3 Header: CSeq: 1 INVITE Header: Contact: Header: Record-Route: ,
User Agent to User Agent: Call Waiting SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 1573383876 1573383876 IN IP4 192.168.66.2 Header: s=VOVIDA Session Header: c=IN IP4 0.0.0.0 Header: t=3177769031 0 Header: m=audio 3456 RTP/AVP 0 Header: ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.66.3:5060->192.168.66.
User Agent to User Agent: Call Waiting Header: Content-Type: application/sdp Header: Content-Type: application/sdp Header: Content-Length: 168 Header: Content-Length: 168 ----------------------------------------------------------------SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 1528076688 1528076688 IN IP4 192.168.66.1 Header: s=VOVIDA Session Header: c=IN IP4 192.168.66.
User Agent to User Agent: Call Waiting sip-req: BYE sip:5220@192.168.66.180:5060;maddr=192.168.66.180 SIP/2.0 [192.168.66.2:5060->192.168.66.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.2:5060 Header: From: 5220 Header: To: UserAgent Header: Call-ID: 4732a6465cfdffbdc0d38708c0728708@192.168.66.1 Header: CSeq: 4 BYE Header: Route: ,
SIP IP Phone to SIP IP Phone: Forward to Voice Mail SIP IP Phone to SIP IP Phone: Forward to Voice Mail Call Scenario Figure C-29 illustrates the following call scenario: • User A calls User B • User B does not answer the call • The call is forwarded to the voice mail feature server VOCAL System Redirect Server User Agent Marshal Server User A Calling Party Call Forward No Answer Feature Server Voice Mail Feature Server User B Called Party Figure C-29.
UA Marshal Redirect Server Feature Server: FNA 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 100 8. INVITE 9. 100 10. INVITE 11. 302 12. ACK 13. INVITE 14. 100 15. 180 Figure C-30.
UA Marshal Redirect Server Feature Server: FNA 16. 180 17. 180 18. 180 19. CANCEL 20. 200 21. CANCEL 22. 200 23. 180 24. 180 25. 200 26. 200 27. ACK 28. ACK 29. BYE 30. BYE 31. 200 Figure C-31.
UA Marshal Redirect Server Feature Server: FNA 32. 200 Figure C-32.
SIP IP Phone to SIP IP Phone: Forward to Voice Mail Call Trace The following call trace shows a SIP IP phone attempting to call another onnetwork SIP IP phone. The second phone is unanswered and the call is reinitiated with the Voice Mail server. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5210@192.168.56.180 SIP/2.0 [192.168.10.18:50443>192.168.56.180:5060] Header: Via: SIP/2.0/UDP 192.
SIP IP Phone to SIP IP Phone: Forward to Voice Mail Header: t=0 0 Header: m=audio 23994 RTP/AVP 0 101 Header: a=rtpmap:0 pcmu/8000 Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 302 Moved Temporarily [192.168.56.200:5060->192.168.56.180:5060] Header: Via: SIP/2.0/UDP 192.168.56.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.10.
SIP IP Phone to SIP IP Phone: Forward to Voice Mail SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5210@192.168.56.180:5060 SIP/2.0 [192.168.56.220:5074>192.168.56.180:5060] Header: Via: SIP/2.0/UDP 192.168.56.220:5074;branch=102 Header: Via: SIP/2.0/UDP 192.168.56.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.10.18:5060 Header: From: Header: To:
SIP IP Phone to SIP IP Phone: Forward to Voice Mail ----------------------------------------------------------------Header: v=0 Header: o=CiscoSystemsSIP-IPPhone-UserAgent 25678 28140 IN IP4 192.168.10.18 Header: s=SIP Call Header: c=IN IP4 192.168.10.
SIP IP Phone to SIP IP Phone: Forward to Voice Mail Header: a=rtpmap:0 pcmu/8000 Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 100 Trying [192.168.10.10:50783->192.168.56.180:5060] Header: Via: SIP/2.0/UDP 192.168.56.180:5060;branch=4,SIP/2.0/UDP 192.168.56.220:5074;branch=102,SIP/2.0/UDP 192.168.56.180:5060;branch=2,SIP/2.
