CA2286132A1 - A system, method and article of manufacture for switched telephony communication - Google Patents

Abstract

A hybrid telecommunication system includes a switched network which transfers information across the Internet to provide multi-routed and multidimensional callback processing. The hybrid network includes one or more switched networks coupled to one or more packet transmission networks, and a call router coupled to the switched communication network and the packet transmission network to route information to the appropriate switched telephony device or Internet device address. A computer with an attached display communicates with the packet transmission network. The computer is used to initiate remote management of the hybrid network, including tests of the hybrid network. The tests include circuit analysis such as selecting signaling states which could be loop start, ground start, or detecting signals such as dual tone multifrequency, multifrequency or dialpulse. The hybrid network includes support for an operator to monitor the management of the hybrid network, and an expert system to regulate the Quality of Service of the hybrid telecommunication system.

Description

CA 02286132 1999-10-14 ' DE~111AIVDES OU BREVETS VOLUMINEUX
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WO 98!47298 PCT/US98/07927 A SYSTEM METHOD AND ARTICLE OF MANUFACTURE FOR
SWITCHED TELEPHONY COMMUNICATION
Field Of The Invention The present invention relates to the integration of the Internet with telephony systems, and more specifically, to a system, method and article of manufacture for using the Internet as the communication backbone of a communication system architecture while maintaining a to rich array of call processing features.
Background of the Invention The Internet has increasingly become the communication network of 1 s choice for the consumer e-mail marketplace. Recently, software companies have begun to investigate the transfer of telephone calls across the Internet. However, the system features that users demand of normal call processing are considered essential far call processing the Internet. Today, those features are not available on the Internet. Thus, 2o a system is required that connects the communication network including telephony capability with the Internet to facilitate callback processing.
Callback scenarios for reserving calls over existing telephony networks 25 have been available for some time. Examples of such service are CSI
Callback, Rumilla Telecommunication for international callback and SummitLink which provides international callback offering distribution, wholesaling and rebilling features. The Internet provides a website ' entitled, "Callback on the Net" which purports to "collect all available 3o information on callback services." This information was accumulated by doing a Yahoo search utilizing the search term "callback".
International callback as provided by the prior art system refers to a user being able to dial a number to connect to a switch overseas. The caller allows the phone to ring twice and hangs up. The switch then utilizes the ANI and/or called number information to query a database of profile information stored on the switch to determine billing and other information on the caller. Then, the switch initiates a call to the caller s and when the caller goes offhook, the switch provides a dialtone allowing the caller to access any number available to the switch. In this way, international or other long distance callers can obtain low cost long distance services so long as they are pre-registered for the service. This service still requires the caller to be responsible for all of the overhead associated with initiating call processing, requires a caller to learn the protocol of interfacing with the switch, does not provide reservation of such services such as conferencing, and it does not allow operator assistance on the calls.
Is Recently, AT&T has announced a service very similar to conferenceMCI
(MCI's Operator Call-in Conference Call capability). This service termed "On-Line Teleconference" capability allows teleconference customers to use an on-line interface to allow customers to pre-arrange an AT&T
Teleconference call through the World Wide Web. However, while 2o conference call definition of a number for each participate to call into to join the conference is provided, "all voice connections are established over the existing telephone network" and require all parties to contact a common number to establish the conference call (AT&T Teleconference Service: On-Line Trial Information, 2/7/97).
2s While this new AT&T service is moving in the direction that the subject invention has already arrived at, it does not provide integration of voice over the Internet with existing network services, nor does it provide any mention of a callback architecture which allows a calling party to pre-3o arrange for a network service to contact one or more parties and effectively eliminate the need for any manual intervention. Moreover, it does not provide an operator on an exception basis for Internet telephony operations. Thus, a true union of the Internet and existing telephony networks is not provided.
Z

"Callback" is a telephony term utilized by remote test systems in reference to accessing and testing customer's voice and data circuits which traverse through a Digital Cross-Connect System (DXC). When a s callback feature is selected, the remote test system bridges a local phone line to a customer's DSO circuit. If the circuit under test is an analog circuit, the remote tester performs monitoring. If the circuit under test is a voice circuit, the remote tester performs voice testing which includes selecting appropriate signaling states for callback to the customer's to phone over the circuit under test by the remote test system. The callback feature enables the remote tester to enter a phone number to a co-located phone residing at the remote tester's location.
The remote test system has local phone lines attached to an internal 15 card. The purpose of the phone lines are to allow the test system to place outbound calls. After a remote user enters a number, which includes an area code or their co-located phone, the remote test system selects one of the local phone lines, goes off hood, and upon detecting dial tone from the local telco central office will dial pulse or DTMF the 2o entered phone number. The remote tester's phone receives the call from the remote test system, goes off hook, and then the call from the remote test system to the remote tester is considered complete.
The remote tester can then either monitor the audible quality or initiate 2s a call to the voice circuit customer's phone by selecting the appropriate signaling state for the circuit under test. Once the appropriate signaling state for the customer is selected, the channel bank card or PBX detects the incoming call and converts it to ring cycle to the customer's phone.
This action initiates a ringing condition to the customer's phone. Upon 3o the customer answering the phone, the remote tester verbally communicates with the customer over the customer's circuit under test.
This testing is routinely performed for circuit assurance verification testing.
,[.' The remote test system has a limit of outbound calls that can be completed. This limit is dependent on the number of phone lines that can be supported by the test system's interface. There are also monthly access charges by the telephone company for each local line terminating s into the test system.

Summary of the Invention According to a broad aspect of a preferred embodiment of the invention, telephone calls, data and other multimedia information are routed s through a hybrid network including a switched network which transfers information across the Internet to provide multirouted and multidimensional callback processing. The hybrid network includes one or more switched networks coupled to one or more packet transmission networks that also couple a call router to the switched communication to network and the packet transmission network to route information to the appropriate switched telephony device or Internet device address. A
computer with an attached display communicates with the packet transmission network, the computer is used to initiate remote management of the hybrid network, including tests of the hybrid ~ 5 network which include circuit analysis such as selecting signaling states which could be loop start, ground start, dual tone multifrequency detection or removing a line from service. The hybrid network includes support for an operator to monitor the management of the mybrid network, and an expert system to regulate the Quality of Service of the zo hybrid telecommunication system.
DESCRIPTION OF THE DRAWINGS
The foregoing and other objects, aspects and advantages are better 25 understood from the following detailed description of a preferred embodiment of the invention, with reference to the drawings, in which:
Figure 1A is a block diagram of a representative hardware environment in accordance with a preferred embodiment;
Figure 1B is a block diagram illustrating the architecture of a typical Common Channel Signaling System #7 (SS7) network in accordance with a preferred embodiment;
S~

Figure 1C is a block diagram of an Internet telephony system in accordance with a preferred embodiment;
Figure 1D is a block diagram of a hybrid switch in accordance with a preferred embodiment;
Figure lE is a block diagram of the connection of a hybrid switch in accordance with a preferred embodiment;
to Figure 1F is a block diagram of a hybrid (Internet-telephony) switch in accordance with a preferred embodiment;
Figure 1G is a block diagram showing the software processes involved in the hybrid Internet telephony switch in accordance with a preferred 1 s embodiment;
Figure 2 is a block diagram illustrating the use of PMUs in a typical SS7 network in accordance with a preferred embodiment;
2o Figure 3 is a block diagram illustrating the systems architecture of the preferred embodiment;
Figure 4 is a high-Ievel process flowchart illustrating the logical systern components in accordance with a preferred embodiment;
Figures 5 - 9 are process flowcharts illustrating the detailed operation of the components illustrated in Figure 4 in accordance with a preferred embodiment;
3o Figure lOA illustrates a Public Switched Telephone Network (PSTN) 1000 comprising a Local Exchange Carrier (LEC) 1020 through which a calling party uses a telephone 1021 or computer 1030 to gain access to a switched network in accordance with a preferred embodiment;

Figure lOB illustrates an Internet routing network in accordance with a preferred embodiment;
Figure 11 illustrates a VNET Personal Computer (PC) to PC Information call s flow in accordance with a preferred embodiment;
Figure 12 illustrates a VNET Personal Computer (PC) to out-of-network PC
Information call flow in accordance with a preferred embodiment;
to Figure 13 illustrates a VNET Personal Computer (PC) to out-of-network Phone Information call flow in accordance with a preferred embodiment;
Figure 14 illustrates a VNET Personal Computer (PC) to in-network Phone Information call flow in accordance with a preferred embodiment;
is Figure 15 illustrates a personal computer to personal computer Internet telephony call in accordance with a preferred embodiment;
Figure 16 illustrates a phone call that is routed from a PC through the 2o Internet to a phone in accordance with a preferred embodiment;
Figure 1? illustrates a phone to PC call in accordance with a preferred embodiment;
25 Figure 18 illustrates a phone to phone call over the Internet in accordance with a preferred embodiment;
Figure 19A and 19B illustrate an Intelligent Network in accordance with a preferred embodiment;
Figure 19C illustrates a Video-Conferencing Architecture in accordance with a preferred embodiment;
Figure 19D illustrates a Video Store and Forward Architecture in accordance with a preferred embodiment;
Figure 19E illustrates an architecture for transmitting video telephony over the Internet in accordance with a preferred embodiment;
s Figure 19F is a block diagram of an Internet telephony system in accordance with a preferred embodiment;
Figure 19G is a block diagram of a prioritizing access/router in accordance to with a preferred embodiment;
Figure 20 is a high level block diagram of a networking system in accordance with a preferred embodiment;
15 Figure 21 is a functional block diagram of a portion of the system shown in Figure 20 in accordance with a preferred embodiment;
Figure 22 is another high level block diagram in accordance with a preferred embodiment of Figure 2I;
Figure 23 is a block diagram of a switchless network system in accordance with a preferred embodiment;
Figure 24 is a hierarchy diagram illustrating a portion of the systems zs shown in Figures 20 and 23 in accordance with a preferred embodiment;
Figure 25 is a block diagram illustrating part of the system portion shown in Figure 24 in accordance with a preferred embodiment;
3o Figure 26 is a flow chart illustrating a portion of a method in accordance with a preferred embodiment;
Figures 2?-39 are block diagrams illustrating further aspects of the systems of Figures 20 and 23 in accordance with a preferred embodiment;

Figure 40 is a diagrammatic representation of a web server logon in accordance with a preferred embodiment;
s Figure 41 is a diagrammatic representation of a server directory structure used with the logon of Figure 40 in accordance with a preferred embodiment;
Figure 42 is a more detailed diagrammatic representation of the logon of Figure 40 in accordance with a preferred embodiment;
Figures 43-50 are block diagrams illustrating portions of the hybrid network in accordance with a preferred embodiment;
is Figure 51 illustrates a configuration of the Data Management Zone (DMZ) 5105 in accordance with a preferred embodiment;
Figures 52A-52C illustrate network block diagrams in connection with a dial-in environment in accordance with a preferred embodiment;
Figure 53 depicts a flow diagram illustrating the fax tone detection in accordance with a preferred embodiment;
Figures 54A through 54E depict a flow diagram illustrating the VFP
Completion process for fax and voice mailboxes in accordance with a preferred embodiment;
Figures 55A and 55B illustrate the operation of the Pager Termination processor in accordance with a preferred embodiment;
Figure 56 depicts the GetCallback routine called from the pager termination in accordance with a preferred embodiment;
Figure 57 shows a user login screen for access to online profile y management m accordance with a preferred embodiment;
Figure 58 shows a call routing screen, used to set or change a user's call routing instructions in accordance with a preferred embodiment;
Figure 59 shows a guest menu configuration screen, used to set up a guest menu for presentation to a caller who is not an account owner in accordance with a preferred embodiment;
1o Figure 60 shows an override routing screen, which allows a user to route all calls to a selected destination in accordance with a preferred embodiment;
Figure 61 shows a speed dial numbers screen, used to set up speed dial in accordance with a preferred embodiment;
Figure 62 shows a voicemail screen, used to set up voicemail in accordance with a preferred embodiment;
2o Figure 63 shov~Ts a faxmail screen, used to set up faxmail in accordance with a preferred embodiment;
Figure 64 shows a call screening screen, used to set up call screening in accordance v~.~ith a preferred embodiment;
Figures 65-67 show supplemental screens used with user profile management in accordance with a preferred embodiment;
Figure 68 is a flow chart showing how the validation for user entered speed 3o dial numbers is carried out in accordance with a preferred embodiment;
Figures 69A-69AI are automated response unit (ARU) call flow charts showing software implementation in accordance with a preferred embodiment;
jo Figures 70A-?OR are console call flow charts further showing software implementation in accordance with a preferred embodiment;
Figure 71 illustrates a typical customer configuration for a VNET to VNET
system in accordance with a preferred embodiment;
Figure 72 illustrates the operation of DAPs in accordance with a preferred embodiment;
to Figure 73 illustrates the process by which a telephone connects to a release link trunk for 1-800 call processing in accordance with a preferred embodiment;
Figure ?4 illustrates the customer side of a DAP procedure request in accordance with a preferred embodiment;
Figure 75 illustrates operation of the switch 10530 to select a particular number or "hotline" for a caller in accordance with a preferred embodiment;
2o Figure ?6 illustrates the operation of a computer-based voice gateway for selectively routing telephone calls through the Internet in accordance with a preferred embodiment;
2s Figure 77 illustrates the operation of the VRU of figure 76 deployed in a centralized architecture in accordance with a preferred embodiment;
Figure ?8 illustrates the operation of the VRU of figure ?6 deployed in a distributed architecture in accordance with a preferred embodiment;
Figure 79A and 79B illustrate the operation of sample applications for Internet call routing in accordance with a preferred embodiment;
Figure 79B illustrates a number of applications for caller-initiated consumer transactions in accordance with a preferred embodiment;
Figure 80 illustrates a configuration of a switching network offering voice mail and voice response unit services, as well as interconnection into a s service provider, in accordance with a preferred embodiment;
Figure 81 illustrates an inbound shared Automated Call Distributor (ACD) call with data sharing through a database in accordance with a preferred embodiment;
to Figure 82 is a block diagram of an exemplary telecommunications system in accordance with a preferred embodiment;
Figure 83 is a block diagram of an exemplary computer system in Is accordance with a preferred embodiment;
Figure 84 illustrates the CDR and PNR call record formats in accordance with a preferred embodiment;
2o Figures 85(A) and 85(B) collectively illustrate the ECDR and EPNR call record formats in accordance with a preferred embodiment;
Figure 86 illustrates the OSR and POSR call record formats in accordance with a preferred embodiment;
Figures 8?(A) and 87(B) collectively illustrate the EOSR and EPOSR call record formats in accordance with a preferred embodiment;
Figure 88 illustrates the SER call record format in accordance with a preferred embodiment;
Figures 89(A) and 89(B) are control flow diagrams illustrating the conditions under which a switch uses the expanded record format in accordance with a preferred embodiment;
~z Figure 90 is a control flow diagram illustrating the Change Time command in accordance with a preferred embodiment;
Figure 91 is a control flow diagram illustrating the Change Daylight Savings Time command in accordance with a preferred embodiment;
Figure 92 is a control flow diagram illustrating the Network Call Identifier (NCLD) switch call processing in accordance with a preferred embodiment;
io Figure 93 is a control flow diagram illustrating the processing of a received Network Call Identifier in accordance with a preferred embodiment;
Figure 94(A) is a control flow diagram illustrating the generation of a ~5 Network Call Identifier in accordance with a preferred embodiment;
Figure 94(B) is a control flow diagram illustrating the addition of a Network Call Identifier to a call record in accordance with a preferred embodiment;
2o Figure 95 is a control flow diagram illustrating the transport of a call in accordance with a preferred embodiment;
Figure 96 shows a hardware component embodiment for allowing a video operator to participate in a video conferencing platform, providing services 2s including but not limited to monitoring, viewing and recording any video conference call and assisting the video conference callers in accordance with a preferred embodiment;
Figure 97 shows a system for enabling a video operator to manage video 3o conference calls which includes a video operator console system in accordance with a preferred embodiment;
Figure 98 shows a system for enabling a video operator to manage video conference calls which includes a video operator console system in i ,y accordance with a preferred embodiment;
Figure 99 shows how a video conference call initiated by the video operator in accordance ~rith a preferred embodiment;
Figure 100 shows the class hierarchy for video operator software system classes in accordance with a preferred embodiment;
Figure 101 shows a state transition diagram illustrating the state changes 1o that may occur in the VOCaII object's m_state variable in accordance with a preferred embodiment;
Figure 102 shot;Ts a state transition diagram illustrating the state changes that may occur in the VOConnection object's m_state variable ("state variable") in accordance with a preferred embodiment;
Figure 103 shov~~s a state transition diagram illustrating the state changes that may occur in the VOConference object's m_state variable {"state variable") in accordance with a preferred embodiment;
Figure 104 shows a state transition diagram illustrating the state changes that may occur in the VORecorder object's m_state variable ("state variable") in accordance with a preferred embodiment;
Figure 105 shows a state transition diagram illustrating the state changes that may occur in the VORecorder object's m_state variable ("state variable") in accordance with a preferred embodiment;
Figure 106 shows the class hierarchy for the video operator graphics user 3o interface ("GUl") classes in accordance with a preferred embodiment;
Figure 10'7 shows a database schema for the video operator shared database in accordance with a preferred embodiment;

WO 98147298 PCT/US98/~7927 Figure 108 shows one embodiment of the Main Console window in accordance with a preferred embodiment;
Figure 109 shows one embodiment of the Schedule window in accordance s with a preferred embodiment;
Figure 110 shows vne embodiment of the Conference window 41203, which is displayed when the operator selects a conference or playback session in the Schedule window in accordance with a preferred 1 o embodiment;
Figure 111 shows one embodiment of the Video Watch window 41204, which displays the H.320 input from a selected call of a conference connection or a separate incoming or outgoing call in accordance with a t s preferred embodiment;
Figure 112 shows one embodiment of the Console Output window 41205 which displays all error messages and alerts in accordance with a preferred embodiment; and 2o Figure 113 shows a Properties dialog box in accordance with a preferred embodiment.
Figure 114A is a block diagram of an access/router system in accordance 25 with a preferred embodiment.
Figure 114B is a block diagram of the architecture in accordance with a preferred embodiment.
3o Figure 115 is a block diagram of an Internet based callback architecture in accordance with a preferred embodiment.
/ S' DETAILED DESCRIPTION
INDEX OF CONTENTS
I. THE COMPOSITION OF THE INTERNET
II. PROTOCOL STANDARDS
A. Internet Protocols B. International Telecommunication Union-Telecommunication Standardization Sector ("ITU-T") Standards III. TCP/IP FEATURES
to IV. INFORMATION TRANSPORT IN COMMUNICATION NETWORKS
A. Switching Techniques B. Gateways and Routers C. Using Network Level Communication for Smooth User Connection D. Datagrams and Routing V. TECHNOLOGY INTRODUCTION
A. ATM
B. Frame Relay C. ISDN
VI. MCI INTELLIGENT NETWORK
2o A. Components of the MCI Intelligent Network 1. MCI Switching Network 2. Network Control System/ Data Access Point (NCS / DAP) 3. Intelligent Services Network (ISN) 4 4. Enhanced Voice Services (EVS) 9 S. Additional Components B. Intelligent Network System Overview C. Call Flow Example VII. ISP FRAMEWORK
A. Background 1. Broadband Access 2.Internet Telephony System 3. Capacity 4. Future Services B. ISP Architecture Framework ~6 C. ISP Functional Framework D. ISP Integrated Network Services E. ISP Components F. Switchless Communications Services s G. Governing Principles l.Architectural Principles 2. Service Feature Principles 3. Capability Principles 4. Service Creation, Deployment, and Execution Principles to 5.Resource Management Model 2150 Principles 6. Data Management 2138 Principles 7.Operational Support Principles 8. Physical Model Principles H. ISP Service Model t s 1. Purpose 2. Scope of Effort 3. Service Model Overview 4. Service Structure 5. Service 2200 Execution 20 6. Service Interactions 7. Service Monitoring I. ISP Data Management Model 1. Scope 2. Purpose 2s 3. Data management Overview 4. Logical Description S.Physical Description 6.Technology Selection 7. Implementations 30 8. Security 9. Meta-Data 10. Standard Database Technologies J. ISP Resource Management Model 2.The Local Resource Manager (LRM):
l3 3.The Global Resource Manager (GRM) 2188:
4.The Resource Management Model (RMM) 5. Component Interactions K . Operational Support Model s l.Introduction 2.The Operational Support Model 3. The Protocol Model 4.The Physical Model S. Interface Points to 6. General L. Physical Network Model 1.Introduction 2.Information Flow 3. Terminology is 4.Entity Relationships VIII. INTELLIGENT NETWORK

A. Network Management B. Customer Service C. Accounting 2o D. Commissions E. Reporting F. Security G. Trouble Handling IX. ENHANCED PERSONAL SERVICES
zs A. Web Server Architecture i . Welcome Server 450 2.Token Server 454 3. Application Servers B. Web Server System Environment 3o 1. Welcome Servers 2.Token Servers 454 3.Profile Management Application Servers C. Security D. Login Process E. Service Selection F. Service Operation 1. NIDS Server 2.TOKEN database service.
' S 3. SERVERS database service.
4. HOSTILE IP database service.
S.TOKEN HOSTS database service.
6. SERVER ENV database service.
7. Chron Jobs) io G. Standards H. System Administration I. Product/ Enhancement .J. Interface Feature Requirements (Overview) l.The User Account Profile 15 2.The Database of Messages K. Automated Response Unit (ARU) Capabilities 1. User Interface L. Message Management 1. Multiple Media Message Notification 20 2. Multiple Media Message Manipulation 3.Text to Speech 4. Email Forwarding to a Fax Machine S. Pager Notification of Messages Received 6. Delivery Confirmation of Voicemail 25 7. Message Prioritization M. Information Services N. Message Storage Requirements O. Profile Management P. Call Routing Menu Change 3a Q. Two-way Pager Configuration Control and Response to Park and Page R. Personalized Greetings S. List Management T. Global Message Handling X. INTERNET TELEPHONY AND RELATED SERVICES

A. System Environment for Internet Media 1. Hardware 2.Object-Oriented Software Tools B. Telephony Over The Internet 1. Introduction 2.IP Phone as a Commercial Service 3. Phone Numbers in the Internet 4.Other Internet Telephony Carriers S.International Access to C. Internet Telephony Services D. Call Processing 1. VNET Call Processing 2. Descriptions of Block Elements E. Re-usable Call Flow Blocks 1. VNET PC connects to a corporate intranet and logs in to a directory seance 2. VNET PC queries a directory service for a VNET translation 3. PC connects to an ITG
4.ITG connects to a PC
S.VNET PC to PC Call Flow Description 6. Determining best choice for Internet client selection of an Internet Telephony Gateway server on the Internet:
7. Vnet Call Processing XI. TELECOMMUNICATION NETWORK MANAGEMENT
A. SNMS Circuits Map B. SNMS Connections Map C. SNMS Nonadjacent Node Map D. SNMS LATH Connections Map E. NPA-NXX Information List 3o F. End Office Information List G. Trunk Group Information List H. Filter Definition Window I. Trouble Ticket Window XII. VIDEO TELEPHONY OVER POTS

A. Components of Video Telephony System 1. DSP modem pools with ACD.
2 . Agent 3. Video on Hold Server ' s 4. Video Mail Server 5. Video Content Engine 6. Reservation Engine 7. Video Bridge B. Scenario to C. Connection Setup D. Calling the Destination E. Recording Video-Mail, Store & Forward Video and Greetings F. Retrieving Video-Mail and Video On Demand G. Video-conference Scheduling is XIII. VIDEO TELEPHONY OVER THE INTERNET
A. Components 1. Directory and Registry Engine 2. Agents 3. Video Mail Server 20 4. Video Content Engine 5. Conference Reservation Engine 6. MCI Conference Space 7. Virtual Reality Space Engine B. Scenario 25 C. Connection Setup D. Recording Video-Maii, Store 8s Forward Video and Greetings E. Retrieving Video-Mail and Video On Demand F. Video-conference Scheduling ' G. Virtual Reality 3o XIV. VIDEO-CONFERENCING ARCHITECTURE
A. Features B. Components l.End-User Terminals 2. LAN Interconnect System 3.ITU H.323 Server 4. Gatekeeper S. Operator Services Module 6. Multipoint Control Unit (MCU) 7. Gateway 8. Support Service Units C. Overview D. Call Flow Example 1. Point-to-Point Calls l0 2. Multipoint Video-Conference Calls E. Conclusion XV. VIDEO STORE AND FORWARD ARCHITECTURE
A. Features B. Architecture ~5 C. Components 1. Content Creation and Transcoding 2. Content Management and Delivery 3. Content Retrieval and Display D. Overview 2o XVI. VIDEO OPERATOR
A. Hardware Architecture B. Video Operator Console C. Video Conference Call Flow D. Video Operator Software System 25 1. Class Hierarchy 2.Class and Object details E. Graphical User Interface Classes 1. Class Hierarchy 2. Class and Object details 3o F. Video Operator Shared Database 1. Database Schema G. Video Operator Console Graphical User Interface Windows 1. Main Console Window 2. Schedule Window zz 3. Conference Window 4. Video Watch Window 5. Console Output Window 6. Properties Dialog Box s XVII. WORLD WIDE WEB (WWW) BROWSER CAPABILITIES
A. User Interface B. Performance C. Personal Home Page 1. Storage Requirements to 2.On Screen Help Text 3. Personal Home Page Directory 4. Control Bar 5. Home Page 6. Security Requirements ~5 7.On Screen Help Text 8. Profile Management 9.Information Services Profile Management 10. Personal Home Page Profile Management 1 1. List Management 20 12. Global Message Handling D. Message Center 1. Storage Requirements E. PC Client Capabilities 1. User Interface 25 2. Security 3. Message Retrieval 4. Message Manipulation F. Order Entry Requirements 1. Provisioning and Fulfillment 3o G. Traffic Systems H. Pricing I. Billing XVIII. DIRECTLINE MCI
A. Overview z3 l.The ARU (Audio Response Unit) 502 2.The VFP (Voice Fax Platform) 504 3.The DDS (Data Distribution Service) 506 B . Rationale C . Detail 1. Call Flow Architecture 520 2. Network Connectivity 3. Call Flow 4. Data Flow Architecture 1o D . Voice Fax Platform (VFP) 504 Detailed Architecture i . Overview 2. Rationale 3. Detail E. Voice Distribution Detailed Architecture 1. Overview 2. Rationale F. Login Screen G. Call Routing Screen H. Guest Menu Configuration Screen 2o I. Override Routing Screen J. Speed Dial Screen K. ARU CALL FLOWS

XIX.INTERNET FAX

A. Introduction B. Details XX. INTERNET SWITCH TECHNOLOGY

A. An Embodiment B. Another Embodiment XXI.BILLING

3o A. An Embodiment 1. Call Record Format 2. Network Call Identifier B. [Another Embodiment) 1. Call Record Format 2~

2. Network Call Identifier XXII. PRIORITIZING ACCESS/ROUTER
A. Prioritizing Access/Router Overview B. Prioritizing Access/Router Process s XXIII. CALLBACK TELEPHONY SYSTEM
A. Introduction to a Callback Telephony System in Accordance with a preferred Embodiment B. Internet-Based Callback Architecture C. Callback Service Potential 1o D. Internet Service Potential E. Internet-Based Callback Architecture F. Self Regulating System zS-INTRODUCTION TO THE INTERNET
I. THE COMPOSITION OF THE INTERNET
The Internet is a method of interconnecting physical networks and a set of s conventions for using networks that allow the computers they reach to interact. Physically, the Internet is a huge, global network spanning over 92 countries and comprising 59,000 academic, commercial, government, and military networks, according to the Government Accounting Office (GAO), with these numbers expected to double each year. Furthermore, there are about 10 million host computers, 50 million users, and 76,000 World-Wide Web servers connected to the Internet. The backbone of the Internet consists of a series of high-speed communication links between major supercomputer sites and educational and research institutions within the U.S. and throughout the world.
is Before progressing further, a common misunderstanding regarding the usage of the term "internet" should be resolved. Originally, the term was used only as the name of the network based upon the Internet Protocol, but now, Internet is a generic term used to refer to an entire class of networks.
zo An "Internet" (lowercase "i") is any collection of separate physical networks, interconnected by a common protocol, to form a single logical network, whereas the "Internet" (uppercase "I") is the worldwide collection of interconnected networks that uses Internet Protocol to link the large number of physical networks into a single logical network.
II. PROTOCOL STANDARDS
A. Internet Protocols govern the behavior along the Internet backbone and thus set down the key rules for data communication. Transmission Control 3o Protocol/Internet Protocol {TCP/IP) has an open nature and is available to everyone, meaning that it attempts to create a network protocol system that is independent of computer or network operating system and architectural differences. As such, TCP/IP protocols are publicly available in standards documents, particularly in Requests for Comments (RFCs). A requirement z6 WO 98!47298 PCT/US98/07927 for Internet connection is TCP/IP, which consists of a large set of data communications protocols, two of which are the Transmission Control Protocol and the Internet Protocol. An excellent description of the details associated with TCP/IP -and UDP/IP is provided in TCP/IP Illustrated, W.
s Richard Stevens, Addison-Wesley Publishing Company ( 1996).
B. International Telecommunication Union-Telecommunication Standardization Sector ("ITU T") Standards The International Telecommunication Union-Telecommunication Standardization Sector ("ITU-T") has established numerous standards governing protocols and line encoding for telecommunication devices.
Because many of these standards are referenced throughout this document, summaries of the relevant standards are listed below for t s reference.
ITU 6.711 Recommendation for Pulse Code Modulation of 3kHz Audio Channels.
ITU 6.722 Recommendation for 7kHz Audio Coding within a 64kbit/ s zo channel.
ITU 6.723 Recommendation for dual rate speech coder for multimedia communication transmitting at 5.3 and 6.3 kbits.
ITU G.?28 Recommendation for coding of speech at l6kbit/ s using low-delay code excited linear prediction (LD-CELP) 2s ITU H.221 Frame Structure for a 64 to 1920 kbit/ s Channel in Audiovisual Teleservices ITU H.223 Multiplexing Protocols for Low Bitrate Multimedia Terminals ITU H.225 ITU Recommendation for Media Stream Packetization and Synchronization on non-guaranteed quality of service LANs.
3o ITU H.230 Frame-synchronous Control and Indication Signals for Audiovisual Systems ITU H.231 Multipoint Control Unit for Audiovisual Systems Using Digital Channels up to 2 Mbit/ s ITU H.242 System for Establishing Communication Between Audiovisual 2~

Terminals Using Digital Channels up to 2Mbits ITU H.243 System for Establishing Communication Between Three or More Audiovisual Terminals Using Digital Channels up to 2 Mbit/s ITU H.245 Recommendation for a control protocol for multimedia s communication ITU H.261 Recommendation for Video Coder-Decoder for audiovisual services supporting video resolutions of 352x288 pixels and 176x144 pixels.
ITU H.263 Recommendation for Video Coder-Decoder for audiovisual services supporting video resolutions of 128x96 pixels, 176x144 pixels, 352x288 pi~~els, 704x576 pixels and 1408x1152 pixels.
ITU H.320 Recommendation for Narrow Band ISDN visual telephone systems.
ITU H.321 Visual Telephone Terminals over ATM
is ITU H.322 Visual Telephone Terminals over Guaranteed Quality of Service LANs ITU H.323 ITU Recommendation for Visual Telephone Systems and Equipment for Local Area Networks which provide a non-guaranteed quality of service.
Zo ITU H.324 Recommendation for Terminals and Systems for low bitrate(28.8 Kbps) multimedia communication on dial-up telephone lines.
ITU T.120 Transmission Protocols for Multimedia Data.
In addition, several other relevant standards are referenced in this 25 document:
ISDN Integrated Services Digital Network, the digital communication standard for transmission of voice, video and data on a single communications link.
3o RTP Real-Time Transport Protocol, an Internet Standard Protocol for transmission of real-time data like voice and video over unicast and multicast networks.
IP Internet Protocol, an Internet Standard Protocol for transmission and delivery of data packets on a packet switched network of interconnected z~

computer systems.
PPP Point-to-Point Protocol MPEG Motion Pictures Expert Group, a standards body under the International Standards Organization(ISO), Recommendations for ' s compression of digital Video and Audio including the bit stream but not the compression algorithms.
SLIP Serial Line Internet Protocol RSVP Resource Reservation Setup Protocol UDP User Datagram Protocol io III. TCP/IP FEATURES
The popularity of the TCP/ IP protocols on the Internet grew rapidly because they met an important need for worldwide data communication and had several important characteristics that allowed them to meet this need.
1s These characteristics, still in use today, include:
A common addressing scheme that allows any device running TCP/IP to uniquely address any other device on the Internet.
Open protocol standards, freely available and developed independently of any hardware or operating system. Thus, TCP/IP is capable of being used 2o with different hardware and software, even if Internet communication is not required.
Independence from any specific physical network hardware, allows TCP/IP
to integrate many different kinds of networks. TCP/IP can be used over an zs Ethernet, a token ring, a dial-up line, or virtually any other kinds of physical transmission media.
IV. INFORMATION TRANSPORT IN COMMUNICATION NETWORKS
A. Switching Techniques 3o An understanding of how information travels in communication systems is required to appreciate the recent steps taken by key players in today's Internet backbone business. The traditional type of communication network is circuit switched. The U.S. telephone system uses such circuit switching techniques. When a person or a computer makes a telephone z~

call, the switching equipment within the telephone system seeks out a physical path from the originating telephone to the receiver's telephone. A
circuit-switched network attempts to form a dedicated connection, or circuit, between these two points by first establishing a circuit from the s originating phone through the local switching office, then across trunk lines, to a remote switching office, and finally to the destination telephone.
This dedicated connection exists until the call terminates.
The establishment of a completed path is a prerequisite to the transmission to of data for circuit switched networks. After the circuit is in place, the microphone captures analog signals, and the signals are transmitted to the Local Exchange Carrier (LEC) Central Office (CO) in analog form over an analog loop. The analog signal is not converted to digital form until it reaches the LEC Co, and even then only if the equipment is modern enough 15 to support digital information. In an ISDN embodiment, however, the analog signals are converted to digital at the device and transmitted to the LEC as digital information.
Upon connection, the circuit guarantees that the samples can be delivered 2o and reproduced by maintaining a data path of 64 Kbps (thousand bits per second). This rate is not the rate required to send digitized voice per se.
Rather, 64Kbps is the rate required to send voice digitized with the Pulse Code Modulated (PCM) technique. Many other methods for digitizing voice exist, including ADPCM (32Kbps), GSM ( 13 Kbps), TrueSpeech 8.5 (8.5 25 Kbps), 6.723 (6.4 Kbps or 5.3 Kbps) and Voxware RT29HQ (2.9 Kbps).
Furthermore, the 64 Kbps path is maintained from LEC Central Office (CO) Switch to LEC CO, but not from end to end. The analog local loop transmits an analog signal, not 64 Kbps digitized audio. One of these analog local loops typically exists as the "last mile" of each of the telephone 3o network circuits to attach the local telephone of the calling party.
This guarantee of capacity is the strength of circuit-switched networks.
However, circuit switching has two significant drawbacks. First, the setup time can be considerable, because the call signal request may find the lines 3~

busy with other calls; in this event, there is no way to gain connection until some other connection terminates. Second, utilization can be low while costs are high. In other words, the calling party is charged for the duration of the call and for all of the time even if no data transmission takes place (i.e. no one speaks). Utilization can be low because the time between transmission of signals is unable to be used by any other calls, due to the dedication of the line. Any such unused bandwidth during the connection is v~~asted.
to Additionally, the entire circuit switching infrastructure is built around Kbps circuits. The infrastructure assumes the use of PCM encoding techniques for voice. However, very high quality codecs are available that can encode voice using less than one-tenth of the bandwidth of PCM.
However, the circuit switched network blindly allocates 64 Kbps of is bandwidth for a call, end-to-end, even if only one-tenth of the bandwidth is utilized. Furthermore, each circuit generally only connects two parties.
Without the assistance of conference bridging equipment, an entire circuit to a phone is occupied in connecting one party to another party. Circuit switching has no multicast or multipoint communication capabilities, 2o except when used in combination with conference bridging equipment.
Other reasons for long call setup time include the different signaling networks involved in call setup and the sheer distance causing propagation delay. Analog signaling from an end station to a CO on a low bandwidth 2s link can also delay call setup. Also, the call setup data travels great distances on signaling networks that are not always transmitting data at the speed of light. When the calls are international, the variations in signaling networks grows, the equipment handling call setup is usually not as fast as modem setup and the distances are even greater, so call setup 3o slows down even more. Further, in general, connection-oriented virtual or physical circuit setup, such as circuit switching, requires more time at connection setup time than comparable connectionless techniques due to the end-to-end handshaking required between the conversing parties.
a9 Message switching is another switching strategy that has been considered.
With this form of switching, no physical path is established in advance between the sender and receiver; instead, whenever the sender has a block of data to be sent, it is stored at the first switching office and retransmitted s to the next switching point after error inspection. Message switching places no limit on block size, thus requiring that switching stations must have disks to buffer long blocks of data; also, a single block may tie up a line for many minutes, rendering message switching useless for interactive traffic.
to Packet switched networks, which predominate the computer network industry, divide data into small pieces called packets that are multiplexed onto high capacity intermachine connections. A packet is a block of data with a strict upper limit on block size that carries with it sufficient identification necessary for delivery to its destination. Such packets t5 usually contain several hundred bytes of data and occupy a given transmission line for only a few tens of milliseconds. Delivery of a larger file via packet switching requires that it be broken into many small packets and sent one at a time from one machine to the other. The network hardware delivers these packets to the specified destination, where the 2o software reassembles them into a single file.
Packet switching is used by virtually ali computer interconnections because of its efficiency in data transmissions. Packet switched networks use bandwidth on a circuit as needed, allowing other transmissions to pass 25 through the lines in the interim. Furthermore, throughput is increased by the fact that a router or switching office can quickly forward to the next stop any given packet, or portion of a large file, that it receives, long before the other packets of the file have arrived. In message switching, the intermediate router would have to wait until the entire block was delivered 3o before forwarding. Today, message switching is no longer used in computer networks because of the superiority of packet switching.
To better understand the Internet, a comparison to the telephone system is helpful. The public switched telephone network was designed with the goal 3~2 WO 98/47298 PCTlUS98/07927 of transmitting human voice, in a more or less recognizable form. Their suitability has been improved for computer-to-computer communications but remains far from optimal. A cable running between two computers can transfer data at speeds in the hundreds of megabits, and even gigabits per second. A poor error rate at these speeds would be only one error per day.
In contrast, a dial-up line, using standard telephone lines, has a maximum data rate in the thousands of bits per second, and a much higher error rate. In fact, the combined bit rate times error rate performance of a local cable could be 11 orders of magnitude better than a voice-grade telephone i o line. New technology, however, has been improving the performance of these Lines.
B. Gateways and Routers The Internet is composed of a great number of individual networks, t5 together forming a global connection of thousands of computer systems.
After understanding that machines are connected to the individual networks, we can investigate how the networks are connected together to form an internetwork, or an Internet. At this point, Internet gateways and Internet routers come into play.
In terms of architecture, two given networks are connected by a computer that attaches to both of them. Internet gateways and routers provide those links necessary to send packets between networks and thus make connections possible. Without these links, data communication through 2s the Internet would not be possible, as the information either would not reach its destination or would be incomprehensible upon arrival. A
gateway may be thought of as an entrance to a communications network that performs code and protocol conversion between two otherwise incompatible networks. For instance, gateways transfer electronic mail and 3o data files between networks over the Internet.
IP Routers are also computers that connect networks and is a newer term preferred by vendors. These routers must make decisions as to how to send the data packets it receives to its destination through the use of ,.3.3 continually updated routing tables. By analyzing the destination network address of the packets, routers make these decisions. Importantly, a router does not generally need to decide which host or end user will receive a packet; instead, a router seeks only the destination network and thus s keeps track of information sufficient to get to the appropriate network, not necessarily the appropriate end user. Therefore, routers do not need to be huge supercomputing systems and are often just machines with small main memories and little disk storage. The distinction between gateways and routers is slight, and current usage blurs the line to the extent that the two to terms are often used interchangeably. In current terminology, a gateway moves data between different protocols and a router moves data between different networks. So a system that moves mail between TCP/IP and OSI
is a gateway, but a traditional IP gateway (that connects different networks) is a router.
is Now, it is useful to take a simplified look at routing in traditional telephone systems. The telephone system is organized as a highly redundant, multilevel hierarchy. Each telephone has two copper wires coming out of it that go directly to the telephone company's nearest end office, also called a 20 local central office. The distance is typically less than 10 km; in the U.S.
alone, there are approximately 20,000 end offices. The concatenation of the area code and the first three digits of the telephone number uniquely specify an end office and help dictate the rate and billing structure.
2s The two-wire connections between each subscriber's telephone and the end office are called local loops. If a subscriber attached to a given end office calls another subscriber attached to the same end office, the switching mechanism within the office sets up a direct electrical connection between the two local loops. This connection remains intact for the duration of the 3o call, due to the circuit switching techniques discussed earlier.
If the subscriber attached to a given end office calls a user attached to a different end office, more work has to be done in the routing of the call.
First, each end office has a number of outgoing lines to one or more nearby switching centers, called toll offices. These lines are called toll connecting trunks. If both the caller's and the receiver's end offices happen to have a toll connecting trunk to the same toll office, the connection may be established within the toll office. If the caller and the recipient of the call s do not share a toll office, then the path will have to be established somewhere higher up in the hierarchy. There are sectional and regional offices that form a network by which the toll offices are connected. The toll, sectional, and regional exchanges communicate with each other via high bandwidth inter-toll trunks. The number of different kinds of s~~itching 1o centers and their specific topology varies from country to country, depending on its telephone density.
C. Using Network Level Communication for Smooth User Connection 15 In addition to the data transfer functionality of the Internet, TCP/IP also seeks to convince users that the Internet is a solitary, virtual network.
TCP/IP accomplishes this by providing a universal interconnection among machines, independent of the specific networks to which hosts and end users attach. Besides router interconnection of physical networks, software 2o is required on each host to allow application programs to use the Internet as if it were a single, real physical network.
D. Datagrams and Routing The basis of Internet service is an underlying, connectionless packet 25 delivery system run by routers, with the basic unit of transfer being the packet. In internets running TCP/IP, such as the Internet backbone, these packets are called datagrams. This section will briefly discuss how these datagrams are routed through the Internet.
30 in packet switching systems, routing is the process of choosing a path over which to send packets. As mentioned before, routers are the computers that make such choices. For the routing of information from one host within a network to another host on the same network, the datagrams that are sent do not actually reach the Internet backbone. This is an example of 3S~

internal routing, which is completely self-contained within the network.
The machines outside of the network do not participate in these internal routing decisions.
s At this stage, a distinction should be made between direct delivery and indirect delivery. Direct delivery is the transmission of a datagram from one machine across a single physical network to another machine on the same physical network. Such deliveries do not involve routers. Instead, the sender encapsulates the datagram in a physical frame, addresses it, 1o and then sends the frame directly to the destination machine.
Indirect delivery is necessary when more than one physical network is involved, in particular when a machine on one network wishes to communicate with a machine on another network. This type of 15 communication is what we think of when we speak of routing information across the Internet backbone. In indirect delivery, routers are required. To send a datagram, the sender must identify a router to which the datagram can be sent, and the router then forwards the datagram towards the destination network. Recall that routers generally do not keep track of the 2o individual host addresses (of which there are millions), but rather just keeps track of physical networks (of which there are thousands).
Essentially, routers in the Internet form a cooperative, interconnected structure, and datagrams pass from router to router across the backbone until they reach a router that can deliver the datagram directly.
V. TECHNOLOGY INTRODUCTION
The changing face of the Internet world causes a steady inflow of new systems and technology. The following three developments, each likely to became more prevalent in the near future, serve as an introduction to the 3o technological arena:
A. ATM
Asynchronous Transfer Mode (ATM) is a networking technology using a high-speed, connection-oriented system for both local area and wide area networks. ATM networks require modern hardware including:
High speed switches that can operate at gigabit {trillion bit) per second speeds to handle the traffic from many computers;
Optical fibers (versus copper wires) that provide high data transfer rates, with host-to-ATM switch connections running at 100 or 155 Mbps (million bits per second);
Fixed size cells, each of which includes 53 bytes.
ATM incorporates features of both packet switching and circuit switching, to as it is designed to carry voice, video, and television signals in addition to data. Pure packet switching technology is not conducive to carrying voice transmissions because such transfers demand more stable bandwidth.
B. Frame Relay is Frame relay systems use packet switching techniques, but are more efficient than traditional systems. This efficiency is partly due to the fact that they perform less error checking than traditional X.25 packet-switching services. In fact, many intermediate nodes do little or no error checking at all and only deal with routing, leaving the error checking to the 2o higher layers of the system. With the greater reliability of today's transmissions, much of the error checking previously performed has become unnecessary. Thus, frame relay offers increased performance compared to traditional systems.
25 C. ISDN
An Integrated Services Digital Network is an "international telecommunications standard for transmitting voice, video, and data over digital lines," most commonly running at 64 kilobits per second. The traditional phone network runs voice at only 4 kilobits per second. To 3o adopt ISDN, an end user or company must upgrade to ISDN terminal equipment, central office hardware, and central office software. The ostensible goals of ISDN include the following:
To provide an internationally accepted standard for voice, data and signaling;
.'3 ~