SIP IP Phone to SIP IP Phone: Forward to Voice Mail Header: Via: SIP/2.0/UDP 192.168.56.220:5074;branch=102 Header: From: Header: To: Header: Call-ID: c2943000-482e7-61caa-2e323931@192.168.10.
SIP IP Phone to SIP IP Phone: Forward to Voice Mail Header: Contact: Header: Record-Route: ,,
SIP IP Phone to SIP IP Phone: Forward to Voice Mail Header: Route: , Header: Proxy-Authorization: Basic VovidaClassXSwitch Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: BYE sip:5210@192.168.56.180:5060 SIP/2.0 [192.168.10.18:50443>192.168.56.180:5060] Header: Via: SIP/2.0/UDP 192.168.10.
User Agent to User Agent: Consulted Transfer User Agent to User Agent: Consulted Transfer Call Scenario Figure C-33 illustrates the following call scenario: • User A calls User B. • User B puts User B on hold and notifies User C about User A’s call. • User B transfers the call to User C. VOCAL System Redirect Server User Agent Marshal Server User A Calling Party Call Transfer Feature Server User B Called Party User C Call Transferred Party Figure C-33.
UA Marshal VOCAL User Agent VOCAL User Agent 1. INVITE 2. 100 3. INVITE 4. 200 5. 200 6. ACK 7. ACK 8. INVITE 9. 100 10. INVITE 11. 200 12. 200 13. ACK 14. ACK 15. INVITE Figure C-34.
UA Marshal VOCAL User Agent VOCAL User Agent 16. 100 17. INVITE 18. 180 19. 180 20. 200 21. 200 22. ACK 23. ACK 24. INVITE 25. 100 26. INVITE 27. 200 28. 200 29. ACK 30. ACK 31. TRANSFER C-105 Figure C-35.
UA Marshal VOCAL User Agent VOCAL User Agent 32. 100 33. TRANSFER 34. 100 35. INVITE 36. 100 37. INVITE 38. 200 39. 200 40. 200 41. 200 42. ACK 43. ACK 44. BYE 45. BYE 46. BYE 47. 200 Figure C-36.
UA Marshal VOCAL User Agent VOCAL User Agent 48. BYE 49. 200 50. 200 51. 200 52. BYE 53. BYE 54. 200 55. 200 Figure C-37.
User Agent to User Agent: Consulted Transfer Call Trace The following call trace shows a consulted call transfer between two SIP IP phones. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5221@192.168.26.180:5060;user=phone SIP/2.0 [192.168.66.1:5060>192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent
User Agent to User Agent: Consulted Transfer Header: ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.66.2:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221
User Agent to User Agent: Consulted Transfer SIP Headers ----------------------------------------------------------------sip-req: ACK sip:5221@192.168.66.2:5060 SIP/2.0 [192.168.26.180:5060>192.168.66.2:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221
User Agent to User Agent: Consulted Transfer Header: c=IN IP4 0.0.0.0 Header: t=3174939344 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 Header: jË ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.66.2:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.
User Agent to User Agent: Consulted Transfer ----------------------------------------------------------------sip-req: ACK sip:5221@192.168.66.2:5060 SIP/2.0 [192.168.26.180:5060>192.168.66.2:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221 Header: Call-ID: 1614879410580032@192.168.
User Agent to User Agent: Consulted Transfer Header: o=- 1113249245 1113249245 IN IP4 192.168.66.1 Header: s=VOVIDA Session Header: c=IN IP4 192.168.66.1 Header: t=3174939385 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 Header: om ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 180 Ringing [192.168.66.3:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.
User Agent to User Agent: Consulted Transfer Header: Record-Route: , Header: Content-Type: application/sdp Header: Content-Length: 168 ----------------------------------------------------------------SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 1113249245 1113249245 IN IP4 192.168.66.3 Header: s=VOVIDA Session Header: c=IN IP4 192.168.66.
User Agent to User Agent: Consulted Transfer SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 100 Trying [192.168.26.180:5060->192.168.66.1:5060] Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5222 Header: Call-ID: 1614879410580032@192.168.66.
User Agent to User Agent: Consulted Transfer Header: Contact: Header: Content-Type: application/sdp Header: Content-Length: 166 ----------------------------------------------------------------SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 698447251 698447251 IN IP4 192.168.66.3 Header: s=VOVIDA Session Header: c=IN IP4 192.168.66.