To make all transmission circuits end-to-end digital;
To adopt a standard out-of-band signaling system; and To bring significantly more bandwidth to the desktop.
s VI. MCI INTELLIGENT NETWORK
The MCI Intelligent Network is a call processing architecture for processing voice, fax and related services. The Intelligent Network comprises a special purpose bridging switch with special capabilities and a set of general purpose computers along with an Automatic Call Distributor (ACD). The 1o call processing including number translation services, automatic or manual operator services, validation services and database services are carried out on a set of dedicated general purpose computers with specialized software.
New value added services can be easily integrated into the system by enhancing the software in a simple and cost-effective manner.
is Before proceeding further, it will be helpful to establish some terms.
ISP Intelligent Services Platform NCS Network Control System DAP Data Access Point 2o ACD Automatic Call Distributor ISN Intelligent Services Network (Intelligent Network) ISNAP Intelligent Services Network Adjunct Processor MTOC Manual Telecommunications Operator Console ARU Audio Response Unit 25 ACP Automatic Call Processor NAS Network Audio Server EVS Enhanced Voice Services POTS Plain Old Telephone System ATM Asynchronous Transfer Mode The Intelligent Network Architecture has a rich set of features and is very flexible. Addition of new features and services is simple and fast. Features and services are extended utilizing special purpose software running on general purpose computers. Adding new features and services involves upgrading the special purpose software and is cost-effective.
Intelligent Network Features and Services include Call type identification;
s Call Routing and selective termination;
Operator selection and call holding;
Manual and Automated Operator;
Voice Recognition and automated, interactive response;
Customer and customer profile verification and validation;
io Voice Mail;
Call validation and database;
Audio Conference reservation;
Video Conference reservation;
Fax delivery and broadcasting;
is Customer Billing;
Fraud Monitoring;
Operational Measurements and Usage Statistics reporting; and Switch interface and control.
2o A. Components of the MCI Intelligent Figure 19A illustrates an Intelligent Network in accordance with a preferred embodiment. The MCI Intelligent Network is comprised of a large number of components. Major components of the MCI Intelligent Network include the 2s MCI Switching Network 2 Network Control System (NCS)/Data Access Point(DAP) 3 ISN - Intelligent Services Network 4 EVS - Enhanced Voice Services 9 30 1. MCI Switching Network.
The MCI switching network is comprised of special purpose bridging switches 2. These bridging switches 2 route and connect the calling and the called parties after the call is validated by the intelligent services network 4. The bridging switches have limited programming capabilities 3%

and provide the basic switching services under the control of the Intelligent Services Network (ISN) 4.
2. Network Control System/Data Access Point (NCS/DAP).
The NCS/DAP 3 is an integral component of the MCI Intelligent Network.
The DAP offers a variety of database services like number translation and also provides services for identifying the switch ID and trunk ID of the terminating number for a call.
1o The different services offered by NCS/DAP 3 include:
Number Translation for 800, 900, VNET Numbers;
Range Restrictions to restrict toll calling options and advanced parametric routing including Time of Day, Day of Week/Month, Point of Origin and percentage allocation across multiple sites;
is Information Database including Switch ID and Trunk ID of a terminating number for a given call;
Remote Query to Customer Databases;
VNET/950 Card Validation Services; and VNET ANI/DAL Validation Services.
3. Intelligent Services Netv~~ork (ISN) The ISN 4 includes an Automatic Call Distributor (ACD) for routing the calls. The ACD communicates with the Intelligent Switch Network Adjunct Processor (ISNAP) 5 and delivers calls to the different manual or automated 2s agents. The ISN inciudes the ISNAP 5 and the Operator Network Center (ONC). ISNAP 5 is responsible for Group Select and Operator Selection for call routing. The /SNAP communicates with the ACD for call delivery to the different agents. The ISNAP is also responsible for coordinating data and voice for operator-assisted calls. The ONC is comprised of Servers, 3o Databases and Agents including Live Operators or Audio Response Units (ARU) including Automated Call Processors (ACP)s, MTOCs and associated NAS 7. These systems communicate with each other on an Ethernet LAN
and provide a variety of services for call processing.
~D

The different services offered by the ONC include:
Validation Services including call-type identification, call verification and call restrictions if any;
Operator Services, both manual and automated, for customer assistance;
Database Services for a variety of database lookups;
Call Extending Capabilities;
Call Bridging Capabilities;
Prompt for User Input; and Play Voice Messages.
to 4. Enhanced Voice Services (EVS) Enhanced Voice Services offer menu-based routing services in addition to a number of value-added features. The EVS system prompts the user for an input and routes calls based on customer input or offers specialized 1s services for voice mail and fax routing. The different services offered as a part of the EVS component of the MCI Intelligent Network include:
Play Customer Specific Voice Messages;
Prompt for User Input;
User Input based Information Access;
2o Call Extending Capabilities;
Call Bridging Capabilities;
Audio Conference Capabilities;
Call Transfer Capabilities;
Record User Voice Messages;
25 Remote Update of Recorded Voice; and Send/Receive Fax.
S. Additional Components.
In addition to the above mentioned components, a set of additional 3o components are also architected into the MCI Intelligent Network. These components are:
Intelligent Call Routing (ICR) services are offered for specialized call routing based on information obtained from the calling party either during the call or at an earlier time. Routing is also based on the knowledge of the physical and logical network layout. Additional intelligent routing services based on time of day, alternate routing based on busy routes are also offered.
Billing is a key component of the MCI Intelligent Network. The billing component provides services for customer billing based on call type and call duration. Specialized billing services are additionally provided for value added services like the 800 Collect calls.
Fraud Monitoring component is a key component of the MCI Intelligent Network providing services for preventing loss of revenue due to fraud and illegal usage of the network.
Operational Measurements include information gathering for analysis of t5 product performance. Analysis of response to advertising campaigns, calling patterns resulting in specialized reports result from operational measurements. Information gathered is also used for future product planning and predicting infrastructure requirements.
2o Usage Statistics Reporting includes gathering information from operational databases and billing information to generate reports of usage. The usage statistics reports are used to study call patterns, load patterns and also demographic information. These reports are used for future product plans and marketing input.
B. Intelligent Network System Overview The MCI Call Processing architecture is built upon a number of key components including the MCI Switch Network, the Network Control System, the Enhanced Voice Services system and the Intelligent Services 3o Network. Call processing is entirely carried out on a set of general purpose computers and some specialized processors thereby forming the basis for the MCI Intelligent Network. The switch is a special purpose bridging switch with limited programming capabilities and complex interface.
Addition of new services on the switch is very difficult and sometimes not x~

possible. A call on the MCI Switch is initially verified if it needs a number translation as in the case of an 800 number. If a number translation is required, it is either done at the switch itself based on an internal table or the request is sent to the DAP which is a general purpose computer with s software capable of number translation and also determining the trunk ID
and switch ID of the terminating number.
The call can be routed to an ACD which delivers calls to the various call processing agents like a live operator or an ARU. The ACD communicates to with the ISNAP which does a group select to determine which group of agents are responsible for this call and also which of the agents are free to process this call.
The agents process the calls received by communicating with the NIDS
1 s (Network Information Distributed Services) Server which are the Validation or the Database Servers with the requisite databases for the various services offered by ISN. Once the call is validated by processing of the call on the server, the agent communicates the status back to the ACD. The ACD in turn dials the terminating number and bridges the incoming call 2o with the terminating number and executes a Release Link Trunk (RLT) for releasing the call all the way back to the switch. The agent also generates a Billing Detail Record (BDR) for billing information. When the call is completed, the switch generates an Operation Services Record (OSR) which is later matched with the corresponding BDR to create total billing 2s information. The addition of new value added services is very simple and new features can be added by additional software and configuration of the different computing systems in the ISP. A typical call flow scenario is explained below.
3o C. Call Flow Example The Call Flow example illustrates the processing of an 800 Number Collect Call from phone 1 in Figure 19A to phone 10. The call is commenced when a calling party dials 1-800-COLLECT to make a collect call to phone 10 the Called Party. The call is routed by the Calling Party's Regional Bell Operating Company (RBOC), which is aware that this number is owned by MCI, to a nearest MCI Switch Facility and lands on an MCI switch 2.
The switch 2 detects that it is an 800 Number service and performs an 800 s Number Translation from a reference table in the switch or requests the Data Access Point (DAP) 3 to provide number translation services utilizing a database lookup.
The call processing is now delegated to a set of intelligent computing Io systems through an Automatic Call Distributor (ACD) 4. In this example, since it is a collect call, the calling party has to reach a Manual or an Automated Operator before the call can be processed further. The call from the switch is transferred to an ACD 4 which is operational along with an Intelligent Services Network Adjunct Processor (ISNAP) 5. The ISNAP 5 is determines which group of Agents are capable of processing the call based on the type of the call. This operation is referred to as Group Select. The agents capable of call processing include Manual Telecommunications Operator Console {MTOC)s 6 or Automated Call Processors (ACP)s 7 v~~ith associated Network Audio Servers (NAS)s 7a. The ISNAP 5 determines 2o which of the Agents is free to handle the call and routes the voice call to a specific Agent.
The Agents are built with sophisticated call processing software. The Agent gathers all the relevant information from the Calling Party including the 2s telephone number of the Called Party. The Agent then communicates with the database servers with a set of database lookup requests. The database lookup requests include queries on the type of the call, call validation based on the telephone numbers of both the calling and the called parties and also call restrictions, if any, including call blocking restrictions based on 3o the called or calling party's telephone number. The Agent then signals the ISNAP-ACD combination to put the Calling Party on hold and dial the called party and to be connected to the Called Party. The Agent informs the called party about the Calling Party and the request for a Collect Call. The Agent gathers the response from the Called Party and further processes the call.

If the CaIIed Party has agreed to receive the call, the Agent then signals the ISNAP-ACD combination to bridge the Called Party and the Calling Party.
The Agent then cuts a BDR which is used to match with a respective OSR
s generated by the switch to create complete billing information. The ISNAP-ACD combination then bridges the Called Party and the Calling Party and then releases the line back to the switch by executing a Release Trunk (RLT). The Calling Party and the Called Party can now have a conversation through the switch. At the termination of the call by either party, the to switch generates a OSR which will be matched with the BDR generated earlier to create complete billing information for the call. If the Called Party declines to accept the collect call, the Agent signals the ACD-ISNAP
combination to reconnect the Calling Party which was on hold back to the Agent. Finally, the Agent informs the Calling Party about the Called Party's is response and terminates the call in addition to generating a BDR.
MCI Intelligent Network is a scaleable and efficient network architecture for call processing and is based on a set of intelligent processors with specialized software, special purpose bridging switches and ACD's. The 2o Intelligent Network is an overlay network coexisting with the MCI Switching Network and is comprised of a large number of specialized processors interacting with the switch network for call processing. One embodiment of Intelligent Network is completely audio-centric. Data and fax are processed as voice calls with some specialized, dedicated features and value-added 2s services.
In another embodiment, the Intelligent Network is adapted for newly emerging technologies, including POTS-based video-phones and Internet telephony for voice and video. The following sections describe in detail the 3o architecture, features and services based on the emerging technologies.
COMPATIBILITY OF ISN WITH EMERGING TECHNOLOGIES
The following sections describe in detail the architecture, features and services based on several emerging technologies, all of which can be integrated into the Intelligent Network.
VII. ISP FRAMEWORK
s A. Background The ISP is composed of several disparate systems. As ISP integration proceeds, formerly independent systems now become part of one larger whole with concomitant increases in the level of analysis, testing, scheduling, and training in all disciplines of the ISP.
to 1. Broadband Access.
A range of high bandwidth services are supported by a preferred embodiment. These include: Video on Demand, Conferencing, Distance Learning, and Telemedicine.
ATM (asynchronous transfer mode) pushes network control to the periphery of the network, obviating the trunk and switching models of traditional, circuit-based telephony. It is expected to be deployed widely to accommodate these high bandwidth services.
2. Internet Telephony System.
The Internet and with it, the World Wide Web, offers easy customer access, widespread commercial opportunities, and fosters a new role for successful telecommunications companies. The ISP platform offers many features which can be applied or reapplied from telephony to the Internet. These include access, customer equipment, personal accounts, billing, marketing (and advertising) data or application content, and even basic telephone seance.
3o The telecommunication industry is a major transmission provider of the Internet. A preferred embodiment which provides many features from telephony environments for Internet clients is optimal.
Figure 19F is a block diagram of an Internet telephony system in accordance with a preferred embodiment. A number of computers 1900, 1901, 1902 and 1903 are connected behind a firewall 1905 to the Internet 1910 via an Ethernet or other network connection. A domain name system 1906 maps names to IP addresses in the Internet 1910. Individual s systems for billing 1920, provisioning 1922, directory services 1934, messaging services 1930, such as voice messaging 1932 are all attached to the Internet 1910 via a communication link. Another communication link is also utilized to facilitate communications to a satellite device 1940 that is used to communicate information to a variety of set top devices 1941-to 1943. A web server 1944 provides access for an order entry system 1945 to the Internet 1910.
In an embodiment, the order entry system 1945 generates complete profile information for a given telephone number, including, name, address, fax is number, secretary's number, wife's phone number, pager, business address, e-mail address, IP address and phonemail address. This information is maintained in a database that can be accessed by everyone on the network with authorization to do so. In an alternate embodiment, the order entry system utilizes a web interface for accessing an existing 2o directory service database 1934 to provide information for the profile to supplement user entered information.
The Internet 1910 is tied to the Public Switched Network (PSTN) 1960 via a gateway 1950. The gateway 1950 in a preferred embodiment provides a 25 virtual connection from a circuit switched call in the PSTN 1960 and some entity in the Internet 1910.
The PSTN 1960 has a variety of systems attached, including a direct-dial input 1970, a Data Access Point (DAP) 1972 for facilitating 800 number 3o processing and Virtual NETwork (VNET) processing to facilitate for example a company tieline. A Public Branch Exchange (PBX) 1980 is also attached via a communication link far facilitating communication between the PSTN
1960 and a variety of computer equipment, such as a fax 1981, telephone 1982 and a modem 1983. An operator 1973 can also optionally attach to a call to assist in placing a call or conference call coming into and going out of the PSTN 1960 or the Internet 1910.
Various services are attached to the PSTN through individual s communication links including an attachment to the Intelligent Services Network (ISN) 1990, direct-dial plan 1991, provisioning 19?4, order entry 19?5, billing i9?6, directory services 19??, conferencing services 1978, and authorization / authentication services 1979. All of these services can communicate between themselves using the PSTN 1960 and the Internet l0 1910 via a gateway 1950. The functionality of the ISN 1990 and the DAP
19?2 can be utilized by devices attached to the Internet 1910.
Figure 19G is a block diagram of a Prioritizing Access/Router in accordance with a preferred embodiment. A prioritizing access router (PAR) is designed ~5 to combine the features of an Internet access device and an Internet Protocol (IP) Router. It enables dial-up modem access to the Internet by performing essential modem and PPP/ SLIP to IP and the reverse IP to PPP/SLIP conversion. It also analyzes IP packet source/destination addresses and UPD or TCP ports and selects appropriate outgoing network 2o interfaces for each packet. Lastly, it uses a priority routing technique to favor packets destined for specific network interfaces over packets destined for other network interfaces.
The design goal of the prioritizing access/router is to segregate real-time 2s traffic from the rest of the best- effort data traffic on Internet networks.
Real-time and interactive multimedia traffic is best segregated from traffic without real-time constraints at the access point to the Internet, so that greater control over quality of service can be gained. The process that a prioritizing access/router utilizes is presented below with reference to 3o Figure 19G.
First, at 2010, a computer dials up the PAR via a modem. The computer modem negotiates a data transfer rate and modem protocol parameters with the PAR modem. The computer sets up a Point to Point Protocol (PPP) ~g session with the PAR using the modem to modem connection over a Public Switched Telephone Network (PSTN) connection.
The computer transfers Point-to-Point (PPP) packets to the PAR using the modem connection. The PAR modem 2010 transfers PPP packets to the s PPP to IP conversion process 2020 via the modem to host processor interface 2080. The modem to host processor interface can be any physical interface presently available or yet to be invented. Some current examples are ISA, EISA, VME, SCbus, MVIP bus, Memory Channel, and TDM buses.
There is some advantage in using a multiplexed bus such as the Time to Division Multiplexing buses mentioned here, due to the ability to devote capacity for specific data flows and preserve deterministic behavior.
The PPP to IP conversion process 2020 converts PPP packets to IP packets, and transfers the resulting IP packets to the packet classifier 2050 via the is process to process interface 2085. The process to process interface can be either a physical interface between dedicated processor hardware, or can be a software interface. Some examples of process to process software interfaces include function or subroutine calls, message queues, shared memory, direct memory access (DMA), and mailboxes.
The packet classifier 2085 determines if the packet belongs to any special prioritized group. The packet classifier keeps a table of flow specifications, defined by destination IP Address source IP address combined source/destination IP Address combined destination IP Address/UDP Port combined destination IP Address/TCP Port combined source IP address/UDP Port 3o combined source IP Address/TCP Port combined source IP Address and TCP or UDP port with destination IP
address combined destination IP Address and TCP or UDP port with source IP
address ~9 combined source IP Address and TCP or UDP port with destination IP
address and TCP/UDP Port.
The packet classifier checks its table of flow specifications against the IP
s addresses and UDP or TCP ports used in the packet. If any match is found, the packet is classified as belonging to a priority flow and labeled as with a priority tag. Resource Reservation Setup Protocol techniques may be used for the packet classifier step.
to The packet classifier 2050 hands off priority tagged and non-tagged packets to the packet scheduler 2060 via the process to process interface (90). The process to process interface 2090 need not be identical to the process to process interface 2085, but the same selection of techniques is available. The packet scheduler 2060 used a priority queuing technique is such as Weighted Fair Queueing to help ensure that prioritized packets (as identified by the packet classifier) receive higher priority and can be placed on an outbound network interface queue ahead of competing best-effort traffic.
2o The packet scheduler 2060 hands off packets in prioritized order to any outbound network interface (2010, 2070, 2071 or 2072) via the host processor to peripheral bus 2095. Any number of outbound network interfaces may be used.
25 IP packets can arrive at the PAR via non-modem interfaces (2070, 2071 and 2072). Some examples of these interfaces include Ethernet, fast Ethernet, FDDI, ATM, and Frame Relay. These packets go through the same steps as IP packets arriving via the modem PPP interfaces.
3o The priority flow specifications are managed through the controller process 2030. The controller process can accept externally placed priority reservations through the external control application programming interface 2040. The controller validates priority reservations for particular flows against admission control procedures and policy procedures, and if S'D

the reservation is admitted, the flow specification is entered in the flow specification table in the packet classifier 2050 via the process to process interface 2065. The process to process interface 2065 need not be identical to the process to process interface 2085, but the same selection of techniques is available.
Turning now to Figure 20, there is shown an architectural framework for an Intelligent Services Platform (ISP) 2100, used in the present invention.
The architecture of the ISP 2100 is intended to define an integrated to approach to the provision and delivery of intelligent services to the MCI
network across all the components of the ISP.
Each of the existing communication network systems has its own way of providing service management, resource management, data management, security, distributed processing, network control, or operations support.
The architecture of the ISP 2100 defines a single cohesive architectural framework covering these areas. The architecture is focused on achieving the following goals:
~ Develop global capabilities;
~ Deliver enhanced future services;
~ Make efficient use of resources;
~ Improve time to market;
~ Reduce maintenance and operations costs;
~ Increase overall product quality; and z5 ~ Introduce scalability both upward and downward capabilities.
The target capabilities of the ISP 2100 are envisioned to provide the basic building blocks for very many services. These services are characterized as - providing higher bandwidth, greater customer control or personal flexibility, 3o and much reduced , even instantaneous, provisioning cycles.
3. Capacity .
The ISP 2100 has a reach that is global and ubiquitous. Globally, it will reach every country through alliance partners' networks. In breadth, it WO 98/47298 PCT/US98/0?927 reaches alI business and residential locales through wired or wireless access.
4. F~.tture Services.
The above capabilities will be used to deliver:
~ Telephony and messaging services beyond what we have today;
~ Emerging video and multi-media offerings;
~ Powerful data services, including enhanced private networks; and ~ Software and equipment to enable end users to gain complete 1o control over their services.
Services provided by the ISP 2100 will span those needed in advertising, agriculture, education, entertainment, finance, government, law, manufacturing, medicine, network transmission, real estate, research, t5 retailing, shipping, telecommunications, tourism, wholesaling, and many others.
Services:
~ Customizable: customer is able to tailor the service offerings to their 20 own needs.
~ Customer managed: customer has direct (network-side) access for the administration and control of their service.
~ Loosely Coupled: services obtain and use network resources only when needed; customers pay for only what they use. Bandwidth is 25 available on demand, and without pre-allocation.
~ Secure & Private: customer privacy and confidentiality is paramount in the networked world. Commercial interests are guaranteed safe, secure transactions. Users and customers are identified and authenticated, and the network protected from tampering or 3o corruption.
B. ISP Architecture Framework The following section describes the role of the ISP Platform 2100 in providing customer services.
S'~2 The ISP 2100 provides customer services through an intelligent services infrastructure, including provider network facilities 2102, public network facilities 2104, and customer equipment 2106. The services infrastructure ensures the end-to-end quality and availability of customer service.
The following section describes the relationship of the ISP platform 2100 to various external systems both within and outside a provider.
1o The provider components 2108 in Figure 20 are:
~ Intelligent Services 2110 - responsible for service provisioning, service delivery, and service assurance, including the internal data communications networks 2102. This represents the ISP's role.
is ~ Revenue Management 2112 - responsible for financial aspects of customer services.
~ Network Management 2114 - responsible for the development and operation of the physical networks 2102.
~ Product Management 2116 - responsible for the creation and marketing of zo customer services.
The entities external to the ISP 2100 depicted in Figure 20 are:
~ Networks 2104- this represents all the network connections and access methods used by customers 2106 for service. This includes a provider's circuit switched network, packet switched networks, internal extended wide 2s area network, the Internet, a provider's wireless partners' networks, a provider's global alliance and national partner networks, broadband networks, as well as the customer premises equipment 2118 attached to these networks.
~ 3rd party Service Providers 2120 - this represents those external 30 organizations which deliver services to customers via the provider's Intelligent Services Platform 2100.
~ Service Resellers 2122 - this represents those organizations which have customers using the facilities 2100.
~ Global Alliance Partners 2124 - organizations which have shared facilities S'~3 and exchange capabilities of their networks and service infrastructures.
C. ISP Functional Figure 21 shows components of the ISP 2100 in more detail. Shown is the set of logical components comprising the ISP 2100 architecture. None of these components is a single physical entity; each typically occurs multiple times in multiple locations. The components work together to provide a seamless Intelligent Services 2110 environment. This environment is not fixed; it is envisioned as a flexible evolving platform capable of adding new 1o services and incorporating new technologies as they become available. The platform components are linked by one or more network connections which include an internal distributed processing infrastructure.
The ISP 2100 Functional Components are:
t5 ~ Inbound and Outbound Gateways 2126 - allows access to services provided by other providers, and allows other providers to access the provider's services.
~ Marketable Service Gateway 2128- interface to a three-tier service creation environment for services the provider sells. Services are deployed 2o and updated through the Marketable Service Gateway 2128. This is actually no different than the Management Service Gateway 2130, except that the services created and deployed through here are for external customers.
25 ~ Management Service Gateway 2130 - illustrates that service creation concepts apply to management of the platform as well as service logic.
Management services are deployed and managed through the Management Service Gateway 2130. Also, interfaces with management systems external to ISP 2100 are realized by the Management Service Gateway 2130. Some 3o examples of management services include the collection, temporary storage, and forwarding of (billable) network events. Other services include collection and filtering of alarm information from the ISP 2100 before forwarding to network management 2132.
S 5( ~ Service Engines 2134 - A Service Logic Execution Environment for either marketable or management services. The Service Engines 2134 execute the logic contained in customer-specific profiles in order to provide unique customized service features.
~ Service Creation Environment 2136 - Creates and deploys management services as well as marketable services, and their underlying features and capabilities.
to ~ Data Management 2138- Where all customer and service profile data is deployed. Data is cached on Service Engines 2134, Statistics Servers 2140, Call Context servers 2142, Analysis Servers 2144, and other specialized applications or servers 2146 requiring ISP 2100 data.
~ Service Select 2148 - Whether the services are accessed via a narrowband or broadband network, circuit-switched, packet-switched, or cell-switched, the services are accessed via a Service Select function 2148. Service Select 2148 is a specialized version of a service engine 2134, designed specifically to choose a service or services to execute.
~ Resource Managers 2150 - manages all resources, including specialized resources 2152 and service instances running on service engines 2134, and any other kind of resource in the ISP 2100 that needs management and allocation.
~ Specialized Resources 2152 - Special network-based capabilities (Internet to voice conversion, DTMF-detection, Fax, Voice Recognition, etc) are shown as specialized resources 2152.
~ Call Context Server 2142 - accepts network event records and service event records in real time, and allows queries against the data. Once all events for a call (or any other kind of network transaction) are generated, the combined event information is delivered en masse to the Revenue Management function 2154. Data is stored short-term.
~S',5~

~ Statistics Server 2140- accepts statistics events from service engines, performs rollups, and allows queries against the data. Data is stored short-term.
s ~ Customer Based Capabilities 2156- software and specialized hardware on the customer premises that enables customer-premises based capabilities, such as ANI screening, Internet access, compression, interactive gaming, videoconferencing, retail access, you name it.
~ Analysis Services 2144- a special kind of service engine that isn't based on network access, but is based on adding value based upon network statistics or call context information in real time or near real time.
Examples include fraud detection and customer traffic statistics.
~ Other Special Services 2146- entail other specialized forms of applications or servers that may or may not be based on the Service Engine model.
These components provide other computing resources and lower-level functional capabilities which may be used in Service delivery, monitoring, or management.
D. ISP Integrated Network Services Figure 22 shows how the ISP architecture 2100 supplies services via different networks. The networks shown include Internet 2160, the public 2s switched telephony network (PSTN) 2162, Metro access rings 2164, and Wireless 2166. Additionally, it is expected that new "switchless"
broadband network architectures 2168 and 2170 such as ATM or ISO
Ethernet may supplant the current PSTN networks 2162.
3o The architecture accommodates networks other than basic PSTNs 2162 due to the fact that these alternative network models support services which cannot be offered on a basic PSTN, often with an anticipated reduced cost structure. These Networks are depicted logically in Figure 22.

Each of these new networks are envisioned to interoperate with the ISP
2100 in the same way. Calls (or transactions) will originate in a network from a customer service request, the ISP will receive the transaction and provide service by first identifying the customer and forwarding the transaction to a generalized service-engine 21?4. The service engine determines what service features are needed and either applies the necessary logic or avails itself of specialized network resources for the needed features.
The ISP 2100 itself is under the control of a series of Resource managers and Administrative and monitoring mechanisms. A single system image is enabled through the concurrent use of a common information base. The information base holds all the Customer, Service, Network and Resource information used or generated by the ISP. Other external applications (from ~5 within MCI and in some cases external to MCI) are granted access through gateways, intermediaries, and sometimes directly to the same information base.
In Figure 22, each entity depicts a single logical component of the ISP.
2o Each of these entities is expected to be deployed in multiple instances at multiple sites.
E. ISP Components Ext App 2I?6- an external application;
25 App 21?8- an internal ISP application (such as Fraud Analysis);
Dc 2180- Data client, a client to the ISP information base which provides a local data copy;
Ds 2182- Data server, one of the master copies of ISP information;
Admin 2184- the ISP administrative functions (for configurations, and 30 maintenance);
Mon 2186- the ISP monitoring functions (for fault, performance, and accounting);
GRM 2188- the global resource management view for selected resources;
LRM 2190- the local resource management view for selected resources;

SR 2192- the pools of specialized resources (such as video servers, ports, speech recognition);
SE 2134- the generalized service engines which execute the desired service logic; and s Service Select 2194- the function which selects the service instance (running on a service engine 2134) which should process transactions offered from the networks.
F. Switchless Communications Services to The switchless network 2168 is a term used for the application of cell-switching or packet-switching techniques to both data and isochronous multimedia communications services. In the past, circuit switching was the only viable technology for transport of time-sensitive isochronous voice.
Now, with the development of Asynchronous Transfer Mode cell switching 1s networks which provide quality of service guarantees, a single network infrastructure which serves both isochronous and bursty data services is achievable.
The switchless network is expected to provide a lower cost model than 2o circuit switched architectures due to:
~ Flexibility to provide exactly the bandwidth required for each application, saving bandwidth when no data is being transferred. A minimum 56 Kbps circuit will not automatically be allocated for every call.
~ Adaptability to compression techniques, further reducing bandwidth 2s requirements for each network session.
~ Lower costs for specialized resource equipment, due to the fact that analog ports do not have to be supplied for access to special DSP capabilities such as voice recognition or conferencing. A single high-bandwidth network port can serve hundreds of "calls" simultaneously.
30 ~ Applicability and ease of adaptation of the switchless networks to advanced high-bandwidth services such as videoconferencing, training on demand, remote expert, integrated video/voice/fax/electronic mail, and information services. Figure 23 illustrates a sample switchless network 2168 in accordance with a preferred embodiment.

G. Governing Principles 1. Architectural Principles.
This section contains a listing of architectural principles which provide the s foundation of the architecture which follows.
Service Principles 1. The Service Model must support seamless integration of new and existing services.
2. Services are created from a common Service Creation Environment (SCE) which provides a seamless view of services.
3. All services execute in common service logic execution environments (SLEEs), which do not require software changes when new services are introduced.
4. All services are created from one or more service features.
5. Data stored in a single customer profile in the ISP Data Servers may be used to drive multiple services.
6. The Service Model must support the specification and fulfillment of quality of service parameters for each service. These quality of service parameters, when taken together, constitute a service level agreement 2o with each customer. Service deployment must take into account specified quality of service parameters.
2. Service Feature Principles.
1. All service features are described by a combination of one or more capabilities.
2. All service features can be defined by a finite number of capabilities.
3. Individual service features must be defined using a standard methodology to allow service designers to have a common understanding of a capability. Each service feature must document their inputs, outputs, 3o error values, display behaviors, and potential service applications.
4. Interaction of physical entities in the network implementation shall not be visible to the user of the service feature through the service feature interfaces.
S. Each service feature should have a unified and stable external ~9 interface. The interface is described as a set of operations, and the data required and provided by each operation.
6. Service features are not deployed into the network by themselves. A
service feature is only deployed as part of a service logic program which invokes the service feature (see Figure 21). Thus, service features linked into service logic programs statically, while capabilities are linked to service logic programs dynamically. This is where the loose coupling of resources to services is achieved.
3. Capability Principles.
1. Capabilities are defined completely independent from consideration of any physical or logical implementation (network implementation independent).
2. Each capability should have a unified and stable interface. The is interface is described as a set of operations, and the data required and provided by each operation.
3. Individual capabilities must be defined using a standard methodology to allow service designers to have a common understanding of a capability. Each capability must document their inputs, outputs, error 2o values, display behaviors, and potential service applications.
4. Interaction of physical entities in the network implementation shall not be visible to the user of the capability through the capability interfaces.
5. Capabilities may be combined to form high-level capabilities.
25 6. An operation on a capability defines one complete activity. An operation on a capability has one logical starting point and one or more logical ending points.
7. Capabilities may be realized in one or more piece of physical hardware or software in the network implementation.
30 8. Data required by each capability operation is defined by the capability operation support data parameters and user instance data parameters.
9. Capabilities are deployed into the network independent of any service.
~U

10. Capabilities are global in nature and their location need not be considered by the service designer, as the whole network is regarded as a single entity from the viewpoint of the service designer.

11. Capabilities are reusable. They are used without modification for other services.
4. Service Creation, Deployment, and Execution Principles.
1. Each Service Engine 2134 supports a subset of the customer base.
to The list of customers supported by a service engine is driven by configuration data, stored on the ISP Data Server 2182.
2. Each Service Engine 2134 obtains its configuration data from the ISP
data servers 2152 at activation time.
3. Service Engines 2134 use ISP database clients 218U (see the data ~5 management section of this description) to cache the data necessary to support the customers configured for that service engine 2134, as needed.
Caching can be controlled by the ISP database server 2182, or controlled by the database of the ISP database server 2182. Data may be cached semi-permanently (on disk or in memory) at a service engine 2134 if it is 2o deemed to be too much overhead to load data from the data server 2182 on a frequent basis.
4. Service Engines 2134 may be expected to execute all of a customer's services, or only a subset of the customer's services. However, in the case of service interactions, one Service Engine 2134 must always be in control 2s of the execution of a service at any given time. Service Engines may hand-off control to other service engines during the course of service execution.
5. Service Engines do not own any data, not even configuration data.
Service Engines 2134 are not targets for deployment of data. Data Servers 2182 are targets for deployment of data.
5. Resource Management Model 2150 Principles.
1. Resources 2152 should be accessible from anywhere on the network.
2. Resources are not service-specific and can be shared across all services if desired.

3. Resources of the same type should be managed as a group.
4. The Resource Management Model 2150 should be flexible enough to accommodate various management policies, including: Least Cost, Round Robin, Least Recently Used, Most Available, First Encountered, Use Until Failure and Exclusive Use Until Failure.
5. The Resource Management Model 2150 should optimize the allocation of resources and, if possible, honoring a selected policy.
6. The RM 2150 must allow for a spectrum of resource allocation techniques ranging from static configuration to fully dynamic allocation of to resources on a transaction by transaction basis.
7. The Resource Management Model 2150 must allow for the enforcement of resource utilization policies such as resource time out and preemptive reallocation by priority.
8. The Resource Management Model 2150 must be able to detect and ~s access the status, utilization and health of resources in a resource pool.
9. All Resources 2152 must be treated as managed objects.
10. All resources must be able to register with the RM 2150 to enter a pool, and de-register to leave a pool.
I 1. The only way to request, acquire and release a resource 2152 is 2o through the RM 2150.

12. The relationship between resources should not be fixed, rather individual instances of a given resource should be allocated from a registered pool in response to need or demand.

13. All specialized resources 2152 must be manageable from a consistent 25 platform-wide viewpoint.

14. All specialized resources 2152 must offer SNMP or CMIP agent functionality either directly or through a proxy.

15. Every specialized resource 2152 shall be represented in a common management information base.
30 16. All specialized resources shall support a standard set of operations to inquire, probe, place in or out of service, and test the item.
17. All specialized resources shall provide a basic set of self-test capabilities which are controlled through the standard SNMP or CMIP
management interfaces.

6. Data Management 2138 Principles.
1. Multiple copies of any data item are allowed.
2. Multiple versions of the value of a data item are possible, but one s view is considered the master.
3. Master versions of a given data item are under a single jurisdiction.
4: Multiple users are allowed to simultaneously access the same data.
5. Business rules must be applied uniformly across the ISP 2100 to ensure the validity of all data changes.
6. Users work on local copies of data; data access is location independent and transparent.
7. From the data management point of view, users are applications or other software components.
8. Data access should conform to a single set of access methods which ~s is standardized across the ISP 2100.
9. Private data is allowed at a local database, but cannot be shared or distributed.
10. Only master data can be shared or distributed.
11. Private formats for a shared data item are allowed at the local 2o database.
12. Transactional capabilities can be relaxed at end-user discretion if allowed within the business rules.
I3. Rules-based logic and other meta-data controls provide a flexible means to apply policy.
25 14. Data Replication provides reliability through duplication of data sources.
15. Database Partitioning provides scalability by decreasing the size of any particular data store, and by decreasing the transaction rate against any particular data store.
30 16. Data Management 2138 must allow both static and dynamic - configuration of data resources.
17. Common data models and common schemas should be employed.
18. Logical application views of data are insulated from physical data operations such as relocation of files, reloading of databases, or l~

reformatting of data stores.
19. Audit trails, and event histories, are required for adequate problem resolutions.
20. On-line data audits and reconciliation are required to ensure data s integrity.
21. Data recovery of failed databases is needed in real time.
22. Data metrics are needed for monitoring, trending, and control purposes.
23. 7 by 24 operation with 99.9999 availability is required.
to 24. Data Management 2138 mechanisms must scale for high levels of growth.
25. Data Management 2138 mechanisms must provide cost effective solutions for both Large-scale and small-scale deployments.
26. Data Management mechanisms must handle overload conditions is gracefully.
27. Data processing and data synchronization must occur in real-time to meet our business needs.
28. Trusted order entry and service creation should work directly on the ISP databases rather than through intermediary applications whenever 2o possible.
29. All data must be protected; additionally customer data is private and must retain its confidentiality.
30. Configurations, operational settings, and run-time parameters are mastered in the ISP MIB (management information base).
25 31. Wherever possible, off the shelf data solutions should be used to meet Data Management needs.
The following principles are stated from an Object-oriented view:
32. Data items are the lowest set of persistent objects; these objects encapsulate a single data value.
3o 33. Data items may have a user defined type.
34. Data items may be created and deleted.
35. Data items have only a single get and set method.
36. The internal value of a data item is constrained by range restrictions and rules.

37. Data items in an invalid state should be inaccessible to users.
7. Operational Support Principles.
i. Common View - All ISP 2100 Operational Support User Interfaces - 5 should have the same look 8v feel.
2. Functional Commonality - The management of an object is represented in the same manner throughout the ISP Operational support environment.
3. Single View - A distributed managed object has a single to representation at the ISP Operational Support User Interfaces, and the distribution is automatically.
4. OS/DM Domain - Data within the Operational support domain should be managed with the ISP Data Management 2138 Mechanisms.
5. Global MIB - There is a logical Global MIB which represents is resources in the entire ISP.
6. External MIBs - Embedded MIBs that are part of a managed component are outsider of Operational Support and Data Management.
Such MIBs will be represented to the OS by a Mediation Device.
7. System Conformance - System conformance to the ISP OS standards zo wili be gained through Mediation Layers.
8. Operational Functions - Operational personnel handle the Network Layer & Element Management for physical & logical resources.
9. Administration Functions - Administration personnel handle the Planning & Service Management.
2s 10. Profile Domain - Service & customer profile data bases are managed by administration personnel under the domain of the Data Management system.
11. Telecommunication Management Network (TMN) compliance - TMN
compliance will be achieved through a gateway to any TMN system.
30 12. Concurrent - Multiple Operators & Administrators must be able to simultaneously perform operations from the ISP OS Interfaces.
8. Physical Model Principles.
1. Compatibility: The physical network model provides backward WO 98/47298 PCTlUS98/07927 compatibility for existing telecommunications hardware and software.
2. Scaleable: The physical network model is scaleable to accommodate a wide range of customer populations and service requirements.
3. Redundant: The physical network model provides multiple paths of s information flow across two network elements. Single points of failure are eliminated.
4. Transparent: Network elements is transparent to the underlying network redundancy. In case of a failure, the switchover to redundant links is automatic.
to 5. Graceful Degradation: The physical network model is able to provide available services in a gradual reduction of capacity in the face of multiple network failures.
6. Interoperable: The physical network model allows networks with different characteristics to interoperate with different network elements.
is 7. Secure: The physical network model requires and provides secure transmission of information. It also has capabilities to ensure secure access to network elements.
8. Monitoring: The physical network model provides well-defined interfaces and access methods for monitoring the traffic on the network.
2o Security (see above) is integrated to prevent unauthorized access to sensitive data.
9. Partitionable: The physical network model is (logically) partitionable to form separate administrative domains.
10. Quality of Service: The physical network model provides QOS
2s provisions such as wide range of qualities, adequate QOS for legacy applications, congestion management and user-selectable QOS.
I 1. Universal Access: The physical network model does not prevent access to a network element due to its location in the network. A service is able to access any resource on the network.
30 12. Regulatory awareness: The physical network model is amenable at all levels to allow for sudden changes in the regulatory atmosphere.
13. Cost Effective: The physical network model allows for cost effective implementations by not being reliant on single vendor platforms or specific standards for function.
6~

H. ISP Service This section describes the Service model of the Intelligent Services Platform Architecture Framework.
° 5 1. Purpose.
The ISP Service Model establishes a framework for service development which supports:
~ rapid service creation and deployment;
~ efficient service execution;
~ complete customization control over services for customers;
~ total service integration for a seamless service view for customers;
~ improved reuse of ISP capabilities through loose coupling of those capabilities;
15 ~ reduced cost of service implementation; and ~ vendor-independence.
2. Scope of Effort.
The ISP Service Model supports all activities associated with Services, 2o including the following aspects:
~ provisioning;
~ creation;
~ deployment;
~ ordering;
2s ~ updating;
~ monitoring;
~ execution;
~ testing or simulation;
~ customer support and troubleshooting;
30 ~ billing;
° ~ trouble ticket handling; and ~ operations support.
This model covers both marketable services and management services.
b~

~ Marketable services are the services purchased by our customers ~ Management services are part of the operation of the MCI network, and are not sold to customers.
s The Service Model also defines interactions with other parts of the ISP
Architecture, including Data Management, Resource Management, and Operational Support.
3. Service Model Overview.
io Central to the Intelligent Services Platform is the delivery of Services (Figure 24). Services are the most critical aspect in a telecommunication service provider's ability to make money. The following definition of services is used throughout this service model:
A service 2200 is a set of capabilities combined with well-defined logic 15 structures and business processes which, when accessed through a published interface, results in a desired and expected outcome on behalf of the user.
One of the major differences between a Service 2200 and an Application 20 2176 or 2178 (Figure 22) is that a Service 2200 includes the business processes that support the sale, operation, and maintenance of the Service.
The critical task in developing a Service is defining what can be automated, and clearly delineating how humans interact with the Service.
25 4. Service Structure.
The vocabulary we will use for describing services includes the services themselves, service features, and capabilities. These are structured in a three-tier hierarchy as shown in Figure 24.
3o A service 2200 is an object in a sense of an object-oriented object as described earlier in the specification. An instance of a service 2200 contains other objects, called service features 2202. A service feature 2202 provides a well defined interface which abstracts the controlled interaction of one or more capabilities 2204 in the ISP Service Framework, on behalf of a seance.
Service features 2202, in turn, use various capability 2204 objects.
Capabilities 2204 are standard, reusable, network-wide building blocks - s used to create service features 2202. The key requirement in Service Creation is for the engineers who are producing basic capability objects to insure each can be reused in many different services as needed.
a) Services 2200 Services 2200 are described by "service logic," which is basically a program io written in a very high-level programming language or described using a graphical user interface. These service logic programs identify:
~ what service features 2202 are used;
~ the order in which service features are invoked;
~ the source of input service data;
~5 ~ the destination for output service data;
~ error values and error handling;
~ invocation of other services 2200;
~ interaction with other services; and ~ the interactions with other services;
The service logic itself is generally not enough to execute a service 2200 in the network. Usually, customer data is needed to define values for the points of flexibility defined in a service, or to customize the service for the customer's particular needs. Both Management and Marketable Services 2s are part of the same service model. The similarities between of Management and Marketable Services allow capabilities to be shared. Also, Management and Marketable Services represent two viewpoints of the same network: Management Services represent and operational view of the network, and Marketable Services represent an external end-user or 3o customer view of the network. Both kinds of services rely on network data which is held in common.
Every Marketable Service has a means for a customer to order the service, a billing mechanism, some operational support capabilities, and service 6y monitoring capabilities. The Management Services provide processes and supporting capabilities for the maintenance of the platform.
b) Service Features 2202 s Service features 2202 provide a well-defined interface of function calls.
Service features can be reused in many different services 2200, just as capabilities 2204 are reused in many different service features 2202.
Service features have specific data input requirements, which are derived from the data input requirements of the underlying capabilities. Data output behavior of a service feature is defined by the creator of the service feature, based upon the data available from the underlying capabilities.
Service Features 2202 do not rely on the existence of any physical resource, rather, they call on capabilities 2204 for these functions, as shown in Figure 25.
Some examples of service features are:
~ Time-based Routing - based on capabilities such as a calendar, date/time, and call objects, this feature allows routing to different locations based upon time.
~ Authentication - based upon capabilities such as comparison and database lookup, this function can be used to validate calling card use by prompting for a card number and/or an access number (pin number), or to validate access to a virtual private network.
~ Automated User Interaction - based upon capabilities such as voice objects (for recording and playback of voice), call objects (for transferring and bridging calls to specialized resources), DTMF
objects (for collection or outpulsing of DTMF digits), vocabulary objects (for use with speech recognition), this feature allows automated interaction with the user of a service. This service 3o feature object can be extended to include capabilities for video interaction with a user as well.
c) Capabilities 2204 A capability 2204 is an object, which means that a capability has internal, ~o ~........~...~,..,~...... ........,_,...~._ . ..."*........ _.p... ..,.

private state data, and a well-defined interface for creating, deleting, and using instances of the capability. Invoking a capability 2204 is done by invoking one of its interface operations. Capabilities 2204 are built for reuse. As such, capabilities have clearly defined data requirements for - s input and output structures. Also, capabilities have clearly defined error handling routines.
Capabilities may be defined in object-oriented class hierarchies whereby a general capability may be inherited by several others.
Some examples of network-based capability objects are:
~ voice (for recording or playback), ~ call (for bridging, transferring, forwarding, dial-out, etc}, ~ DTMF (for collection or outpulsing), and ~ Fax (for receive, send, or broadcast).
is Some capabilities are not network-based, but are based purely on data that has been deployed into our platform. Some examples of these capabilities are:
~ calendar (to determine what day of the week or month it is), zo ~ comparison (to compare strings of digits or characters), ~ translation (to translate data types to alternate formats), and ~ distribution (to choose a result based on a percentage distribution).
d) Service Data 2s There are three sources for data while a service executes:
~ Static Data defined in the service template, which include default values for a given service invocation.
~ Interactive Data obtained as the service executes, which may be explicit user inputs or derived from the underlying network 30 connections.
~ Custom Data defined in User Profiles, which is defined by customers or their representatives when the service is requested (i.e. at creation time).