User Agent to User Agent: Consulted Transfer sip-req: TRANSFER sip:5221@192.168.66.2:5060 SIP/2.0 [192.168.26.180:5060>192.168.66.2:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221 Header: Transfer-To: Header: Call-ID: 1614879410580032@192.
User Agent to User Agent: Consulted Transfer Header: CSeq: 1 INVITE Header: Subject: VovidaINVITE Header: Record-Route: , Header: Contact:
User Agent to User Agent: Consulted Transfer Header: s=VOVIDA Session Header: c=IN IP4 192.168.66.3 Header: t=3174939395 0 Header: m=audio 23466 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 Header: bc ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.66.2:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.
User Agent to User Agent: Consulted Transfer ----------------------------------------------------------------sip-req: BYE sip:5222@192.168.26.180:5060;maddr=192.168.26.180 SIP/2.0 [192.168.66.1:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5222 Header: Call-ID: 1614879410580032@192.168.66.1 Header: CSeq: 5 BYE Header: Route:
User Agent to User Agent: Consulted Transfer Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.180:5060->192.168.66.1:5060] Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5222 Header: Call-ID: 1614879410580032@192.168.66.
User Agent to User Agent: Blind Transfer User Agent to User Agent: Blind Transfer Call Scenario Figure C-38 illustrates a call scenario in which: • User A calls User B. • User B transfers the call to User C without consulting User C. VOCAL System Redirect Server User Agent Marshal Server User A Calling Party Call Transfer Feature Server User B Called Party User C Call Transferred Party Figure C-38.
UA Marshal VOCAL User Agent VOCAL User Agent 1. INVITE 2. 100 3. INVITE 4. 180 5. 180 6. 200 7. 200 8. ACK 9. ACK 10. INVITE 11. 100 12. INVITE 13. 200 14. 200 15. ACK Figure C-39.
UA Marshal VOCAL User Agent VOCAL User Agent 16. ACK 17. INVITE 18. 100 19. INVITE 20. 180 21. 180 22. INVITE 23. 100 24. INVITE 25. 180 26. 180 27. TRANSFER 28. 100 29. TRANSFER 30. 100 31. INVITE Figure C-40.
UA Marshal VOCAL User Agent VOCAL User Agent 32. 100 33. INVITE 34. 180 35. 180 36. 200 37. 200 38. BYE 39. CANCEL 40. 200 41. BYE 42. 200 43. CANCEL 44. 200 45. 200 46. 200 47. 200 C-125 Figure C-41.
UA Marshal VOCAL User Agent VOCAL User Agent 48. ACK 49. ACK 50. BYE 51. BYE 52. 200 53. 200 Figure C-42.
User Agent to User Agent: Blind Transfer Call Trace The following call trace shows an unconsulted call transfer. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5221@192.168.26.180:5060;user=phone SIP/2.0 [192.168.66.1:5060>192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221
User Agent to User Agent: Blind Transfer ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 180 Ringing [192.168.66.2:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221
User Agent to User Agent: Blind Transfer Header: s=VOVIDA Session Header: c=IN IP4 192.168.66.2 Header: t=3174939460 0 Header: m=audio 23466 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 Header: }! ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: ACK sip:5221@192.168.26.180:5060;maddr=192.168.26.180 SIP/2.0 [192.168.66.1:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.
User Agent to User Agent: Blind Transfer ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5221@192.168.66.2:5060 SIP/2.0 [192.168.26.180:5060>192.168.66.2:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent
User Agent to User Agent: Blind Transfer Header: c=IN IP4 192.168.66.2 Header: t=3174939460 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 Header: ‚ ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: ACK sip:5221@192.168.26.180:5060;maddr=192.168.26.180 SIP/2.0 [192.168.66.1:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.
User Agent to User Agent: Blind Transfer ----------------------------------------------------------------sip-req: INVITE sip:5222@192.168.66.3:5060 SIP/2.0 [192.168.26.180:5060>192.168.66.3:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5222 Header: Call-ID: 507936623238514@192.168.
User Agent to User Agent: Blind Transfer Header: s=VOVIDA Session Header: c=IN IP4 0.0.0.0 Header: t=3174939482 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 Header: ÄÌ ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 100 Trying [192.168.26.180:5060->192.168.66.1:5060] Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent
User Agent to User Agent: Blind Transfer ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: TRANSFER sip:5221@192.168.26.180:5060;maddr=192.168.26.180 SIP/2.0 [192.168.66.1:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221 Header: Transfer-To:
User Agent to User Agent: Blind Transfer Header: o=- 1986829226 1986829226 IN IP4 192.168.66.2 Header: s=VOVIDA Session Header: c=IN IP4 192.168.66.2 Header: t=3174939488 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 100 Trying [192.168.26.180:5060->192.168.66.2:5060] Header: Via: SIP/2.0/UDP 192.168.66.
User Agent to User Agent: Blind Transfer ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.66.2:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221
User Agent to User Agent: Blind Transfer SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.66.2:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.1:5060 Header: From: UserAgent Header: To: 5221 Header: Call-ID: 507936623238514@192.168.66.
User Agent to User Agent: Blind Transfer Header: ÷Ï ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.180:5060->192.168.66.2:5060] Header: Via: SIP/2.0/UDP 192.168.66.2:5060 Header: From: UserAgent Header: To: Header: Call-ID: 507936623238514@192.168.66.1 Header: CSeq: 1 INVITE Header: Contact:
User Agent to User Agent: Blind Transfer sip-req: BYE sip:5222@192.168.66.3:5060 SIP/2.0 [192.168.26.180:5060>192.168.66.3:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.66.2:5060 Header: From: UserAgent Header: To: Header: Call-ID: 507936623238514@192.168.66.
JTAPI JTAPI Call Scenario Figure C-43 illustrates the following call scenario: • A user uses a JTAPI User Agent on a PC to remotely instruct SIP Phone A to call SIP Phone B. VOCAL System Redirect Server User Agent Marshal Server JTAPI Feature Server PC with JTAPI UA Controlling Party SIP Phone A Calling Party SIP Phone B Called Party Figure C-43. JTAPI Call Flow Diagram C-140 Figures C-44, C-45, C-46 and C-47 show third party call control through a JTAPI server.
Redirect Server UA Marshal 1. INVITE 2. 302 3. ACK 4. INVITE 5. 100 6. INVITE 7. 302 8. ACK 9. INVITE 10. 180 11. 180 12. 200 13. 200 14. ACK 15. INVITE Figure C-44.
Redirect Server UA Marshal 16. 302 17. 200 18. 200 19. ACK 20. ACK 21. 100 22. 100 23. INVITE 24. 302 25. INVITE 26. 302 27. ACK 28. INVITE 29. 100 30. 302 31. ACK Figure C-45.
Redirect Server UA Marshal 32. 100 33. 180 34. 180 35. 180 36. 180 37. 200 38. 200 39. BYE 40. BYE 41. 200 42. 200 43. 200 44. 200 45. 200 46. 200 47. ACK C-143 Figure C-46.
Redirect Server UA Marshal 48. ACK 49. ACK 50. 200 51. ACK 52. ACK 53. BYE 54. BYE 55. BYE 56. BYE 57. 200 58. 200 59. 200 60. 200 Figure C-47.
JTAPI Call Trace The following call trace shows third party call control through a JTAPI server. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:6710@192.168.26.200:5060;user=phone SIP/2.0 [192.168.5.11:25060>192.168.26.200:5060] Header: Via: SIP/2.0/UDP 192.168.5.11:25060;branch=201 Header: From: Header: To: 6710
JTAPI Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:6710@192.168.26.200:5060;user=phone SIP/2.0 [192.168.26.180:5060>192.168.26.200:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=3 Header: Via: SIP/2.0/UDP 192.168.5.11:25060;branch=301 Header: From: Header: To: 6710
JTAPI Header: Via: SIP/2.0/UDP 192.168.5.11:25060;branch=301 Header: From: Header: To: 6710 Header: Call-ID: 1087893762978930@192.168.5.11 Header: CSeq: 1 INVITE Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.22.36:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.
JTAPI SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 1735072859 1735072859 IN IP4 192.168.22.36 Header: s=VOVIDA Session Header: c=IN IP4 192.168.22.36 Header: t=3174942917 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:6710@192.168.26.180:5060 SIP/2.0 [192.168.