5. Service 2200 Execution.
Services 2200 execute in Service Logic Execution Environments (SLEEs).
A SLEE is executable software which allows any of the services deployed into the ISP 2100 to be executed. In the ISP Architecture, Service Engines s 2134 (Figure 21) provide these execution environments. Service Engines 2134 simply execute the services 2200 that are deployed to them.
Service templates and their supporting profiles are deployed onto database servers 2182 (Figure 22). When a SLEE is started on a Service Engine to 2134, it retrieves its configuration from the database server 2182. The configuration instructs the SLEE to execute a Iist of services 2200. The software for these services is part of the service templates deployed on the database servers. If the software is not already on the Service Engine 2134, the software is retrieved from the database server 2182. The 15 software is executed, and service 200 begins to run.
In most cases a service 2200 will first invoke a service feature 2202 (Figure 24) which allows the service to register itself with a resource manager 2188 or 2190. Once registered, the service can begin accepting transactions.
2o Next, a service 2200 will invoke a service feature 2202 which waits on an initiating action. This action can be anything from an Internet logon, to an 800 call, to a point of sale card validation data transaction. Once the initiating action occurs in the network, the service select function 2148 (Figure 21) uses the Resource Manager 2150 function to find an instance 25 of the executing service 2200 to invoke. The initiating action is delivered to the service 2200 instance, and the service logic (from the service template) determines subsequent actions by invoking additional service features 2202.
3o During service 2200 execution, profile data is used to determine the behavior of service features 2202. Depending on service performance requirements, some or all of the profile data needed by a service may be cached on a service engine 2134 from the ISP 2100 database server 2182 to prevent expensive remote database lookups. As the service executes, information may generated by service features 2202 and deposited into the Context Database. This information is uniquely identified by a network transaction identifier. In the case of a circuit-switched call, the already-defined Network Call Identifier will be used as the transaction identifier.
s Additional information may be generated by network equipment and deposited into the Context Database as well, also indexed by the same unique transaction identifier. The final network element involved with the transaction deposits some end-of transaction information into the Context Database. A linked list strategy is used for determining when all to information has been deposited into the Context Database for a particular transaction. Once all information has arrived, an event is generated to any service which has subscribed to this kind of event, and services may then operate on the data in the Context Database. Such operations may include extracting the data from the Context Database and delivering it to billing ~ s systems or fraud analysis systems.
6. Service Interactions.
In the course of a network transaction, more than one service can be invoked by the network. Sometimes, the instructions of one service may 2o conflict with the instructions of another service. Here's an example of such a conflict: a VNET caller has a service which does not allow the caller to place international calls. The VNET caller dials the number of another VNET user who has a service which allows international dialing, and the called VNET user places an international call, then bridges the first caller 2s with the international call. The original user was able to place an international call through a third party, in defiance of his company's intention to prevent the user from dialing internationally. In such circumstances, it may be necessary to allow the two services to interact with each other to determine if operation of bridging an international call 3o should be allowed.
The ISP service model must enable services 2200 to interact with other services. There are several ways in which a service 2200 must be able to interact with other services (see Figure 26):

~ Transfer of Control 2210: where a service has completed its execution path and transfers control to another service;
~ Synchronous Interaction 2212: where a service invokes another service and waits for a reply;
s ~ Asynchronous Interaction 2214: where a service invokes another service, performs some other actions, then waits for the other service to complete and reply; or ~ One Way Interaction 2216: where a service invokes another service but does not wait for a reply.
In the example of interacting VNET services above, the terminating VNET
service could have queried the originating VNET service using the synchronous service interaction capability. The interesting twist to this idea is that service logic can be deployed onto both network-based Is platforms and onto customer premises equipment. This means that service interaction must take place between network-based services and customer-based services.
7. Service Monitoring.
2o Services 2200 must be monitored from both the customer's viewpoint and the network viewpoint. Monitoring follows one of two forms:
~ The service 2200 can generate detailed event-by-event information for delivery to the transaction context database ~ The service can generate statistical information for delivery periodically to 25 a statistics database, or for retrieval on demand by a statistics database.
Analysis services can use the Statistics Database or the Context Database to perform real time or near real time data analysis services.
The Context Database collects all event information regarding a network 3o transaction. This information will constitute all information necessary for network troubleshooting, billing, or network monitoring.
I. ISP Data Management Mode This section describes the Data Management 2138 aspects of the Intelligent Services Platform (ISP) 2100 Target Architecture.
1. Scope.
The ISP Data Management 2138 Architecture is intended to establish a model which covers the creation, maintenance, and use of data in the ' S production environment of the ISP 2100, including all transfers of information across the ISP boundaries.
The Data Management 2138 Architecture covers all persistent data, any copies or flows of such data within the ISP, and all flows of data across the to ISP boundaries. This model defines the roles for data access, data partitioning, data security, data integrity, data manipulation, plus database administration. It also outlines management policies when appropriate.
2. Purpose.
t5 The objectives of this architecture are to:
~ Create a common ISP functional model for managing data;
~ Separate data from applications;
~ Establish patterns for the design of data systems;
~ Provide rules for systems deployment;
20 ~ Guide future technology selections; and ~ Reduce redundant developments and redundant data storage.
Additional goals of the target architecture are:
~ Ensure data flexibility;
25 ~ Facilitate data sharing;
~ Institute ISP-wide data control and integrity;
~ Establish data security and protection;
~ Enable data access and use;
' ~ Provide high data performance and reliability;
30 ~ Implement data partitioning; and ~ Achieve operational simplicity.
3. Data management Overview.
In one embodiment, the Data Management Architecture is a framework ~S~

describing the various system components, how the systems interact, and the expected behaviors of each component. In this embodiment data is stored at many locations simultaneously, but a particular piece of data and all of its replicated copies are viewed logically as a single item. A key s difference in this embodiment is that the user (or end-point) dictates what data is downloaded or stored locally.
a) Domains Data and data access are characterized by two domains 2220 and 2222, as io shown in Figure 27. Each domain can have multiples copies of data within it. Together, the domains create a single logical global database which can span international boundaries. The key aspect to the domain definitions below is that all data access is the same. There is no difference in an Order Entry feed from a Call Processing lookup or Network side data 1 s update.
Central domain 2220 controls and protects the integrity of the system.
This is only a logical portrayal, not a physical entity. Satellite domain 2222 provides user access and update capabilities. This is only a logical 2o portrayal, not a physical entity.
b) Partitions In general, Data is stored at many locations simultaneously. A particular 25 piece of data and all of its replicated copies are viewed logically as a single item. Any of these copies may be partitioned into physical subsets so that not all data items are necessarily at one site. However partitioning preserves the logical view of only one, single database.
3o c) Architecture The architecture is that of distributed databases and distributed data access with the following functionality:
~ Replication and Synchronization;
~ Partitioning of Data Files;

~ Concurrency Controls;
~ Transactional Capability; and ~ Shared common Schemas.
' s Figure 28 shows logical system components and high-level information flows. None of the components depicted is physical. Multiple instances of each occur in the architecture.
The elements in Figure 28 are:
~ NETWK 2224 - external access to the ISP 2100 from the network side;
to ~ SVC I/F 2226 -the network interface into ISP;
~ SYSTMS 2228 - external application such as Order Entry;
~ G/W 2230 - a gateway to the ISP 2100 for external applications;
~ dbAppl 2232 - a role requiring data access or update capabilities;
~ dbClient 2234- the primary role of the satellite domain;
~5 ~ dbServer 2236- the primary role of the central domain;
~ dbAdmin 2238- an administrative role for Data;
~ dbMon 2240- a monitoring role;
~ I / F Admin 2242 administrative role for interfaces; and ~ Ops 2244- operations console.
d) Information Flow The flows depicted in Figure 28 are logical abstractions; they are intended to characterize the type of information passing between the logical components.
The flows shown above are:
~ Rest - data requests to the ISP from external systems;
~ Resp -responses from the ISP to external requests;
~Access - data retrieval by applications within the ISP;
' ~ Updates -data updates from applications within ISP;
~ Evts, data related events sent to the monitor;
~ Meas - data related metrics sent to the monitor;
~ New Data -additions to ISP master data;
~ Changed Data changes to ISP master data;
~ Views - retrieving ISP master data;
~t ~ Subscriptions -asynchronous stream of ISP master data;
~ Cache copies- a snapshot copy of ISP master data;
~ Actions- any control activity; and ~ Controls any control data.
e) Domain Associations In general the Satellite domains 2222 of Data Management 2138 encompass:
~ iSP Applications;
~ External systems ;
~ Network interfaces 2226 and system gateways 2230; and ~ Database client (dbClient) 2234.
The Central domain for Data Management 2138 encompasses:
~ Monitoring (dbMon) 2240;
~ Administration (dbAdmin) 2238; and ~ Database masters (dbServer) 2236 4. Logical Description .
2o The behavior of each Architecture component is described separately below:
a) Data Applications (dbAppl) 2232 This includes any ISP applications which require database access.
Examples are the ISN NIDS servers, and the DAP Transaction Servers, The applications obtain their required data from the dbClient 2234 by attaching to the desired databases, and providing any required policy instructions.
These applications also provide the database access on behalf of the external systems or network element such as Order Entry or Switch requested translations. Data applications support the following 3o functionality:
~ Updates: allow an application to insert, update, or delete data in an ISP
database.
~ Access requests allow an application to search for data, list multiple items, select items from a list or set, or iterate through members of a set.

~ Events and Measurements are special forms of updates which are directed to the monitoring function (dbMon) 2240.
b) Data Management 2138 (1) Client Databases ( dbClient) 2234 The dbClients represent satellite copies of data. This is the only way for an application to access ISP data. Satellite copies of data need not match the format of data as stored on the dbServer 2236.
io The dbClients register with master databases (dbServer) 2236 for Subscriptions or Cache Copies of data. Subscriptions are automatically maintained by dbServer 2236, but Cache Copies must be refreshed when the version is out of date.
A critical aspect of dbClient 2234 is to ensure that data updates by applications are serialized and synchronized with the master copies held by dbServer 2236. However, it is just as reasonable for the dbClient to accept the update and only later synchronize the changes with the dbServer (at 2o which time exception notifications could be conveyed back to the originating application). The choice to update in lock-step, or not, is a matter of application policy not Data Management 2138.
Only changes made to the dbServer master copies are forwarded to other dbClients.
If a dbClient 2234 becomes inactive or loses communications with the dbServer; it must resynchronize with the master. In severe cases, operator intervention may be required to reload an entire database or selected 3o subsets.
The dbClient 2234 offers the following interface operations:
~ Attach by an authorized application to a specified set of data;
~ Policy preferences to be set by an authorized application;
~9 ~ Select a specified view of the local copy of data;
~ Insert, Update, or Delete of the local copy of data;
~ Synchronize subscripted data with the dbServer; and ~ Expiration notifications from dbServer for cached data.
Additionally, the dbClients submit Logs or Reports and signal problems to the monitor (dbMon) 2240.
(2) Data Masters ( dbServer) 2236 The dbServers 2236 play a central role in the protection of data. This is Io where data is 'owned' and master copies maintained. At least two copies of master data are maintained for reliability. Additional master copies may be deployed to improve data performance.
These copies are synchronized in lock-step. That is each update is required to obtain a corresponding master-lock in order to prevent update conflicts.
The strict implementation policies may vary, but in general, all master copies must preserve serial ordering of updates, and provide the same view of data and same integrity enforcement as any other master copy. The internal copies of data are transparent to the dbClients 2234.
The dbServer 2236 includes the layers of business rules which describe or enforce the relationships between data items and which constrain particular data values or formats. Every data update must pass these rules or is rejected. In this way dbServer ensures all data is managed as a single copy and all business rules are collected and applied uniformly.
The dbServer 2236 tracks when, and what kind of, data changes are made, and provides logs and summary statistics to the monitor (dbMon) 2240.
Additionally these changes are forwarded to any active subscriptions and 3o Cache-copies are marked out of date via expiration messages.
The dbServer also provides security checks and authorizations, and ensures that selected items are encrypted before storage.
The dbServer supports the following interface operations:
Sb ~. ...... ._ ,. , , WO 98/47298 PCTlUS98/07927 ~ View selected data from dbServer;
~ Subscribe to selected data from dbServer;
~ Copy selected data into a cache-copy at a dbClient 2234;
~ Refresh a dbClient cache with the current copy on demand;
~ New data insertion across all dbServer copies of the master;
~ Change data attributes across all dbServer copies; and ~ Cancel previous subscriptions and drop cache-copies of data.
(3) Data Administration (dbAdmin) 2238 to Data Administration (dbAdmin) 2238 involves setting data policy, managing the logical and physical aspect of the databases, and securing and configuring the functional components of the Data Management 2138 domain. Data Management policies include security, distribution, integrity rules, performance requirements, and control of replications and partitions.
is dbAdmin 2238 includes the physical control of data resources such as establishing data locations, allocating physical storage, allocating memory, loading data stores, optimizing access paths, and fixing database problems.
dbAdmin 2238 also provides for logical control of data such as auditing, reconciling, migrating, cataloguing, and converting data.
The dbAdmin 2238 supports the following interface operations:
~ Define the characteristics of a data type;
~ Create logical containers of given dimensions;
~ Relate two or more containers through an association;
2s ~ Constrain data values or relations through conditional triggers and actions;
~ Place physical container for data in a given location;
~ Move physical containers for data to new locations;
~ Remove physical containers and their data;
~ Load data from one container to another;
~ Clear the data contents of a container; and ~ Verify or reconcile the data contents of a container.
(4) Data Monitoring (dbMon) 2240 The dbMon 2240 represents a monitoring function which captures all data-related events and statistical measurements from the ISP boundary gateways, dbClients 2234 and dbServers 2236. The dbMon 2240 mechanisms are used to create audit trails and logs.
The dbMon typically presents a passive interface; data is fed to it. However monitoring is a hierarchical activity and further analysis and roll-up (compilation of data collected at intervals, such as every minute, into longer time segments, such as hours or days) occurs within dbMon. Additionally 1o dbMon will send alerts when certain thresholds or conditions are met.
The rate and count of various metrics are used for evaluating quality of Service (QOS) , data performance, and other service level agreements. All exceptions and date errors are logged and flow to the dbMon for inspection, storage, and roll-up.
dbMon 2240 supports the following interface operations:
~ Setting monitor controls, filters, and thresholds;
~ Logging of data related activity;
~ Reports of status, metrics, or audit results; and ~ Signaling alarms, or alerts.
(5) Data Management operations (Ops) 2244 The Operations consoles (Ops) 2244 provide the workstation-interface for the personnel monitoring, administering, and otherwise managing the system. The Ops consoles provide access to the operations interfaces for dbMon 2240, dbAdmin 2238, and dbServer 2236 described above. The Ops consoles 2244 also support the display of dynamic status through icon based maps of the various systems, interfaces, and applications within the 3o Data management domain 2138.
5. Physical Description.
This section describes the Data Management 2138 physical architecture. It describes how a set of components could be deployed. A generalized deployment view is shown in Figure 29. In Figure 29:
~ circles are used to represent physical sites, ~ boxes or combined boxes are computer nodes, and ~ functional roles are indicated by abbreviations.
s The abbreviations used in Figure 29 are:
~ OE - order entry systems 2250;
~ GW - ISP gateway 2230;
~ APP - application (dbAppl) 2232;
to ~ CL- a dbClient 2234;
~ SVR- a dbServer 2236;
~ ADM- a dbAdmin component 2238;
~ MON- a dbMon component 2240; and ~ Ops - operations console.
1s The functional roles of these elements were described above (see Logical Description of the Target Architecture) in connection with Figure 28.
Each of the sites shown in Figure 29 is typically linked with one or more of the other sites by wide area network (WAN) links. The exact network 2o configuration and sizing is left to a detailed engineering design task. It is not common for a database copy to be distributed to the Order Entry (OE) sites 2251, however in this architecture, entry sites are considered equivalent to satellite sites and will contain the dbClient functionality.
25 On the network-side of the ISP 2100, Satellite sites 2252 each contain the dbClient 2234 too. These sites typically operate local area networks (LANs). The dbClients act as local repositories for network or system applications such as the ISN operator consoles, ARUs, or NCS switch requested translations.
The Central sites 2254 provide redundant data storage and data access paths to the dbClients 2234. Central sites 2254 also provide roll-up monitoring (dbMon) functions although dbMon components 2240 could be deployed at satellite sites 2252 for increased performance.
~' ~3 The administrative functions are located at any desired operations or administration site 2254 but not necessarily in the same location as the dbMon. Administrative functions require the dbAdmin 2238, plus an s operations console 2244 for command and control. Remote operations sites are able to access the dbAdmin nodes 2238 from wide-area or local-area connections. Each of the sites is backed-up by duplicate functional components at other sites and are connected by diverse, redundant links.
to 6. Technology Selection.
The following section describes the various technology options which should be considered. The Data Management 2138 architecture does not require any particular technology to operate; however different technology choices will impact the resulting performance of the system.
is Figure 30 depicts a set of technologies which are able to provide a very-high performance environment. Specific application requirements will determine the minimum level of acceptable performance. Three general environments are shown.
~ In the upper part, a mufti-protocol routed network 2260 connects external and remote elements with the central data sites. Administrative terminals, and smaller mid-range computers are shown, plus a high-availability application platform such as Order Entry.
~ In the center are large-scale high-performance machines 2262 with large data-storage devices; these would be typical of master databases and data processing, and data capture/tracking functions such as dbServer 2236 and dbMon 2240.
~ In the lower part of the diagram are local area processing and network interfaces 2264, such as the ISN operator centers or DAP sites.
7. Implementations.
.~ . , , While much is known of the current ISP data systems, additional detailed requirements are necessary before any final implementations are decided.
These requirements must encompass existing ISN, NCS, EVS, NIA, and TMN system needs, plus all of the new products envisioned for Broadband, s Internet, and Switchless applications.
8. Security.
ISP data is a protected corporate resource. Data access is restricted and authenticated. Data related activity is tracked and audited. Data to encryption is required for all stored passwords, PINS {personal identification numbers), private personnel records, and selected financial, business, and customer information. Secured data must not be transmitted in clear-text forms.
~s 9. Meta-Data .
Meta-data is a form of data which comprises the rules for data driven logic.
Meta-data is used to describe and manage (i.e. manipulate] operational forms of data. Under this architecture, as much control as possible is intended to be driven by meta-data. Meta-data (or data-driven logic) zo generally provides the most flexible run-time options. Meta-data is typically under the control of the system administrators.
10. Standard Database Technologies.
Implementation of the proposed Data Management Architecture should 25 take advantage of commercially available products whenever possible.
Vendors offer database technology, replication services, Rules systems, Monitoring facilities, Console environments, and many other attractive offerings.
3o J. ISP Resource Management Modet This section describes the Resource Management 2150 Model as it relates to the ISP 2100 Architecture.
a) Scope f~.S--The Resource Management Model covers the cycle of resource allocation and de-allocation in terms of the relationships between a process that needs a resource, and the resource itself. This cycle starts with Resource Registration and De-registration and continues to Resource Requisition, s Resource Acquisition, Resource Interaction and Resource Release.
b) Purpose The Resource Management 2150 Model is meant to define common architectural guidelines for the ISP development community in general, and to for the ISP Architecture in particular.
c) Objectives In the existing traditional ISP architecture, services control and manage their own physical and logical resources. Migration to an architecture that 15 abstracts resources from services requires defining a management functionality that governs the relationships and interactions between resources and services. This functionality is represented by the Resource Management 2150 Model.
The objectives of the Resource Management Model are designed to allow for zo network-wide resource management and to optimize resource utilization, to enable resource sharing across the network:
~ Abstract resources from services;
~ Provide real-time access to resource status;
~ Simplify the process of adding and removing resources;
25 ~ Provide secure and simple resource access; and ~ Provide fair resource acquisition, so that no one user of resources may monopolize the use of resources.
d) Background Concepts 3o Generally, the Resource Management 2150 Model governs the relationships and interactions between the resources and the processes that utilize them.
Before the model is presented, a solid understanding of the basic terminology and concepts used to explain the model should be established.
The following list presents these terms and concepts:
~6 ( 1 ) Definitions ~ Resource: A basic unit of work that provides a specific and well-defined capability when invoked by an external process. Resources s can be classified as logical, like a service engine and a speech recognition algorithm, or physical, like CPU, Memory and Switch ports. A resource may be Shared like an ATM link bandwidth or Disk space, or Dedicated like a VRU or a Switch port.
~ Resource Paol: A set of registered resource members that share to common capabilities.
~ Service: A logical description of all activities and the interaction flow between the user of the network resources and the resources themselves.
~ Policy: A set of rules that governs the actions taken on resource ~5 allocation and de-allocation, resource pool size thresholds and resource utilization thresholds.
"' (2) Concepts ~ The Resource Management Model is a mechanism which governs 2o and allows a set of functions to request, acquire and release resources to/from a resource pool through well-defined procedures and policies. The resource allocation and de-allocation process involves three phases:
~ Resource Requisition is the phase in which a process requests a 2s resource from the Resource Manager 2150.
~ Resource Acquisition: If the requested resource is available and the requesting process has the privilege to request it, the Resource Manager 2150 will grant the resource and the process can utilize it.
3o Otherwise, the process has the choice to either abandon the resource allocation process and may try again later, or it may request that the Resource Manager 2150 grant it the resource whenever it becomes available or within a specified period.
cg ~ Resource Release: The allocated resource should be put back into the resource pool once the process no longer needs it. Based on the resource type, the process either releases the resource and the resource informs the Resource Manager of its new status, or the s process itself informs the Resource Manager that the resource is available. In either case, the Resource Manager will restore the resource to the resource pool.
The Resource Management Model allows for the creation of resource pools to and the specification of the policies governing them. The Resource Management Model allows resources to register and de-register as legitimate members of resource pools.
Resource Management Model policies enforce load balancing, failover and least cost algorithms and prevent services from monopolizing resources.
The Resource Management Model tracks resource utilization and automatically takes corrective action when resource pools are not sufficient to meet demand. Any service should be able to access and utilize any available resource across the network as long as it has the privilege to do zo so.
The Resource Management Model adopted the OSI Object Oriented approach for modeling resources. Under this model, each resource is represented by a Managed Object (MO). Each MO is defined in terms of the 2s following aspects:
~ Attributes: The attributes of a MO represent its properties and are used to describe its characteristics and current states. Each attribute is a associated with a value, for example the value CURRENT_STATE attribute of a MO could be IDLE.
30 ~ Operations: Each MO has a set of operations that are allowed to be performed on it. These operations are:
~ Create: to create a new MO
~ Delete: to delete an existing MO
~ Action: to perform a specific operation such as SHUTDOWN.

~ Get Value: to obtain a specific MO attribute value ~ Add Value: to add specific MO attribute value ~ Remove Value: to delete a specific MO attribute value from a set of values.
~ Replace Value: to replace an existing MO attribute values) with a new one.
~ Set Value: to set a specific MO attribute to its default value.
~ Notification: Each MO can report or notify its status to the management entity. This could be viewed as triggers or traps.
~ Behavior: The behavior of an MO is represented by how it reacts to a specific operation and the constraints imposed on this reaction.
The MO may react to either external stimuli or internal stimuli. An external stimuli is represented by a message that carries an operation. The internal stimuli, however, is an internal event that i5 occurred to the MO like the expiration of a timer. A constraint on how the MO should react to the expired timer may be imposed by specifying how many times the timers has to expire before the MO
can report it.
2o All elements that need to utilize, manipulate or monitor a resource need to treat it as a MO and need to access it through the operations defined above.
Concerned elements that need to know the status of a resource need to know how to receive and react to events generated by that resource.
25 Global and Local Resource Management:
The Resource Management Model is hierarchical with at least two levels of management: Local Resource Manager (LRM) 2190 and Global Resource Manager (GRM) 2188. Each RM, Local and Global, has its own domain and 3o functionality.
2. The Local Resource Manager (LRM):.
~ Domain: The domain of the LRM is restricted to a specific resource pool (RP) that belongs to a specific locale of the network. Multiple 8 r,~

LRMs could exist in a single locale, each LRM may be responsible for managing a specific resource pool.
~ Function: The main functionality of the LRM is to facilitate the resource allocation and de-allocation process between a process and a resource according the Resource Management Model guidelines.
3. The Global Resource Manager (GRM) 2188:.
~ Domain: The domain of the GRM 2188 covers all registered to resources in all resource pools across the network.
~ Function: The main function of the GRM is to help the LRM 2190 locate a resource that is not available in the LRM domain.
Figure 31 illustrates the domains of the GRM 2188 and LRM 2190 within ~ s network 2270.
4. The Resource Management Model (RMM).
The Resource Management Model is based on the concept of Dynamic 2o Resource Allocation as opposed to Static Configuration. The Dynamic Resource Allocation concept implies that there is no pre-defined static relationship between resources and the processes utilizing them. The allocation and de-allocation process is based on supply and demand. The Resource Managers 2150 will be aware of the existence of the resources 25 and the processes needing resources can acquire them through the Resource Managers 2150. On the other hand, Static Configuration implies a pre-defined relationship between each resource and the process that needs it. In such a case, there is no need for a management entity to manage these resources. The process dealing with the resources can 3o achieve that directly. Dynamic Resource Allocation and Static Configuration represent the two extremes of the resource management paradigms.
Paradigms that fall between these extremes may exist.
The Resource Management Model describes the behavior of the LRM 2190 q0 . ._.. . , , , and GRM 2188 and the logical relationships and interactions between them. It also describes the rules and policies that govern the resource allocation and de-allocation process between the LRM/GRM and the processes needing the resources.
s a) Simple Resource Management Model Realizing that resource allocation and de-allocation could involve a complex process, a simple form of this process is presented here as an introduction to the actual model. Simple resource allocation and de-allocation is achieved through six steps. Figure 32 depicts these steps.
1. A process 22?1 requests the resource 21?3 from the resource manager 2150.
2. The resource manager 2150 allocates the resource 21?3.
3. The resource manager 2150 grants the allocated resource 2173 to 1s the requesting process 2271.
4. The process 2271 interacts with the resource 2273.
5. When the process 22?1 is finished with the resource 2273, it informs the resource.
6. The resource 2273 releases itself back to the resource manager 20 2150.
b) The Resource Management Model Logical Ele me nts:
The Resource Management Model is represented by a set of logical elements 2s that interact and co-operate with each other in order to achieve the objectives mentioned earlier. These elements are shown in Figure 33 and include: Resource Pool (RP) 2272, LRM 2190, GRM 2188 and Resource Management Information Base (RMIB) 2274.
30 {1) Resource Pool {RP) 2272 All resources that are of the same type, share common attributes or provide the same capabilities, and are located in the same network locale may be logically grouped together to form a Resource Pool (RP) 2272. Each RP will have its own LRM 2190.

(2) The Local Resource Manager (LRM) 2190 The LRM 2190 is the element that is responsible for the management of a specific RP 2272. All processes that need to utilize a resource from a RP
s that is managed by a LRM should gain access to the resource through that LRM and by using the simple Resource Management Model described above.
(3) The Global Resource Manager (GRM) 2188 The GRM 2188 is the entity that has a global view of the resource pools across the network. The GRM gains this global view through the LRMs 2190. All LRMs update the GRM with RP 2272 status and statistics.
There are cases where a certain LRM can not allocate a resource because all local resources are busy or because the requested resource belongs to 1s another locale. In such cases, the LRM can consult with the GRM to locate the requested resource across the network.
(4) The Resource Management Information Base (RMIB) 2274 2o As mentioned above, all resources will be treated as managed objects (MO).
The RMIB 2274 is the database that contains all the information about all MOs across the network. MO information includes object definition, status, operation, etc. The RMIB is part of the iSP Data Management Model. All LRMs and the GRM can access the RMIB and can have their own view and 2s access privileges of the MO's information through the ISP Data Management Model.
5. Component Interactions.
To perform their tasks, the Resource Management Model elements must 3o interact and co-operate within the rules, policies and guidelines of the Resource Management Model. The following sections explain how these entities interact with each other.
a) Entity Relationship (ER) Diagram (Figure 33):
~z In Figure 33, each rectangle represents one entity, the verb between the "<>" implies the relationship between two entities and the square brackets "(J" imply that the direction of the relationship goes from the bracketed number to the non bracketed one. The numbers imply is the relationship is s 1-to-1, 1-to-many or many-to-many.
Figure 33 can be read as follows:
1. One LRM 2190 manages one RP 22?2.
2. Many LRMs 2190 access the RMIB 22?4.
3. Many LRMs 2190 access the GRMs 2188.
4. Many GRMs 2188 access the RMIB 2274.
b) Registration and De-registration Resource registration and de-registration applies only on the set of resources that have to be dynamically managed. There are some cases ~s where resources are statically assigned.
LRMs 2190 operate on resource pools 2272 where each resource pool contains a set of resource members. In order for the LRM to manage a certain resource, the resource has to inform the LRM of its existence and 2o status. Also, the GRM 2188 needs to be aware of the availability of the resources across the network in order to be able to locate a certain resource. The following registration and de-registration guidelines should be applied on all resources that are to be dynamically managed:
~ All resources must register to their LRM 2190 as members of a 25 specific resource pool 22T2.
~ All resources must de-register from their LRM 2190 if, for any reason, they need to shutdown or be taken out of service.
~ All resources must report their availability status to their LRM
2190.
30 ~ All LRMs must update the GRM 2188 with the latest resource availability based on the registered and de-registered resources.
c) GRM, LRM and RP Interactions Every RP 22?2 will be managed by an LRM 2190. Each process that needs y3 a specific resource type will be assigned an LRM that will facilitate the resource access. When the process needs a resource it must request it through its assigned LRM. When the LRM receives a request for a resource, two cases may occur:
1. Resource is available: In this case, the LRM allocates a resource member of the pool and passes a resource handle to the process. The process interacts with the resource until it is done with it. Based on the resource type, once the process is done with the resource, it either informs the resource that it is done with it, and the resource itself informs its LRM
to that it is available, or it releases the resource and informs the LRM that it is no longer using the resource.
2. Resource is not available: In this case, the LRM 2190 consults with the GRM 2188 for an external resource pool that contains the requested resource. If no external resource is available, the LRM informs the ~5 requesting process that no resources are available. In this case, the requesting process may:
~ give up and try again, ~ request that the LRM allocate the resource whenever it becomes available, or 20 ~ request that the LRM allocates the resource if it becomes available within a specified period of time.
If an external resource is available, the GRM 2188 passes location and access information to the LRM 2190. Then the LRM either:
2s . ~ allocates the resource on the behalf of the requesting process and passes a resource handle to it (In this case the resource allocation through the GRM is transparent to the process), or ~ advises the requesting process to contact the LRM that manages the located resource.
d) GRM, LRM and RMI8 Interactions The RMIB 2274 contains alI information and status of all managed resources across the network. Each LRM 2190 will have a view of the RMIB 274 that maps to the RP 22?2 it manages. The GRM 2188, on the 9~

other hand, has a total view of all resources across the network. This view consists of all LRMs views. The GRM's total view enables it to locate resources across the network.
In order for the RMIB 22?4 to keep accurate resource information, each LRM 2190 must update the RMIB with the latest resource status. This includes adding resources, removing resources and updating resource states.
Both the LRM 2190 and GRM 2188 can gain their access and view of the RMIB 22?4 through the ISP Data Management entity. The actual management of the RMIB data belongs to the ISP Data Management entity.
The LRM and GRM are only responsible for updating the RMIB.
is K. Operational Support Model 1. Introduction.
Most of the existing ISP service platforms were developed independently, each with it's own set of Operational Support features. The amount of time required to learn how to operate a given set of platforms increases with the 2o number of platforms. The ISP service platforms need to migrate to an architecture with a common model for ail of its Operational Support features across all of its products. This requires defining a model that will support current needs and will withstand or bend to the changes that will occur in the future. The Operational Support Model (OSM) defines a 25 framework for implementation of management support for the ISP 2100.
a) Purpose The purpose of the Operational Support Model is to:
~ achieve operational simplicity by integrating the management platform for 3o ISP resources;
~ reduce the learning curve for operational personnel by providing a common management infrastructure;
~ reduce the cost of management systems by reducing overlapping management system development;
9s-~ improve time to market for ISP services by providing a common management infrastructure for all of the ISP services and network elements; and ~ provide a framework for managing ISP physical resources (hardware) and s logical resources (software).
b) Scope The OSM described here provides for the distributed management of ISP
physical network elements and the services that run on them. The management framework described herein could also be extended to the management of logical (software) resources. However, the architecture presented here will help map utilization and faults on physical resources to their resulting impact on services.
The management services occur within four layers 15 ~ Planning, ~ Service Management, ~ Network Layers, and ~ Network Elements.
Information within the layers falls into four functional areas:
20 ~ Configuration Management, ~ Fault Management, ~ Resource Measurement, and ~ Accounting.
The use of a common Operational Support Model for all of the ISP will 25 enhance the operation of the ISP, and simplify the designs of future products and services within the ISP. This operational support architecture is consistent with the ITU Telecommunications Management Network (TMN) standards.
3o c) Definitions Managed Object: A resource that is monitored, and controlled by one or more management systems Managed objects are located within managed systems and may be embedded in other managed objects. A managed object may be a logical or physical resource, and a resource may be ~6 represented by more than one managed object (more than one view of the object).
Managed System: One or more managed objects.
Management Sub-Domain: A Management domain that is wholly located s within a parent management domain.
Management System: An application process within a managed domain which effects monitoring and control functions on managed objects and/or management sub-domains.
Management Information Base : A MIB contains information about io managed objects.
Management Domain: A collection of one or more management systems, and zero or more managed systems and management sub-domains.
Network Element: The Telecommunications network consist of many types of analog and digital telecommunications equipment and associated ~5 support equipment, such as transmission systems, switching systems, multiplexes, signaling terminals, front-end processors, mainframes, cluster controllers, file servers, LANs, WANs, Routers, Bridges, Gateways, Ethernet Switches, Hubs, X.25 links, SS7 links, etc. When managed, such equipment is generally referred to as a network element (NE).
2o Domain: The management environment may be partition in a number a ways such as functionally (fault, service....), geographical, organizational structure, etc.
Operations Systems: The management functions are resident in the Operations System.
2. The Operational Support Model.
Figure 34 shows the four management layers 2300, 2302, 2304 and 2306 of the Operational Support Model 2308 over the network elements 2310.
The Operational Support Model 2308 supports the day to day management of the ISP 2100. The model is organized along three dimensions. Those dimensions are the layers 2300-2306, the functional area within those layers, and the activities that provide the management services. Managed objects (a resource) are monitored, controlled, and altered by the management system.
9~

a) The Functional Model The following sections describe the functional areas as they occur within the management layers 2300-2306.
s { 1 ) Planning The ISP Planning Layer 2300 is the repository for data collected about the ISP 2100, and the place where that data is to provide additional value.
~ Configuration Management 2312: Setting of policy, and goals.
~ Fault Management 2314: Predicting of mean time to failure.
~ Resource Measurement 2316: Predicting future resource needs (trending, capacity, service agreement compliance, maintenance agreement, work force).
~ Accounting: Determine cost of providing services in order to support i s service pricing decisions.
(2) Service Management The Service Ordering, Deployment, Provisioning, Quality of Service agreements, and Quality of service monitoring are in the ISP Service 2o Management layer 2302. Customers will have a restricted view of the SM
layer 2302 to monitor and control their services. The SM layer provides a managers) that interacts with the agents in the NLMs. The SM layer also provides an agents) that interacts with the managers) in the Planning layer 2300. Managers within the SM layer may also interact with other 2s managers in the SM layer. In that case there are manager-agent relationships at the peer level.
~ Configuration Management 2320: Service Definition, Service Activation, Customer Definition, Customer Activation, Service Characteristics, Customer Characteristics, hardware provisioning, software provisioning, 3o provisioning of other data or other resources.
~ Fault Management 2322: Monitor and report violations of service agreement. Testing.
~ Resource Measurement 2324: Predict the violation of a service agreement and flag potential resource shortages. Predict the needs of current and future (trending) services.
~ Accounting 2326: Process and forward Accounting information.
Network Layer Management:
The ISP Network Layer Management (NLM) Layer 2304 has the responsibility for the management of all the network elements, as presented by the Element Management, both individually and as a set. It is not concerned with how a particular element provides services internally. The NLM layer 2304 provides a managers) that interacts with the agents in the EMs 2306. The NLM layer also provides an agents) that interacts with the managers) in the SM layer 2302. Managers within the NLM layer 2304 may also interact other managers in the NLM layer. In that case there are manager agent relationships at the peer level.
~ Configuration Management 2328 provides functions to define the characteristics of the local and remote resources and services from a network wide perspective.
~ Fault Management 2330 provides functions to detect, report, isolate, and correct faults that occur across multiple NEs.
~ Resource Measurement 2332 provides for the network wide measurement, 2o analysis, and reporting of resource utilization from a capacity perspective.
~ Accounting 2334 consolidates Accounting information from multiple sources.
(3) Element Management 2s The Element Management Layer 2306 is responsible for the NEs 2310 on an individual basis and supports an abstraction of the functions provided by the NEs The EM Layer 2306 provides a managers) that interact with the agents in the NEs. The EM layer also provides an agents) that interact with the managers) in the NLM Layer 2304. Managers within the EM layer 30 2306 may also interact other managers in the EM layer. In that case there are manager agent relationships at the peer level.
~ Configuration Management 2336 provides functions to define the characteristics of the local and remote resources and services.