JTAPI Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.180:5060->192.168.5.11:25060] Header: Via: SIP/2.0/UDP 192.168.5.11:25060;branch=301 Header: From: Header: To: 6710 Header: Call-ID: 1087893762978930@192.168.5.
JTAPI Header: Call-ID: 1087893762978930@192.168.5.11 Header: CSeq: 2 TRANSFER Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:6711@192.168.26.180:5060;user=phone SIP/2.0 [192.168.22.36:5060>192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.22.36:5060 Header: From: UserAgent Header: To:
JTAPI Header: Via: SIP/2.0/UDP 192.168.5.11:15060;branch=202 Header: From: UserAgent Header: To: 6711 Header: Call-ID: 458898268105186@192.168.5.11 Header: CSeq: 1 INVITE Header: Contact: Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: ACK sip:6711@192.168.26.
JTAPI sip-req: ACK sip:6711@192.168.26.200:5060;user=phone SIP/2.0 [192.168.26.180:5060>192.168.26.200:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=3 Header: From: UserAgent Header: To: 6711 Header: Call-ID: 458898268105186@192.168.5.
JTAPI sip-res: SIP/2.0 200 OK [192.168.22.36:5060->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.5.11:25060 Header: From: Header: To: 6710 Header: Call-ID: 1087893762978930@192.168.5.
JTAPI Header: From: UserAgent Header: To: 6711;tag=c29430003e2620-0 Header: Call-ID: 458898268105186@192.168.5.11 Header: Server: Cisco IP Phone/ Rev. 1/ SIP enabled Header: Contact: sip:6711@192.168.26.10:5060 Header: Record-Route:
JTAPI Header: o=CiscoSystemsSIP-IPPhone-UserAgent 26487 28247 IN IP4 192.168.26.10 Header: s=SIP Call Header: c=IN IP4 192.168.26.10 Header: t=0 0 Header: m=audio 23206 RTP/AVP 0 101 Header: a=rtpmap:0 pcmu/8000 Header: a=rtpmap:101 telephone-event/8000 Header: a=fmtp:101 0-11 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.180:5060->192.168.22.36:5060] Header: Via: SIP/2.
JTAPI Header: CSeq: 1 ACK Header: Route: Header: Proxy-Authorization: Basic vovidaClassXswitch Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.10:50373->192.168.26.180:5060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=4,SIP/2.0/UDP 192.168.5.11:15060;branch=302 Header: From: UserAgent
JTAPI Header: CSeq: 2 BYE Header: Route: Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: BYE sip:js@192.168.5.11:15060 SIP/2.0 [192.168.26.180:5060>192.168.5.11:15060] Header: Via: SIP/2.0/UDP 192.168.26.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.26.10:5060 Header: From: 6711
JTAPI Header: From: 6711;tag=c29430003e2620-0 Header: To: UserAgent Header: Call-ID: 458898268105186@192.168.5.11 Header: CSeq: 2 BYE Header: Contact: Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.26.180:5060->192.168.26.10:5060] Header: Via: SIP/2.0/UDP 192.
Ad Hoc Conference Call Between User Agents Ad Hoc Conference Call Between User Agents Call Scenario Figure C-48 illustrates the following call scenario: • User A calls User B • User A brings User C into the conversation via ad hoc conferencing. VOCAL System Redirect Server User Agent Marshal Server User A Calling Party Conference Bridge Marshal Server Conference Bridge User B Called Party User C Conferenced Party Figure C-48.
UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 1. INVITE 2. 100 3. INVITE 4. 302 5. ACK 6. INVITE 7. 302 8. ACK 9. INVITE 10. 180 11. 180 12. 200 13. 200 14. ACK 15. ACK Figure C-49.
UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 16. INVITE 17. 100 18. INVITE 19. 200 20. 200 21. ACK 22. ACK 23. INVITE 24. 100 25. INVITE 26. 302 27. ACK 28. INVITE 29. 302 30. ACK 31. INVITE C-161 Figure C-50.
UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 32. 180 33. 180 34. 200 35. 200 36. ACK 37. ACK 38. INVITE 39. 100 40. INVITE 41. 200 42. 200 43. ACK 44. ACK 45. TRANSFER 46. 100 47. TRANSFER Figure C-51.
UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 48. 100 49. INVITE 50. 100 51. INVITE 52. 302 53. ACK 54. TRANSFER 55. INVITE 56. 100 57. 200 58. 200 59. 200 60. ACK 61. BYE 62. ACK 63. BYE C-163 Figure C-52.
UA Marshal Redirect Server VOCAL User Agent VOCAL User Agent 64. 200 65. TRANSFER 66. INVITE 67. 100 68. 200 69. 200 70. 200 71. ACK 72. BYE 73. ACK 74. BYE Figure C-53.
Ad Hoc Conference Call Between User Agents Call Trace The following call trace shows an ad hoc conference call between three users. ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5203@192.168.46.180:5060;user=phone SIP/2.0 [192.168.46.1:5060>192.168.46.180:5060] Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour
Ad Hoc Conference Call Between User Agents sip-res: SIP/2.0 302 Moved Temporarily [192.168.46.200:5060>192.168.46.180:5060] Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=1 Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 5203 Header: Call-ID: 64133833318102@192.168.46.1 Header: CSeq: 1 INVITE Header: Contact:
Ad Hoc Conference Call Between User Agents Header: From: seymour Header: To: 5203 Header: Call-ID: 64133833318102@192.168.46.1 Header: CSeq: 1 ACK Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5203@192.168.46.3:5060 SIP/2.0 [192.168.46.180:5060>192.168.46.3:5060] Header: Via: SIP/2.
Ad Hoc Conference Call Between User Agents Header: Record-Route: , Header: Content-Type: application/sdp Header: Content-Length: 168 ----------------------------------------------------------------SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 1554681096 1554681096 IN IP4 192.168.46.3 Header: s=VOVIDA Session Header: c=IN IP4 192.168.46.
Ad Hoc Conference Call Between User Agents ================================================================= First call established ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5203@192.168.46.180:5060;maddr=192.168.46.180 SIP/2.0 [192.168.46.1:5060->192.168.46.180:5060] Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour
Ad Hoc Conference Call Between User Agents Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 5203 Header: Call-ID: 64133833318102@192.168.46.1 Header: CSeq: 2 INVITE Header: Contact:
Ad Hoc Conference Call Between User Agents ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5202@192.168.46.180:5060;user=phone SIP/2.0 [192.168.46.1:5060>192.168.46.180:5060] Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 5202 Header: Call-ID: 64133833318102@192.168.46.
Ad Hoc Conference Call Between User Agents Header: To: 5202 Header: Call-ID: 64133833318102@192.168.46.1 Header: CSeq: 3 INVITE Header: Contact: Header: Content-Length: 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: ACK sip:5202@192.168.46.200:5060;user=phone SIP/2.0 [192.168.46.180:5060>192.168.46.200:5060] Header: Via: SIP/2.
Ad Hoc Conference Call Between User Agents ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: INVITE sip:5202@192.168.46.2:5060 SIP/2.0 [192.168.46.180:5060>192.168.46.2:5060] Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour
Ad Hoc Conference Call Between User Agents SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 979501686 979501686 IN IP4 192.168.46.2 Header: s=VOVIDA Session Header: c=IN IP4 192.168.46.2 Header: t=3177798665 0 Header: m=audio 23466 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.
Ad Hoc Conference Call Between User Agents sip-req: INVITE sip:5202@192.168.46.180:5060;maddr=192.168.46.180 SIP/2.0 [192.168.46.1:5060->192.168.46.180:5060] Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 5202 Header: Call-ID: 64133833318102@192.168.46.1 Header: CSeq: 4 INVITE Header: Route: ,
Ad Hoc Conference Call Between User Agents Header: Content-Length: 168 ----------------------------------------------------------------SDP Headers ----------------------------------------------------------------Header: v=0 Header: o=- 1554681096 1554681096 IN IP4 192.168.46.2 Header: s=VOVIDA Session Header: c=IN IP4 192.168.46.