~ Fault Management 2338 provides functions to detect, report, isolate, and correct faults.
~ Resource Measurement 2340 provides for the measurement, analysis, and reporting of resource utilization from a capacity perspective.
~ Accounting 2342 provides for the measurement and reporting of resource utilization from an accounting perspective.
b) Network Element The computers, processes, switches, VRUs, Internet gateways, and other equipment that provide the network capabilities are Network Elements 2310. NEs provide agents to perform operations on the behalf of the Element Management Layer 2306.
c) Information Model Figure 35 shows manager agent interaction. Telecommunications network management is a distributed information application process. It involves the interchange of management information between a distributed set of management application processes for the purpose of monitoring and controlling the network resources (NE) 2310. For the purpose of this 2o exchange of information the management processes take on the role of either manager 2350 or agent 2352. The manager 2350 role is to direct management operation requests to the agent 2352, receive the results of an operation, receive event notification, and process the received information. The role of the agent 2352 is to respond to the manager's request by performing the appropriate operation on the managed objects 2354, and directing any responses or notifications to the manager. One manager 2350 may interact with many agents 2352, and the agent may interact with more than one manager. Managers may be cascaded in that a higher level manager acts on managed objects through a lower level 3o manager. In that case the lower level manager acts in both manager and agent roles.
3. The Protocol Model.
a) Protocols Iob The exchange of information between manager and agent relies on a set of communications protocols. TMN, which offers a goad model, uses the Common Management Information Services (CMIS) and Common Management Information Protocol (CMIP) as defined in Recommendations X.710, and X.? 11. This provides a peer-to-peer communications protocol based on ITU's Application Common Service Element (X.217 service description 8~ X.227 protocol description) and Remote Operation Service Element (X.219 service description & X.229 protocol description) . FTAM is also supported as an upper layer protocol for file transfers. The use of these upper layer protocols is described in Recommendation X.812. The transport protocols are described in Recommendation X.811.
Recommendation X.811 also describes the interworking between different lower layer protocols. This set of protocols is referred to as Q3.
is h) Common context In order to share information between processes there needs to be a common understanding of the interpretation of the information exchanged.
ASN.1 (X.209) with BER could be used to develop this common understanding for all PDU exchanged between the management processes 20 (manager/agent).
c) Services of the upper layer The following identifies the minimum services required of the service layer and is modeled after the TMN CMIS services.
2s SET: To add, remove, or replace the value of an attribute .
GET: To read the value of an attribute.
CANCEL-GET: To cancel a previously issued GET.
ACTION: To request an object to perform a certain action.
3o CREATE: To create an object.
DELETE: To remove an object.
EVENT-REPORT: Allows the network resource to announce an event.
4. The Physical Model.
fof Figure 36 shows the ISP 2100 physical model.
5. Interface Points .
Mediation Device 2360 provides conversion from one information model to s the ISP information model. Gateways 2362 are used to connect to management systems outside of the ISP. These gateways will provide the necessary functions for operation with both ISP compliant systems, and non-compliant systems. The gateways may contain mediation devices 2360. Figure 36 identifies nine interface points. The protocols associated with those interface points are:
1. There are two upper layer protocols. The protocol for communications with the workstation and the ISP upper layer for all other operational support communications. The lower layer is TCP/IP over Ethernet.
~s 2. The upper layer is the protocol for communications with workstation 2364, and the lower layer is TCP/IP over Ethernet.
3,4. The upper layer is the ISP upper layer, and the lower layer is TCP/IP
20 over Ethernet.
5. The proprietary protocols are the of legacy systems that are not compatible with the supported interfaces. Equipment that provides a Simple Network Management Protocol (SNMP) interface will be supported 25 with Mediation Devices.
6,7,8,9. Gateways by their nature will support ISP compliant and non-compliant interfaces. Gateways to enterprise internal systems could include such as the Order Entry system, or an enterprise wide TMN system.
The ISP Realization of the Operational Support Model Figure 37 shows operational support realization.
~ O:L

6. General.
The Operational Support Model provides a conceptual framework for building the Operational Support System. Figure 37 represents an ISP
realization of this conceptual model. In this implementation of that model all the ISP Network Elements would be represented to the Operational Support System by a Management Information Base (MIB} 2370 and the agent process that acts upon the objects in the MIB.
Field support personnel have two levels from which the ISP 2100 will be to managed.
1. For trouble-shooting, the Network Layers Manager 23?2 gives field support a picture of the ISP as a whole. The process of detecting, isolating, and correcting problems begins from there. From that layer, problems could be isolated to a single Network Element. Individual Network t5 Elements are accessible from the Network Element Managers 2374 and would allow a more detailed level of monitoring, control, configuration, and testing. The centralized view of the ISP is missing from today's ISP, but many recognize its importance.
2o For configuration the Network Layers Manager 2370 provides an ISP-wide view, and interacts with the Network Element Managers 2374 to configure Network Elements in a consistent manner. This will help insure that the ISP configuration is consistent across all platforms. The ability to change a piece of information in one place and have it automatically distributed 25 ISP- wide is a powerful tool that has not been possible with the current ISP
management framework.
Once a service definition has been created from the Service Creation Environment 23?6, the Service Manager 2378 is used to place it in the ISP
3o network, and provision the network for the new service. Customers for a service are provisioned through the Service Manager 2378. As a part of provisioning customers the Service Manager predicts resource utilization, and determines if new resources need to be added to handle the customer's use of a service. It uses the current utilization statistics as a basis for that I ~,3 determination. Once a customer is activated, the Service Manager monitors the customer's usage of the service to determine if the quality of service agreement is being met. As customer utilization of the services increases the Service Manager 2378 predicts the need to add resources to the ISP
s network. This Service Management, with appropriate restrictions, can be extended to customers as another service. While Service Creation is the talk of the IN world, it needs a Service Manager that is integrated with the rest of the system, and that is one of the purposes of this model.
1o Finally, for planning personnel (non-field support), the Planning Manager 2380 analyzes the ISP-wide resource utilization to determine future needs, and to allocate cost to different services to determine the cost of a service as the basis for future service pricing.
L. Physical Network Model 1. Introduction.
This section describes the Physical Network aspects of the Intelligent Services Platform (ISP) 2100 Architecture.
2o a) Purpose The Physical Network Model covers the:
~ Logical Architecture Mapping;
~ Information Flows; and ~ Platform Deployment in the production environment of the architecture.
bj Scope This model defines the terminology associated with the physical network, describes the interactions between various domains and provides examples of realizations of the architecture.
3o c) Objectives The objectives of this model are to:
~ Create a model for identifying various network platforms;
~ Classify Information Flow;
~ Provide standard nomenclature;
( b 1-~

~ Provide rules for systems deployment; and ~ Guide future technology selections.
2. Information Flow.
- s One of the key aspects of the intelligent network (IN} is the Information Flow across various platforms installed in the network. By identifying types of information and classifying them, the network serves the needs of IN.
Customers interact with IN in a series of call flows. Calls may be audio-to centric (as in the conventional ISP products), multimedia-based (as in internetMCI user using the web browser), video-based (as in video-on-demand) or a combination of contents.
Information can be classified as follows:
15 ~ Content;
~ Signaling; or ~ Data.
Normally, a customer interacting with the intelligent network will require all three types of information flows.
a) Content Content flows contain the primary information being transported.
Examples of this are analog voice, packet switched data, streamed video and leased line traffic. This is customer's property that IN must deliver with minimum loss, minimum latency and optimal cost. The IN elements are standardized such that the transport fabric supports more connectivity suites, in order to allow content to flow in the same channels with flow of other information.
3o b~ Signaling Signaling flows contain control information used by network elements.
ISUP RLT/IMT, TCP/IP domain name lookups and ISDN Q.931 are all instances of this. The IN requires, uses and generates this information.
Signaling information coordinates the various network platforms and allows coy WO 98!47298 PCT/US98/07927 intelligent call flow across the network. In fact, in a SCE-based IN, service deployment will also require signaling information flowing across the fabric.
c) Data Data flows contain information produced by a call flow, including crucial billing data records often produced by the fabric and certain network platforms.
3. Terminology.
to Network: A set of interconnected network elements capable of transporting content, signaling and/or data. MCI's IXC switch fabric, the ISP extended WAN, and the Internet backbone are classic examples of networks. Current installations tend to carry different contents on different networks, each of which is specialized for specific content transmission. Both technology and ~5 customer requirements (for on-demand high bandwidth) will require carriers to use more unifed networks for the majority of the traffic. This will require the fabric to allow for different content characteristics and protocols along the same channels. Another aspect of this will be more uniform content-independent signaling.
2o Site: A set of physical entities collocated in a geographically local area.
In the current ISP architecture, instances of sites are Operator Center, ISNAP
Site (which also has ARU's) and an EVS site. By the very definition, the NT
and DSC switches are NOT part of the site. They are instead part of the Transport Network (see below). In the architecture, a group of 25 (geographically collocated) Service Engines (SE), Special Resources (SR), Data Servers (DS) along with Network Interfaces and Links form a site.
Network Element: A physical entity connecting to the Transport Networks through Network Interfaces. Examples of this are ACP, EVS SIP, MTOC, Videoconference Reservation Server, DAP Transaction Server, and NAS. In 3o the next few years, elements such as web servers, voice authentication servers, video streamers and network call record stores will join the present family of network elements.
Network Interface: Equipment enabling connectivity of Network Elements to the Transport Networks. DS1 CSU/DSU, lOBaseT Ethernet interface 1~6 card and ACD ports are network interfaces. With the architecture of the preferred embodiment, network interfaces will provide a well-understood uniform set of API's for communication.
Link: Connection between 2 or more Network Interfaces which are at s different sites. A link may be a segment of OC-12 SONET Fiber or 100mbps dual ring FDDI section. In the coming years, IN must handle network links such as ISO Ethernet WAN hub links and gigabit rate OC-48's.
Connection: an attachment of two or more Network Interfaces which are at to the same site.
Figure 38 shows a representation of a physical network 2400 schematic.
Networks 2401 contain network elements 2402 at sites 2404 are interconnected through network interfaces 2406 and one or more gateways i s 2408.
4. Entity Relationships.
Entity relationships as shown in Figure 39 have been arrived at as part of the physical network modeling rules. Some of these rules allow for 2o generalities that future demands and some will constrain definitions to avoid conflicts.
1. A Network 2401 spans one or more sites 2404, and contains one or more network elements 2402.
2. A Site 2404 contains one or more network elements 2402.
zs 3. A Network Element 2402 is located in only one Site 2404.
4. A Link 2420 connects two or more Sites 2404.
5. A Connection 2422 connects two or more Network Elements.
6. A Network Element 2402 contains one or more Network Interfaces 2406.
The preferred embodiment integrates product and service offerings for MCI's business customers. The initial embodiment focuses on a limited product set. Requirements for an interface have been identified to capitalize on the integration of these services. The interface provides user-f o~

manageability of features, distribution list capabilities, and a centralized message database.
VIII. INTELLIGENT NETWORK
s All of the platform's support services have been consolidated onto one platform. The consolidation of platforms enables shared feature/functionality of services to create a common look and feel of features.
i o A. Network Management The architecture is designed such that it can be remotely monitored by an MCI operations support group. This remote monitoring capability provides MCI the ability to:
15 ~ Identify degraded or broken connectivity between:
-platforms, servers or nodes that must pass information (i.e., objects) to the "universal inbox", -platforms, servers or nodes responsible for retrieving messages and delivering messages, 20 -the "universal inbox" and the PC Client messaging interface, -the "universal inbox" and the Message Center interface, -platforms, servers or nodes that must pass profile information to Profile, and -platforms, servers or nodes that must pass profile information to the ARU;
2s ~ Identify degraded application processes and isolate the process that is degraded;
~ Identify hardware failure; and ~ Generate alarms that can be detected and received by an internal MCI monitoring group for all application process, hardware or 3o interface failures.
In addition, remote access to system architecture components is provided to the remote monitoring and support group such that they can perform remote diagnostics to isolate the cause of the problem.

B. Customer Service Customer Service teams support all services. Customer support is provided to customers in a seamless manner and encompasses the complete product - 5 life cycle including:
~ Alpha tests;
~ Beta tests;
~ Commercial release; and ~ Identification of enhancements to address customer feedback or to additional customer support requirements Comprehensive and coordinated support procedures ensure complete customer support from inception to termination. Customer service is provided from the time the Account Team submits the order until the customer cancels the account. Comprehensive and coordinated customer t5 support entails the following:
~ A one-stop, direct access, customer service group to support ARU or VRU problems, WWW Browser problems or PC Client problems.
~ A staff that is well trained on diagnosing problems associated with access (ARU, WWW Browser or PC Client), the user interface (ARU, WWW
2o Browser or PC Client), the application ( Message Center or Profile Management) or the back-end system interfaces (universal inbox, directlineMCI voicemail/faxmail platform, Fax Broadcast System, SkyTel Paging server, order entry systems, billing systems, etc.) ~ A staff that has on-line access to databases with information about 25 ARU or VRU capabilities, WWW Browser capabilities, identified hardware issues and identified application issues ~ 7 x 24 customer support ~ a single toll free number (800 or 888) with direct access to the customer service group 30 ~ seamless first, second and third level support for most troubles where:
- Level 1 support is the first support representative answering the telephone. They are expected to be able to resolve the most commonly asked questions or problems reported by customers. These questions or I o'~

problems typically deal with access type (ARU, WWW Browser, PC Client), dial-up communication for the WWW Browser or PC Client, installation or basic computer (PC, workstation, terminal) hardware questions.
Additionally they are able to open and update trouble tickets, and s reactivate customers' passwords.
- Level 2 support is provided within the customer support group when referrals to more experienced technical experts is necessary.
- Level 3 support may involve an outside vendor for on-site hardware support for the customer or an internal MCI engineering or support group to depending on the nature of the problem. The customer support group will be able to track the status of the customer visit and add the identified problem to both the customer support databases.
- Level 4 support will continue to be provided by the Systems ~5 Engineering programmers.
~ Staffing levels to provide acceptable customer hold times and abandon rates.
~ A staff that has on-line access to the order entry and billing systems.
~ Automatically generate weekly reports that detail volume of calls 2o made, received, average hold-time of calls and number of trouble tickets opened/closed/escalated.
C. Accounting Accounting is supported according to current MCI procedures.
D. Commissions Commissions are supported according to current MCI procedures.
E. Reporting 3o Reporting is required for revenue tracking, internal and external customer installation / sales, usage and product/ service performance.
Weekly and monthly fulfillment reports are required from the fulfillment house(s). These fulfillment reports correlate the number of orders received and number of orders delivered. In addition, reporting identifies the f /D

number of different subscribers accessing Profile Management or the Message Center through the WWW Site.
F. Security Security is enforced in accordance with MCI's published policies and procedures for Internet security. In addition, security is designed into the WWW Browser and ARU interface options to verify and validate user access to directlineMCI profiles, Message Center, Personal Home Page calendars and Personal Home Page configurations.
G. Trouble Handling Trouble reporting of problems is documented and tracked in a single database. All troubles are supported according to the Network Services Trouble Handling System (NSTHS) guidelines. Any Service Level Agreements (SLAs) defined between MCI organizations are structured to support NSTHS.
Any troubles that require a software fix are closed in the trouble reporting database and opened as a Problem Report (PR) in the Problem Tracking System. This Problem Tracking System is used during all test 2o phases of and is accessible by all engineering and support organizations.
IX. ENHANCED PERSONAL SERVICES
Throughout this description, the following terms will be used:
Term Represents Server Both the hardware platform and a TCP service Web Server AIX 4.2 system running Netscape Commerce Server HTTP Daemon - Welcome Server 3o Application Server The Web Servers running as Welcome Servers will be running the Netscape Commerce Server HTTP Daemon in secure as well as normal mode. The Web Servers operating as various application servers will run this daemon in secure mode only. The Secure Mode uses SSLv2.

A. Web Server Architecture The Web Servers are located in a DMZ. The DMZ houses the Web Servers and associated Database Clients as required. The database clients do not hold any data, but provide an interface to the data repositories behind the s corporate firewall.
The Web space uses Round-Robin addressing for name resolution. The Domain name is registered with the administrators of mci.com domain, with a sub-netted (internally autonomous) address space allocated for gaiileo.mci.com domain.
Figure 40 shows the sequence of events leading to a successful login.
1. Welcome Server 450.
1s This Web Server runs both the secure and normal HTTP daemons. The primary function of this server is to authenticate user 452 at login time.
The authentication requires the use of Java and a switch from normal to secure mode operation. There are one or more Welcome servers 450 in the DMZ. The information provided by the Welcome server 450 is stateless. The 2o statelessness means that there is no need to synchronize multiple Welcome Servers 450.
The Welcome server's first task is to authenticate the user. This requires the use of single use TOKENS, Passcode authentication and Hostile IP
25 filtering. The first is done using a Token Server 454, while the other two will be done using direct database 456 access.
In case of failed authentication, the user 452 is shown a screen that mentions alI the reasons (except Hostile-IP) why the attempt may have 3o failed. This screen automatically leads the users back to the initial login screen.
Welcome server 450's last task, after a successful authentication, is to send a service selection screen to the user 452. The Service Selection IIZ

screen directs the user to an appropriate Application Server. The user selects the Application, but an HTML file in the Server Section page determines the Application Server. This allows the Welcome Servers 450 to do rudimentary load balancing.
All the Welcome Servers 450 in the DMZ are mapped to www.galileo.mci.com. The implementation of DNS also allows galileo.mci.com to map to www.galileo.mci.com.
2. Token Server 454.
This is a database client and not a Web Server. The Token servers 454 are used by Welcome Servers 450 to issue a TOKEN to login attempts. The issued TOKEN, once validated, is used to track the state information for a connection by the Application Servers. The TOKEN information is be is maintained in a database on a database server 456 (repository) behind the corporate firewall.
The Token Servers 454 do the following tasks:
1. Issue single use TOKEN during authentication phase.
2. Validate single use TOKEN (mark it for multi use).
20 3. Validate mufti-use TOKEN.
Re-validate mufti-use TOKEN.
The Token Servers 454 are required to issue a unique TOKEN on every new request. This mandates a communication link between multiple Token 25 Servers in order to avoid conflict of TOKEN values issued. This conflict is eliminated by assigning ranges to each Token Server 454.
The TOKEN is a sixteen character quantity made up of 62 possible character values in the set [0-9A-Za-z]. The characters in positions 0,1 and 30 2 for each TOKEN issued by the Token Server are fixed. These character - values are assigned to each Token Server at configuration time. The character at position 0 is used as physical location identifier. The character at position 1 identifies the server at the location while the character at position 2 remains fixed at '0'. This character could be used to identify the I!3 version number for the Token Server.
The remaining 13 characters of the TOKEN are generated sequentially using the same 62 character set described above. At startup the TOKEN
s servers assign the current system time to the character positions 15-10, and set positions 9-3 to '0'. The TOKEN values are then incremented sequentially on positions 15-3 with position 3 being least significant. The character encoding assumes the following order for high to low digit values 'z'-'a' 'Z'-'A' '9'-'0'.
> >
l0 The above scheme generates unique tokens if the system time is computed in 4 byte values, which compute to 6 base-62 characters in positions 15-10. The other assumption is that the scheme does not generate more than 62~7 (35* 10~ 12) TOKENS in one second on any given Token Server in any t s embodiment.
The use of TOKEN ranges allows the use of multiple Token Servers in the Domain without any need for explicit synchronization. The method accommodates a maximum 62 sites, each having no more than 62 Token 2o Servers. An alternate embodiment would accommodate more sites.
All of the Token Servers in the DMZ are mapped to token. galileo. mci. com.
The initial embodiment contains two Token Servers 454. These Token Servers 454 are physically identical to the Welcome Servers 450, i.e., the 2s Token Service daemon will run on the same machine that also runs the I-ITTP daemon for the Welcome service. In another embodiment, the two run on different systems.
The Welcome Servers) 450 use the Token Servers) 454 to get a single use 3o TOKEN during the authentication phase of the connection. Once authenticated, the Welcome Server 450 marks the TOKEN valid and marks it for multiple use. This mufti-use TOKEN accompanies the service selection screen sent to the user by the Welcome Server.

The design of TOKEN database records is discussed in detail below.
3. Application Servers.
The Application servers are Web servers that do the business end of the s user transaction. The Welcome Server's last task, after a successful authentication, is to send a service selection screen to the user. The service selection screen contains the new multi-use TOKEN.
When the user selects a service, the selection request, with its embedded 1o TOKEN, is sent to the appropriate Application Server. The Application Server validates the TOKEN using the Token Server 454 and, if valid, serves the request. A Token Server can authenticate a TOKEN issued by any one of the Token Servers on the same physical site. This is possible because the Token Servers 454 are database clients for the data 1s maintained on a single database repository behind the corporate firewall.
An invalid TOKEN (or a missing TOKEN) always leads to the "Access Denied" page. This page is served by the Welcome Servers) 450. All denial of access attempts are logged.
The actual operation of the Application Server depends on the Application itself. The Application Servers in the DMZ are mapped to <appName><num>.galileo.mci.com. Thus, in an embodiment with multiple applications (e.g., Profile Management, Message Center, Start Card Profile, Personal Web Space etc.), the same Welcome and Token servers 450 and 454 are used and more Applications servers are added as necessary.
Another embodiment adds more servers for the same application. If the work load on an application server increases beyond its capacity, another 3o Application Server is added without any changes to existing systems. The SERVERS and TOKEN HOSTS databases (described below) are updated to add the record for the new server. The <num> part of the host name is used to distinguish the Application Servers.
I f5 There is no need to use DNS Round-robin on these names. The Welcome server 450 uses a configuration table (The SERVERS database loaded at startup) to determine the Application Server name prior to sending the service selection screen.
s B. Web Server System Environment All the Web servers run the Netscape Commerce Server HTTP daemon. The Welcome Servers 450 run the daemon in normal as well as secure mode, while the Application Servers only run the secure mode daemon.
io The Token Servers) run a TCP service that runs on a well known port for ease of connection from within the DMZ. The Token Service daemon uses tcp_wrapper to deny access to all systems other than Welcome and Application server(s). In order to speed this authentication process, the list 1s of addresses is loaded by these servers at configuration time, instead of using reverse name mapping at every request. The use of tcp_wrapper also provides the additional tools for logging Token Service activity.
The Application servers mostly work as front-ends for database services 2o behind the firewall. Their main task is to validate the access by means of the TOKEN, and then validate the database request. The database requests are to Create, Read, Update or Delete exiting records or data fields on behalf of the user. The Application Servers do the necessary validation and authority checks before serving the request.
2s 1. Welcome Servers.
The Welcome Servers serve the HTML pages described below to the user at appropriate times. The pages are generated using Perl-based Common Gateway Interface (CGI) scripts. The Scripts reside in a directory which is 3o NOT in the normal document-root directory of the HTTP daemon. The normal precautions regarding disabling directory listing and removing all backup files etc. are taken to ensure that CGI scripts are not readable to the user. Figure 41 shows the directory structure 455 on the Welcome Server 450.
~~6 Figure 41 shows that the <document root> 456 is separated from the <server_root> 458. It also shows that the <document root> directory holds only the welcome and access failure HTML pages.
The HTTP Server maps all requests to the "cgi" directory 460 based on the URL requested. The CGI scripts use the HTML templates from the "template" directory 462 to create and send the HTML output to the users on fly.
io The use of the URL to map to a CGI script out of the <document root> 456 blocks access to the <document root> directory 456 by a malicious user.
Since every access to the Welcome Server 450 maps to a CGI script in the cgi directory 460 of the Welcome Server 450, security is ensured by calling is the authentication function at start of every script.
The user Authentication libraries are developed in Perl to authenticate the user identity. NSAPI's authentication phase routines also add features for TOKEN verification and access mode detection in the servers themselves.
The Welcome Servers 450 read their operating parameters into their environment from the database 456 at startup. It is necessary to keep this information in the common database in order to maintain the same environment on multiple Welcome Servers 450.
a) Welcome Page The welcome page is sent as the default page when the Welcome Server 450 is first accessed. This is the only page that is not generated using a cgi script, and it is maintained in the <document root> directory 456. This 3o page does the following:
~ Confirms that the browser can display Frames. If the browser fails to display Frames correctly, this page will display an appropriate error message and direct the user to down load Microsoft Internet Explorer V3.0 or later.
l I ..~

~ Confirms that the browser can run Java. A failure will result in the user being directed to Microsoft Internet Explorer V3.0 or later.
~ If the browser successfully displays Frames and runs Java, then this page will automatically request the Welcome Server 450 to send a login page.
The last action by the Welcome page is done using the Java applet embedded in page. This also switches the user's browser from normal to secure mode.
to b) Login Page The Login Page is a cgi-generated page that contains an embedded single use TOKEN, a Java applet, and form fields for the user to enter a User Id and Passcode. The page may display a graphic to emphasize service.
The processing of this page is padded to introduce an artificial delay. In the initial embodiment, this padding is set to zero.
The response from this page contains the TOKEN, a scrambled TOKEN
value generated by the applet, User Id and Passcode. This information is 2o sent to the Welcome server using a POST HTTP request by the Java applet.
The POST request also contains the Applet signature.
If the login process is successful the response to this request is the Server Selection page. A failure at this stage results in an Access Failed page.
c) Server Selection Page The Server Selection Page is a cgi-generated page which contains an embedded multi-use TOKEN. This page also shows one or more graphics to indicate the types of services available to the user. Some services are not 3o accessible by our users. In other embodiments, when more than one service exists, a User Services Database keyed on the User Id is used to generate this page.
The Welcome server uses its configuration information to embed the names t l S

of appropriate Application Servers with the view to sharing the load among all available Application Servers. This load sharing is done by using the configuration data read by the Welcome Servers) during startup.
s The Welcome Server selects an Application Server based upon entries in its configuration file for each of the services. These entries list the names of Application Servers) for each application along with their probability of selection. This configuration table is loaded by the Welcome Servers at startup.
to d) Access Failed Page The Access Failed Page is a static page. That displays a message indicating that the login failed because of an error in User Id, Passcode or both. This page automatically loads the Login Page after a delay of 15 seconds.
~s e) Access Denied Page The Access Denied Page is a static page that displays a message indicating that an access failed due to authentication error. This page automatically loads the Login Page after a delay of 15 seconds. The Access Denied page is 2o called by the Application Servers when their authentication service fails to recognize a TOKEN. All loads of this page will be logged and monitored.
2. Token Servers 454.
The TOKEN service on the Web site is the only source of TOKEN generation 25 and authentication. The Tokens themselves are stored in a shared Database 456. This database can be shared among all Token servers. The Token Database is behind the firewall out of the DMZ.
The Token service provides the services over a well-known (> 1024) TCP
3o port. These services are provided only to a trusted host. The list of trusted hosts is maintained in a configuration database. This database is also maintained behind the firewall outside of the DMZ. The Token servers read their configuration database only on startup or when they receive a signal to refresh. The Token services are:
1 1 q ~ Grant a single use TOKEN for login attempt.
~ Validate a single use TOKEN.
~ Validate a TOKEN.
~ Re-Validate a TOKEN.
s TOKEN aging is implemented by a separate service to reduce the work load on the Token servers.
All access to the Token Servers) is logged and monitored. The Token Service itself is written using the tcp_wrapper code available from MCI's internal security groups.
io 3. Profile Management Application Servers.
The profile management application servers) are the only type of Application servers implemented in the first embodiment. These servers have the same directory layout as the Welcome Servers. This allows the ~s same system to be used for both services if necessary.
C. Security The data trusted by subscribers to the Web server is sensitive to them. They would like to protect it as much as possible. The subscribers 2o have access to this sensitive information via the Web server(s). This information may physically reside on one or more database servers, but as far as the subscribers are concerned it is on Servers) and it should be protected.
2s Presently only the following information needs to be protected in an embodiment:
In other embodiments, profile information for directline account additional information is protected, including Email, Voice Mail, Fax Mail, and 3o Personal Home Page information.
The protection is offered against the following type of attackers:
~ People with access to Web;
~ Other subscribers;
Izo ~ MCI personnel;
~ People with access to Subscriber's network;
~ People with access to Subscriber's system;
~ People looking over the shoulder of the Subscriber; and s ~ Other systems pretending to be Server(s).
The project implements the security by using the following schemes:
~ Single use TOKENS for login attempts;
~ Validated TOKENS will accompany all transactions;
to ~ TOKEN aging to invalidate a TOKEN if it has not been used for ten minutes;
~ TOKEN is associated with the IP Address of the calling machine, so TOKEN stealing is not an easy option;
~ Use of SSL prevents TOKEN or DATA stealing without having is physical access to the customer's display;
~ Use of TOKEN in a form analogous to the Netscape Cookie gives us the option to switch to cookies at a later date. Cookies offer us the facility to hide the TOKEN even further into the document for one extra layer of security; and 20 ~ Use of Hostile-IP table to block multiple offenders without detection by them.
In addition to the security implemented by TOKEN as described above, the Web Servers) are in a Data Management Zone for further low Ievel security.
2s The DMZ security is discussed below.
D. Logirc Process Figure 42 shows the Login Process. The sequence of events leading to a successful login is:
3o 1. The user requests a connection to www.galileo.mci.com.
2. A server is selected from a set using DNS Round-robin.
3. An HTML Page is sent to the user's browser.
4. The Page checks the browser for JAVA Compliance and displays a welcome message.
iz1 WO 98147298 PCT/US9810?927 5. If the browser is not Java compliant, the process stops with an appropriate message.
6. If the browser is Java compliant, it automatically issues a "GET Login Screen" request to the www.galileo.mci.com server. This request also switches the browser to SSL v2. It will fail if the Browser is not SSL
compliant.
7. The Web Server does the following:
A. The Web server gets a Single Use Token from its internal Token service.
~o B. The Web server picks one applet from a large set.
C. The Web server Records the Applet, Token, and Client IP
address in a Database.
D. The Web server sends back the Login Screen, with Applet &
Token.
is 8. User fills in the Login Screen fields - User Id and Passcode.
A. The User Id is the user's Directline number (printed on User's Business cards and is in public domain).
B. The Passcode is a Six digit number known only to the User.
9. When the User presses Enter (or clicks on the LOGIN button) the 2o Java Applet sends the UserId, Passcode, Token, and Scrambled Token back. The Scrambling Algorithm is specific to the Applet that was sent in Step ?D.
10. If the browser's IP address is in the Hostile-IP table, the server goes back to Step ?.
25 I 1. The Web server authenticates the Login request against what it recorded in Step ?C.
12. If the test is invalid: if this is the third successive failed attempts from the same IP address server records the Address in Hostile-IP
table.
30 13. The server goes back to Step ?.
14. If the test is valid; The server sends a select services screen to the Browser with an embedded Token. The Token is still associated with the Browser's IP address, but it now has an expiration time.
a zz ,.

E. Service Selection When the user selects an option from the Service selection screen, the request is accompanied by the Token. The token is validated before the service is accessed, as shown in Figure 43.
s F. Service The screens generated by the Application Servers all contain the Token issued to the user when the Login process was started. This Token has an embedded expiration time and a valid source IP Address. All operation to requests include this token as a part of the request.
The service requests are sent by the browser as HTML forms, APPLET
based forms or plain Hyper Links. In the first two instances, the Token is sent back as a Hidden field using the HTTP-POST method. The Hyper-Links Is use either the HTTP-GET method with embedded Token or substitute the Cookie in place of a Token. The format of the Token is deliberately chosen to be compatible with this approach.
1. NIDS Server.
2o The NIDS server in the system is isolated from the Web Servers by a router-based firewall. The NIDS server runs the NIDSCOMM and ASCOMM
services that allow TCP clients access to databases on the NIDS server. The NIDSCOMM and ASCOMM services do not allow connectivity to databases not physically located on the NIDS Server.
The following databases (C-tree services) on the NIDS server are used by the Welcome Server, Token Server and Profile Management Application Server:
~ 800_PIN_ 1 CALL (this is a partitioned database);
~ 1 CALL TRANS;
~ COUNTRY;
~ COUNTRY_SET;
~ COUNTRY2 (maybe);
~ COUNTRY CITY (maybe);

~ NPA_CITY;
~ NPACITY_OA300 (maybe); and ~ OP153T00.
In addition to the C Tree services named above the following new C tree services will be defined in the SERVDEF and used only on the NIDS server dedicated to the system:
~ TOKEN;
~ SERVERS;
to ~ HOSTILE IP;
~ TOKEN HOSTS; and ~ SERVER ENV.
The following descriptions for these databases do not show the filler field ~5 required at the first byte of each record, nor do they attempt to show any other filler fields that may be required for structure alignment along the 4-byte boundaries. This omission is made only for clarity. The numbers in parentheses next to the field definitions are the number of bytes required to hold the field value.
2. TOKEN database service.
The TOKEN database service is accessed by the Token Servers. The primary operations on this service are Create a new record, read a record for a given Token value and update a record for the given Token value.
A separate chron job running on the NIDS Server itself also accesses this database and deletes obsolete records on a periodic basis. This chron job runs every hour. It does a sequential scan of the database and deletes records for expired tokens.
The TOKEN database service contains the TOKEN records. The TOKEN
records use a single key (the TOKEN) and have the following fields:
1. Version ( 1 );
2. Use Flag (Single/Multi) (1);
I 2 ~-I-3. Token Value ( I6);
4. IP Address ( I6);
5. User Id (16);
6. Time Granted (4); and 7. Time expires (4).
The key field is the Token Value.
3. SERVERS database service.
The Servers Database Service is accessed by the Welcome Server at configuration time. The records in this database contain the following 1o fields:
1. Application Name (16);
2. Application Server Host Name (32);
3. Application Server Domain Name (32);
4. Weight (1);
5. Application Icon File URL (64); and 6. Application Description File URL (64).
The key field is the combination of Application Name, Server Host Name, and Server Domain Name. This database is read by the Welcome Servers sequentially. This database is also accessed by the Web Administrators to 2o Create, Read, Update and Delete records. This access is via the ASCOMM
interface. The Web Administrators use the a HTML form and CGI script for their administration tasks.
4. HOSTILE IP database service.
This database is accessed by the Welcome servers to create new records or read existing records based on IP address as the key. The read access is very frequent. This database contains the following fields:
1. IP Address ( 16);
2. Time entered (4); and 3. Time expires (4).
The key field is the IP Address. All three values are set by the Welcome Server when creating this record. If the entry is to be over-ridden, the service doing the over-ride will only be allowed to change the Time expires value to <epoch_start>, thus flagging the entry as over-ride.
i2~

This database is also accessed by the Web Administrators to Create, Read, Update, and Delete records. Access is via the ASCOMM interface. The Web Administrators use the HTML form and CGI script for their administration tasks.
Customer Service uses a specially developed tool to access this database and access is allowed only from within the corporate firewall.
to A chron job running on the NIDS server also accesses this database and deletes all obsolete records from this database. This job logs all its activity.
The log of this job is frequently examined by the Web Administrators all the time.
S. TOKEN HOSTS database service.
This database service lists IP Addresses of the hosts trusted by the Token Servers. This database is read by the Token Service at configuration time.
The records in this database contain the following fields:
1. IP Address (16);

2. Authority ( 1 );

3. Host Name (32);

4. Host Domain Name (32);
and S. Host description (64).

The key field is the IP Address. The Authority binary flag determines the access level. The low access level only allows validate/re-validate commands on an existing TOKEN; the high access level additionally allows Grant and Validate single use TOKEN commands as well.
This database is also accessed by the Web Administrators to Create, Read, 3o Update and Delete records. Access is via the ASCOMM interface. The Web Administrators use the HTML form and CGI script for their administration tasks.
6. SERVER ENV database service.
iz6 ,.

This database is read by the Welcome and Application servers at startup. It defines the starting environment for these servers. In ane embodiment, only one field (and only for the Welcome Servers) is designed to be used. This is expanded in other embodiments.
s The records in this database contain the following fields:
1. Sequence Number (4);
2. Application Name ( 16);
3. Environment Name (32); and 4. Environment Value (64).
The key field is Sequence Number. Environment values may refer to other environment variables by name. The values are evaluated at run time by the appropriate CGI scripts. The Welcome Servers are assigned the pseudo is Application Name of WELCOME.
This database is also accessed by the Web Administrators to Create, Read, Update and Delete records. This access is via the ASCOMM interface. The Web Administrators use the HTML form and CGI script for their 2o administration tasks.
7. Chron Job(s).
The NIDS Server runs a cleanup chron job. This job is scheduled to run every hour. The main tasks for this job are the following:
25 1. Scan the HOSTILE_IP database and report on all records. This report contains all records. The aim to track repeat offenders based on this report.
2. Scan the HOSTILE IP database and report on records with <epoch_time> as their expiration time.
30 3. Scan the HOSTILE_IP database and delete obsolete records.
4. Scan the TOKEN database and report on all records. This report format will be geared towards traffic reporting rather than scanning each entry.
5. Scan the TOKEN database to delete obsolete records.
j2=~

G. Standards The following coding standards have been developed:
1. HTML Look and Feel standards;
s 2. Java Look and Feel standards (derived from the HTML look and feel standards, these are the new class libraries used in development to force a common look and feel on the site's pages); and 3. HTML Programming standards.
H. System The system administration tasks require reporting of at least the following System Operating Parameters to the System Administrators:
~ System stats and disk usage with time stamps;
~ Network operating parameters with time stamps;
1s ~ Web page usage and access statistics with time stamps;
~ TOKEN usage statistics;
~ Hostile IP alarms and statistics;
The following tools and utilities are on the Servers in DMZ;
~ Time synchronization;
20 ~ Domain Name Servers;
~ System Log Monitoring;
~ Alarm reporting; and ~ Secure Shell.
The system generates alarms for the following conditions:
2s ~ Incorrect use of TOKENS;
~ Hostile IP table changes;
~ TOKEN Expiration; and ~ Login attempts.
3o The alarms will be generated at different levels. The Web Servers use the following broad guidelines:
1. The servers run in a root environment.
2. The administrators are able to start a staging server on a non-standard port to test a new (staged) service.
~ 2~

3. The staging server is accessible from Internet during the staging run.
The Administrators have the option to move the staging software from staging area to production area with a single command. There are suitable checks to make sure this is not done accidentally.
s I. Product/
A preferred embodiment enables directlineMCI customers additional control over their profile by providing a graphical user interface, and a common messaging system. The capability to access the power of a preferred to embodiment exists in the form of a directlineMCI profile and common messaging system. The user is able to modify his account, customizing his application by making feature/functionality updates. The application enables the power of the future capabilities that a preferred embodiment integration will provide by allowing the user to run his application.
The user is able to access ail of his messages by connecting with just one location. FAX, email, page and voice messages will be accessed through a centralized messaging interface. The user is able to call into the centralized messaging interface through his message center interface to retrieve 2o messages. A centralized message interface provides the user the capability to manage his communications easily and effectively.
The user interface has two components, the user's application profile and message center. The interface is accessible through PC software (i.e., PC
Client messaging interface), an ARU or a VRU, and a World Wide Web (WWW) Browser. The interface supports the customization of applications and the management of messages.
The feature/functionality requirements for an embodiment will be 3o presented below. The first piece to be described is the ARU interface and its requirements for the user interface, message management and profile management. Following the ARU requirements, requirements are also provided for the WWW Browser and PC Client interfaces.
2y J. Interface Feature Requirements (Overview) A front-end acts as an interface between the user and a screen display server in accordance with a preferred embodiment. The user is able to access the system and directly access his profile and messages. The user interface is used to update his profile and to access his messages. The user's profile information and the user's messages may reside in different locations, so the interface is able to connect to both places. Profile and messaging capabilities are separate components of the interface and have different requirements.
Through his interface, the user is able to update his profile in real-time through profile management. The application profile is the front-end to the user account directory, which is where all of the user account information resides in a virtual location. Also, a user is able to manage his messages (voicemail, faxmail, email, pager recall) through his message center. The message center is the front-end to the centralized messaging database, which is where all of the user's messages may reside, regardless of message content.
2o Three user interfaces are supported:
~ DTMF access to an ARU or VRU;
~ WWW Browser access to a WWW Site; and ~ PC Client access to a Messaging Server.
2s From the ARU, the users are able to update their profiles (directlineMCI
only), retrieve voicemail messages and pager recall messages, and retrieve message header (sender, subject, date/time) information for faxmail and email messages. Through the PC Client, the user is limited to message retrieval and message manipulation. The WWW Browser provides the user a 30 comprehensive interface for profile management and message retrieval.
Through the WWW Browser, the users are able to update their profiles (directiineMCI, Information Services, List Management, Global Message Handling and Personal Home Pages) and retrieve all message types.

1. The User Account Profile.
The user is able to access account information through the application profile. The application profile provides an intelligent interface between the user and his account information, which resides in the user account s directory. The User Account Directory accesses the individual account information of users. Users are able to read and write to the directory, making updates to their accounts. The directory allows search capabilities, enabling customer service representatives to search for a specific account when assisting a customer.
When a customer obtains a phone number, the user account directory reflects the enrollment, and the user is able to access and update features through his user account profile. If a customer withdraws, the user directory will reflect the deactivation, and the service will be removed from is the user's application profile.
In summary, the user account directory provides account information for each of the user's services. However, the user account directory is limited to: directlineMCI profile, Information Services profile, Global Message 2o Handling, List Management and Personal Home Page profiles. This information determines the feature/functionality of the user's application and provides the user with the flexibility that is necessary to customize his application, allowing MCI to meet his continuously changing communication needs.
2. The Database of Messages.
An important feature that is offered is the integration of messages.
Messages of similar and dissimilar content are consolidated in one virtual location. Through a call, the message center provides the user with a review of all of his messages, regardless of content or access. Through the interface messaging capabilities, the user is also able to maintain an address book and distribution lists.
This message database is a centralized information store, housing (3I

messages for users. The message database provides common object storage capabilities, storing data files as objects. By accessing the message database, users retrieve voicemail, faxmail, email and pager recall messages from a single virtual location. In addition, by using common object storage capabilities, message distribution is extremely efficient.
K. Automated Response Unit (ARUJ Capabilities K
1. User Interface .
The ARU interface is able to perform directlineMCI Profile Management, 1o Information Services Profile Management, message retrieval and message distribution. The DTMF access provided through the ARU is applied consistently across different components within the system. For example, entering alphabetic characters through the DTMF keypad is entered in the same manner regardless if the user is accessing Stock Quote information or broadcasting a fax message to a distribution list.
Voicemail Callback Auto Redial provides the capability to prompt for and collect a DTMF callback number from a guest leaving a voicemail and automatically launch a return call to the guest call back number when 2o retrieving messages. Upon completing the callback, the subscriber will be able to return to the same place where they left off in the mailbox.
Music On-Hold provides music while a guest is on-hold.
Park and Page provides a guest an option to page a directlineMCI
subscriber, through the directlineMCI gateway, then remain on-hold while the subscriber is paged. The subscriber receives the page and calls their directlineMCI number, where they can select to be connected with the guest on hold. Should the subscriber fail to connect a call with the guest, the 3o guest will receive an option to be forwarded to voicemail. If the subscriber does not have voicemail as a defined option, then the guest a final message will be played for the guest.
Note: The guest has the ability to press an option to be forwarded to voicemail at any time while on hold.
13z Call Screening with Park and Page An embodiment provides the subscriber with functionality for responding to a park and page, the identity of the calling party (i.e., guest). This provides the subscribers the ability to choose whether they wish to speak to the guest ar transfer the guest to voicemail, prior to connecting the call. Specifically, guests are ARU
prompted to record their names when they select the park and page option.
When the subscriber respond to the park and page, they will hear an ARU
prompt stating, "You have a call from RECORDED NAME", then be presented with the option to connect with the calling party or transfer the party to voicemail. If the subscriber does not have voicemail as a defined option, then the guest will be deposited to a final message. The guest also will have the ability to press an option to be forwarded to voicemail at any time while on hold.
Two-way Pager Configuration Control and Response to Park and Page The system also allows a subscriber to respond to a park and page notification by instructing the ARU to route the call to voicemail or final message or continue to hold, through a command submitted by a two-way 2o pager.
Text Pager Support The system allows a subscriber to page a directlineMCI subscriber, through the directlineMCI gateway, and a leave a message to be retrieved by a text pager. Specifically, upon choosing the appropriate option, the guest will be transferred to either the networkMCI Paging or the SkyTel message center where an operator will receive and submit/create a text-based message to be retrieved by the subscriber's text pager.
3o Forward to the Next Termination Number The system provides the capability for the party answering the telephone, to which a directlineMCI call has been routed, to have the option to have the call routed to the next termination number in the directlineMCI routing sequence. Specifically, the called party will receive a prompt from the r33 directlineMCI ARU gateway, which indicates that the call has been routed to this number by directlineMCI and providing the called party with the option to receive the incoming call or have the call routed to the next termination number or destination in the routing sequence. The options presented to a called party include:
~ Press an option to accept the call ~ Press an option to send the call to the next termination ~ Let the call time-out (i.e., no action taken) and then proceed to the next termination.
io Less Than 2 Second # Reorigination An embodiment also provides the capability to reoriginate an outbound call, from the directlineMCI gateway, by pressing the pound ( # ) key for less than two seconds. Currently, directlineMCI requires the # key to be depressed for two seconds or more before the subscriber can reoriginate a call.
L. Message 1. Multiple Media Message Notification .
The subscriber can receive an accounting of current messages across a number of media, to include voicemail, faxmail, email, paging. Specifically, 2o the subscriber will hear an ARU script stating, for example, "You have 3 new voicemail messages, 2 new faxmail messages, and 10 new email messages."
2. Multiple Media Message Manipulation .
2s A subscriber is allowed to access the Universal Inbox to perform basic message manipulation, of messages received through multiple media (voicemail, faxmaii, email, paging), through the directlineMCI ARU gateway.
Subscribers are able to retrieve voicemail messages and pager messages, and retrieve message header (priority, sender, subject, date/time, size) 3o information for faxmail and email messages. In addition, subscribers are able to save, forward or delete messages reviewed from the ARU interface.
The forward feature is limited to distributing messages as either voicemails or faxmails. Only voicemail messages can be forwarded as voicemails.
Email, faxmail and pager messages can be forwarded as faxmaiis; however, l3~

it may be necessary to convert email and pager messages to a G3 format.
When forwarding messages as faxmails, subscribers have the ability to send messages to distribution lists and Fax Broadcast lists.
3. Text to Speech .
The system converts text messages, received as email, faxmail or pager messages, into audio, which can be played back through the directlineMCI
gateway. Initially, the text-to-speech capability will be limited to message header (priority, sender, subject, date/time, size) information.
Subscribers are provided the option to select whether they want to hear message headers first and then select which complete message they want to be played. The only message type that does not support a text-to-speech capability for the complete message will be faxmail messages. The ~5 capability only exists to play faxmail headers. FAXmail header information includes sender's ANI, date/time faxmail was received and size of faxmail.
4. Email Forwarding to a Fax Machine .
Subscribers can forward an email, retrieved and reviewed through the 2o directlineMCI ARU gateway, to a subscriber-defined termination number.
Specifically, the subscriber has the ability to review an email message through the directlineMCI ARU. After reviewing the message, the subscriber receives, among the standard prompts, a prompt requesting whether he would Iike to forward the email message to a specified termination number 25 or have the option to enter an impromptu number. Upon selecting this option and indicating the termination number, the email message is converted to a G3 format and transmitted to the specified termination number. Email attachments that are binary files are supported. If an attachment cannot be delivered to the terminating fax machine, a text 3o message must be provided to the recipient that the binary attachment could not be forwarded. Forwarding of emails to a fax machine does not result in the message being deleted from the "universal inbox".
5. Pager Notification of Messages Received .
13~

A subscriber can receive a pager notification, on a subscriber-defined interval, indicating the number of messages, by message media, that currently reside in the subscriber's "universal inbox". Specifically, the subscriber will have the ability to establish a notification schedule, through the directlineMCI ARU, to receive a pager message which indicates the number of voicemail, faxmail, email and pager messages that reside in the subscriber's "universal inbox".
6. Delivery Confirmation of Voicemail .
to The system provides the subscriber the ability to receive a confirmation voicemail message when a subscriber-initiated voicemail message was not successfully delivered to the terminating party(s).
7. Message Prioritization .
~5 The system provides the guest the ability to assign either regular or urgent priority to a message. When the subscriber receives an accounting of messages, the prioritization will be indicated, and all urgent messages will be indexed before regular messages. This requirement only applies to voicemails, not faxmails. This will require that the "universal inbox" present 2o the proper message priority for directlineMCI voicemails.
M. Irtforma~iorc Services Through the ARU interface, users will be able to receive content from information services which are configurable through the WWW Browser 2s interface. Information content will be provided as an inbound service and an outbound service. The information content that is defined through the WWW Browser (i.e., Profile Management) is defined as the inbound information content and will be limited to:
~ Stock Quotes and Financial News 30 ~ Headline News.
Subscribers also have the ability to access additional information content through the ARU interface; however, this information is not configurable through the WWW Browser (i.e., Profile Management). This additional I ~~
,.

information content will be referred to as outbound information content and will consist of:
~ Stock Quotes and Financial News;
~ Headline News;
~ Weather;
~ Sports News and Scores;
~ Soap Opera Updates;
~ Horoscopes;
~ Lottery Results;
to ~ Entertainment News; and ~ Traveler's Assist.
The configurable parameters of the inbound information content is defined below. Retrieval of outbound information content will support the entry of ~5 alphabetic characters through a DTMF keypad. Entering of alphabetic characters must be consistent with the manner that alphabetic characters are entered through DTMF for list management.
Access to Traveler's Assist will be bundled with the other outbound 2o information services such that the subscriber only has to dial a single 800/8XX number. The 800/8XX call may extend to different termination depending upon the information content selected.
N. Message Storage Requirements 25 The message storage requirements are consistent with the message storage requirements defined below.
4. Pro, f~t Ie Marcagrement directlineMCI Profile Management 3o Subscribers can also review, update and invoke their directlineMCI account profiles. The directlineMCI profile management capabilities through the ARU interface are consistent with the presentation provided through the WWW Browser and support the following requirements:
~ Create new directlineMCI profiles and assign names to the profile;
(3~

~ Invoke directlineMCI profiles;
~ Voice annotate directlineMCI profile names;
~ Update existing directlineMCI profiles;
~ Support the rules-based logic of creating and updating directlineMCI
profiles (e.g., selection of only one call routing option, like voicemail, will invoke override routing to voicemail; and updates made in one parameter must ripple through all affected parameters, like paging notification);
~ Enable a directlineMCI number;
~ Enable and define override routing number; and ~ Enable and define FollowMe routing.
~ Enable and define final routing (formerly called alternate routing) to:
~ Voicemail and pager;
-Voicemail only;
-Pager only;
-Final message;
~ Invoke menu routing if two or more of the call routing options (FollowMe, voicemail, faxmail or pager) are enabled;
~ Define the default number for faxmail delivery;
~ Activate paging notification for voicemail;
~ Activate paging notification for faxmail; and ~ Provide guest option to classify voicemails for urgent delivery;
~ Define call screening parameters for:
-Name and ANI;
-ANI only;
-Name only; and ~ Enable or disable park and page.
P. Call Routing Menu Change The system also provides the capability for subscribers to modify their call 3o routing termination numbers without having to re-enter termination numbers which they do not wish to change. Specifically, the directlineMCI
routing modification capability requires the subscriber to re-enter all termination numbers in a routing sequence should they wish to change any of the routing numbers. This capability permits the subscriber to change t3S

only the termination numbers they wish to change, and indicate by pressing the "#" key when they do not wish to change a specific number in the routing sequence.
Q. Two-way Pager Configuration Control and Response to Park and Page The system can also enable or disable predefined directlineMCI profiles through a command submitted by a two-way pager.
io R. Personalized Greetings The system provides subscribers the ability to review and update the personalized greeting that will be played from the ARU or displayed from their Personal Home Page. Each greeting is maintained separately and customized to the features available through each interface (ARU or is Personal Home Page).
S. List Management The system also provides the subscriber the ability to create and update lists, and create a voice annotation name for a list. Fax Broadcast list 2o management capabilities are integrated with directlineMCI list management capabilities to provide a single database of lists. From the ARU interface, subscribers have the ability to review, update, add or delete members on a list. In addition, subscribers are able to delete or create lists. The ARU
interface is able to use the lists to distribute voicemail and faxmail 2s messages.
Access to distribution lists supports alphabetic list names such that lists are not limited to list code names. Entering of alphabetic characters through DTMF to the ARU for list names is consistent with the manner that 3o alphabetic characters are entered through DTMF for Information Services.
The List Management requirements are discussed in greater detail below.
In addition to providing message manipulation capabilities, the PC Client also provides an address book and access to lists. The user is able to make i ~ '~

modifications to the address book and manage distribution lists for voice, fax, email and paging messages. In one embodiment, lists created or maintained through the PC Client interface are not integrated with lists created or maintained through the WWW Browser or ARU interfaces, but s such integration can be implemented in an alternative embodiment. The subscriber is able to send a message to a distribution list from the PC
Client. This requires a two-way interface between the PC Client and the List Management database whereby the PC Client can export a comma delimited or DBF formatted file to the database of lists.
to The user is able to create and modify recipient address information through his interface PC software. The user is able to record multiple types of addresses in his address book, including 10 digit ANIs, voice mailbox ids, fax mailbox ids, paging numbers and email addresses (MCIMaiI and is Internet). This information should is saved onto the PC. The address information retained on the PC Client is classified and sorted by recipient's name.
T. Global Message Handling 2o From the ARU interface, subscribers are able to define which message types can be accessed from the "universal inbox". The global message handling requirements are consistent with the requirements defined below.
X. INTERNET TELEPHONY AND RELATED SERVICES
2s The discussion thus far has provided an introduction to the Internet, and therefore Internet telephony, but Internet telephony encompasses quite a few areas of development. The following is a summary of Internet telephony, divided into six key areas. The first area consists of access to Internet telephony services. This area involves accessing and utilizing the Internet 3o using such mechanisms as satellites, dialup services, T1, T3, DS3, OC3, and OC 12 dedicated lines, SMDS networks, ISDN B-channels, ISDN D-channeis, multirate ISDN, multiple B-channel bonded ISDN systems, Ethernet, token ring, FDDI GSM, LMDS, PCS, cellular networks, frame relay, and X.25.
~-I-O
,. , , The second area involves sharing Internet telephony. Multimedia data can utilize circuit-switched networks quite readily due to the high reliability and throughput potential. Issues include shared data, pushing URL data s between parties, data conferencing, shared whiteboarding, resource collaboration, and ISDN user-user signaling.
The third area deals with routing Internet telephony. Issues include the time-of day, the day-of-week, the day-of month, and the day-of-year, in 1o addition to geographic points of origin, network point of origin, and time zone of origin. Analysis of routing also includes user data, destination parties, telephone numbers, lines of origin, types of bearer service, presubscribed feature routing, ANI, and IP addresses. Also, VNET plans, range privileges, directory services, and Service Control Points (SCP)s fall is into routing Internet telephony.
The fourth category deals with quality of service. Analysis must include switched networks, ISDN, dynamic modifications, Internet telephony, RSVP, and redundant network services. In addition, this category includes 2o hybrid Internet/telephony switches, Ethernet features, ISDN features, analog local loops and public phones, and billing for reserved and/or utilized services.
The fifth category is composed of directory services, profiles, and 2s notifications. Examples are distributed directories, finding-me and follow-me services, directory management of telephony, and user interfaces.
Calling party authentication security is also included. Hierarchical and object-oriented profiles exist, along with directory service user profiles, network profile data structures, service profiles, and order entry profiles.
The sixth category consists of hybrid Internet telephony services. Areas include object directed messaging, Internet telephony messaging, Internet conferencing, Internet faxing, information routing (IMMR), voice communications, and intranets (such as those that exist within a l ~-f~ ~

company). Other services include operator services, management service, paging services, billing services, wireless integration, message broadcasts, monitoring and reporting services, card services, video-mail services, compression, authorization, authentication, encryption, telephony application builders, billing, and data collection services.
The seventh category consists of hybrid Internet media services, which include areas of collaborative work which involve a plurality of users. Users can collaborate on Audio, Data and Video. This area includes media 1o conferencing within the Hybrid network. Then there is a broadly related area of Reservations mechanism, Operator-assisted conferencing, and the introduction of content into conferences. The Virtual locations of these conferences will assume importance in the future. The next-generation Chat Rooms will feature virtual conference spaces with simulated Office Environments.
A. System Enrr~rorcmerct for Internet Media 1. Hardware.
A preferred embodiment of a system in accordance with the present 2o invention is preferably practiced in the context of a personal computer such as the IBM PS/2, Apple Macintosh computer or UNIX based workstation. A
representative hardware environment is depicted in Figure lA, which illustrates a typical hardware configuration of a workstation 99 in accordance with a preferred embodiment having a central processing unit 10, such as a microprocessor, and a number of other units interconnected via a system bus 12. The workstation shown in Figure lA includes a Random Access Memory (RAM) 14, Read Only Memory (ROM) 16, an I / O
adapter 18 for connecting peripheral devices such as a communication network (e.g., a data processing network) 81, printer 30 and a disk storage 3o unit 20 to the bus I2, a user interface adapter 22 for connecting a keyboard 24, a mouse 26, a speaker 28, a microphone 32, and/or other user interface devices such as a touch screen (not shown} to the bus 12, and a display adapter 36 for connecting the bus 12 to a display device 38.
The workstation typically has resident thereon an operating system such as ~ ~- z the Microsoft Windows NT or Windows/95 Operating System (OS), the IBM
OS/2 operating system, the MAC System/? OS, or UNIX operating system.
Those skilled in the art will appreciate that the present invention may also be implemented on platforms and operating systems other than those s mentioned.
2. Object-Oriented Software Tools.
A preferred embodiment is written using JAVA, C, and the C++ language and utilizes object oriented programming methodology. Object oriented to programming (OOP) has become increasingly used to develop complex applications. As OOP moves toward the mainstream of software design and development, various software solutions require adaptation to make use of the benefits of OOP. A need exists for these principles of OOP to be applied to a messaging interface of an electronic messaging system such that a set 1s of OOP classes and objects for the messaging interface can be provided.
OOP is a process of developing computer software using objects, including the steps of analyzing the problem, designing the system, and constructing the program. An object is a software package that contains both data and a zo collection of related structures and procedures. Since it contains both data and a collection of structures and procedures, it can be visualized as a self sufficient component that does not require other additional structures, procedures or data to perform its specific task. OOP, therefore, views a computer program as a collection of largely autonomous components, 2s called objects, each of which is responsible for a specific task. This concept of packaging data, structures, and procedures together in one component or module is called encapsulation.
In general, OOP components are reusable software modules which present 3o an interface that conforms to an object model and which are accessed at run-time through a component integration architecture. A component integration architecture is a set of architectural mechanisms which allow software modules in different process spaces to utilize each other's capabilities or functions. This is generally done by assuming a common I~3 component object model on which to build the architecture.
It is worthwhile to differentiate between an object and a class of objects at this point. An object is a single instance of the class of objects, which is s often just called a class. A class of objects can be viewed as a blueprint, from which many objects can be formed.
OOP allows the programmer to create an object that is a part of another object. For example, the object representing a piston engine is said to have 1o a composition-relationship with the object representing a piston. In reality, a piston engine comprises a piston, valves and many other components; the fact that a piston is an element of a piston engine can be Iogicaliy and semantically represented in OOP by two objects.
is OOP also allows creation of an object that "derived from" another object.
If there are two objects, one representing a piston engine and the other representing a piston engine wherein the piston is made of ceramic, then the relationship between the two objects is not that of composition. A
ceramic piston engine does not make up a piston engine. Rather it is 2o merely one kind of piston engine that has one more limitation than the piston engine; its piston is made of ceramic. In this case, the object representing the ceramic piston engine is called a derived object, and it inherits all of the aspects of the object representing the piston engine and adds further limitation or detail to it. The object representing the ceramic 25 piston engine "derives from" the object representing the piston engine. The relationship between these objects is called inheritance.
When the object or class representing the ceramic piston engine inherits all of the aspects of the objects representing the piston engine, it inherits the 3o thermal characteristics of a standard piston defined in the piston engine class. However, the ceramic piston engine object overrides these ceramic specific thermal characteristics, which are typically different from those associated with a metal piston. It skips over the original and uses new functions related to ceramic pistons. Different kinds of piston engines have y4 ,.

different characteristics, but may have the same underlying functions associated with them (e.g., number of pistons in the engine, ignition sequences, lubrication, etc.). To access each of these functions in any piston engine object, a programmer would identify the same functions with s the same names, but each type of piston engine may have different/overriding implementations of functions behind the same name.
This ability to hide different implementations of a function behind the same name is called polymorphism and it greatly simplifies communication among objects.
io With the concepts of composition-relationship, encapsulation, inheritance and polymorphism, an object can represent just about anything in the real world. In fact, our logical perception of the reality is the only limit on determining the kinds of things that can become objects in object-oriented is software. Some typical categories are as follows:
? Objects can represent physical objects, such as automobiles in a traffic-flow simulation, electrical components in a circuit-design program, countries in an economics model, or aircraft in an air-traffic-control system.
2o ? Objects can represent elements of the computer-user environment such as windows, menus or graphics objects.
? An object can represent an inventory, such as a personnel file or a table of the latitudes and longitudes of cities.
? An object can represent user-defined data types such as time, angles, 2s and complex numbers, or points on the plane.
With this enormous capability of an object to represent just about any logically separable matters, OOP allows the software developer to design and implement a computer program that is a model of some aspects of 3o reality, whether that reality is a physical entity, a process, a system, or a composition of matter. Since the object can represent anything, the software developer can create an object which can be used as a component in a larger software project in the future.
f ~~

If 90% of a new OOP software program consists of proven, existing components made from preexisting reusable objects, then only the remaining 10% of the new software project has to be written and tested from scratch. Since 90% already came from an inventory of extensively s tested reusable objects, the potential domain from which an error could originate is IO% of the program. As a result, OOP enables software developers to build objects out of other, previously built, objects.
This process closely resembles complex machinery being built out of to assemblies and sub-assemblies. OOP technology, therefore, makes software engineering more Iike hardware engineering in that software is built from existing components, which are available to the developer as objects. All this adds up to an improved quality of the software as well as an increased speed of its development.
Programming languages are beginning to fully support the OOP principles, such as encapsulation, inheritance, polymorphism, and composition-relationship. With the advent of the C++ language, many commercial software developers have embraced OOP. C++ is an OOP language that offers a fast, machine-executable code. Furthermore, C++ is suitable for both commercial-application and systems-programming projects. For now, C++ appears to be the most popular choice among many OOP
programmers, but there is a host of other OOP languages, such as Smalltalk, common lisp object system (CLOS), and Eiffel. Additionally, OOP
capabilities are being added to more traditional popular computer programming languages such as Pascal.
The benefits of object classes can be summarized, as follows:
Objects and their corresponding classes break down complex 3o programming problems into many smaller, simpler problems.
Encapsulation enforces data abstraction through the organization of data into small, independent objects that can communicate with each other. Encapsulation also protects the data in an object from accidental damage, but allows other objects to interact with that data ~ ~6 by calling the object's member functions and structures.
Subclassing and inheritance make it possible to extend and modify objects through deriving new kinds of objects from the standard classes available in the system. Thus, new capabilities are created without having to start from scratch.
Polymorphism and multiple inheritance make it possible for different programmers to mix and match characteristics of many different classes and create specialized objects that can still work with related objects in predictable ways.
1o Class hierarchies and containment hierarchies provide a flexible mechanism for modeling real-world objects and the relationships among them.
Libraries of reusable classes are useful in many situations, but they also have some limitations. For example:
Complexity. In a complex system, the class hierarchies for related classes can become extremely confusing, with many dozens or even hundreds of classes.
Flow of control. A program written with the aid of class libraries is still responsible for the flow of control (i.e., it must control the 2o interactions among all the objects created from a particular library).
The programmer has to decide which functions to call at what times for which kinds of objects.
Duplication of effort. Although class libraries allow programmers to use and reuse many small pieces of code, each programmer puts those pieces together in a different way. Two different programmers can use the same set of class libraries to write two programs that do exactly the same thing but whose internal structure (i.e., design) may be quite different, depending on hundreds of small decisions each programmer makes along the way. Inevitably, similar pieces of code 3o end up doing similar things in slightly different ways and do not work as well together as they should.
Class libraries are very flexible. As programs grow more complex, more programmers are forced to reinvent basic solutions to basic problems over I ~-I~ ~

and over again. A relatively new extension of the class library concept is to have a framework of class libraries. This framework is more complex and consists of significant collections of collaborating classes that capture both the small scale patterns and major mechanisms that implement the common requirements and design in a specific application domain. They were first developed to free application programmers from the chores involved in displaying menus, windows, dialog boxes, and other standard user interface elements for personal computers.
to Frameworks also represent a change in the way programmers think about the interaction between the code they write and code written by others. In the early days of procedural programming, the programmer called libraries provided by the operating system to perform certain tasks, but basically the program executed down the page from start to finish, and the programmer is was solely responsible for the flow of control. This was appropriate for printing out paychecks, calculating a mathematical table, or solving other problems with a program that executed in just one way.
The development of graphical user interfaces began to turn this procedural 2o programming arrangement inside out. These interfaces allow the user, rather than program logic, to drive the program and decide when certain actions should be performed. Today, most personal computer software accomplishes this by means of an event loop which monitors the mouse, keyboard, and other sources of external events and calls the appropriate 25 parts of the programmer's code according to actions that the user performs.
The programmer no longer determines the order in which events occur.
Instead, a program is divided into separate pieces that are called at unpredictable times and in an unpredictable order. By relinquishing control in this way to users, the developer creates a program that is much 3o easier to use. Nevertheless, individual pieces of the program written by the developer still call libraries provided by the operating system to accomplish certain tasks, and the programmer must still determine the flow of control within each piece after it's called by the event loop. Application code still "sits on top of the system.
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Even event loop programs require programmers to write a lot of code that should not need to be written separately for every application. The concept of an application framework carries the event loop concept further. Instead - s of dealing with all the nuts and bolts of constructing basic menus, windows, and dialog boxes and then making these things all work together, programmers using application frameworks start with working application code and basic user interface elements in place. Subsequently, they build from there by replacing some of the generic capabilities of the framework 1o with the specific capabilities of the intended application.
Application frameworks reduce the total amount of code that a programmer must write from scratch. However, because the framework is really a generic application that displays windows, supports copy and paste, and so is on, the programmer can also relinquish control to a greater degree than event loop programs permit. The framework code takes care of almost all event handling and flow of control, and the programmer's code is called only when the framework needs it (e.g., to create or manipulate a data structure) .
A programmer writing a framework program not only relinquishes control to the user (as is also true for event loop programs), but also relinquishes the detailed flow of control within the program to the framework. This approach allows the creation of more complex systems that work together 2s in interesting ways, as opposed to isolated programs with custom code being created over and over again for similar problems.
Thus, as explained above, a framework basically is a collection of cooperating classes that make up a reusable design solution for a given problem domain. It typically provides objects that define default behavior (e.g., for menus and windows), and programmers use it by inheriting some of that default behavior and overriding other behavior so that the framework calls application code at the appropriate times.
There are three main differences between frameworks and class libraries:
I ~-I- ~

? Behavior versus protocol. Class libraries are essentially collections of behaviors that you can call when you want those individual behaviors in your program. A framework, on the other hand, provides not only behavior but also the protocol or set of rules that govern the ways in which behaviors can be combined, including rules for what a programmer is supposed to provide versus what the framework provides.
? CaII versus override. With a class library, the code the programmer instantiates objects and calls their member functions. It's possible to to instantiate and call objects in the same way with a framework (i.e., to treat the framework as a class library), but to take full advantage of a framework's reusable design, a programmer typically writes code that overrides and is called by the framework. The framework manages the flow of control among its objects. Writing a program involves dividing responsibilities among the various pieces of software that are called by the framework rather than specifying how the different pieces should work together.
? Implementation versus design. With class libraries, programmers reuse only implementations, whereas with frameworks, they reuse 2o design. A framework embodies the way a family of related programs or pieces of software work. It represents a generic design solution that can be adapted to a variety of specific problems in a given domain. For example, a single framework can embody the way a user interface works, even though two different user interfaces created 2s with the same framework might solve quite different interface problems.
B. Telephony Over The Internet Voice over the Internet has become an inexpensive hobbyist commodity.
3o Several firms are evolving this technology to include interworking with the PSTN. This presents both a challenge and an opportunity for established carriers like MCI and BT especially in the IDDD arena. This discussion explores how a carrier class service could be offered based on this evolving technology. Of particular interest are ways to permit interworking between SO
,.

the PSTN and the Internet using 1 plus dialing.
The introductory discussion considers the technical requirements to support PC to PC connectivity in a more robust manner than presently offered, in addition to the technical requirements for a PSTN to Internet voice gateway. Consideration is given to how calls can be placed from PCs to a PSTN destination and visa versa. The case of PSTN to PSTN
communications, using the Internet as a long distance network is also explored.
io It is shown how such services can be offered in a way that will complement existing PSTN services, offering lower prices for a lower quality of service.
At issue in the longer term is the steady improvement in quality for Internet telephony and whether this will ultimately prove competitive with conventional voice services.
1. Introduction.
In the mid-late 1970s, experiments in the transmission of voice over the Internet were conducted as part of an ongoing program of research 2o sponsored by the US Defense Advanced Research Projects Agency. In the mid-1980s, UNIX-based workstations were used to conduct regular audio/video conferencing sessions, in modest quantities, over the Internet.
These experimental applications were extended in the late 1980s with larger scale, one-way multicasting of voice and video. In 1995 a small 2s company, VocalTec (www.vocaltec.com), introduced an inexpensive software package that was capable of providing two way voice communications between mufti-media PCs connected to the Internet. Thus was born a new generation of telephony over the Internet.
3o The first software package, and its immediate followers, provided a hobbyist tool. A meeting place based on a Internet Relay Chat "room" (IRC) was used to establish point to point connections between end stations for the voice transfer. This resulted in chance meetings, as is common in chat rooms, or a prearranged meeting, if the parties coordinated ahead of time, by email or ~3 f other means.
a~ How it Works A user with a mufti-media PC and an Internet connection can add the s Internet Telephony capability by loading a small software package. In the case of VocalTec, the package makes a connection to the meeting place (IRC
server), based on a modified chat server. At the IRC the user sees a list of all other users connected to the IRC.
1o The user calls another user by clicking on his name. The IRC responds by sending the IP address of the called party. For dial in users of the Internet, an IP address is assigned at dial in time, and consequently will change between dial in sessions. If the destination is not already engaged in a voice connection, its PC beeps a ring signal. The called user can answer 15 the phone with a mouse click, and the calling party then begins sending traffic directly to the IP address of the called party. A mufti-media microphone and speakers built into or attached to the PC are used as a speakerphone. The speaker's voice is digitized, compressed and packetized for transmission across the Internet. At the other end it is decompressed 2o and converted to sound through the PC's speakers.
b) Implications Telephony over the Internet offers users a low cost service, that is distance and border insensitive. For the current cost of Internet access (at low hourly rates, or in some cases unlimited usage for a flat fee) the caller can 25 hold a voice conversation with another PC user connected to the Internet.
The called party contributes to the cost of the conversation bx paying for his Internet access. In the case that one or both ends are LAN connected to the Internet by leased lines the call is free of additional charges. All of this is in contrast to the cost of a conventional long distance, possibly 3o international, call.
c) Quality of Service The voice quality across the Internet is good, but not as good as typical telephone toll quality. In addition, there are significant delays experienced tSz during the conversation. Trying to interrupt a speaker in such an environment is problematic. Delay and quality variations are as much a consequence of distance and available capacity as they are a function of compression, buffering and packetizing time.
Delays in the voice transmission are attributable to several factors. One of the biggest contributors to delays is the sound card used. The first sound cards were half duplex and were designed for playback of recorded audio.
Long audio data buffers which helped ensure uninterrupted audio playback io introduced real time delays. Sound card based delays are being reduced over time as full duplex cards designed for "speakerphone" applications are brought to the market.
Other delays are inherent in the access line speeds (typically 14.4-28.8 is kbps for dial-up Internet access) and in the packet forwarding delays in the Internet. Also there is delay inherent in filling a packet with digitized encoded audio. For example, to fill a packet with 90 ms of digitized audio, the application must wait at least 90 ms to receive the audio to digitize.
Shorter packets reduce packet-filling delays, but increase overhead by 2o increasing the packet header to packet payload data ratio. The increased overhead also increases the bandwidth demands for the application, so that an application which uses short packets may not be able to operate on a 14.4 kbps dial-up connection. LAN-based PCs suffer less delay, but everyone is subject to 2s variable delays which can be annoying.
Lastly, there are delays inherent in audio codecs. Codec delays can vary from 5 to 30 ms for encoding or decoding. Despite the higher latencies associated with Internet telephony, the price is right, and this form of voice 3o communication appears to be gaining in popularity.
2. IP Phone as a Commercial Service.
IP telephony technology is here whether the established carriers Iike it or not. Clearly the use of the Internet to provide international voice calls is a t!~3 potential threat to the traditional International Direct Distance Dialing (IDDD) revenue stream. Although it may be several years before there is an appreciable revenue impact, it cannot be stopped, except perhaps within national borders on the basis of regulation. The best defense by the s carriers is to offer the service themselves in an industrial strength fashion.
To do this requires an improved call setup facility and an interface to the PSTN.
Facilitating PC to PC connections is useful for cases in which the voice to conversation needs to be conducted during a simultaneous Internet data packet communication, and the parties don't have access to separate telephone facilities. Dial-up Internet subscribers with only one access circuit might find themselves in that position. Cost considerations may also play a role in dictating the use of PC to PC telephony. The larger use of 15 this technology will occur when the Internet can be used in place of the long distance network to interconnect ordinary telephone hand sets. The number of multi-media Internet connected PCs in the world (estimated at million) is minuscule compared to the number of subscriber lines worldwide (estimated at 660 million). This service is in the planning stages of several companies.
In the sections below we look at each of the end point combinations possible in a full Internet telephony service. The most important aspects relate to the PSTN to Internet gateway capabilities. Of particular interest is the possibility of providing the PSTN caller with one-step dialing to his called party. The one-step dialing solutions discussed below are in the context of the North American numbering plan. There are essentially four cases:
PC to PC;
3o PC to PSTN;
PSTN to PC; and PSTN to PSTN.
The first case is addressed by today's IP Phone software. The second and third case are similar but not identical and each requires a gateway 15~

WO 98/47298 PCTlUS98/07927 between the PSTN and the Internet. The last case uses the Internet as a long distance network for two PSTN telephones.
aj PC to PC
(1) Directory Service To facilitate PC to PC Internet Telephony a directory service is needed to find the IP address of the called party based on a name presented by the calling party Early Internet telephony software utilized a modified Internet chat server as a meeting place. More recently, Internet telephony software is replacing the chat server with a directory service which will uniquely to identify Internet telephone users (perhaps by email address). To receive calls, customers would register with the directory service (for a fee, with recurring charges) and would make their location (IP address) known to the directory system whenever they connect to the Internet and want to be available for calls. The best way to accomplish automatic notification is to get agreement between the vendors of IP phone software on a protocol to notify the directory service whenever the software is started (automatic presence notification). It would also be desirable, as an option, to find a way to automatically invoke the IP phone software when the IP stack is started.
The directory service is envisioned as a distributed system, somewhat like the Internet Domain Name System, for scalability. This is not to imply, necessarily, the user@foo.com format for user identification.
Theoretically only the called parties need to be registered. If the calling party is not registered, then the charge for the call (if there is one) could be made to the called party (a collect call). Alternatively, we can insist that the caller also be registered in the directory and billed through that mechanism (this is desirable since we charge for the registration and avoid the complications that collect calls require). A charge for the call setup is billed, 3o but not for the duration, over and above the usual Internet charges.
Duration charges already apply to the dial up Internet user and Internet usage charges, both for dial up and dedicated usage, are probably not too far away.
r ,~5 Collect calls from a registered user may be required to meet market demand. A scheme for identifying such calls to the called party must be devised, along with a mechanism for the called party to accept or reject the collect call. The directory service will track the ability of the called software s to support this feature by version number (or, alternatively, this could be a matter for online negotiation between the IP telephony software packages).
In the event of collect calls (assuming the caller is not registered), the caller could claim to be anyone she chooses. The directory service will force the caller to take on a temporary "assigned" identity (for the duration of the to call) so the called party will know this is an unverified caller. Since IP
addresses are not necessarily fixed, one cannot rely on them to identify parties.
(2) Interoperability is Nearly all IP phone software packages on the market today use different voice encoding and protocols to exchange the voice information. To facilitate useful connections the directory will store the type and version (and possibly options) of Internet phone software being used. To make this work effectively software vendors will report this information automatically 2o to the directory service. This information will be used to determine interoperability when a call is placed. If the parties cannot interoperate, an appropriate message must be sent to the caller. As an alternative, or in addition to registration of software type, a negotiation protocol could be devised to determine interoperability on the fly, but all packages would 25 have to "speak" it.
There is a question of whether translations between IP phone encoding can be performed with acceptable quality to the end user. Such a service could have a duration and or volume fee associated with it, which might limit the 3o desirability of its use. Also, after a shake out period we expect only a few different schemes to exist and they will have interoperability, perhaps through an industry agreed lowest common denominator compression and signaling protocol. So far, all the IP phone software vendors we have contacted are in favor of an Esperanto that will permit interoperability. If ~~6 this comes to pass the life span of the translation services will be short, probably making them not economically attractive.
We can help the major software vendors seek consensus on a "common"
s compression scheme and signaling protocol that will provide the needed interoperability. Once the major vendors support this method the others will follow. This is already happening, with the recent announcements from Intel, Microsoft, Netscape, and VocalTec that they will all support the H.323 standard in coming months. This can be automatically detected at call setup time. The directory service would keep track of which versions of which software can interoperate. To facilitate this functionality the automatic notification of presence should include the current software version. This way upgrades can be dynamically noted in the directory service. Some scheme must also be defined to allow registration 15 information to be passed between software packages so if a user switches packages she is able to move the registration information to the new application. There is no reason to object if the user has two applications each with the same registration information. The directory service will know what the user is currently running as part of the automatic presence 2o notification. This will cause a problem only if the user can run more than one IP phone package at the same time. If the market requires this ability the directory service could be adapted to deal with it. The problem could also be overcome through the use of negotiation methods between interacting IP phone software packages.
(3) Call Progress Signaling If the user is reachable through the directory system, but is currently engaged in a voice connection, then a call waiting message (with caller ID, something which is not available in the PSTN call waiting service) is sent to 3o the called party and a corresponding message is sent back to the caller.
If the user is reachable through the directory system, but is currently not running his voice software (IP address responds, but not the application --see below for verification that this is the party in question) then an S~

appropriate message is returned to the caller. (As an option an email could be sent to the called party to alert him to the call attempt. An additional option would be to allow the caller to enter a voice message and attach the "voice mail" to the email. The service could also signal the caller to s indicate: busy, unreachable, active but ignored call waiting, etc. Other notification methods to the called party can also be offered, such as FAX or paging. In each case, the notification can include the caller's identity, when known.) Once the directory system is distributed it will be necessary to query the other copies if contact cannot be made based on local io information. This system provides the ability to have various forms of notification, and to control the parameters of those forms.
(4) Party Identification A critical question is how will the directory service know that a called party is is no longer where she was last reported (i.e., has "gone away"). The dialed in party might drop off the network in a variety of ways (dialed line dropped, PC hung, Terminal Server crashed) without the ability to explicitly inform the directory service of his change in status. Worse yet, the user might have left the network and another user with a voice application might 2o be assigned the same IP address. (This is OK if the new caller is a registered user with automatic presence notification; the directory service could then detect the duplicate IP address. There may still be some timing problems between distributed parts of the directory service.) Therefore, some scheme must exist for the directory service to determine that the 25 customer is still at the last announced location.
One approach to this is to implement a shared secret with the application, created at registration time. Whenever the directory system is contacted by the software (such as automatic presence notification or call initialization) 30 or attempts to contact the called party at the last known location, it can send a challenge (like CHAP) to the application and verify the response.
Such a scheme eliminates the need for announcing "I am no longer here", or wasteful keep alive messages. A customer can disconnect or turn off his IP phone application at any time without concern for notification to the ~' directory system. If multiple IP phone applications are supported, by the directory service, each may do the challenge differently.
(5) Other Services - s Encrypted Internet telephone conversations will require a consensus from the software vendors to minimize the number of encryption setup mechanisms. This will be another interoperability resolution function for the directory service. The directory service can provide support for public key applications and can provide public key certificates issued by suitable to certificate authorities.
The user can also specify on the directory service, that his PC be called (dial out) if she is not currently on-line. Charges for the dial out can be billed to the called party, just as would happen for call forwarding in POTS. The call is detail record (CDR) for the dial out needs to be associated with the call detail of an entity in the IP Phone system (the called party). Note that this is different than the PC to PSTN case in that no translation of IP encoded voice to PCM is required, indeed the dial out will use TCP/IP over PPP. If the dial out fails an appropriate message is sent back.
The dial out could be domestic or international. It is unlikely that the international case will exist in practice due to the cost. However, there is nothing to preclude that case and it requires no additional functionality to perform.
b) PC to PSTN
The PSTN to Internet gateway must support translating PCM to multiple encoding schemes to interact with software from various vendors.
Alternatively the common compression scheme could be used once it is 3o implemented. Where possible, the best scheme, from a quality stand point, . should be used. In many cases it will the software vendor's proprietary version. To accomplish that, telcos will need to license the technology from selected vendors. Some vendors will do the work needed to make their scheme work on telco platforms.
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(1) Domestic PSTN Destination The PC caller needs to be registered to place calls to the PSTN. The only exception to this would be if collect calls from the Internet are to be allowed. This will add complications with respect to billing. To call a PSTN
destination the PC caller specifies a domestic E.164 address. The directory system maps that address to an Internet dial out unit based on the NPA-NXX. The expectation is that the dial out unit will be close to the destination and therefore will be a local call. One problem is how to handle the case where there is no "local" dial out unit. Another problem is what to l0 do if the "local" out dial unit is full or otherwise not available.
Three approaches are possible. One approach is to offer the dial out service only when local calls are possible. A second approach is to send a message back to the caller to inform him that a long distance call must be placed on his behalf and request permission to incur these charges. A third approach t5 is to place the call regardless and with no notification. Each of these cases requires a way to correlate the cost of the dial out call (PSTN CDR) with the billing record of the call originator (via the directory service).
2o The third approach will probably add to the customer support load and result in unhappy customers. The first approach is simple but restrictive.
Most users are expected to be very cost conscious, and so might be satisfied with approach one. Approach two affords flexibility for the times the customer wants to proceed anyway, but it adds complexity to the 25 operation. A possible compromise is to use approach one, which will reject the call for the reason that no local out dial is available. We could also add an attribute in the call request that means "I don't care if this ends up as a long distance call." In this case the caller who was rejected, but wants to place the call anyway makes a second call attempt with this attribute set.
3o For customers with money to spare, all PSTN calls could be made with that attribute set.
Placing domestic PSTN calls supports the international calling requirement for Internet originated calls from Internet locations outside the US.
t6o (2) International PSTN Destinations Calls to an international PSTN station can be done in one of two ways.
First, an international call could be placed from a domestic dial out station.
s This is not an attractive service since it saves no money over the customer making an international telephone call himself. Second, the Internet can be used to carry the call to the destination country and a "local" dial out can be made there.
This situation is problematic for it must be agreed to by the carrier at the international destination. This case may be viable in one of two ways.
Both ways require a partner at the international destination. One option would be to use a local carrier in the destination country as the partner. A
second option would be to use an Internet service provider, or some other service provider connected to the Internet in the destination country.
is c) PSTN to PC
This case appears to be of least interest, although it has some application and is presented here for completeness.
2o As noted in the PC to PSTN case the PSTN to Internet gateway will need to support translating PCM to multiple encoding schemes to interwork with software from various vendors. The directory service is required to identify the called PC. Automatic notification of presence is important to keep the called party reachable. The PSTN caller need not be registered with the 2s directory service, for caller billing will be based on PSTN information.
The caller has an E.164 address that is "constant" and can be used to return . calls as well as to do billing. Presumably we can deliver the calling number to the called party as an indication of who is calling. The calling number will not always be available, for technological or privacy reasons. It must 3o be possible to signal the PC software that this is a PSTN call and provide the E.164 number or indicate that it is unavailable.
The service can be based on charging the calling phone. This can be done as if the Internet were the long distance portion of the call. This is possible 16( with a second dial tone. If an 800 or local dial service is used it is necessary for the caller to enter billing information. Alternatively a 900 service will allow PSTN caller-based billing. In either case the caller will need to specify the destination "phone number" after the billing information or after dialing the 900 number.
A major open issue is how the caller will specify the destination at the second dial tone. Only touch tones are available at best. To simplify entry we could assign an E.164 address to each directory entry. To avoid to confusion with real phone numbers (the PSTN to PSTN case) the numbers need to be under directory control. Perhaps 700 numbers could be used, if there are enough available. Alternatively a special area code could be used.
Spelling using the touch tone PAD is a less "user friendly" approach.
t5 3. Phone Numbers in the Internet.
The best approach is to have an area code assigned. Not only will this keep future options open, but it allows for simpler dialing from day one. Given a legitimate area code the PSTN caller can directly dial the E.164 address of the PC on the Internet. The telephone system will route the call to an MCI
2o POP where it will be further routed to a PSTN-to-Internet voice gateway.
The called number will be used to place the call to the PC, assuming it is on-line and reachable. This allows the PSTN caller to dial the Internet as if it were part of the PSTN. No second dial tone is required and no billing information needs to be entered. The call will be billed to the calling PSTN
25 station, and charges will accrue only if the destination PC answers. Other carriers would be assigned unique area codes and directories should be kept compatible.
For domestically originated calls, all of the billing information needed to bill 3o the caller is available and the intelligent network service functionality for third party or other billing methods is available via the second dial tone.
4. Other Internet Telephony Carriers.
All this will get more complicated when number portability becomes I (~ 2 required. It may be desirable to assign a country code to the Internet.
Although this would make domestic dialing more complex (it appears that dialing anything other than 1 plus a ten digit number significantly reduces the use of the service) it may have some desirable benefits. In any event s the assignment of an area code (or several) and the assignment of a country code are not mutually exclusive. The use of a country code would make dialing more geographically uniform.
5. International Access.
to It is unlikely that an international call will be made to the US to enter the Internet in the US. If it happens, however, the system will have enough information to do the caller-based billing for this case without any additional functionality.
15 Another possibility is that we will (possibly in partnership) set up to handle incoming calls outside the US and enter the Internet in that country to return to the US, or go anywhere else on the Internet. If the partner is a local carrier, then the partner will have the information needed for billing the PSTN caller.
a) Collect Calls PSTN to PC collect calls require several steps. First, the call to the PSTN to Internet gateway must be collect. The collect call could then be signaled in the same way as PC to PC calls. It will be necessary to indicate that the 2s caller is PSTN based and include the calling E.164 address if it is available.
b~ PSTN to PSTN
The choice of voice compression and protocol scheme for passing voice between PSTN to Internet gateways is entirely under the carrier's control.
3o Various service levels could be offered by varying the compression levels . offered. Different charges could associated with each level. The caller would select a quality level; perhaps by dialing different 800 number services first.
(G3 (1) Domestic Destination Neither the calling nor the called parties need be registered with the directory service to place calls across the Internet. The caller dials a PSTN-to-Internet gateway and receives a second dial tone and specifies, using s touch tones, the billing information and the destination domestic E.164 address. 900 service could be used as well. The directory service (this could be separate system, but the directory service already has mapping functionality to handle the PC to PSTN dial out case) will be used to map the call to an out dialer to place a local call, if possible. Billing is to the caller and the call detail of the out dial call needs to be associated with the call detail of the inbound caller.
An immediate question is how to deal with the case where the nearest dial out unit to the number called results in a long distance or toll call, as 15 discussed in PC to PSTN case. The situation here is different to the extent that notification must be by voice, and authorization to do a long distance, or toll call dial out must be made by touch tones. In the event of a long distance dial out the Internet could be skipped altogether and the call could go entirely over the PSTN. It is not clear that there is any cost savings by 2o using the Internet in this case.
(2) One Step Dialing The problem is that the destination PSTN number needs to be entered and, somehow, it needs to be indicated that the destination is to be reached via 25 the Internet rather than the conventional long distance network.
This selection criteria can be conveyed according to the following alternatives:
Assign a new lOXXX number that is the carrier's Internet.
By subscription.
3o The first method allows the caller to select the Internet as the long distance carrier on a call by call basis. The second method makes the Internet the default long distance network. In the second case a customer can return to the carrier's conventional long distance network by dialing the carrier's lOXXX code.
1 E ~-The first method has the draw back that the caller must dial an extra five digits. Although many will do this to save money, requiring any extra dialing will reduce the total number of users of the service. The second method avoids the need to dial extra digits, but requires a commitment by the subscriber to predominately use the Internet as his long distance network. The choice is a lower price with a lower quality of service.
In the PSTN to PSTN case it is possible to consider offering several grades of io service at varying prices. These grades will be based on a combination of the encoding scheme and the amount of compression (bandwidth) applied, and will offer lower cost for lower bandwidth utilization.
To signal the grade of service desired three lOXXX codes could be used. By is subscription a particular grade would be the default and other service grades would be selected by a lOXXX code.
(3) Service Quality The service quality will be measured by two major factors. First, sound 2o quality, the ability to recognize the caller's voice, and second by the delays that are not present in the PSTN.
On the first point we can say that most of the offerings available today provide an acceptable level of caller recognition. Delay, however, is another 25 story. PC to PC users experience delays of a half second to two seconds.
As noted in the introduction much of the delay can be attributed to the sound cards and the low speed dial access. In the case of PSTN to PSTN
service both these factors are removed.
3o The use of DSPs in the PSTN to Internet voice gateway will keep compression and protocol processing times very low. The access to the gateway will be at a full 64 kbps on the PSTN side and likely Ethernet on the Internet side. Gateways will typically be located close to the backbone so the router on the Ethernet will likely be connected to the backbone by a ~ ~.5 T3 line. This combination should provide a level of service with very Iow delays. Some buffering will be needed to mask the variable delays in the backbone, but that can likely be kept to under a quarter of a second in the domestic carrier backbone.
The main differentiation of quality of service will be voice recognition which will be related to bandwidth usage. If needed, the proposed IETF Resource reSerVation setup Protocol (RSVP) can be used to assure lower delay variation, but the need for the added complexity of RSVP is yet to be ~o established. Also, questions remain regarding the scalability of RSVP for large-scale Internet telephony.
(4} Costs An open question is whether using the Internet for long distance voice in place of the switched telephone network is actually cheaper. Certainly it is is priced that way today, but do current prices reflect real costs? Routers are certainly cheaper than telephone switches, and the 10 kbps (or so) that the IP voice software uses {essentially half duplex} is certainly less than the dedicated 128 kbps of a full duplex 64 kbps DSO. Despite these comparisons the question remains.
Although routers are much cheaper than telephone switches, they have much less capacity. Building large networks with small building blocks gets not only expensive, but quickly reaches points of diminishing returns.
We already have seen the Internet backbone get overloaded with the current crop of high end routers, and they are yet to experience the significant traffic increase that a successful Internet Telephony offering would bring. We are saying two things here.
1. It is unlikely that the current Internet backbone can support a major 3o traffic increase associated with a successful Internet telephony service.
We need to wait for the technology of routers to improve.
2. The second issue raised above was that of bandwidth usage. Indeed 10 kbps half duplex (a little more when both parties occasionally speak at the same time, but much less during periods of silence) is considerably less than 64 kbps full duplex dedicated capacity. Two points should be noted on this argument.
s First, bandwidth is cheap, at least, when there is spare fiber in the ground.
Once the last strand is used the next bit per second is very expensive.
Second, on transoceanic routes, where bandwidth is much more expensive, we are already doing bandwidth compression of voice to 9.6 kbps. This is essentially equivalent to the 10 kbps of Internet Telephony.
to Why is IP capacity priced so much cheaper than POTS? The answer is that the pricing difference is partly related to the subsidized history of the Internet. There is a process in motion today, by the Internet backbone providers, to address some of the cost issues of the Internet. The essence ~s of the process is the recognition that the Internet requires a usage charge.
Such charges already apply to some dial up users, but typically do not apply to users with dedicated connections.
If PC to PC Internet Telephony becomes popular, users will tend to keep 2o their PCs connected for long periods. This will make them available to receive calls. It will also drive up hold times on dial in ports. This will have a significant effect on the capital and recurring costs of the Internet.
(5) Charges 25 A directory service must provide the functions described above and collect enough information to bill for the service. A charge can be made for directory service as well as for registration (a one time fee plus a monthly fee), call setup, but probably not for duration. Duration is already charged for the Internet dial in user and is somewhat bundled for the LAN-attached 3o user. Usage charges for Internet service may be coming soon (as discussed . above). Duration charges are possible for the incoming and outgoing PSTN
segments.
Incoming PSTN calls may be charged as the long distance segment by using ~6~