Ad Hoc Conference Call Between User Agents Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 5203 Header: Transfer-To: Header: Call-ID: 64133833318102@192.168.46.1 Header: CSeq: 2 TRANSFER Header: Require: cc Transfer Header: Route: ,
Ad Hoc Conference Call Between User Agents Header: t=3177798665 0 Header: m=audio 23456 RTP/AVP 0 Header: a=rtpmap:0 PCMU/8000 Header: a=ptime:20 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 100 Trying [192.168.46.180:5060->192.168.46.1:5060] Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 818883831000
Ad Hoc Conference Call Between User Agents ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: TRANSFER sip:5203@192.168.46.3:5060 SIP/2.0 [192.168.46.180:5060>192.168.46.3:5060] Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour
Ad Hoc Conference Call Between User Agents ----------------------------------------------------------------Header: v=0 Header: o=CiscoSystemsSIP-GW-UserAgent 1397 1625 IN IP4 192.168.5.169 Header: s=SIP Call Header: c=IN IP4 192.168.5.169 Header: t=0 0 Header: m=audio 20246 RTP/AVP 0 ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-res: SIP/2.0 200 OK [192.168.46.3:5060->192.168.46.180:5060] Header: Via: SIP/2.
Ad Hoc Conference Call Between User Agents sip-req: BYE sip:5203@192.168.46.3:5060 SIP/2.0 [192.168.46.180:5060>192.168.46.3:5060] Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 5203 Header: Call-ID: 64133833318102@192.168.46.
Ad Hoc Conference Call Between User Agents Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=4 Header: Via: SIP/2.0/UDP 192.168.46.180:5060;branch=2 Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 5202 Header: Call-ID: 64133833318102@192.168.46.
Ad Hoc Conference Call Between User Agents ----------------------------------------------------------------SIP Headers ----------------------------------------------------------------sip-req: BYE sip:5203@192.168.46.180:5060;maddr=192.168.46.180 SIP/2.0 [192.168.46.1:5060->192.168.46.180:5060] Header: Via: SIP/2.0/UDP 192.168.46.1:5060 Header: From: seymour Header: To: 5203 Header: Call-ID: 64133833318102@192.168.46.
Ad Hoc Conference Call Between User Agents C-184
Index Symbols *69 call flow C-56 Numerics 1xx and 2xx B-3 3xx Responses B-3 4xx Responses B-3 5xx Responses B-4 6xx Responses B-4 900 # Admin Block 1-20 900 # User Block 1-23 A Access Level 1-4 Access list C-4 ACK B-2 adding users, see New Adhoc conferencing A-6 Administrator 1-4 alias 1-29 Aliases 1-32 Authentication access list C-4 digest C-6 Authentication Type 1-21 B BYE B-2 C Call Block Enabled 1-20 Call Block Group 1-20 Call blocking 1-32, A-3 Call flow *69 C-56 ad hoc conference call C-159 analog ph
Index (Continued) Long Distance Admin Block 1-20 Long Distance User Block 1-22 Marshal 1-19 Name 1-19 Password 1-21 Static Reg Enabled 1-21 Terminating Host 1-21 Terminating Port 1-21 User Group 1-19 Delete 1-8 procedure for deleting users 1-27 Digest C-6 E Edit 1-8 multiple users 1-28 users 1-28 Edit user Aliases 1-32 call blocking 1-32 call screening 1-33 caller ID blocking 1-33 forward all calls 1-32 forward no answer busy 1-33 JTAPI 1-32 Edit User Screen 1-31 F Failure Case 1-20 Features core system fea
Index (Continued) R U REGISTER B-2 Registration C-3 Right mouse click menu 1-16 User Configuration screen buttons 1-6 data field descriptions 1-19 editing: show alias 1-29 features: user controlled 1-30 User Configuration Admin data fields 900 # Admin Block 1-20 Authentication Type 1-21 Call Block Enabled 1-20 Call Block Group 1-20 Call Return Enabled 1-21 Call Return Group 1-21 Call Screen Enabled 1-20 Call Screen Group 1-20 Caller ID Block Enabled 1-21 Caller ID Group 1-21 Failure Case 1-20 Forward All
Index (Continued) Users adding new users 1-10 deleting 1-27 edit multiple users 1-28 finding users 1-26 Load all users 1-24–1-27 procedure for editing 1-28 viewing individually 1-16–1-18 V View 1-8 viewing users 1-16 Voice mail A-4 Index-4