a special area code. Other direct billing options are 900 calls and calling card (or credit card) billing options (both require a second dial tone).
Requiring all callers (except incoming PSTN calls) to be registered with the directory service will eliminate the immediate need for most collect calling.
This will probably not be a great impediment since most users of the IP
Phone service will want to receive as well as originate calls, and registration is required for receiving calls. Callers could have unlisted entries which would be entries with an E.164 address, but no name. People given this Io E.164 address could call the party (from the PSTN or from a PC), as is the case in the present phone system.
Different compression levels can be used to provide different quality of voice reproduction and at the same time use more or less Internet transit is resources. For PC to PC connections the software packages at both ends can negotiate the amount of bandwidth to be used. This negotiation might be facilitated through the directory service.
(6) Technical Issues 2o It will be necessary to coordinate with IP Phone vendors to implement the registration, automatic presence notification, and verification capabilities.
We will also need to add the ability to communicate service requests. These will include authorization for collect calls specifying attributes such as "place a dial out call to the PSTN even if it is long distance" and others to be 2s determined.
Registration with a directory is a required feature that will be illuminated below. Using the DNS model for the distributed directory service will likely facilitate this future requirement. Assignment of a pseudo E.164 number 3o to directory entries will work best if a real area code is used. If each carrier has an area code it will make interworking between the directory systems much easier. An obvious complication will arise when number portability becomes required.
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WO 98147298 PCT/US98l07927 IP Telephony, in accordance with a preferred embodiment, is here and will stay for at least the near future. A combination of a carrier level service, based on this technology, and a growth in the capacity of routers may lead to the Internet carrying a very significant percentage of future long distance traffic.
The availability of higher speed Internet access from homes, such as cable modems, will make good quality consumer IP Telephony service more easily attained. The addition of video will further advance the desirability of the to service.
More mundane, but of interest, is FAX services across the Internet. This is very similar to the voice service discussed above. Timing issues related to FAX protocols make this a more difficult offering in some ways.
Conferencing using digital bridges in the Internet make voice and video services even more attractive. This can be done by taking advantage of the multi-casting technology developed in the Internet world. With multi-casting the cost of providing such services will be reduced.
C. Internet Telephony Services Figure 1C is a black diagram of an Internet telephony system in accordance with a preferred embodiment. Processing commences when telephone 200 is utilized to initiate a call by going off hook when a party dials a telephone 2s number. Telephone 200 is typically connected via a conventional two-wire subscriber loop through which analog voice signals are conducted in both directions. One of ordinary skill in the art will readily realize that a phone can be connected via fiber, ISDN or other means without departing from the teaching of the invention. Alternatively, a person could dial a phone 3o number from a computer 210, paging system, video conferencing system or other telephony capable devices. The call enters a Local Exchange Carrier (LEC) 220 which is another name for a Regional Bell Operating Company (RBOC) central switch. The call is terminated by a LEC at a leased Common Business Line (CBL) 230 of an interchange carrier such as MCI.
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As a result of the termination to the CBL, the MCI Switch 221 receives an offhook indication.
The Switch 221 responds to the offhook by initiating a DAL Hotline s procedure request to the Network Control System (NCS) which is also referred to as a Data Access Point (DAP) 240. The switch 221 is simplified to show it operating on a single DS 1 line, but it will be understood that switching among many lines actually occurs so that calls on thousands of individual subscriber lines can be routed through the switch on their way to to ultimate destinations. The DAP 240 returns a routing response to the originating switch 221 which instructs the originating switch 221 to route the call to the destination switch 230 or 231. The routing of the call is performed by the DAP 240 translating the transaction information into a specific SWitch ID (SWID) and a specific Terminating Trunk Group (TTG) ~s that corresponds to the route out of the MCI network necessary to arrive at the appropriate destination, in this case either switch 230 or 231. An alternative embodiment of the hybrid network access incorporates the Internet access facility into a switch 232. This integrated solution allows the switch 232 to attach directly to the Internet 295 which reduces the 2o number of network ports necessary to connect the network to the Internet 295. The DAP sends this response information to the originating switch 221 which routes the original call to the correct Terminating Switch 230 or 231. The terminating switch 230 or 231then finds the correct Terminating Trunk Group (TTG) as indicated in the original DAP response and routes zs the call to the ISN 250 or directly to the modem pool 2?O based on the routing information from the DAP 240. If the call were destined for the Intelligent Services Network (ISN) 250, the DAP 240 would instruct the switch to terminate at switch 230.
3o Based upon analysis of the dialed digits, the ISN routes the call to an Audio Response Unit (ARU) 252. The ARU 252 differentiates voice, fax, and modem calls. If the call is a from a modem, then the call is routed to a modem poal 2?1 for interfacing to an authentication server 291to authenticate the user. If the call is authenticated, then the call is ! ~ c'7 forwarded through the UDP/IP or TCP/IP LAN 281 or other media communication network to the Basic Internet Protocol Platform (BIPP) 295 for further processing and ultimate delivery to a computer or other media capable device.
s If the call is voice, then the ARU prompts the caller for a card number and a terminating number. The card number is validated using a card validation database. Assuming the card number is valid, then if the terminating number is in the US (domestic), then the call would be routed over the current MCI voice lines as it is today. If the terminating number is international, then the call is routed to a CODEC 260 that converts the voice to TCP/IP or UDP/IP and sends it via the LAN 280 to the Internet 295. The call is routed through a gateway at the terminating end and ultimately to a phone or other telephony capable device.
is Figure 1D is a block diagram of a hybrid switch in accordance with a preferred embodiment. Reference numbers have been conserved from Figure 1C, and an additional block 233 has been added. Block 233 contains the connecting apparatus for attaching the switch directly to the 2o Internet or other communication means. The details of the connecting apparatus are presented in Figure IE. The principal difference between the hybrid switch of Figure 1D and the switches presented in Figure IC is the capability of switch 221 attaching directly to the Internet 295.
25 Figure lE is a block diagram of the connecting apparatus 233 illustrated in Figure 1D in accordance with a preferred embodiment. A message bus 234 connects the switch fabric to an internal network 236 and 237. The internal network in turn receives input from a Dynamic Telephony Connection (DTC) 238 and 239 which in turn provides demuxing for 3o signals originating from a plurality of DS 1 lines 242, 243, 244 and 245.
DS 1 lines, described previously, refer to the conventional bit format on the T 1 lines.
To accommodate the rapidly diversifying telephony / media environment, a preferred embodiment utilizes a separate switch connection for the other internal network 23?. A Spectrum Peripheral Module (SPM) 24? is utilized to handle telephony/ media signals received from a pooled switch matrix 248, 249, 251, 254, 261-268. The pooled switch matrix is managed by the SPM 24? through switch commands through control lines. The SPM
247 is in communication with the service provider's call processing system which determines which of the lines require which type of hybrid switch processing. For example, fax transmissions generate a tone which identifies the transmission as digital data rather than digitized voice. Upon 1o detecting a digital data transmission, the call processing system directs the call circuitry to allow the particular input Iine to connect through the pooled switch matrix to a corresponding line with the appropriate processing characteristics. Thus, for example, an Internet connection would be connected to a TCP/IP Modem line 268 to assure proper t5 processing of the signal before it was passed on through the internal network 237 through the message bus 234 to the originating switch 221 of Figure 1D.
Besides facilitating direct connection of a switch to the Internet, the pooled 2o switch matrix also increases the flexibility of the switch for accommodating current communication protocols and future communication protocols.
Echo cancellation means 261 is efficiently architected into the switch in a manner which permits echo cancellation on an as-needed basis. A
relatively small number of echo cancellers can effectively service a relatively 25 large number of individual transmission lines. The pooled switch matrix can be configured to dynamically route either access-side transmissions or network-side transmissions to OC3 demux, DSP processing or other specialized processing emanating from either direction of the switch.
3o Moreover, a preferred embodiment as shown in Figure lE provides additional system efficiencies such as combining multiplexer stages in a port device on one side of a voice or data circuit switch to enable direct connection of a fiber-optic cable to the multiplexed output of the port device. Moreover, redundancy is architected into the switch through the ~~z alternate routes available over CEM 248 / 249 and RM 251 / 254 to alternate paths for attaching various communication ports.
When the switch 221 of Figure 1D, is connected to the Internet 295, the - 5 processing is provided as follows. A line from the Internet 295 enters the switch through a modem port 268 and enters the pooled switch matrix where demux and other necessary operations are performed before the information is passed to the switch 221 through the internal network 237 and the message bus 234. The modules 261-268 provide plug and play capability for attaching peripherals from various communication disciplines.
Figure 1F is a block diagram of a hybrid (Internet-telephony) switch in accordance with a preferred embodiment. The hybrid switch 221 switches is circuits on a public switched telephone network (PSTN) 256 with TCP/IP
or UDP/ IP ports on an Internet network 295. The hybrid switch 221 is composed of PSTN network interfaces (24?, 260}, high-speed Internet network interfaces (271, 2?2, 2T4), a set of Digital Signal Processor (DSP)s (259, 263), a time-division multiplexed bus 262, and a high-speed data 2o bus 275.
The hybrid Internet telephony switch 221 grows out of the marriage of router architectures with circuit switching architectures. A call arriving on the PSTN interface 25? is initiated using ISDN User Part (ISUP) signaling, 25 with an Initial Address Message (IAM), containing a called party number and optional calling party number. The PSTN interface 257 transfers the IAM to the host processor 2?O. The host processor 2?0 examines the PSTN
network interface of origin, the called party number and other IAM
- parameters, and selects an outgoing network interface for the call. The 3o selection of the outgoing network interface is made on the basis of routing tables. The switch 221 may also query an external Service Control Point (SCP) 2?6 on the Internet to request routing instructions. Routing instructions, whether derived locally on the switch 221 or derived from the SCP 276, may be defined in terms of a subnet to use to reach a particular l~3 destination.
Like a router, each of the network interfaces in the switch 221 is labeled with a subnet address. Internet Protocol (IP) addresses contain the subnet address on which the computer is located. PSTN addresses do not contain IP subnet addresses, so subnets are mapped to PSTN area codes and exchanges. The switch 221 selects routes to IP addresses and PSTN
addresses by selecting an interface to a subnet which will take the packets closer to the destination subnet or local switch.
The call can egress the switch via another PSTN interface 258, or can egress the switch via a high-speed Internet network interface 2?3, If the call egresses the switch via the PSTN interface 258, the call can egress as a standard PCM Audio call, or can egress the switch as a modem call 1s carrying compressed digital audio.
In the case where the call egresses the switch 221 as a standard PCM
audio call, the PCM audio is switched from PSTN Interface 257 to PSTN
Interface 258 using the TDM bus 260. Similarly, PCM audio is switched 2o from PSTN Interface 258 to PSTN Interface 25? using the TDM bus 260.
In the case where the call egresses the switch 221 as a modem call carrying compressed digital audio, the switch 221 can initiate an outbound call to a PSTN number through a PSTN interface 258, and attach across the TDM
zs Bus 260 a DSP resource 259 acting as a modem. Once a modem session is established with the destination, the incoming PCM audio on PSTN
interface 25? can be attached to a DSP Resource 263 acting as an audio codec to compress the audio. Example audio formats include ITU 6.729 and 6.723. The compressed audio is packetized into Point to Point Protocol 30 (PPP) packets on the DSP 263, and transferred to DSP 259 for modem delivery over the PSTN Interface 258.
In the case where the call egresses the switch 221 on a high speed Internet interface 2?2, the switch 221 attaches the PSTN Interface 25? to the DSP
! ~'-~

resource 263 acting as an audio codec to compress the PCM audio, and packetize the audio into UDP/IP packets for transmission over the Internet network. The UDP/IP packets are transferred from the DSP resource 263 over the high-speed data bus 2?5 to the high-speed Internet network s interface 272.
Figure 1G is a block diagram showing the software processes involved in the hybrid Internet telephony switch 221. Packets received on the Internet network interface 296 are transferred to the packet classifier 293. The to packet classifier 293 determines whether the packet is a normal IP packet, or is part of a routing protocol (ARP, RARP, RIP, OSPF, BGP, CIDR) or management protocol (ICMP). Routing and management protocol packets are handed off to the Routing Daemon 294. The Routing Daemon 294 maintains routing tables for the use of the packet classifier 293 and packet i5 scheduler 298. Packets classified as normal IP packets are transferred either to the packetizer/depacketizer 292 or to the packet scheduler 298.
Packets to be converted to PCM audio are transferred to the packetizer/ depacketizer 292. The packetizer/ depacketizer takes packet contents and hands them to the codec 291, which converts compressed 2o audio into PCM Audio, then transfers PCM audio to the PSTN Interface 290.
Normal IP packets to be sent to other Internet devices are handed by the packet classifier 293 to the packet scheduler 298, which selects the 25 outgoing network interface for the packet based on the routing tables. The packets are placed upon an outbound packet queue for the selected outgoing network interface, and the packets are transferred to the high speed network interface 296 for deliver across the Internet 295.
D. Call 3o This section describes how calls are processed in the context of the networks described above.
1. VNET Call Processing.
Figure lOA illustrates a Public Switched Network (PSTN) 1000 comprising a local exchange (LEC) 1020 through which a calling party uses a I ~ .~

telephone 1021 or computer 1030 to gain access to a switched network including a plurality of MCI switches 1011, 1010. Directory services for routing telephone calls and other information is provided by the directory services 1031 which is shared between the Public Branch Exchanges s 1041, 1040 and the PSTN.
This set of scenarios allows a subscriber to use either a PC, telephone or both to make or receive VNET calls. In this service, the subscriber may have the following equipment:
A telephone that uses VNET routing is available today in MCI's network. In 1o this case, VNET calls arriving in the MCI PSTN network using the subscriber's VNET number are routed with the assistance of the DAP just as they are routed today.
A PC that is capable of Internet telephony. Calls are routed into and out of this PC with the assistance of an Internet or Intranet Directory Service that 15 tracks the logged-in status and current IP address of the VNET user.
A PC and a telephone is used to receive and make calls. In this case, a user profile will contain information that allows the DAP and Directory Service to make a determination whether to send an incoming call to the PC or to the telephone. For example, the user may always want calls to go 2o to their PC when they are logged-in and to their phone at all other times.
Or, they may want their calls to always go to their PC during normal work hours and to their phone at other times. This type of control over the decision to send incoming calls to a phone or PC may be controlled by the subscriber.
The following scenarios apply to this type of service.
A PC to PC call where the Directory service is queried for the location of the terminating PC:
PCs connected to an Intranet using the Intranet as transport.
3o Both PC's connected to a corporate Intranet via dial up access.
Both PCs on separate Intranets with the connection made through the Internet.
Both PCs on the Internet through a dial-up connection.
One PC directly connected to a corporate Intranet and the other PC using a yb dial-up connection to the Internet.
One PC using a dial up connection to a corporate Intranet and the other PC
using a dial-up connection to the Internet.
Both PCs on separate Intranets with the connection made through the PSTN.
One or both PCs connected to a corporate Intranet using dial-up access.
One or both of the PCs connected to an Internet Service Provider.
One or both of the ITGs as an in-network element.

to A PC to phone call where a directory service is queried to determine that the terminating VNET is a phone. The PC then contacts an Internet Telephony Gateway to place a call to the terminating phone.
PC on an intranet using a private ITG connected to the PSTN with the ITG
as an out of network element. The destination phone is connected to a PBX.
~s The PC may also be using a public ITG that must be access through the Internet.
The PC may be connected to the corporate Intranet using dial-up access.
PC on an intranet using a private ITG connected to the PSTN with the ITG
as an in-network element. The destination phone is connected to a PBX.
20 .The PC may also be using a public ITG that must be accessed through the Internet.
.The PC may be connected to the corporate Intranet using dial-up access.
.PC on an intranet using a private ITG connected to the PSTN with the ITG as an in-network element. The destination phone is connected to the 2s PSTN.
.The PC may also be using a public ITG that must be accessed through the Internet.
.The PC may be connected to the corporate Intranet using dial-up access.
.The ITG may be an in-network element.
30 .PC on an intranet using a private ITG connected to a PBX with the traffic carried over the Intranet.
.PC is at a different site than the destination phone with the traffic carried over the Internet or intranet.
.The PC may be using a dial-up connection to the corporate Intranet.
l~~

A phone to PC call where the DAP or PBX triggers out to the Internet Directory Service to identify the terminating IP address and ITG for routing the call. The call is then routed through the PSTN to an ITG and a s connection is made from the iTG to the destination PC.
Possible Variations:
Same variations as the PC to phone.
A Phone to Phone call where the DAP or PBX must query the Directory Service to determine whether the call should be terminated to the subscriber's phone or PC.
Possible Variations:
Both Phones are on a PBX;
One phone is on a PBX and the other phone is on the PSTN; and is Both phones are on the PSTN.
For each of these variations, the DAP and Directory Service may be a single entity or they may be separate entities. Also, the directory service may be a private service or it may be a shared service. Each of the scenarios will be discussed below with reference to a call flow description in accordance with 2o a preferred embodiment. A description of the block elements associated with each of the call flow diagrams is presented below to assist in understanding the embodiments.
2. Descriptions of Block Elements.
Element Description 25 Ph 1 Traditional analog phone connected to a Local Exchange Carrier.
For the purposes of these VNET scenarios, the phone is capable of making VNET calls, local calls or DDD calls. In some scenarios the VNET
access may be done through The customer dials a 700 number with the last seven digits being the destination VNET number for the call.
3o The LEC will know that the phone is picked to MCI and route the call to the MCI switch. The MCI switch will strip off the "700", perform and ANI
lookup to identify the customer ID and perform VNET routing using the VNET number and customer ID. The customer dials an 800 number and is prompted to enter their Social Security number (or other i~-~

unique id) and a VNET number. The switch passes this information to the DAP which does the VNET translation.
PC 1 PC2 Personal computer that has the capability to dial in to an Internet service provider or a corporate intranet for the purpose of s making or receiving Internet telephony calls. The following access methods might be used for this PC Internet service provider The PC
dials an 800 number (or any other dial plan) associated with the service provider and is routed via normal routing to the modem bank for that provider. The user of the PC then follows normal log-on procedures to connect to the Internet. Corporate Intranet The PC dials an 800 number (or any other dial plan) associated with the corporate Intranet and is routed via normal routing to the modem bank for that Intranet.
The user of the PC then follows normal log-on procedures to connect to the Intranet.
1s LEC SF1 Switching fabric for a local exchange carrier. This fabric provides the connection between Ph 1 / PC 1 / PC2 and MCI's telephone network. It also provides local access to customer PBXs.
MCI SF1 MCI SF2 Switching fabric for MCI (or for the purpose of patenting, any telephony service provider). These SFs are capable of 2o performing traditional switching capabilities for MCI's network. They are able to make use of advanced routing capabilities such as those found in MCI's NCS (Network Control System).
NCS The NCS provides enhanced routing services for MCI. Some of the products that are supported on this platform are: 800, EVS, Universal 2s Freephone, Plus Freephone, Inbound International, SAC(ISAC) Codes, Paid 800, 8XX/Vnet Meet Me Conference Call, 900, 700, PCS, Vnet, Remote Access to Vnet, Vnet Phone Home, CVNS, Vnet Card, MCI Card (950 Cards), Credit Card and GETS Card. In support of the existing VNET services, the DAP provides private dialing plan capabilities to Vnet 3o customers to give them a virtual private network. The DAP supports digit translation, origination screening, supplemental code screening, 800 remote access, and some special features such as network call redirect for this service. To support the call scenarios in this document, the NCS also has the capability to made a data query to directory services in order to route calls to PCs.
Dir Svc 1 Dir Svc 2 Internet Directory Services. The directory service performs: Call routing - As calls are made to subscribers using Internet telephony services from MCI, the directory service must be queried to determine where the call should terminate. This may be done based upon factors such as the logged-in status of the subscriber, service subscriptions identifying the subscriber as a PC or phone only user preferred routing choices such as "route to my PC always if I am logged in", or "route to my PC from 8-5 on weekdays, phone all to other times", etc. Customer profile management - The directory service must maintain a profile for each subscriber to be able to match VNET
numbers to the service subscription and current state of subscribers.
Service authorization - As subscribers connect their PCs to an IP
telephony service, they must be authorized for use of the service and t5 may be given security tokens or encryption keys to ensure access to the service. This authorization responsibility might also place restrictions upon the types of service a user might be able to access, or introduce range privileges restricting the ability of the subscriber to place certain types of calls.
2o ITG 1 ITG 2 Internet Telephony Gateway - The Internet Telephony Gateway provides a path through which voice calls made be bridged between an IP network and a traditional telephone network. To make voice calls from an IP network to the PSTN, a PC software package is used to establish a connection with the ITG and request that the ITG dial 25 out on the PSTN on behalf of the PC user. Once the ITG makes the connection through the voice network to the destination number, the ITG provides services to convert the IP packetized voice from the PC to voice over the PSTN. Similarly, the ITG will take the voice from the PSTN and convert it to IP packetized voice for the PC. To make voice 3o calls from the PSTN to the IP network, a call will be routed to the ITG via PSTN routing mechanisms. Once the call arrives, the ITG identifies the IP address for the destination of the call, and establishes an IP
telephony session with that destination. Once the connection has been established, the ITG provides conversion services between IP packetized I~~

voice and PCM voice.
ITG 3 ITG 4 These ITGs act in a similar capacity as the ITGs connected to the PSTN, but these ITGs also provide a connection between the corporate Intranet and the PBX.
s IAD 1 IAD 2 The Internet access device provides general dial-up Internet access from a user's PC to the Internet. This method of connecting to the Internet may be used for Internet telephony, but it may also be simply used for Internet access. When this device is used for Internet telephony, it behaves differently than the ITG. Although the to IAD is connected to the PSTN, the information traveling over that interface is not PCM voice, it is IP data packets. In the case of telephony over the IAD, the IP data packets happen to be voice packets, but the IAD has no visibility into those packets and cannot distinguish a voice packet from a data packet. The IAD can be thought of as a modem pool that provides access to the Internet.
PBX 1 PBX 2 Private Brach Exchange - This is customer premise equipment that provides connection between phones that are geographically co-located. The PBX also provides a method from those phones to make outgoing calls from the site onto he PSTN. Most PBXs 2o have connections to the LEC for local calls, and a DAL connection to another service provider for VNET type calls. These PBXs also show a connection to a Directory Service for assistance with call routing. This capability does not exist in today's PBXs, but in the VNET call flows for this document, a possible interaction between the PBX and the Directory 2s Service is shown. These PBXs also show a connection to an ITG. These ITGs provide the bridging service between a customer's Intranet and the traditional voice capabilities of the PBX.
Ph 11 Ph 12 Ph21 Ph22 These are traditional PBX connected phones.
3o PC 11 PC 12 PC21 PC22 These are customer premises PCs that are connected to customer Intranets. For the purposes of these call flows, the PCs have Internet Telephony software that allow the user to make or receive calls.
(gl E. Re-ustacble Call Flow Blocks 1. VNET PC connects to a corporate intranet and logs in to a directory seance.
Direcmry Services VNET. ~PauwordIP
1 ) PC Online Authenticate user VNET PC connects U date Profile with IP to corporate lntranet 2) PC Online Adc and Config data ' Optional data depending upon implementation I. The user for a PC connects their computer to an IP network, turns on the computer and starts an IP telephony software package. The software package sends a message to a directory service to register the computer as "on-line" and available to to receive calls. This on-line registration message would most likely be sent to the directory service in an encrypted format for security. The encryption would be based upon an common key shared between the PC and the directory service. This message contains the following information:
Some sort of identification of the computer or virtual private network number that may be used to address this computer. In this VNET scenario, this is the VNET number assigned to the individual using this PC. This information will be used to identify the customer profile associated with 2o this user. It may also be some identification such as name, employee id, or any unique ID which the directory service can associate with a VNE?' customer profile.
A password or some other mechanism for authenticating the user identified by the VNET number.
zs The IP address identifying the port that is being used to connect this computer to the network. This address will be used by other IP telephony software packages to establish a connection to this computer.
The message may contain additional information about the 1~2 specifics of the software package or PC being used for IP
telephony and the configuration/ capabilities of the software or PC. As an example it might be important for a calling PC
to know what type of compression algorithms are being used, . s or other capabilities of the software or hardware that might affect the ability of other users to connect to them or use special features during a connection.
The location of the directory service to receive this "on-line"
message will be determined by the data distribution 1o implementation for this customer. In some cases this may be a private database for a company or organization subscribing to a VNET service, in other cases it might be a national or worldwide database for all customers of a service provider (MCI). This location is configured in the telephony software package running t5 on the PC.
2. When the directory service receives this message from the PC, it validates the user by using the VNET number to look up a user profile and comparing the password in the profile to the password received. Once the user has been validated, the 2o directory service will update the profile entry associated with the VNET number (or other unique ID) to indicate that the user is "on-line" and is located at the specified IP address. The directory service will also update the profile with the configuration data sent during the login request. Upon successful update of the, the 25 directory service sends a response back to the specified IP
address indicating that the message was received and processed.
This acknowledgment message may also contain some sort of security or encryption key to guarantee secure communication with the directory service when issuing additional commands.
3o When the PC receives this response message it may choose to notify the user via a visual or audible indicator.
Variation for On-line registration The call flow segment shown earlier in this section showed a PC on-line registration where the PC simply sends a password to the directory service to log-on. A variation for this log-on procedure would be the following call flow segment where the directory service presents a challenge and the PC user must respond to the challenge to complete the log-in sequence. This variation on the log-in sequence is not shown in any of the call flows contained within this document, but it could be used in any of them.
PC Directory Services VNET.IP.
t)PC Online Calculate Challenge 2) Directory Service Challenge Calculate Resoors:e 3) Challenge Raponx ' Authenticate user update Profile with IP
4) PC Online Ack and Config data Optional data depending upon implementation 1. The user for a PC connects their computer to an IP network, turns on the computer and starts an IP telephony software package. The software package sends a message to a directory service to register the computer as "on-Iine" and available to 2o receive calls. This on-line registration message would most likely be sent to the directory service in an encrypted format for security. The encryption would be based upon an common key shared between the PC and the directory service. This message contains the following information:
Some sort of identification of the computer or virtual private network number that may be used to address this computer. In this VNET scenario, this is the VNET number assigned to the individual using this PC. This information will be used to identify the customer profile associated with 3o this user. It may also be some identification such as name, employee id, or any unique ID which the directory service can associate with a VNET customer prof le.
The IP address identifying the port that is being used to connect this computer to the network. This address will be used by other IP telephony software packages to establish a connection to this computer.
The message may contain additional information about the specifics of the software package or PC being used for IP
- s telephony and the configuration/capabilities of the software or PC. As an example it might be important for a calling PC
to know what type of compression algorithms are being used, or other capabilities of the software or hardware that might affect the ability of other users to connect to them or use special features during a connection.
The location of the directory service to receive this "on-Line"
message will be determined by the data distribution.
implementation for this customer. In some cases this may be a private database for a company or organization subscribing to a is VNET service, in other cases it might be a national or worldwide database for all customers of a service provider (MCI). This location is configured in the telephony software package running on the PC.
2. In this scenario the PC did not provide a password in the initial 2o registration message. This is because the directory service uses a challenge/response registration process. In this case, the directory service will use a shared key to calculate a challenge that will be presented to the PC
3. The PC receives this challenge and presents it to the user of the 25 PC. The PC user uses the shared key to calculate a response to the challenge and send the response back to the directory service.
4. When the directory service receives this response from the PC, it validates the user. Once the user has been validated, the 3o directory service will update the profile entry associated with the VNET number (or other unique ID) to indicate that the user is "on-line" and is located at the specified IP address. The directory service will also update the profile with the configuration data sent during the login request. Upon successful update of the, the ( gs directory service sends a response back to the specified IP
address indicating that the message was received and processed.
This acknowledgment message may also contain some sort of security or encryption key to guarantee secure communication with the directory service when issuing additional commands.
When the PC receives this response message it may choose to notify the user via a visual or audible indicator.
2. VNET PC queries a directory service for a VNET
~o translation PC Directory Scrvice Source VNET, IP, pest VNET'Co~g Data t ) VNET Transta4on Req !Match ViJET to profile Determrtte route tl~.'E'f PC trutslation 1 s 2) VNET Transtauon Ren, 'Check configuration IP. 'Config Dau or 1P. Disted Number * Optional data depending upon implementation 1. A PC uses an Internet telephony software package to attempt to connect to a VNET number. To establish this connection, the user of the PC dials theVNET number (or other unique ID such as name, employee ID, etc). Once the telephony software package has identified this call as a VNET
type call , it will send a translation request to the directory service. At a minimum, this translation request will contain the following information:
The IP address of the computer sending this request The VNET number of the PC sending this request.
The Vnet number (or other ID) of the computer to be dialed.
3o A requested configuration for the connection. For example, the calling PC might want to use white-board capabilities within the telephony software package and may wish to verify this capability on the destination PC before establishing a connection. If the VNET number does not translate to a PC, i ~' b this configuration information may not provide any benefit, but at the time of sending this request the user does not know whether the VNET number will translate to a PC or phone.
2. When the directory service receives this message, it uses the - 5 Vnet number (or other ID) to determine if the user associated with that VNET number (or other ID) is "on-line" and to identify the IP address of the location where the computer may be contacted. This directory service may also contain and make use of features like time of day routing, day of week routing, ANI screening, etc.
io If the VNET number translates into a PC that is "on-line", the directory service will compare the configuration information in this request to the configuration information available in the profile for the destination PC. When the directory service returns the response to the translation request from the originating PC, the response will include 15 The registered "on-line" IP address of the destination PC. This is the IP address that the originating PC may use to contact the destination PC
Configuration information indicating the capabilities of the destination PC and maybe some information about which 2o capabilities are compatible between the origination and destination PC.
If the VNET number translates to a number that must be dialed through the PSTN, the response message to the PC will contain the 25 following An IP address of an Internet Telephony gateway that may be used to get this call onto MCI's PSTN. The selection of this gateway may be based upon a number of selection algorithms. This association between the caller and the ITG to be used is made 3o based upon information in the profile contained within the directory service.
A VNET number to be dialed by the ITG to contact the destination phone. In the case of this call flow this is the VNET number of the destination phone. This allows the call to use the existing l8~

VNET translation and routing mechanisms provided by the DAP.
If the VNET number translates to a phone which is reachable through a private ITG connected to the customer's PBX, the directory service will return the following.
The VNET number of an ITG gateway that is connected to the PBX
serving the destination phone. This association between the destination phone the ITG connected to its serving PBX is made by the directory service.
1o The VNET number to be dialed by the ITG when it offers the call to the PBX. In most cases this will just be an extension number.
3. PC connects to an ITG.
Source Internet PC Telephony Gateway 1) ~ Telephony Dial 2) Call ack 3) ~ Telephony Answer 4)Voice path established 1. A PC uses its telephony software package to send a "connection"
message to an ITG. This IP address is usually returned from the directory service in response to a VNET translation. The specific format and contents of this message is dependent upon the software sending 2s the message or the ITG software to receive the message. This message may contain information identifying the user of the PC or it may contain information specifying the parameters associated with the requested connection.
2. The ITG responds to the connect message by responding to the 3o message with an acknowledgment that a call has been received. This step of call setup may not be necessary for a PC calling an iTG, but it is shown here in an attempt to maintain a consistent call setup procedure that is independent of whether the PC is connecting to an ITG or to another PC. When connecting to a PC, this step of the procedure allows the calling PC to know that the destination PC is ringing.
3. The ITG accepts the call.
4. A voice path is established between the ITG and the PC.
4. ITG connects to a PC.
lntemet Desunauon i Telephony pC
Gateway OITer call 1 ) ~ Telephony Dial 1 O 2) Call Ack Call accept 3) fP Telephony Answer 4)Voice path established 1. An ITG uses its telephony software to send a "connection" message to a PC. The ITG must know the IP address of the PC to which it is connecting. The specific format and contents of this message is ' dependent upon the ITG software sending the message or the PC
software to receive the message. This message may contain information identifying this call as one being offered from an ITG, or it may contain information specifying the requested configuration for the 2o call (i.e. voice only call).
2. The message from step 1 is received by the PC and the receipt of this message is acknowledged by sending a message back to the ITG
indicating that the PC is offering the call to the user of the PC
3. The user of the PC answers to call and a message is sent back to the originating PC indicating that the call has been accepted.
4. A voice path is established between the ITG and the PC.
. 5. VNET PC to PC Call Flow Description.
The user for PC 12 1051 connects the computer to an Internet Protocol (IP) 3o network I07I, turns on the computer and starts an IP telephony software protocol system. The system software transmits a message to a directory service 1031 to register the computer as "on-line" and available to receive calls. This message contains IP address identifying the connection that is being used to connect this computer to the network. This address may be f~9 used by other IP telephony software packages to establish a connection to this computer. The address comprises an identification of the computer or virtual private network number that may be used to address this computer 1051. In this VNET scenario, the address is a VNET number assigned to s the individual using this PC. VNET refers to a virtual network in which a particular set of telephone numbers is supported as a private network of numbers that can exchange calls. Many corporations currently buy communication time on a trunk that is utilized as a private communication channel for placing and receiving inter-company calls. The address may to also be some identification such as name, employee id, or any other unique ID.
The message may contain additional information regarding the specifics of the system software or the hardware configuration of PC I 1 1051 utilized ~5 for IP telephony. As an example, it is important for a calling PC to know what type of compression algorithms are supported and active in the current communication, or other capabilities of the software or hardware that might affect the ability of other users to connect or use special feature during a connection.
20 6. Determining best choice for Internet client selection of an Internet Telephony Gateway server on the Internet:.
Figure 10B illustrates an Internet routing network in accordance with a preferred embodiment. If a client computer 1080 on the Internet needs to 2s connect to an Internet Telephony Gateway 1084, the ideal choice for an Gateway to select can fall into two categories, depending on the needs of the client:
If the client 1080 needs to place a telephone call to a regular PSTN phone, 3o and PSTN network usage is determined to be less expensive or higher quality than Internet network usage, it is the preferred choice to select a gateway that allows the client to access the PSTN network from a point "closest" to the point of Internet access. This is often referred to as Head-End Hop-Off (HEHO), where the client hops off the Internet at the "head i qc~

end" or "near end" of the Internet.
If the client 1080 needs to place a telephone call to a regular PSTN phone, and the PSTN network is determined to be more expensive than Internet s network usage, it is the preferred choice to select a gateway that allows the client to access the PSTN from the Internet at a point closest to the destination telephone. This is often referred to as Tail-End Hop-Off (TEHO), where the client hops off the Internet at the "tail end" or "far end" of the Internet.
to a) Head-End Hop-Off Methods ( 1 ) Client Ping Method This method selects the best choice for a head-end hop-off Internet telephony gateway by obtaining a list of candidate Internet telephony gateway addresses, and pinging each to determine the best choice in terms ~s of latency and number of router hops. The process is as follows:
Client Computer 1080 queries a directory service 1082 to obtain a list of candidate Internet telephony gateways.
The directory service 1082 looks in a database of gateways and selects a list of gateways to offer the client as candidates. Criteria for selecting zo gateways as candidates can include last gateway selected.
matching 1, 2, or 3 octets in the IPv4 address.
last client access point, if known.
selection of at least one gateway from all major gateway sites, if practical.
25 The directory service 1082 returns a list of "n" candidate IP addresses to the client 1080 in a TCP/IP message.
The client 1080 simultaneously uses the IP ping to send an echo-type message to each candidate Internet Telephony Gateway, 1084, 1081, 1086. The "-r" option will be used with the ping command to obtain a trace 3o route.
Based upon the ping results for each Internet Telephony Gateway, the client 1080 will rank order the ping results as follows:
If any Internet Telephony Gateways are accessible to the client 1080 without traveling through any intervening router as revealed by the ping ~ 9r trace route, they are ranked first.
The remaining Internet Telephony Gateways are ranked in order of lowest latency of round-trip ping results.
s Using the Client Ping Method with the Sample Network Topology above, the Client Computer 1080 queries the Directory Service 1082 for a list of Internet Telephony Gateways to ping. The Directory Service 1082 returns the list:
166.37.61.117 166.25.27. i01 i 66.37.27.205 The Client Computer 1080 issues the following three commands simultaneously:
i 5 ping 166.37.61.117 -r 1 ping 166.25.27.101 -r 1 ping 166.37.27.205 -r 1 The results of the ping commands are as follows:
Pinging 166.37.61.117 with 32 bytes of data:
Reply from 166.37.61.117: bytes=32 time=3ms TTL=30 Route: 166.37.61.101 Reply from 166.37.61.117: bytes=32 time=2ms TTL=30 Route: 166.37.61.101 Reply from 166.37.61.117: bytes=32 time=2ms TTL=31 Route: 166.37.61.101 3o Reply from 166.37.61.117: bytes=32 time=2ms TTL=30 Route: 166.37.61.101 Pinging 166.25.27.101 with 32 bytes of data:
«z Reply from 166.25.27.101: bytes=32 time= l4ms TTL=30 Route: 166.37.61.101 Reply from 166.25.27.101: bytes=32 time=2ms TTL=30 - 5 Route: 166.37.61.101 Reply from 166.25.27.101: bytes=32 time=3ms TTL=31 Route: 166.37.61.101 Reply from 166.25.27.101: bytes=32 time=4ms TTL=30 Route: 166.37.61.101 Pinging 166.37.27.205 with 32 bytes of data:
Reply from 166.37.27.205: bytes=32 time=1 ms TTL=126 1s Route: 166.37.27.205 Reply from 166.37.27.205: bytes=32 time=lms TTL=126 Route: 166.37. 27.205 Reply from 166.37. 27.205: bytes=32 time=lms TTL=126 Route: 166.37. 27.205 2o Reply from 166.37. 27.205: bytes=32 time= lms TTL=126 Route: 166.37. 27.205 Since the route taken to 166.37.27.205 went through no routers (route and 2s ping addresses are the same), this address is ranked first. The remaining Internet Telephony Gateway Addresses are ranked by order of averaged latency. The resulting preferential ranking of Internet Telephony Gateway addresses is 166.37.27.205 30 166.37.61.117 166.25.27.101 The first choice gateway is the gateway most likely to give high quality of service, since it is located on the same local area network. This gateway i93 will be the first the client will attempt to use.
(2) Access Device Location Method The method for identifying the most appropriate choice for an Internet s Telephony Gateway utilizes a combination of the Client Ping Method detailed above, and the knowledge of the location from which the Client Computer 1080 accessed the Internet. This method may work well for clients accessing the Internet via a dial-up access device.
to A client computer 1080 dials the Internet Access Device. The Access Device answers the call and plays modem tone. Then, the client computer and the access device establishes a PPP session. The user on the Client Computer is authenticated (username/password prompt, validated by an authentication server). Once the user passes authentication, the Access 1s Device can automatically update the User Profile in the Directory Service for the user who was authenticated, depositing the following information "User Name" "Account Code" "online timestamp"
"Access Device Site Code"
Later, when the Client Computer requires access through an Internet Telephony Gateway, it queries the Directory Service 1082 to determine the best choice of Internet Telephony Gateway. If an Access Device Site Code is found in the User's Profile on the Directory Service, the Directory Service 1082 selects the Internet Telephony Gateway 1084, 1081 and 1086 at the same site code, and returns the IP address to the Client Computer 1080. If an Internet Telephony Gateway 1084, 1081 and 1086 is unavailable at the same site as the Access Device Site Code, then the next best choice is selected according to a network topology map kept on the directory server.
If no Access Device Site Code is found on the directory server 1082, then the client 1080 has accessed the network through a device which cannot update the directory server 1082. If this is the case, the Client Ping Method described above is used to locate the best alternative Internet r 9 s~

telephony gateway 1084.
(3) User Profile Method Another method for selection of an Internet Telephony Gateway 1084, 1081 - s and 1086 is to embed the information needed to select a gateway in the user profile as stored on a directory server. To use this method, the user must execute an Internet telephony software package on the client computer. The first time the package is executed, registration information is gathered from the user, including name, email address, IP Address (for to fixed location computers), site code, account code, usual Internet access point, and other relevant information. Once this information is entered by the user, the software package deposits the information on a directory server, within the user's profile.
~s Whenever the Internet Telephony software package is started by the user, the IP address of the user is automatically updated at the directory service.
This is known as automated presence notification. Later, when the user needs an Internet Telephony Gateway service, the user queries the directory service for an Internet Telephony Gateway to use. The directory service 2o knows the IP address of the user and the user's usual site and access point into the network. The directory service can use this information, plus the network map of all Internet Telephony Gateways 1084, 1081 and 1086, to select the best Internet Telephony Gateway for the client computer to use.
(4) Gateway Ping Method 2s The last method selects the best choice for a head-end hop-off Internet telephony gateway by obtaining a list of candidate Internet telephony gateway addresses, and pinging each to determine the best choice in terms of latency and number of router hops. The process is as follows:
Client Computer queries a directory service to obtain a best-choice Internet 3o telephony gateway.
The directory service looks in a database of gateways and selects a list of candidate gateways. Criteria for selecting gateways as candidates can include last gateway selected.
9.S

matching 1, 2, or 3 octets in the IPv4 address.
last client access point, if known.
selection of at least one gateway from all major gateway sites, if practical.
The directory sends a message to each candidate gateway, instructing each candidate gateway to ping the client computer's IP Address.
. Each candidate gateway simultaneously uses the IP ping to send an echo-type message to the client computer. The "-r" option will be used with the ping command to obtain a trace route. The ping results are 1o returned from each candidate gateway to the directory service.
. Based upon the ping results for each Internet Telephony Gateway, the directory service will rank order the ping results as follows:
If any Internet Telephony Gateways are accessible to the client without traveling through any intervening router as revealed by the ping trace route, they are ranked first.
The remaining Internet Telephony Gateways are ranked in order of lowest averaged latency of round-trip ping results.
The Client Ping Method and Gateway Ping Method may use the traceroute program as an alternative to the ping program in determining best choice ~ for a head-end hop-off gateway.
b) Tail-End Hop-Off Methods Tail-End Hop-Off entails selecting a gateway as an egress point from the Internet where the egress point is closest to the terminating PSTN location as possible. This is usually desired to avoid higher PSTN calling rates. The Internet can be used to bring the packetized voice to the local calling area of the destination telephone number, where lower local rates can be paid to carry the call on the PSTN.
(1) Gateway Registration 3o One method for Tail-End Hop-Off service is to have Internet Telephony Gateways 1084, 1081 and 1086 register with a directory service. Each Internet Telephony Gateway will have a profile in the directory service which lists the calling areas it serves. These can be listed in terms of Country Code, Area Code, Exchange, City Code, Line Code, Wireless Cell, ~q~

LATA, or any other method which can be used to subset a numbering plan.
The gateway, upon startup, sends a TCP/ IP registration message to the Directory Service 1082 to list the areas it serves.
s When a Client Computer wishes to use a TEHO service, it queries the directory service for an Internet Telephony Gateway 1084 serving the desired destination phone number. The directory service 1082 looks for a qualifying Internet Telephony Gateway, and if it finds one, returns the IP
address of the gateway to use. Load-balancing algorithms can be used to 1o balance traffic across multiple Internet Telephony Gateways 1084, 1081 and 1086 serving the same destination phone number.
If no Internet Telephony Gateways 1084, 1081 and 1086 specifically serve the calling area of the given destination telephone number, the directory 1s service 1082 returns an error TCP/IP message to the Client Computer 1080. The Client 1080 then has the option of querying the Directory Service for any Internet Telephony Gateway, not just gateways serving a particular destination telephone number.
2o As a refinement of this Gateway Registration scheme, Gateways can register calling rates provided for all calling areas. For example, if no gateway is available in Seattle, it may be less expensive to call Seattle from the gateway in Los Angeles, than to call Seattle from the gateway in Portland.
The rates registered in the directory service can enable the directory service 2s the lowest cost gateway to use for any particular call.
7. Vnet Call Processing.
Figure 11 is a callflow diagram in accordance with a preferred embodiment.
Processing commences at 1101 where the location of the directory service 3o to receive this "on-Iine" message will be determined by the data distribution implementation for this customer. In some cases this may be a private database for a company ar organization subscribing to a VNET service, in other cases it might be a national or worldwide database for all customers of a service provider (MCI). When the directory service receives this message ~9~

from PC 12 1051, it will update a profile entry associated with the unique ID to indicate that the user is "on-line" and is located at the specified IP
address. Then, at 1102, after successful update of the prof le associated with the ID, the directory service sends a response (ACK) back to the s specified IP address indicating that the message was ~~~~iv~~ arid processed. i9Vhen the computer (PC 12) receives this response rs'ressage it may choose to notify the user via a visual or audible indicator.
At 1103, a user of PC 11 1052 connects a computer to an IP network, turns to on the computer and starts telephony system software. The registration process for this computer follows the same procedures as those for PC 12 1051. In this scenario it is assumed that the directory service receiving this message is either physically or logically the same directory service that received the message from PC 12 1051.

At 1104, when the directory service 1031 receives a message from PC 1 I
1052, it initiates a similar procedure as it followed for a message for PC I2 1051. However, in this case it will update the profile associated with the identifier it received from PC 11 1052, and it will use the IP address it 2o received from PC 11 1052. Because of the updated profile information, when the acknowledgment message is sent out from the directory service, it is sent to the IP address associated with PC 11 1052. At this point both computers (PC 12 1051 and PC 11 1052) are "on-line" and available to receive calls, 2s At 1105, PC 12 1051 uses its telephony system software to connect to computer PC 1 I 1052. To establish this connection, the user of PC 12 1051 dials the VNET number (or other unique ID such as name, employee ID, etc). Depending upon the implementation of the customer's network, 3o and software package, a unique network identifier may have to be placed in this dial string. As an example, in a telephony implementation of a VNET, a subscriber rnay be required to enter the number $ prior to dialing the VNET
number to signal a PBX that they are using the VNET network to route the call. Once the telephony software package has identified this call as a I~~

VNET type call, it will send a translation request to the directory service.
At a minimum, this translation request will contain the following information:
- The IP address of the computer (PC12 1051) sending this request, and s The VNET number (or other ID) of the computer to be dialed.
At 1106, when the directory service receives this message, it uses the VNET
number (or other ID) to determine if the user associated with the VNET
number (or other ID) is "on-line" and to identify the IP address of the location where the computer may be contacted. Any additional information that is available about the computer being contacted (PC 11 1052), such as compression algorithms or special hardware or software capabilities, may also be retrieved by the directory service 1031. The directory service 1031 then returns a message to PC 12 1051 with status information for PC 11 ~s 1052, such as whether the computer is "on-line," its IP address if it is available and any other available information about capabilities of PC 11 1052. When PC 12 1051 receives the response, it determines whether PC 11 1052 may be contacted. This determination will be based upon the "on-line" status of PC 11 1052, and the additional information about 2o capabilities of PC 11 1052. If PC 12 1051 receives status information indicating that PC 11 1052 may not be contacted, the call flow stops here, otherwise it continues.
The following steps 1107 through 1111 are "normal" IP telephony call 2s setup and tear-down steps. At 110?, PC 12 1051 transmits a "ring"
message to PC 11 1052. This message is directed to the IP address received from the directory service 1031 in step 1106. This message can contain information identifying the user of PC 12 1051, or it may contain information specifying the parameters associated with the requested 30 connection.
At 1108, the message from step 1107 is received by PC 11 1052 and the receipt of this message is acknowledged by sending a message back to PC 12 1051 indicating that the user of PC 11 1052 is being notified of an t incoming call. This notification may be visible or audible depending upon the software package and its configurations on PC I I 1052.
At 1109, if the user of PC 11 1052 accepts the call, a message is sent back to PC 12 1051 confirming "answer" for the call. If the user of PC 11 1052 does not answer the call or chooses to reject the call, a message will be sent back to PC 12 1051 indicative of the error condition. If the call was not answered, the call flow stops here, otherwise it continues.
i o At 11 IO, the users of PC I2 1051 and PC I 1 1052 can communicate using their telephony software. Communication progresses until at 1111 a user of either PC may break the connection by sending a disconnect message to the other call participant. The format and contents of this message is dependent upon the telephony software packages being used by PC 12 1051 i5 and PC 11 1052. In this scenario, PC 11 1052 sends a disconnect message to PC 12 1051, and the telephony software systems on both computers discontinue transmission of voice.
Figure 12 illustrates a VNET Personal Computer (PC) to out-of-network PC
2o Information call flow in accordance with a preferred embodiment. In this flow, the Internet telephony gateway is an out-of-network element. This means that the Internet Telephony Gateway cannot use SS7 signaling to communicate with the switch, it must simply outpulse the VNET number to be dialed. An alternate embodiment facilitates directory services to do a 25 translation of the VNET number directly to a Switch/Trunk and outpulse the appropriate digits. Such processing simplifies translation in the switching network but would require a more sophisticated signaling interface between the Internet gateway and the switch. This type on "in-network" Internet gateway scenario will be covered in another call flow.
This scenario assumes that there is no integration between the Internet and a customer premises Public Branch Exchange {PBX). If there were integration, it might be possible for the PC to go through the Internet (or intranet) to connect to an ITG on the customers PBX, avoiding the use of ~oo the PSTN. Figure 12 is a callflow diagram in accordance with a preferred embodiment. Processing commences at 1201 where the location of the directory service to receive this "on-line" message will be determined by the data distribution implementation for this customer. In some cases this may be a private database for a company or organization subscribing to a VNET service, in other cases it might be a national or worldwide database for all customers of a service provider (MCI).
When the directory service receives this message from PC 12 1051, it will update a profile entry associated with the unique ID to indicate that the user is "on-line" and is located at the specified IP address. Then, at 1202, after successful update of the profile associated with the ID, the directory service sends a response (ACK) back to the specified IP address indicating that the message was received and processed. When the computer (PC 12) t5 receives this response message it may choose to notify the user via a visual or audible indicator.
At 1203, a VNET translation request is then sent to the directory services to determine the translation for the dial path to the out of network Internet 2o gateway phone. A response including the IP address and the DNIS is returned at 1204. The response completely resolves the phone addressing information for routing the call. Then, at 1205, an IP telephony dial utilizing the DNIS information occurs. DNIS refers to Dialed Number Information Services which is definitive information about a call for use in 25 routing the call. At 1206 an ACK is returned from the IP telephony, and at 1207 an IP telephony answer occurs and a call path is established at 1208.
1209a shows the VNET PC going offhook and sending a dial tone 1209b, and outpulsing digits at 1210. Then, at 1211, the routing translation of 3o the DNIS information is used by the routing database to determine how to route the call to the destination telephone. A translation response is received at 1212 and a switch to switch outpulse occurs at 1213. Then, at 1215, a ring is transmitted to the destination phone, and a ringback to the PC occurs. The call is transmitted out of the network via the Internet 10!

gateway connection and answered at 1216. Conversation ensues at 1217, until one of the parties hangs up at 1218.
Figure 13 illustrates a VNET Personal Computer (PC) to out-of-network Phone Information call flow in accordance with a preferred embodiment. In this call flow, the use of the PSTN is avoided by routing the call from the PC
to the Internet/Intranet to an Internet gateway directly connected to a PBX.
Figure 14 illustrates a VNET Personal Computer (PC) to in-network Phone to Information call flow in accordance with a preferred embodiment. In this call flow, the Internet telephony gateway is an in-network element. This requires that the Internet gateway can behave as if it were a switch and utilize SS7 signaling to hand the call off to a switch. This allows the directory service to return the switch/trunk and outpulse digits on the first ~ s VNET lookup. This step avoids an additional lookup by the switch. In this case the directory service must have access to VNET routing information.
a) PC to PC
Figure 15 illustrates a personal computer to personal computer Internet 2o telephony call in accordance with a preferred embodiment. In step 1501, a net phone user connects through the Internet via an IP connection to the step 1502 MCI directory service where a Iook up is performed to determine how to route the call. In step 1503, the call is terminated in the Intelligent System Platform (ISP) to determine where to send the call. IP Router is the 25 gateway that goes into the MCI ISP to determine via the Intelligent Services Network (ISN) feature engine how to get the call through the network. In step 1504, the call is connected through the Internet to the Net Phone user. In alternative scenario step 1504 the person at the phone is unavailable, so the calling party desired to speak with an MCI operator and 3o the IP Router goes through the Net-Switch (interface to the voice world.) In step 1505, the netswitch queries the call processing engine to do DSP
Engine functions. In step 1506, the call is routed through the WAN Hub to a MCI switch to an MCI Operator or voicemail in step 1507. This preferred embodiment utilizes the existing infrastructure to assist the call.
~oZ

b) PC TO PHONE
Figure 16, illustrates a phone call that is routed from a PC through the Internet to a phone. In step 1602, the MCI Directory is queried to obtain s ISN information for routing the call. Then the call is redirected in step 1603 to the ISP Gateway and routed utilizing the IP router to the call processing engine in steps 1604 and 1605. Then, in step 1606, the call is routed to the WAN and finally to the RBOC where Mainframe billing is recorded for the call.
IO
c) Phone to PC
Figure 17 illustrates a phone to PC call in accordance with a preferred embodiment. In step 1701, a phone is routed into a special net switch where in step 1?02, a call processing engine determines the DTMF tones t5 utilizing a series of digital signal processors. Then, at step 1703, the system looks up directory information and connects the call. If the caller is not there, or busy, then at step 1704, the call is routed via an IP router over the switch utilizing the call processing engine in step 1705.
2o d) Phone to Phone Figure 18 illustrates a phone to phone call over the Internet in accordance with a preferred embodiment. A call comes into the switch at step 1801, and is processed by the call logic program running in the call processing engine in step 1802. In step 1803, a lookup is performed in the directory 2s information database to determine routing of the call as described above.
The routing includes storing a billing record in the mainframe billing _ application 1808. All of the ISN features are available to the call even thought the call is routed through the Internet. An IP router is used at each end of the Internet to facilitate routing of the call through the Internet 30 1804 and into the network switch. From the network switch the call is routed to a call processing engine through a WAN hub 1806 through the RBOC 1807 to the target telephone. Various DSP Engines 1803 are utilized to perform digital transcoding, DTMF detection, voice recognition, call progress, VRU functions and Modem functions.
zoo XI. TELECOMMUNICATION NETWORK
A preferred embodiment utilizes a network management system for a telecommunication network for analyzing, correlating, and presenting s network events. Modern telecommunications networks utilize data signaling networks, which are distinct from the call-bearing networks, to carry the signaling data that are required for call setup, processing, and clearing. These signaling networks use an industry-standard architecture and protocol, collectively referred to as Common Channel Signaling System #7, or Signaling System #7 (SS7) for short. SS7 is a significant advancement over the previous signaling method, in which call signaling data were transmitted over the same circuits as the call. SS7 provides a distinct and dedicated network of circuits for transmitting call signaling data. Utilizing SS7 decreases the call setup time (perceived by the caller as is post-dial delay) and increases capacity on the call-bearing network. A
detailed description of SS7 signaling is provided in Signaling System #7, Travis Russell, Mcgraw Hill (1995).
The standards for SS7 networks are established by ANSI for domestic (U.S.) 2o networks, by ITU for international connections, and are referred to as ANSI
SS7 and ITU C7, respectively. A typical SS7 network is illustrated in Figure 1B. A call-bearing telecommunications network makes use of matrix switches 102a/ 102b for switching customer traffic. These switches 102a/ 102b are conventional, such as a DMS-250 manufactured by 25 Northern Telecom or a DEX-600 manufactured by Digital Switch Corporation. These switches 102a/ 102b are interconnected with voice-grade and data-grade call-bearing trunks. This interconnectivity, which is not illustrated in Figure 1B, may take on a large variety of configurations.
3o Switches in telecommunications networks perform multiple functions. In addition to switching circuits for voice calls, switches must relay signaling messages to other switches as part of call control. These signaling messages are delivered through a network of computers, each of which is called a Signaling Point (SP) 102a/ 102b. There are three kinds of SPs in ,Zc, an SS7 network:
- Service Switching Point (SSP) - Signal Transfer Point (STP) - Service Control Point (SCP) s The SSPs are the switch interface to the SS7 signaling network.
Signal Transfer Points (STPs) 104a...104f (collectively referred to as 104) are packet-switching communications devices used to switch and route SS7 signals. They are deployed in mated pairs, known as clusters, for 1o redundancy and restoration. For example, in Fig. 1B, STP 104a is mated with STP 104b in Regional Cluster 1, STP 104c is mated with STP 104d in Regional Cluster 2, and STP 104e is mated with STP 104f in Regional Cluster 3. A typical SS7 network contains a pluralit~~ of STP clusters 104;
three are shown in Fig. 1 for illustrative purposes. Each STP cluster 104 is serves a particular geographic region of SSPs I02. A plurality of SSPs 102 have primary SS7 links to each of two STPs 104 in a cluster. This serves as a primary homing arrangement. Only two SSPs 102 are shown homing to Regional Cluster 2 in Fig. 1B for illustrative purposes; in reality, several SSPs 102 will home on a particular STP cluster 104. SSPs 102 will also 2o generally have a secondary SS7 link to one or both STPs 104 in another cluster. This serves as a secondary homing arrangement.
The SS7 links that connect the various elements are identified as follows:
25 A links connect an SSP to each of its primary STPs (primary homing).
B links connect an STP in one cluster to an STP in another cluster.
_ C links connect one STP to the other STP in the same cluster.
D links connect STPs between different carrier networks (not illustrated).
_ E links connect an SSP to an STP that is not in its cluster (secondary 3o homing) .
F links connect two SSPs to each other.
To interface two different carriers' networks, such as a Local Exchange Carrier (LEC) network with an Interchange Carrier (IXC) network, STP
.26,r clusters 104 from each carriers' network may be connected by D links or A
links. SS7 provides standardized protocol for such an interface so that the signaling for a call that is being passed between an LEC and an IXC may also be transmitted.
When a switch receives and routes a customer call, the signaling for that call is received {or generated) by the attached SSP 102. While intermachine trunks that connect the switches carry the customer's call, the signaling for that call is sent to an STP 104. The STP 104 routes the signal to either the to SSP 102 for the call-terminating switch, or to another STP 104 that will then route the signal to the SSP 102 for the call-terminating switch.
Another element of an SS7 network are Protocol Monitoring Units {PMU) 106, shown in Figure 2. PMUs 106 are deployed at switch sites and provide an independent monitoring tool for SS7 networks. These devices, ~s such as those manufactured by INET Inc. of Richardson, TX., monitor the A, E, and F links of the SS7 network, as shown in Figure 2. They generate fault and performance information for SS7 links.
As with any telecommunications network, an SS7 network is vulnerable to 2o fiber cuts, other transmission outages, and device failures. Since an SS7 network carries all signaling required to deliver customer traffic, it is vital that any problems are detected and corrected quickly. Therefore, there is an essential need for a system that can monitor SS7 networks, analyze fault and performance information, and manage corrective actions.
Prior art SS7 network management systems, while performing these basic functions, have several shortcomings. Many require manual configuration of network topology, which is vulnerable to human error and delay topology updates. Configuration of these systems usually requires that the system 3o be down for a period of time. Many systems available in the industry are intended for a particular vendor's PMU i06, and actually obtain topology data from their PMUs 106, thereby neglecting network elements not connected to a PMU 106 and other vendors' equipment.

WO 98!47298 PCT/US98/07927 Because prior art systems only operate with data received from proprietary PMUs 106, they do not provide correlation between PMU events and events generated from other types of SS7 network elements. They also provide inflexible and proprietary analysis rules for event correlation.
s A system and method for providing enhanced SS7 network management functions are provided by a distributed client/ server platform that can receive and process events that are generated by various SS7 network elements. Each network event is parsed and standardized to allow for the to processing of events generated by any type of element. Events can also be received by network topology databases, transmission network management systems, network maintenance schedules, and system users.
Referring to Figure 3, the systems architecture of the preferred embodiment of the present invention, referred to as an SS7 Network Management 15 System (SNMS), is illustrated. SNMS consists of four logical servers 302/304/306/308 and a plurality of client workstations 312a/312b/312c connected via a Network Management Wide Area Network (WAN) 310. The four logical SNMS servers 302/304/306/308 may all reside on a single or a plurality of physical units. In the preferred 2o embodiment, each logical server resides on a distinct physical unit, for the purpose of enhancing performance. These physical units may be of any conventional type, such as IBM RS6000 units running with AIX operating system.
2s The client workstations 312 may be any conventional PC running with Microsoft Windows or IBM OS/2 operating systems, a dumb terminal, or a VAX VMS workstation. In actuality, client workstations may be any PC or terminal that has an Internet Protocol (IP) address, is running with X-Windows software, and is connected to the WAN 310. No SNMS-specific 3o software runs on the client workstations 312.
SNMS receives events from various SS7 network elements and other network management systems (NMS) 338. It also receives network topology, configuration, and maintenance data from various external systems, as will z o ~~.

be described. The various network elements that generate events include Network Controllers 314, International and Domestic SPs 316/ 102, STPs 104, and PMUs 106. Network Controllers 314 are devices that switch circuits based on external commands. They utilize SS7 signaling in the same manner as an SSP 102, but are not linked to any STPs 104.
International SPs 316 support switches that serve as a gateway between a domestic and international telecommunications network. The STPs 104 may be domestic or international.
1o The PMUs 106 scan all the SS7 packets that pass across the SS7 circuits, analyze for fault conditions, and generate network events that are then passed onto SNMS. The PMUs 106 also generate periodic statistics on the performance of the SS? circuits that are monitored.
All SPs 102/316, STPs 104, PMU 106, and SS7 Network Controllers 314 transmit network events to SNMS via communications networks. This eliminates the need for SNMS to maintain a session with each of the devices. In one typical embodiment, as illustrated in Fig. 3, an Asynchronous Data Communications Network 320 is used to transport 2o events from Network Controllers 314 and International SPs 316. An IBM
mainframe Front End Processor (FEP) 324, such as IBM's 3708, is used to convert the asynchronous protocol to SNA so it can be received by a IBM
mainframe-based Switched Host Interface Facilities Transport (SWIFT) system 326. SWIFT 326 is a communications interface and data distribution application that maintains a logical communications session with each of the network elements.
In this same embodiment, an X.25 Operational Systems Support (OSS) Network 328 is used to transport events from STPs 104, SPs 102, and 3o PMUs 106. These events are received by a Local Support Element (LSE) system 330. The LSE 330, which may be a VAX/VMS system, is essentially a Packet Assembler/ Disassembler (PAD) and protocol converter used to convert event data from the X.25 OSS Network 328 to the SNMS servers 302/304. It also serves the same function as SWIFT 326 in maintaining zoo communication sessions with each network element, thus eliminating the need for SLAMS to do so. The need for both SWIFT 326 and LSE 330 illustrates one embodiment of a typical telecommunications network in which different types of elements are in place requiring different transport s mechanisms. SLAMS supports all these types of elements.
All network events are input to the SLAMS Alarming Server 302 for analysis and correlation. Some events are also input to the SLAMS Reporting Server 304 to be stored for historical purposes. A Control system 332, which 1o may be a VAX/VMS system, is used to collect topology and configuration data from each of the network elements via the X.25 OSS Network 328.
Some elements, such as STPs 104 and SPs 102, may send this data directly over the X.25 OSS Network 328. Elements such as the International SSP 316, which only communicates in asynchronous mode, ~s use a Packet Assembler/Disassembler (PAD) 318 to connect to the X.25 OSS Network 328. The Control system 332 then feeds this topology and configuration data to the SLAMS Topology Server 306.
Network topology information is used by SLAMS to perform alarm correlation 2o and to provide graphical displays. Most topology information is received from Network Topology Databases 334, which are created and maintained by order entry systems and network engineering systems in the preferred embodiment. Topology data is input to the SLAMS Topology Server 306 from both the Network Topology Databases 334 and the Control System 332. An 2s ability to enter manual overrides through use of a PC 336 is also provided to the SLAMS Topology Server 306.
The SLAMS Alarming Server 302 also receives events, in particular DS-3 transmission alarms, from other network management systems (LAMS) 338.
3o Using topology data, SLAMS will correlate these events with events received from SS7 network elements. The SLAMS Alarming Server 302 also receives network maintenance schedule information from a Network Maintenance Schedule system 340. SLAMS uses this information to account for planned network outages due to maintenance, thus eliminating the need to respond ~a 9 to maintenance-generated alarms. SLAMS also uses this information to proactively warn maintenance personnel of a network outage that may impact a scheduled maintenance activity.
The SLAMS Alarming Server 302 has an interface with a Trouble Management System 342. This allows SLAMS users at the client workstations 312 to submit trouble tickets for SLAMS-generated alarms.
This interface, as opposed to using an SLAMS-internal trouble management system, can be configured to utilize many different types of trouble management systems. In the preferred embodiment, the SLAMS Graphics Server 308 supports all client workstations 312 at a single site, and are therefore a plurality of servers. The geographical distribution of SLAMS
Graphics Servers 308 eliminates the need to transmit volumes of data that support graphical presentation to each workstation site from a central t5 location. Only data from the Alarming Server 302, Reporting Server 304, and Topology Server 306 are transmitted to workstation sites, thereby saving network transmission bandwidth and improving SLAMS performance.
In alternative embodiments, the Graphics Servers 308 may be centrally located.
Referring now to Figure 4, a high-level process flowchart illustrates the logical system components of SLAMS. At the heart of the process is Process Events 402. This component serves as a traffic cop for SLAMS processes.
Process Events 402, which runs primarily on the SLAMS Alarming Server 302, is responsible for receiving events from other SLAMS components, processing these events, storing events, and feeding processed event data to the Reporting and Display components. The Process Events process 402 is shown in greater detail in Figure 5.
3o The Receive Network Events component 404, which runs primarily on the Alarming Server 302, receives network events from the various SS7 network elements (STPs 104, SPs 102, PMUs 106, etc.) via systems such as SWIFT 326 and LSE 330. This component parses the events and sends them to Process Events 402 for analysis. The Receive Network Events z~c~
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process 404 is shown in greater detail in Figure 6.
The Process Topology component 406, which runs primarily on the Topology Server 306, receives network topology and configuration data from the Network Topology Databases 334, from the SS7 network elements via the Control System 332, and from Manual Overrides 336. This data is used to correlate network events and to perform impact assessments on such events. It is also used to provide graphical presentation of events.
Process Topology 406 parses these topology and configuration data, stores to them, and sends them to Process Events 402 for analysis. The Process Topology process 406 is shown in greater detail in Figure 7.
The Define Algorithms component 408, which runs primarily on the Alarming Server 302, defines the specific parsing and analysis rules to be used by SNMS. These rules are then loaded into Process Events 402 for use in parsing and analysis. The algorithms are kept in a software module, and are defined by programmed code. A programmer simply programs the pre-defined algorithm into this software module, which is then used by Process Events 402. These algorithms are procedural in nature and are based on 2o network topology. They consist of both simple rules that are written in a proprietary language and can be changed dynamically by an SNMS user, and of mare complex rules which are programmed within SNMS software code.
The Receive NMS Data component 410, which runs primarily on the Alarming Server 302, receives events from other network management systems (NMS) 338. Such events include DS-3 transmission alarms. It also receives network maintenance events from a Network Maintenance Schedule system 340. It then parses these events and sends them to 3o Process Events 402 for analysis. The Display Alarms component 412, which runs primarily on the Graphics Server 308 and the Alarming Server 302, includes the Graphical User Interface (GUI) and associated software which supports topology and alarm presentation, using data supplied by Process Events 402. It also supports user interactions, such as alarm ~.//

clears, acknowledgments, and trouble ticket submissions. It inputs these interactions to Process Events 402 for storing and required data updates.
The Display Alarms process 412 is shown in greater detail in Figure 8.
The Report On Data component 414, which runs primarily on the Reporting Server 304, supports the topology and alarm reporting functions, using data supplied by Process Events 402. The Report On Data process 414 is shown in greater detail in Figure 9.
1o Referring now to Figure 5, the detailed process of the Process Events component 402 is illustrated. This is the main process of SNMS. It receives generalized events from other SNMS components, parses each event to extract relevant data, and identifies the type of event. If it is an SS7-related event, Process Events 402 applies a selected algorithm, such as create is alarm or correlate to existing alarm.
The first three steps 502-506 are an initialization process that is run at the start of each SNMS session. They establish a state from which the system may work. Steps 510-542 are then run as a continuous loop.
In step 502, current topology data is read from a topology data store on the Topology Server 306. This topology data store is created in the Process Topology process 406 and input to Process Events 402, as reflected in Figure 4. The topology data that is read has been parsed in Process 2s Topology 406, so it is read in step 502 by Process Events 402 as a standardized event ready for processing.
In step 504, the algorithms which are created in the Define Algorithms component 408 are read in. These algorithms determine what actions 3o SNMS will take on each alarm. SNMS has a rnap of which algorithms to invoke for which type of alarm.
In step 506, alarms records from the Fault Management (FM) reporting database, which is created in the Report on Data process 414, are read in.
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All previous alarms are discarded. Any alarm that is active against a node or circuit that does not exist in the topology (read in step 502 is discarded.
Also, any alarm that does not map to any existing algorithm (read in step 504) is discarded. The alarms are read from the FM reporting database only within initialization. To enhance performance of the system, future alarm records are retrieved from a database internal to the Process Events 402 component. Step 506 concludes the initialization process; once current topology, algorithms, and alarms are read, SNMS may begin the continuous process of reading, analyzing, processing, and storing events.
~o This process begins with step 510, in which the next event in queue is received and identified. The queue is a First In/First Out (FIFO) queue that feeds the Process Events component 402 with network events, topology events, and NMS events. To reiterate, the topology data that are read in i 5 step 502 and the alarm data that are read in step 504 are initialization data read in at startup to create a system state. In step 510, ongoing events are read in continuously from process components 404, 406, and 410.
These events have already been parsed, and are received as standardized SNMS events. SNMS then identifies the type of event that is being received.
2o If the event is found to be older than a certain threshold, for example one hour, the event is discarded.
In steps 512, 520, 524, and 534 SNMS determines what to do with the event based on the event type identification made in step 510.
In step 5I2, if the event is determined to be topology data, SNMS updates the GUI displays to reflect the new topology in step 514. Then in step 516, SNMS performs a reconciliation with active alarms to discard any alarm not mapping to the new topology. In step 518, the new topology data is 3o recorded in a topology data store, which is kept in the SNMS Topology Server 306.
In step 520, if the event is determined to be NMS data, such as DS-3 alarms 338, it is stored in the FM reporting database on the SNMS
zi3 Reporting Server 304 for future reference by SLAMS rules.
In step 524, if the event is determined to be a defined SS7 network event, then in step 526 one or more algorithms will be invoked for the event. Such algorithms may make use of data retrieved from Network Management Systems 338, Network Maintenance Schedules 340, and Network Topology 334.
For example, when each circuit level algorithm generates an alarm, it to performs a check against the Network Maintenance Schedule 340 and LAMS
338 records. Each alarm record is tagged if the specified circuit is within a maintenance window (Network Maintenance Schedule 340) or is transported on a DS-3 that has a transmission alarm (LAMS 338}. While SS7 circuits run at a DS-0 level, the Network Topology Databases 334 provide a DS-3 to DS-0 conversion table. Any DS-0 circuit within the DS-3 is tagged as potentially contained within the transmission fault. Clear records from LAMS 338 will cause active SLAMS circuit level alarms to be evaluated so that relevant LAMS 338 associations can be removed. SLAMS
clear events will clear the actual SLAMS alarm. GUI filters allow users to 2o mask out alarms that fit into a maintenance window or contained within a transmission fault since these alarms do not require SLAMS operator actions.
In step 528, active alarms are reconciled with new alarm generations and zs clears resulting from step 526. In step 530, the GUI displays are updated.
In step 532, the new alarm data is stored in the FM reporting database.
In step 534, the event may be determined to be a timer. SLAMS algorithms sometimes need to delay further processing of specific conditions for a 3o defined period of time, such as for persistence and rate algorithms. A
delay timer is set for this condition and processing of new SLAMS events continues. When the time elapses, SLAMS treats the time as an event and performs the appropriate algorithm.
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For example, an SS7 link may shut down momentarily with the possibility of functioning again within a few seconds, or it may be down for a much greater period of time due to a serious outage that requires action. SLAMS, when it receives this event, will assign a timer of perhaps one minute to the event. If the event clears within one minute, SLAMS takes no action on it.
However, if after the one minute timer has elapsed the event is unchanged (SS7 link is still down), SLAMS will proceed to take action.
In step 536, the appropriate algorithm is invoked to take such action. In to step 538, active alarms are reconciled with those that were generated or cleared in step 536. In step 540, the GUI displays are updated. In step 542, the new alarm data is stored in the FM reporting database. As stated previously, SLAMS operates in a continuous manner with respect to receiving and processing events. After the data stores in steps 518, 522, 532, and 542, the process returns to step 510.
Referring now to Figure 6, the detailed process of the Receive Network Events component 404 is illustrated. This component collects events from the SS7 network elements via data transport mechanisms, such as the 2o Async Data Network 320, SWIFT 326, X.25 OSS network 328, and the LSE
330. These events are received by the SLAMS Alarming Server 302 in a First In/First Out (FIFO) queue. In steps 602 and 604, events from the SS7 network elements are collected by mainframe applications external to SLAMS, such as SWIFT 326 and LSE 330, and the protocol of the event data is converted from the network element-specific protocol to SNA or TCP/IP.
In one embodiment, SLAMS may also have software running on the mainframe that converts the protocol to that recognizable by the SLAMS
Alarming Server 302. The event data is then transmitted via SNA or TCP/IP
to the SLAMS Alarming Server 302. SLAMS maintains a Signaling Event List 608 of all SS7 event types that is to be processed. In step 606, SLAMS
checks the Signaling Event List 608 and if the current event is found in the list, SLAMS traps the event for processing. If the event is not found in the list, SLAMS discards it.
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In step 610, the event is parsed according to defined parsing rules 614.
The parsing rules 614 specify which fields are to be extracted from which types of events, and are programmed into the SLAMS code. The parsing of events in step 610 extracts only those event data fields needed within the alarm algorithms or displays. Also input to step 610 are scheduled events 612 from the Network Maintenance Schedule 340. Scheduled events 612 are used to identify each network event collected in step 602 that may be a result of scheduled network maintenance. This allows SLAMS operators to account for those SS7 network outages that are caused by planned to maintenance.
In step 616, the parsed event data is used to create standardized event objects in SLAMS resident memory for use by other SLAMS processes. Such event objects are read into the main process, Process Events 402, in step t s 510.
Referring now to Figure 7, the detailed process of the Process Topology component 406 is illustrated. This process component retrieves network topology and configuration data from three types of sources, creates 2o standardized topology data records, and stores this data for use by other SLAMS processes. In particular, it feeds active topology data to Process Events 402, running on the Alarming Server 302, in step 502.
In step 702, the SLAMS Topology server 306 collects topology data from 25 three different sources. It collects current connectivity and configuration data generated by the SS? network elements via the Control system 332. It collects topology data that has been entered into order entry and engineering systems and stored in Network Topology Databases 334. It also accepts manual overrides 336 via workstation. The collection of data from 3o the Topology Database 334 and the Control system 332 occurs on a periodic basis, and is performed independently of the SLAMS Alarming server 302. Unlike prior art systems that use data retrieved from PMUs 106, SLAMS receives topology data from all types of network elements, including those that are not connected to a PMU 106 such as that of Figure ;Z l ~~

2. SNMS also uses data reflecting the topology of foreign networks, such as those of a Local Exchange Carrier (LEC) or an international carrier. This data is used to perform impact assessments that will allow the SNMS user to determine facts such as which end customers may be impacted by an s SS7 link outage. The type of topology data collected and used by SNMS, and for example, for the SS7 linkage of an STP 104 with a Switch/ SSP 102, data is received by network order entry and engineering systems. The data and a brief description of its contents is provided below.
STP Link ID Identifies each SS7 link to the STP
Switch Link ID Identifies each SS7 link to the Switch/SP
STP Linkset Identifies a trunk grouping of SS7 links to the STP
Switch Linkset Identifies a trunk grouping of SS7 links to the Switch / SP
~s MCI/Telco Circuit ID Identifies the SS7 link to external systems. For interfaces between two different networks, each ID
(MCI ID and Telco ID) provides an identification of the SS7 link for each network (MCI and a Telco in this example).
Link Type Identifies the type of SS7 link 2o SLC Signal Link Code For the switched voice network supported by SS7, data is received by network order entry and engineering systems and used to perform SS7 event impact assessments:
2s Voice Trunk Groups Voice trunk group supported by each SSP 102 For the SS7 linkage of a domestic STP 104g to an international STP 104h, data is received by network order entry and engineering systems:
Circuit ID Identifies the SS7 link to external systems SLC Signal Link Code For the purpose of performing impact assessments, Local Exchange Carrier .Z l (LEC) NPA/NXX assignments and End Office to Access Tandem homing arrangements are received by a calling area database that is populated by Bellcore's Local Exchange Routing Guide (LERG).
LATA Local Access Transport Area (conventional) NPA/ NXX Numbering Plan Area/ prefix (conventional) End Office LEC customer serving node Access Tandem LEC end office hub to Foreign network STP 104 clustering and SSP 102 homing arrangements are received by SS7 network elements via a control system.
Point Code Identifies SS7 node (conventional) i > Data identifying certain aspects of each network element are received by a Switch Configuration File, which resides in an external system.
Data mapping each network DS-0 onto a DS-3 is received by Network Topology Databases. This data is used to assign DS-3 alarms received by 2o NMS to DS-0 level circuits.
Data needed to overwrite data acquired through automated processes are provided by manual overrides.
2s Referring now back to Figure 7 in step 704, the various topology data are parsed to extract the data fields that are needed by SNMS algorithms. The data are then standardized into event records that can be processed by Process Events 402.
3o In step 706, the standardized data records are validated against other data.
For example, circuit topology records are validated against node topology records to ensure that end nodes are identified and defined.
In step 708, the topology data are stored on the Topology server 306 of Figure 3 in a relational database, such as that offered by Sybase.
In step ?10, the new topology records are passed from the Topology server 306 to the main SNMS process running on the Alarming server 302 and s compared against the active configuration (i.e. configuration that is currently loaded into memory). Active alarm and GUI displays are reconciled to remove alarms that pertain to non-existent topology entries.
In step ?12, the topology is stored on the Alarming Server 302 (for use by to Process Events 402) in the form of flat files for performance reasons. At this time the flat file mirrors the Topology server 306 database from step 708. This flat file is only accessible by the main process. In step ?14, the new topology records are loaded into active SNMS memory and new processes which require topology data now use the new configuration.
is Referring now to Figure 8, the detailed process of the Display Alarms component 412 is illustrated. This process component provides the results of SNMS processing to the user (referred to as the "operator"), and accepts operator input as actions to be performed within SNMS. Therefore, the 2o process between Display Alarms 412 and Process Events 402 is two-way. It is important to note that while there is a single Process Events process 402 running for the entire SNMS system, there is a different instance of the Display Alarms process 412 running for each operator that is logged onto SNMS. That is, each operator instigates a separate execution of Display 2s Alarms 4I2.
When an operator logs on SNMS, the first four steps, 802 - 808, execute as an initialization. From there, steps 810 - 838 operate as a continuous loop. The initialization provides each operator with a system state from 3o which to work. In step 802, the current topology is read in and displayed via Graphical User Interface (GUI). Each operator has its own GUI process that is initialized and terminated based upon an operator request. Each GUI process manages its displays independently. Any status change is handled by the individual processes.
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In step 804, a filter that defines the specific operator view is read in. Each operator can define the view that his/her GUI process will display. Filter parameters include:
Traffic Alarms, Facility alarms, or both Acknowledged Alarms, Unacknowledged Alarms, or both Alarms on circuits within maintenance windows, Alarms on circuits that are not within a maintenance window, or bath.
1o Alarms on circuits that have associated transmission alarms (DS-3 alarms via outage ids), Alarms on circuits that do not have associated transmission alarms, or both.
Alarms with a specified severity.
Alarms on nodes/circuits ov~~ned by a specified customer id.
t5 Alarms on International circuits, Alarms on Domestic circuits, or both.
The operator's GUI displays are updated both upon initialization in step 804 and when filter changes are requested in steps 828 and 830. Each specific operator's instance of the Display Alarms 412 process opens a 2o connection with Process Events 402 so that only alarm records relevant to the specific operator's filter are transmitted. In step 806, the specific operator's process registers itself with Process Events 402 to identify which alarms are to be sent. In step 808, the GUI display is presented to the operator.
The continuous execution of Display Alarms 412 begins in step 810. Each event that is to be retrieved and presented, as defined by the operator filter, is received and identified. In steps 812, 816, 820, 826, and 836 SNMS
determines what to do with the event based on the event type identification 3o made in step 810. In steps 812 and 816, if the event is determined to be an alarm update or a topology update, the operator's GUI display is updated to reflect this, in steps 814 and 818, respectively. Then the next event is received, in step 810.
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_.____.._~ _.._. .._. , , . _ In step 820, if the event is determined to be an operator action, two activities are required. First, in step 822, the operator's GUI display is updated to reflect the status change. Then, in step 824, a status change update is sent to the main process, Process Events 402, so that the status s change may be reflected in SNMS records and other GUI processes (for other operators) can receive and react to the status changes.
In step 826, if the event is determined to be an operator display action, then it is determined if the action is a filter change request or a display to request. In step 828, if it is determined to be a filter change request, then in step 830 the GUI process registers with Process Events 402 so that the appropriate alarms records are transmitted. In step 832, if it is determined to be an operator display request, then in step 834 the requested display is presented to the operator. Display requests may include:
is node detail and connection circuit connection linkset connection unknown topology alarms (alarms on objects that are not defined in the 2o topology databases) STP pair connections Nodes contained within a LATA
Home/Mate connections (for non-adjacent nodes}
NPA/ NXX lists 2s trunk group lists end office access tandem rules definition help screens (aid the operator in understanding the actual algorithm used in generating the alarm recommended actions (operator defined actions that should be taken when 3o a specific alarm is received) In step 836, if the event is determined to be a termination request, then the specific operator's GUI process is terminated in step 838. Otherwise, the next event is received in step 810. Within the Display Alarm process, .~z~

SLAMS provides several unique display windows which support fault isolation, impact assessments, and trouble handling. All of the GUI displays which contain node and circuit symbols are "active" windows within SLAMS
(i.e. screens are dynamically updated when alarm status of the node or s circuit change). All the displays are possible due to the set of MCI
topology sources used within SLAMS. SLAMS has extensive topology processing of SLAMS which is used in operator displays.
A. SLAMS Circuits Map to This window displays topology and alarm status information for a selected linkset. As network events are received, SLAMS recognizes the relationships between endpoints and isolates the fault by reducing generated alarms.
This display allows the operator to monitor a linkset as seen from both sides of the signaling circuit (from the perspective of the nodes).
is B. SLAMS Connections This window presents a cluster view of MCI's signaling network. All MCI
and non-MCI nodes connected to the MCi STPs in the cluster are displayed along with the associated linksets. A cluster view is important since a 2o single STP failure/isolation is not service impacting, but a cluster failure is since all MCI SPs have connectivity to both MCI STPs in the cluster.
C. SLAMS Nonadjacent Node This window presents a STP pair view of a selected LEC signaling network.
2s All LEC SPs, STPs, and SCPs (with signaling relationships to the MCI
network) connected LEC STP pair are displayed. MCI's area of responsibility does not include the LEC STP to LEC SSP signaling links, so no linksets are displayed here. This display allows the SLAMS operator to monitor a LEC signaling network as seen by the MCI nodes.
D. SLAMS LATA Connections This window presents a map of all LEC owned nodes that are located within a specified LATA. As well, the MCI STP pair that serves the LATA is also displayed along with the associated linksets (where applicable). This zzz display allows the operator to closely monitor a specific LATA if/when problems surface within the LATA. LATA problems, while outside MCI's domain of control, can introduce problems within the MCI network since signaling messages are shared between the networks. As well, MCI voice s traffic which terminates in the specified LATA can be affected by LATH
outages.
E. NPA-1VXX Information List This window presents a list of NPX-NXX's served by a specified LEC switch.
to This display is very valuable during impact assessment periods (i.e. if the specified LEC switch is isolated, which NPA-NXX's are unavailable).
F. End Office Information List This window presents a list of LEC end office nodes which are homed to the 15 specific LEC access tandem. This display is very valuable during impact assessment periods (i.e. if the specified LEC tandem switch is isolated, which end offices are unavailable).
G. Trunk Group Information List 2o This window presents a list of MCI voice trunks, connected to a specified MCI switch, and the LEC end office switches where they terminate. This display is very valuable during impact assessment periods (i.e. what end offices are impacted when a MCI switch is isolated). This display is also available for selected LEC end office switches.
2s H. Filter Definition Window The SNMS operator can limited the scope of his displays to:
d type of alarms that should be presented 3o severity of alarms that should be presented acknowledged alarms, unacknowledged alarms, or both alarms on circuits inside a planned outage window, alarms on circuits outside a planned outage window or both alarms that are not the result of a specified transmission network outage z ,z 3 alarms on specified customer nodes or alarms on circuits connected to specified customer I. Trouble Ticket Window The SNMS operator can open trouble tickets on signaling alarms. These trouble tickets are opened in MCI's trouble ticketing system. Operators can also display the status of existing trouble tickets.
Referring now to Figure 9, the detailed process of the Report On Data to component 414 is illustrated. This process component, which runs on the Reporting server 304, stores SNMS-processed data and provides reports.
Standardized Network Element (NE) Event Records 914 are received with location specific time stamps. In step 902, the time stamps are converted ~s into Greenwich Mean Time (GMT) so that standardized reports can be produced.
In step 904, all data received are stored in individual database tables. Data may also be archived for long-term storage to tape or disk. This data 2o includes SNMS-generated alarms 916, standardized topology records 918, and performance statistics from PMUs 920. It may also include non-processed data, such as DS-3 alarms from NMS 338 and network maintenance schedule data 340.
25 In step 906, reports are produced. These reports may be custom or form reports. They may also be produced on demand, or per a schedule. These reports may be presented in a number of ways, including but not limited to electronic mail 908, X-terminal displays 910, and printed reports 912.
XII. VIDEO TELEPHONY OVER POTS
3o The next logical step from voice over the POTS is video. Today, computers are capable of making video "calls" to each other when connected to some type of computer network. However, most people only have access to a computer network by making a call from their modem on the POTS with another modem on a computer connected to a network, so that they can ~z z ~r then "call" another computer on the network, which is in turn connected by a modem to another network computer. It is much simpler (and efficient) to call another person directly on the POTS and have the modems communicate with each other, without network overhead. ITU
s recommendation H.324 describes terminals for low bitrate (28.8kbps modem) multimedia communication, utilizing V.34 modems operating over the POTS. H.324 terminals may carry real-time voice, data, and video, or any combination, including video telephony. H.324 terminals may be integrated into personal computers or implemented in stand-alone devices to such as videotelephones and televisions. Support for each media type (voice, data, video) is optional, but if supported, the ability to use a specified common mode of operation is required, so that all terminals supporting that media type can interwork. H.324 allows more than one channel of each type to be in use. Other Recommendations in the H.324 t5 series include the H.223 multiplex (combination of voice, data and video), H.245 control, H.263 video codec (digital encoder and decoder), and 6.723.1.1 audio codec.
H.324 makes use of the logical channel signaling procedures of ITU
2o Recommendation H.245, in which the content of each logical channel is described when the channel is opened. Procedures are provided for allowing each caller to utilize only the multimedia capabilities of their machine. For example a person trying to make a video (and audio) call to someone who only has audio and not video capabilities can still 25 communicate with the audio method (6.723.1.1) H.324 by definition is a point-to-point protocol. To conference with more than one other person an MCU (Multipoint Control Unit) is needed to act as a video-call bridge. H.324 computers may interwork with H.320 computers 30 on the ISDN, as well as with computers on wireless networks.
A. Components of Video Telephony 1. DSP modem pools with ACD..
A Digital Signal Processor (DSP) modem pool is a modem bank, with each 2 z.5 modem having the ability to be programmed for extra functions (like new V.
modem protocols, DTMF detection, etc.) A call is routed from the MCI
switch to an ACD. The ACD keeps a matrix of which DSP modems are available. The ACD also communicates with the ISNAP which does a group s select to determine which group of Agents are responsible for this call and also which of the agents are free to process this call. In an alternative embodiment, DSP resources can be deployed without an ACD, directly connected to a switch. In this embodiment, the DSP resources are managed using an NCS-based routing step.
to 2. Agent.
An Agent can be a human Video Operator (video capable MTOC), or an Automated program (video ARU). The ACD knows which Agent ports are available and connects an Agent to an Agent Port.
3. Video on Hold Server.
If the ACD has no Agent ports available, then the caller is connected to the Video On Hold Server, which has the ability to play advertisements and other non-interactive video, until the ACD finds a free Agent port.
4. Video Mail Server.
Video-mail messages are stored here. Customers can manage their mail and record greetings to be stored on this server.
5. Video Content Engine.
Video On Demand content resides on the Video Content Engine. Video stored here can be previously recorded video-conferences, training videos, etc.
6. Reservation Engine.
When people want to schedule a multi-party video-conference, they can specify the participants and time of the conference on this system.
Configuration can be done with the help of a human Video Operator or by some other form entry method.
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7. Video Bridge.
Because H.324 is a point-to-point protocol, a Multi-point Conferencing Unit (MCU) needs to manage each participants call and re-direct the video streams appropriately. MCU conferencing will be available for customers with H.324 and H.320 compliant systems.
B. Scenario A computer or set-top TV has H.324 compliant software, and a modem for use over POTS, most likely to be 28.8kbps (V.34) or higher. One objective is to call another party. If they do not answer or are busy, the originator has the option of leaving video-mail for the destination party. Another objective is to schedule and participate in a conference with more than two participants.
C. Connection Setup Figure 19B illustrates a call connection setup in accordance with a preferred embodiment. There are three methods for making a video-call to someone. The first method is to simply call them (from 1 and 7 of Figure 19B. If the destination is busy or doesn't answer, then the caller can make 2o another call to 1 800 VID MAIL and perform the appropriate procedures as follows.
When a user dials " 1 800 VID MAIL" at 1, the ACD on the DSP modem pool will connect a switch to a modem 2 and a port to an Agent 3. Then the user logs-in to the system with a special, custom terminal program that utilizes the data stream part of the H.324 bandwidth (using the ITU T.I20 . standard), called the V-mail Data Interface (VMDI). From a graphical user interface, icon or other menu, the caller can choose to - browse a.nd search a directory of video-capable MCI customers, - call another H.324 compliant software program, - create a video-mail for Store & Forward for later delivery, - personalize and record their video-mail greeting messages, - view and manage their video-mail, or - view selections from a library of recordings (Video On Demand).
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In an alternate embodiment, a user can dial " 1 800 324 CALL" to call a number. Then, if the destination number was 1 319 375 1772, the modem dial string would be "ATDT 1 800 324 CALL ", 1 319 375 1772" (the comma s ',' tells the modem to do a short pause while dialing.) When the connection to 1 800 324 CALL is made, a connection is made from the originator, to an MCI switch 1, to an ARU 5a, selected by an ACD 2a, 3a.
The ARU 5a detects DTMF tones entered through a telephone keypad or other device for generating DTMF tones to get the destination number. The originator remains on hold while the ARU 5a makes a separate call to the destination number 5a, 6a and 7. If the destination answers, the originator is connected to the destination, both party's modems can connect, and the ARU 5a is released. If the destination is busy or does not answer, the call is is transferred to 1 800 VID MAIL or an Agent through the DSP modem pool 2. If there are no DTMF tones detected, the call is transferred to an Agent through the DSP modem pool 2. The Agent will make an H.324 connection with the caller and ask for their destination number (or provide help.) The architecture for this alternative is similar to how faxes are detected and 2o transmitted in the directlineMCI system as discussed with respect to an alternative embodiment.
D. Calling the Destination When the destination number is known, the Video On Hold Server provides 2s the video input for the H.324 connection 4. A new call is made from the Agent 5, 6 to the destination number 7. One concern that required analysis while working out a detailed embodiment required determining if modems could re-synchronize after a switch operation without going off line. If the destination number answers and is a modem, a connection 3o MUST be made at the same speed as the originator modem speed. After modem handshaking is performed, the ACD instructs the switch to release the agent 3, 5 and releases the modems 2 and 6 and connects the originator to the destination 1 and 7. The destination PC realizes that the connection is an H.324 call (not a fax or otherwise) and the video-call ~ZZ~

proceeds.
In an alternate embodiment, if the destination answers and is a modem, a connection is made. Then, two H.324 calls are using two DSP modems.
The Agent can be released from both calls 3 and 5. The incoming data from each call is copied to the other call 2 and 6. This way, an Agent can monitor the video call for Video Store & Forward 9. When one connection drops carrier, the video-call is complete, and the modem carrier for the remaining call is dropped.
E. Recording Video-Mail, Store 8z Forward Video and Greetings If a destination number does not answer or is busy, the Video Mail Server will play the appropriate Video-Mail greeting for the owner of the is destination number 8. The caller then leaves a video-message, which is stored on the Video Mail Server. The recording of video for Store & Forward Video is exactly the same as leaving a video-message, described above.
Parameters such as destination number, forwarding time, and any other audio S8vF features currently available are entered through the VMDI or 2o communicated with a human video operator (or automated video ARU.) To record a personalized greeting for playback when someone cannot reach you because you are busy or do not answer, is similar to leaving Video-Mail. The option to do this is done through the VMDI or communicated 25 with a human video operator.
F. Retrieving Video-Mait and Video On Demand Users have the choice of periodically polling their video-mail for new messages, ar have the video-mail server call them periodically when they 3o have a new message waiting. Configuration is done through the VMDI or human video operator. Managing and checking video-mail is also performed through the VMDI or communicated with a human video operator.
Choice of video to view for Video On Demand (VOD) is through the VMDI.
.~ z 9 These videos can be previously recorded video-conferences, training videos, etc. and are stored on the Video Content Engine 9.
G. Video-conference Scheduling s A user can navigate through the VMDI or Internet 10 WWW forms, or communicate with a human video operator to schedule a mufti-point conference. This information is stored on the Reservation Engine 11. The other conference participants are notified of the schedule with a video-mail, e-mail message or otherwise. There will be an option to remind all 1o registered conference participants at a particular time (e.g. 1 hour before the conference), through video-mail (or e-mail, voice-mail, paging service or any other available notification method). The MCU (video bridge) can call each participant 12, or H.324 users can dial In to the MCU at the scheduled time 12.
is XIII. VIDEO TELEPHONY OVER THE INTERNET
Figure 19E illustrates an architecture for transmitting video telephony over the Internet in accordance with a preferred embodiment. Real-time Transmission Protocol (RTP) based video-conferencing refers to the 2o transmission of audio, video and data encapsulated as RTP messages. For a RTP-based video-conferencing session, a end-user station first establishes a dial-up Point-to-Point (PPP) connection with the Internet which is then used to transport the RTP messages. Audio information is compressed as per 6.723.1.1 audio codec (coder - decoders standards, Video is compressed 2s in accordance with ITU H.263 video codec standards and data is transmitted as per ITU-T.120 standards.
RTP is a protocol providing support for applications with real-time properties. While UDP/IP is its initial target networking environment, RTP
3o is transport-independent so that it can be used over IPX or other protocols.
RTP does not address the issue of resource reservation or quality of service control; instead, it relies on resource reservation protocols such as RSVP.
The transmission service with which most network users are familiar is point-to-point, or unicast service. This is the standard form of service 23~

provided by networking protocols such as HDLC and TCP.
Somewhat Iess commonly used (on wire-based networks, at any rate) is broadcast service. Over a large network, broadcasts are unacceptable (because they use network bandwidth everywhere, regardless of whether individual sub-nets are interested in them or not), and so they are usually restricted to LAN-wide use (broadcast services are provided by low-level network protocols such as IP). Even on LANs, broadcasts are often undesirable because they require all machines to perform some processing 1 o in order to determine whether or not they are interested in the broadcast data.
A more practical transmission service for data that is intended for a potentially wide audience is multicast. Under the multicast model on a WAN, only hosts that are actively interested in a particular multicast service will have such data routed to them; this restricts bandwidth consumption to the link between the originator and the receiver of multicast data. On LANs, many interface cards provide a facility whereby they will automatically ignore multicast data in which the kernel has not zo registered an interest; this results in an absence of unnecessary processing overhead on uninterested hosts.
A. Components RSVP Routers with MBONE capability for broadcast of video from the Video Content Engine and the MCI Conference Space network. MCI will have an MBONE network that multicasts locally and transmits many unicasts out the Internet.
RSVP is a network control protocol that will allow Internet applications to obtain special qualities-of-service (QOS's) for their data flows. This will generally (but not necessarily) require reserving resources along the data paths) either ahead of time or dynamically. RSVP is a component of the future "integrated services" Internet, which provides both best-effort and real-time qualities of service. An embodiment is presented in the detailed ,23/

WO 98!47298 PCTIUS98/07927 specification that follows.
When an application in a host (end system) requests a specific QOS for its data stream, RSVP is used to deliver the request to each router along the s paths) of the data stream and to maintain router and host state to provide the requested service. Although RSVP was developed for setting up resource reservations, it is readily adaptable to transport other kinds of network control information along data flow paths.
l0 1. Directory and Registry Engine.
When people are connected to the Internet (whether through modem dial-up, direct connection or otherwise), they can register themselves in this directory. The directory is used to determine if a particular person is available for conferencing.
2. Agents.
An Agent can be a human Video Operator (video capable MTOC), or an Automated program (video ARU). An Internet ACD in accordance with a preferred embodiment is designed so that Agent ports can be managed.
2o The ACD will know which Agent ports are available and connects an Agent to an available Agent Port. If the ACD has no Agent ports available, then the caller is connected to the Video On Hold Server, which has the ability to play advertisements and other non-interactive video, until the ACD finds a free Agent port.
3. Video Mail Server.
Video-mail messages are stored here. Customers can manage their mail and record greetings to be stored on this server.
4. Video Content Engine.
Video On Demand content resides on the Video Content Engine. Video stored here may be previously recorded video-conferences, training videos, etc.
2 3:Z

5. Conference Reservation Engine.
When people want to schedule a multi-party video-conference, they can specify the participants and time of the conference on this system.
Configuration can be done with the help of a human Video Operator or by s some other form entry method.
6. MCI Conference Space.
This is the virtual reality area that customers can be present in. Every participant is personified as an "avatar". Each avatar has many abilities 1o and features, such as visual identity, video, voice, etc. Avatars interact with each other by handling various objects that represent document sharing, file transferring, etc., and can speak to each other as well as see each other.
~s 7. Virtual Reality Space Engine.
The Conference Spaces are generated and managed by the Virtual Reality Engine. The virtual reality engine manages object manipulation and any other logical descriptions of the conference spaces.
20 B. Scenario If a user has a current connection to the Internet. The user will utilize H.263 compliant system software utilizing RTP (as opposed to TCP) over the Internet. If the user also desires to participate in VR MCI conference-space, and create/view video-mail, the user can join a VR session.
2s C. Connection Setup The simplest way to make a video call to another person on the Internet is to simply make the call without navigating through menus and options as an initial telephone call. However, if the destination is busy or not answering, MCI provides services for depositing messages.
A customer can login to a telnet server (e.g. telnet vmail.mci.com), or use a custom-made client, or the WWW (e.g. http:/ /vmail.mci.com). The services menu is referred to as the V-Mail Data Interface (VMDI), similar to the VMDI available when dialing through POTS as described above.
~ 33 From a menu, the caller can choose to:
- browse and search a directory of video-capable MCI customers, - call another H.263 compliant software program, s - create a video-mail for Store 8s Forward for later delivery, - personalize and record their video-mail greeting messages, - view and manage their video-mail, and - view selections from a library of recordings (Video On Demand).
to When a user has specified a party to call by indicating the destination's name, IP address or other identification, the Directory is checked. It is possible to determine if a destination will accept a call without actually calling; so, since it can be determined that the destination will accept a call, the originator's video client can be told to connect to the destination. If the 15 callers are using a WWW browser (e.g. Netscape Navigator, Microsoft Internet Explorer, internetMCI Navigator, etc.) to access the VMDI, then a call can be automatically initiated using Java, JavaScript or Helper App. If a call cannot be completed, there will be a choice to leave video-mail.
D. Recording Video-Mail, Store ~ Forward Video and 2o Greetings If an Agent determines that a destination party is not available (off-Line, busy, no answer, etc.), the Video Mail Server plays an appropriate Video-Mail greeting for the owner of the destination number 8. The caller then leaves a video-message, which is stored on the Video Mail Server. The 25 recording of video for Store & Forward (S&F) Video is exactly the same as leaving a video-message, described above. Parameters such as destination number, forwarding time, and any other audio S8vF features currently available are entered through the VMDI or communicated with a human video operator (or automated video ARU.) Customers may record their own personalized greetings to greet callers that cannot reach them because they are busy or do not answer. This is accomplished in a manner similar to leaving Video-Mail, through the VMDI
or communicated with a human video operator.
2 3 ~.

E. Retrieving Video-Mail and Video On Demand Users have the choice of periodically polling their video-mail for new messages, or having the video-mail server call them periodically when they s have a new message waiting. Configuration is done through the VMDI or human video operator. Managing and checking video-mail is also performed through the VMDI or communicated with a human video operator. A choice of video to view for Video On Demand (VOD) is provided through the VMDI. These videos can be previously recorded video-conferences, training videos, etc. and are stored on the Video Content Engine.
F. Video-conference A user can navigate through the VMDI or Internet 10 WWW forms, or ~5 communicate with a human video operator to schedule a conference in the Conference Space. The information is stored on the Conference Reservation Engine 8. The other conference participants are notified of the schedule with a video-mail, e-mail message or otherwise. An optional reminder is provided for all registered conference participants at a particular time (e.g.
20 1 hour before the conference), through video-mail (or e-mail, voice-mail, paging service or any other available notification method).
G. Virtual Reality For multiple party conferences, a virtual meeting place can be generated by zs the Virtual Reality Space Engine. The implementation of the interface includes an embodiment based on VRML. Each person is in control of an - "avatar." Each avatar can have many different features such as visual representation (static representation or live video "head") and audio (voice or music). Data exchange and collaboration are all actions that can be 3o performed in each VR conference room. The private MBONE network allows the mufti-casting of conference member's data streams. Since everyone has a different view when interacting in VR-space, the VR Space Engine can optimize the broadcast of everyone's incoming H.263 streams to everyone else by mufti-casting only those avatar streams in view for each ~ 3 3-particular avatar.
XIV. VIDEO-CONFERENCING ARCHITECTURE
MCI Video-Conferencing describes an architecture for multimedia communications including real-time voice, video and data , or any s combination, including video telephony. The architecture also defines inter-operation with other video-conferencing standards. The architecture also defines multipoint configurations and control, directory services and video mail services.
A. Features 1o Video-Conferencing architecture is a multimedia services system and is designed to provide a number of features and functions including, Point-to-Point Video Telephony Multimedia video-conferencing with a MCU for control and multimedia information processing t s Support for Gateways for interworking with other video-conferencing systems based on ITU H.320 and ITU H.324 standards Support for real-time voice, video and data or any combination Multimedia information streams are transported between the end-user terminals using standard transport protocol RTP
zo Support for dynamic capability exchange and mode preferences, like ITU
H.263 video and ITU 6.723 audio, between end-user terminals Figure 19C illustrates a Video-Conferencing Architecture in accordance with a preferred embodiment. The components and details of the video-2s conferencing architecture are detailed below.
B. Components The Video-Conferencing System is comprised of a set of components including, 3o End-User Terminals LAN Interconnect System ITU H.323 Server Support Service Units ~ 36 1. End-User Terminals.
The end-user terminals are the end points of communication. Users communicate and participate in video conferences using the end-user terminals. End-user terminals, including ITU H.323 terminals 1 8y 8, ITU
- s H.320 terminal 9 and ITU H.324 terminal Z0, are interconnected through the ITU H.323 Server which provides the call control, multi-point control and gateway functions. End-User terminals are capable of multimedia input and output and are equipped with telephone instruments, microphones, video cameras, video display monitors and keyboards.
2. LAN Interconnect System.
The LAN Interconnect System 3 is the interface system between the MCI
Switch Network 2 and the different H.323 Systems including H.323 Server 4, Video Content Engine 5, Video Mail Server 6 and also the H.323 Directory Server 7.
End-User terminals participating in video-telephony sessions or video-conferencing sessions establish communication links with the MCI switch network and communicate with the H.323 Server through the LAN
Interconnect System. The LAN Interconnect system provides ACD-like 2o functionality for the H.323 video-conferencing system.
3. ITU H.323 Server.
The H.323 Server 4 provides a variety of services including call control, multipoint control, multipoint processing, and gateway services for 2s interworking between terminals supporting different video-conferencing standards like ITU H.320 and ITU H.324.
The H.323 Server is comprised of a set of individual components which - communicate with each other and with the other external systems like end-3o user terminals, video mail server and H.323 directory server. The different components of the H.323 Server include:
H.323 Gatekeeper Operator Services Module H.323 Multipoint Control Unit (MCU) ~3 H.323 Multipoint Control Unit (MCU) H.323 Gateway 4. Gatekeeper.
The H.323 Gatekeeper provides call control services to the H.323 terminals and Gateway units. The Gatekeeper provides a variety of services including:
Call Control Signaling with terminals, gateways and MCU;
Admissions Control for access to the video-conferencing system;
Call Authorization ;
1o Bandwidth control and management;
Transport Address Translation for translating addresses between different kinds of interworking video-conferencing systems;
Call Management of on-going calls;
Interfaces with the Directory Server[7] to provide directory services; and Is Interfaces with the Video Mail Server(6] for video mail services.
The Gatekeeper uses the ITU H.225 stream packetization and synchronization procedures for the different services, and is tightly integrated with the Operator Services Module for offering manual operator 2o services.
5. Operator Services Module.
The Operator Services Module offers manuai/automatic operator services and is tightly integrated with the gatekeeper. The manual or the automatic 25 operator terminal, located elsewhere on the LAN, interacts with the gatekeeper through the Operator Services Module to provide all the required operator services.
6. Multipoint Control Unit (MCU).
The MCU is comprised of the Multipoint Controller and the Multipoint 3o Processor and together provides multipoint control and processing services for video-conferences. The multipoint controller provides control functions to support conferences between three or more terminals. The multipoint controller carries out capabilities exchange with each terminal in a multipoint conference. The muitipoint processor provides for the 23~' processing of audio, video and/or data streams including mixing, switching and other required processing under the control of the multipoint controller. The MCU uses ITU H.245 messages and methods to implement the features and functions of the multipoint controller and the multipoint s processor.
7. Gateway.
The H.323 Gateway provides appropriate translation between the various transmission formats. The translation services include, to Call Signaling message translation between H.225 and H.221 which is the part of the H.320 system;
Communication procedures translation between H.245 and H.242; and Translation between the video, audio and data formats like H.263, H.261, 6.723, 6.728 and T.120.
t5 The H.323 Gateway provides conversion functions for transmission format, call setup and control signals and procedures.
8. Support Service Units.
The Support Service Units include the H.323 Directory Server ?, the Video-2o Mail Server 6 and the Video Content Engine 5 which interact with the H.323 Server for providing different services to the end-user terminals. The H.323 Directory Server provides directory services and interacts with the gatekeeper unit of the H.323 Server. The Video Mail Server is the repository of all the video mail generated by the H.323 system and interacts 2s with the gatekeeper unit of the H.323 server for the creation and playback of video mail. The Video Content Engine is the repository of all other types of video content which can be served to the end-user terminals. The Video Content Engine interacts with the gatekeeper unit of the H.323 Server.
30 C. Overview The H.323 based video-conferencing architecture completely describes an architecture for multimedia communications including real-time voice, video and data, or any combination including video telephony. Users with H.323 terminals can participate in a multimedia video-conferencing z39 session, a point-to-point video telephony session, or an audio only session with other terminal users not equipped with video facilities. The architecture also includes gateways for interworking with other video-conferencing terminals based on standards like ITU H.320 and ITU H.324.
s The architecture includes a directory server for offering complete directory services including search facilities. A video mail server is an integral part of the architecture providing for the recording and playback of video mail. A
video content engine is also part of the overall architecture for offering to multimedia content delivery services.
H.323 terminals participating in a video-conferencing or a video telephony session communicate with the H.323 server through the MCI switch network. The H.323 server offers a variety of services including call control, is information stream delivery, mufti-point control and also gateway services for interworking with H.320 or H.324 terminals. The server also offers directory services and video mail services.
A H.323 terminal initiating a video call establishes a communication link 2o with the H.323 Server through the MCI switch network. On admission to the network by the H.323 server, the server offers a directory of other available terminals to the call initiating terminal which selects a destination terminal or a destination group to participate in a video conference. The server then sets up a communication link with the selected destination 2s terminal or terminals and finally bridges the calling terminal and the called terminal/terminals. If the destination terminal is unavailable or busy, the server offers the calling terminal an option to deposit a video mail. The server also notifies the recipient of the video mail and offers the recipient services for retrieval of the video mail on-demand. Additional services like 3o content delivery on-demand to H.323 terminals are also offered and controlled by the H.323 server.
D. CaII Flow Example The Call Flow for the H.323 architecture based video-conferencing is a Sic ... ......._...~.~.~.". ....-.....- ,.._..... ..... .. . ~, ~. ... . ...

including calls to other H.323, H.320 and H.324 terminals; and Multipoint Video-Conference Calls.
Figure 19C illustrates various call flows in accordance with a preferred embodiment.
1. Point-to-Point Calls.
a) Case 1: H.323 Terminal to another H.323 Terminal A call initiating H.323 terminal 1 initiates a call to another H.323 terminal[8] through the MCI Switch Network. The gatekeeper is involved in to controlling the session including call establishment and call control. The Terminal end-user interface is any commercially available Web-browser.
Calling terminal 1 initiates a dial-up call to the MCI Switch network;
the call is terminated on the H.323 Gatekeeper module of the H.323 Server 4 through the LAN Interconnect 3 system;
a PPP link is established between the calling terminal and the Gatekeeper 4 on a well-know unreliable transport address/port;
Calling terminal sends a admission request message to the Gatekeeper[4]
The Gatekeeper 4 sends an admission confirm message and communicates 2o with the Directory Server 7 and sends back directory information to calling terminal for display at the calling terminal, and the directory information is displayed as a web-page along with a choice of calling modes including Point-to-Point or Conference mode;
the admissions exchange is followed by the setting up of a reliable connection for H.225 call control messaging on a well known port;
the terminal user chooses the point-to-point mode and also chooses the destination of the call. This is the setup request message;
the gatekeeper 4 together with the operator services module/ operator proceeds with calling the called terminal 8 with a setup request;
3o if setup request fails, the gatekeeper 4 informs the calling terminal 1 of the failure and provides an option for the calling terminal 1 to leave a video mail;
if the user at calling terminal 1 chooses to leave a video mail for user at the destination terminal 8, the gatekeeper 4 establishes a connection 2~/

if the user at calling terminal 1 chooses to leave a video mail for user at the destination terminal 8, the gatekeeper 4 establishes a connection with the Video Mail Server 6 and receives a reliable port address from the mail server 6 for a H.245 connection;
s the gatekeeper 4 additionally establishes a connection for H.225 call control with the video mail server 6.
the gatekeeper 4 in-turn sends a reliable port address to calling terminal 1 for H.245 control channel. The gatekeeper 4 may be involved in H.245 control channel communications;
1o the calling terminal 1 establishes a reliable connection for H.245 control channel and H.245 procedures like capability exchange, mode preferences, etc. are carried out;
after the capabilities exchange, H.245 procedures will be used to establish logical channels for the different media streams;
15 the capabilities exchange also involves determination of dynamic port addresses for the transport of the different media streams;
the media streams are transported over the dynamic ports in the various logical channels;
once the terminal has completed the video mail, it closes the logical zo channel for video after stopping transmission of the video stream;
data transmission is stopped and logical channel for data is closed;
audio transmission is stopped and logical channel for audio is closed;
H.245 call clearing message is sent to the peer entity;
calling terminal 1 transmits a disconnect message on the H.225 port to the 2s gatekeeper 7 which in turn sends the disconnect message to the video mail server 6;
the disconnect messages are acknowledged and the call is disconnected;
if the setup request is a success, called terminal 8 responds with a connect message which include a reliable port address for H.245 connection;
3o the gatekeeper 4 responds to the calling terminal 1 with the connect message along with the port address for the H.245 control channel communications;
calling terminal 1 sets up a connection for H.225 call control signaling with the gateway 4, establishes another connection for H.245 control z ~u z ,.

channel communications and responds to the gateway 4 with connect acknowledgment message;
the gatekeeper 4 in-turn sends the connect acknowledgment message to called terminal8.
called terminal 8 now sets up a H.225 call control connection and also establishes another connection for H.245 with the gatekeeper 4 for control channel communications;
the terminals, having established a H.245 control channel for reliable communication, exchange capabilities and other initial procedures of H.245, and an audio channel may be optionally opened before the capabilities exchange;
following the capabilities exchange, logical channels over dynamic ports are established for each of the media streams;
once the media logical channels are open over dynamic ports, media ~5 information can be exchanged;
during the session, H.245 control procedures may be invoked for changing the channel structure like mode control, capability, etc.;
also H.225 control channel is for specific procedures as requested by the gatekeeper[4] including call status, bandwidth allocation, etc.;
2o for termination, either terminal may initiate a stop video message, discontinue video transmission and then close the logical channel for video;
data transmission is discontinued and the logical channel for data is closed;
25 audio transmission is discontinued and logical channel for audio is closed;
H.245 end session message is sent and transmission on the control channel is stopped and the control channel is closed;
terminal receiving the end session message will repeat the closing . procedures and then H.225 call signaling channel is used for call 3o clearing; and terminal initiating the termination will send a disconnect message on the H.225 control channel to the gatekeeper 4 which in turn sends a disconnect message to the peer terminal. The peer terminal acknowledges the disconnect which is forwarded to the initiating ~~3 terminal and the call is finally released.
b) Case 2: H.323 Terminal to H.320 Terminal A call initiated from a H.323 terminal 1 invokes a call to a H.320 terminal 9 through an MCI Switch Network. The gatekeeper along with the gateway is involved in controlling the session including call establishment and call control. A terminal end-user interface is any of the commercially available Web-browsers or a similar interface.
The call flow is similar to a H.323 terminal calling another H.323 terminal as explained in the previous case except that a gateway 4 component is introduced between the gatekeeper 4 and the called terminal 9. The gateway transcodes H.323 messages including audio, video, data and control to H.320 messages and vice-versa. If the H.320 terminal 9 initiates ~s a call to a H.323 terminal[1], the initial dial-up routine is performed by the gateway and then the gatekeeper takes over the call control and the call proceeds as explained in the previous case.
c) Case 3: H.323 Terminal to H.324 Terminal 2o Call initiating H.323 terminal 1 initiates a call to a H.324 terminal 10 through the MCI Switch Network. The gatekeeper along with the gateway is involved in controlling the session including call establishment and call control. The Terminal end-user interface is a Web-browser or a similar interface.
25 The call flow is similar to a H.323 terminal calling another H.323 terminal as explained in the previous case except that a gateway 4 component is introduced between the gatekeeper 4 and the called terminal 9.
The gateway 4 transcodes H.323 messages including audio, video, data and control to H.324 messages and vice-versa.
If the H.324 terminal 10 initiates a call to a H.323 terminal 1, the initial dial-up routine is performed by the gateway and then the gatekeeper takes over the call control and the call proceeds as explained in the previous case.

2. Multipoint Video-Conference Calls.
In the case of multipoint video-conference, all the terminals exchange initial call signaling and setup messages with the gatekeeper 4 and then are s connected to the Multipoint Controller 4 for the actual conference including H.245 control channel messaging through the gatekeeper 4.
The following are the considerations for setting up a conference:
After the initial admission control message exchange, the users are presented with a web page with information about conference type to and a dynamic list of participants.
Participants joining later are presented with a web page with conference information and also are requested to enter authentication information All users get connected to the multipoint controller[4] through the ~ s gatekeeper[4]
The multipoint controller[4] distributes information among the various participants E. Conclusion 2o The video-conferencing architecture is a total solution for multimedia communications including real-time voice, video and data, or any combination, including point-to-point video telephony. The architecture defines interworking with other systems utilizing ITU recommendations.
25 Additional services including directory services and video mail services are also part of the overall architecture.
XV. VIDEO STORE AND FORWARD ARCHITECTURE
3o The Video Store and Forward Architecture describes a video-on-demand content delivery system. The content may include video and audio or audio only. Input source for the content is from the existing video-conferencing facility of MCI or from any video/audio source. Input video is stored in a Digital Library in different standard formats like ITU H.320, ITU H.324, ITU
ass H.263 or MPEG and delivered to the clients in the requested format.
Delivery is at different speeds to the clients either on the Internet or on dial-up lines including ISDN and with a single storage for each of the different formats.
s A. Features The Video Store and Forward Architecture is designed with a rich set of features and functionality including:
Delivers Video and Audio on demand;
1o Supports different compression and transmission standards including ITU
H.320, ITU H.324, MPEG and ITU H.263 on both IP (Internet Protocol) and RTP (Real Time Transport Protocol);
Supports content delivery on the Internet, by dial-up ISDN lines and by low speed (28.8kbps) Analog Telephone lines;
t5 Supports single source of content and multiple storage and delivery formats and multiple delivery speeds; and Supports Content Management and Archival in multiple formats.
B. Architecture 2o Figure 19D is a Video Store and Forward Architecture in accordance with a preferred embodiment.
C. Components The Video Store and Forward architecture can be completely described by the following components.
25 Content Creation and Transcoding.
Content Management and Delivery.
Content Retrieval and Display.
1. Content Creation and Transcoding.
3o Input sources include analog video, video from Multi-Point Control Unit (MCU) and other video sources 1a and lb. Input content is converted to standard formats like ITU H.261, ITU H.263, ITU H.320, ITU H.263, ITU
H.324, MPEG and also formats to support delivery of H.263 over RTP and H.263 over an Internet Protocol 2 and 3. Input can initially be coded as H.263 and optionally transcoded into the various other formats and stored 2. The transcoded content is stored on different servers, one for each content type to serve the various clients each supporting a different format Sa, Sb, 5c, 5d, 5e and 5~
a S
2. Content Management and Delivery.
Content is stored on different servers with each server supporting a specific format and is managed by a Digital Library consisting of:
- Index Server for managing the indexes and archival of content 4, io - Object Servers for storage of content Sa, 5b, 5c, Sd, Se and 5f, - Proxy Client as a front end to the Index and Object Server and interacting with the different clients requesting for content 6.
Content Delivery is by:
15 - Internet, - Dial-up ISDN lines, - Dial-up Analog Telephone lines at 28.8kbps, and Content format is either a MPEG Stream, H.320 Stream, H.324 Stream, or 2o a H.253 Stream transported over IP or RTP.
3. Content Retrieval and Display.
Content Retrieval is by clients supporting various formats:
- MPEG Client - ?a;
25 - ITU H.263 Client supporting RTP - ?b;
- ITU H.263 Client supporting IP - ?c;
- ITU H.320 Client - 7d; and - ITU H.324 Client - ?e.
3o Content is retrieved by the different clients on demand and displayed on a local display.
Clients support VCR like functions like fast-forward, re-wind, etc.

D. Overview Analog Video from different sources and H.320 video from an MCU is received as input and transcoded into various formats as required Iike ITU
H.324, ITU H.261, ITU H.263 or MPEG and stored on the different Object s Servers dedicated for each of the formats. The Object Servers are in turn managed by the Index Server and are together called a Digital Library. Any request from the clients for content is received by the Index Server and in turn serviced by the Object Server through a Proxy Client.
to The Index Server or the Library Server respond to requests from the proxy client and store, update and retrieve objects like H.261, H.263 or MPEG
multimedia information on the object servers. Then they direct the object server to deliver the retrieved information back to the proxy client. The Index Server has the complete index information of all the different objects is stored on the object servers and also information on which of the object server the information is residing on. The index information available on the Index Server is accessible by the proxy client for retrieval of multimedia content from the different object servers. Security and access control is also part of the index server functionality.
The Object Servers are an integral part of the Digital Library providing physical storage and acting as the repository for the multimedia content, including the video-conferencing information stream from the conferencing facilities. The multimedia content is stored in standard formats which can be retrieved by the proxy client on demand. Each of the Object Servers are dedicated for a specific format of multimedia content like H.26I, H.263, MPEG, etc. The organization and index information of the multimedia content including information about the specific object server dedicated for a multimedia format is managed by the index server. The Object Server 3o delivers the stored multimedia content to the proxy client upon receiving specific instructions from the index server.
The Proxy Client is the front end of the digital library and is accessed by all the clients through the Internet for on-demand multimedia content. The Proxy Client also is a World Wide Web (WWW) Server and delivers a page to the clients when accessed. The clients interact with the Proxy Client and thereby with the Digital Library through the WWW pages. Clients request multimedia content by interacting with the WWW pages. The Proxy Client s receives the request from the clients through the WWW pages and processes the request. The Proxy Client then communicates with the index server with object queries as requested by the client. The index server then communicates with one of the object servers dedicated to the requested multimedia format and, based on the index information available at the 1o index server, directs the object servers to deliver the requested multimedia content to the Proxy Client. The Proxy Client receives the multimedia content from the object server and delivers it to the client making the request.
is The Clients connect to the Servers either through the Internet or by dial-up connections on an ISDN line or an Analog line at 28.8 Kbps depending on the video format requested and the client capabilities. A H.320 client connects by an ISDN line and a H.324 client requests services on an analog telephone line at 28.8 Kbps. A MPEG client or a H.263 client using RTP or 2o a H.263 client using IP request services through the Internet. The front-ends for multimedia content query and display like the WWW browsers are integrated as a part of the Client and provide an easy-to-use interface for the end-users.
25 A request for video from the client is received by the proxy client which routes the request to the Index Server which is turn processes the request and communicates with a specific Object Server in addition to indexing the content for delivery. The Object Server delivers the requested content to the client through the Internet. In the case of the dial-up links, the content 3o is delivered back on the already established link.
In sum, the Video Store and Forward architecture describes a comprehensive system for the creation, transcoding, storage, archiving, management and delivery of video and audio or audio on demand. The delivery of video and audio or audio will be on the Internet or by ISDN or Analog Telephone dial-up lines. Content including video and audio or audio is delivered at various data rates from individual storage locations, each serving a different delivery speed.
s XVI. VIDEO OPERATOR
A. Hardware Architecture Figure 96 shows the system hardware for allowing a video operator to participate in a video conference or video call, providing numerous services to to the video callers. Among the services provided are: answering incoming video calls or dialing out to customer sites; accessing a system for maintaining video conference schedules, joining callers using Bandwidth on Demand Interoperability Group ("BONDING") calls or International Telecommunication Union-Telecommunication Standardization Sector 15 ("ITU-T") standard H.320 Multi-rate Bearer Service (MRBS) Integrated Services Digital Network ("ISDN") calls into a video conference or video call;
monitoring, viewing and recording any video conference or video call;
playing back video conferences or video calls recorded earlier; and offering assistance to or responding to inquiries from video conference callers 2o during video conferences or video calls.
The system hardware is comprised of a Video Operator Terminal 40001, a Call Server 40002, a multimedia hub ("MM Hub") 40003, wide area network hubs ("WAN Hubs") 40004, a multi-point conferencing unit 25 ("MCU") 40005, a BONDING Server 40006, a Client Terminal 4000?, and a switching network ("MCI") 40008.
In one embodiment, the Video Operator Terminal 40001 is a Pentium-based personal computer with a processing speed of 90 MHz or greater, 3o 32MB RAM, and a hard disk drive with at least 1.OGB storage space. The operating system in this embodiment is Microsoft's Windows 95. Special features include Incite Multimedia Communications Program ("MCP") software, an H.320 video coder/decoder ("codec") card for audio and video compression (e.g. Zydacron's 2240 codec), and an isochronous Ethernet ~ ,~ o ,.

("isoEthernet") network interface card. Incites MCP manages the isoEthernet network interface card to create the equivalent of 96 ISDN B-channels in isochronous channels for transmission of video signals.
The Call Server 40002 in this embodiment is a Pentium-based personal computer with a processing speed of 90 MHz or greater, 32MB RAM, and a hard disk drive with at least 1.OGB storage space. The operating system is Microsoft's Windows NT Server. Special features include the Incite Call Server services and an Ethernet network interface card.
to Different embodiments of the system accommodate any model of MM Hub 40003 and any model of WAN Hub 40004. In one embodiment, the MM
Hub 40003 is the Incite Multimedia Hub, and the WAN Hub is the Incite WAN Hub. The MM Hub 40003 is a local area network ("LAN") hub that connects, via numerous ports supporting isoEthernet interfaces each with a bandwidth consisting of 96 full-duplex B-channels, to personal computers such as the Video Operator Terminal 40001 and the BONDING Server 40006, to WAN Hubs 40004, or to other cascaded MM Hubs. In addition, the MM Hub 40003 can accept up to ten Mbps of Ethernet data via an 2o Ethernet interface such as the one from the Call Server 40002. The WAN
Hub 40004 acts as an interface between an MM Hub 40003 and a public or private switched network such as MCI 40008, enabling video conferencing to extend beyond the WAN or LAN containing the MM Hub 40003 and WAN Hub 40004. , Different embodiments of the system also accommodate various manufacturers' MCU 40005 devices. The function of an MCU 40005 is to allow video conference callers using a variety of different devices, possibly communicating over different circuit-based digital networks, to 3o communicate with one another in a single video conference. For example, one embodiment employs VideoServer's Multimedia Conference Server ("MCS"), which mixes audio to allow any one video conference caller to hear the complete video conference discussion and processes video to allow each video conference caller to see all other callers simultaneously.
~ S-/

In one embodiment, the BONDING Server 40006 is a Pentium-based personal computer with a processing speed of 90 MHz or greater, 32MB
RAM, and a hard disk drive with at least 1.OGB storage space. The s operating system in this embodiment is Microsoft's Windows 95. Special features include Incite BONDING Server software, a Digital Signal Processor ("DSP") card (such as Texas Instrument's "TMS320C80" DSP), and an isoEthernet network interface card. Where a Client Terminal 40007 makes BONDING or Aggregated video calls, the BONDING Server 40006 converts ~o the calls to mufti-rate ISDN calls used within the video operator platform.
In a preferred embodiment, the Client Terminal a Pentium-based personal computer with a processing speed of 90 MHz or greater, 32MB RAM, and a hard disk drive with at least 1.OGB storage space. The operating system is ~s Microsoft's Windows 95 in this embodiment, and the Client Terminal 40007 is equipped with audio and video equipment making it compatible with ITU-T standard H.320.
In this embodiment, the switching network is an integrated services digital 2o network ("ISDN") provided by MCI 40008.
The Video Operator Terminal 40001 is connected to the MM Hub 40003 via an isoEthernet interface with a bandwidth of 96 full-duplex B-channels, which allows each video operator to manage up to eight video conferencing Zs clients, each client employing a Client Terminal 4000?. The MM Hub 40003 is connected to WAN Hubs 40004 via similar isoEthernet local area network ("LAN") connections. One WAN Hub 40004 connects through MCI
40008 to an MCU 40005 via mufti-rate ISDN interfaces. Another WAN
Hub 40004 connects to MCI 40008 via a mufti-rate ISDN interface, and 3o MCI connects to each Client Terminal 40007 via a BONDING or mufti-rate ISDN interface. In a three-way connection, the MCU 40005, the Cali Server 40002 and the MM Hub 40003 are connected to one another through an Ethernet wide area network ("WAN") 40009. The MM Hub 40003 is also connected to a BONDING Server 40006 via an isoEthernet interface with a bandwidth of 248 B-channels in full "iso" mode.
B. V~Ideo Operator Console . Figure 9? shows one embodiment of the system for enabling a video operator to manage video conference calls, which includes a Video Operator Console system 40101 and external systems and interfaces 40108 through - 401 I7.
The Video Operator Console system 40101 is comprised of a Graphical User Interface ("GUI") 40102, a Software System 40103 and a Media Control system 40107. The GUI 40102 interacts with both the Software System 40103 and the Media Control system 40107 to allow a video operator to perform all functions of the video operator invention from the Video Operator Terminal [40001 Figure 96J using the Video Operator is Console system 40101.
The Software System 40103 implements the following systems: a Scheduling system 40104 which manages the video operator's schedule; a Recording and Playback system 40105 which records the audio and video 2o input from any call and plays back audio and video input through any call;
and a Call System Interface 40106 which acts as an application program interface with the Incite MCP application to manage individual calls by performing switching functions such as dial and hold.
2s The Scheduling system 40104 is connected via an Open Database Connectivity ("ODBC") interface 40108 to a Video Operator Shared Database 40111, which is in turn connected via an interface between VOSD and VRS 40114 to a Videoconference Reservation System ("VRS") 40115. The VRS 40115 submits video conference schedules, conference 3o definitions and site definitions to the Video Operator Shared Database 40111 via the interface 40114 either on a regular basis or on demand by a database agent system within the Video Operator Shared Database 40111.
The Video Operator Shared Database 40111, residing in a different computer from that containing the Video Operator Console 40101 in a :253 preferred embodiment, stores all conference and site information such that each Video Operator Console 40101 can retrieve the necessary conference and site configurations for any video conference call. In an alternative embodiment of the external systems associated with the internal s Scheduling system 40104, the Video Operator Shared Database 40111 and VRS 40115 may be merged into a single system.
The Recording and Playback system 40105 communicates via a Dynamic Data Exchange ("DDE"), Object Linking and Embedding ("OLE") or Dynamic to Link Library ("DLL") interface 40109 with a Video Operator Storage and Playback system 40112 located locally in the Video Operator Terminal [4000? Figure 96). The Video Operator Storage and Playback system is comprised of a uni-directional recording device 40116 conforming to ITU-T
standard H.320 and a uni-directional playback device 40117 conforming to 1s ITU-T standard H.320. Conference calls are recorded by transmitting the digitized audio and video signals from the Video Operator Console 40101 to the H.320 recorder 40116. Conference calls are played back by retrieving a previously recorded conference call from disk storage and transmitting the audio and video signals from the H.320 playback device 40117 to the Video 2o Operator Console.
The Call System Interface system 40106 communicates via a DDE interface 40110 with the Incite MCP application 40113 to manage switching functions such as dial, hold, etc.
The Media Control system 40107 allows the GUI 40102 to communicate directly with external components to manage the GUI 40102 presentation of audio and video. In the embodiment shown in Figure 401, the Media Control system 40107 communicates via a DDE interface 40110 with the 3o Incite MCP application 40113. The Incite MCP application 40113 provides all necessary call setup features and multimedia features such as video window placement and audio control through the DDE interface 40110 to the internal Media Control system 40107, and on to the GUI 40102.

Figure 98 shows a second embodiment of the system for enabling a video operator to manage video conference calls, which includes a Video Operator Console system 40101 and external systems and interfaces 40108 through 40117 and 40203 through 40216. In this embodiment, however, the s Software System 40103 is compatible with not only VideoServer's "MCS"
40215 MCU, but also other manufacturers' MCU applications. Thus the internal software system MCU control 40201, the external software system MCU Control System 40208, the MCUs themselves 40214 and 40215, and the interfaces between them 40206, 40210 and 40211, appear in Figure 98. In addition, because not only the Incite MCP 40113 application but also "Other programs with call control interfaces" 40216 may provide necessary call setup and multimedia features in this embodiment, the external Call Control System 40209 is necessary, as are the intervening DDE, OLE or DLL interfaces 40207, 40212 and 40213. This embodiment is also includes a Video Store and Forward system 40204 and its DDE, OLE
or DLL interface 40203. Finally, the second embodiment adds the internal software system Call Monitor 40202.
As in the first embodiment, the Video Operator Console system 40101 is 2o comprised of a GUI 40102 and a Software System 40103. However, in addition to the Scheduling system 40104, the Recording and Playback system 40105 and the Call System Interface 40106, the software system in the second embodiment includes the MCU control 40201 and the Call Monitor 40202.
The Scheduling system 40104 and associated external systems 40108, 40111, 40114 and 40115 are identical to the those in the first embodiment, pictured in Figure 9? and described above.
3o The internal MCU control 40201 communicates via a DDE, OLE or DLL
interface 40206 with the external MCU Control System 40208 to manage resources and features specific to various different MCU systems. The MCU Control System 40208 communicates either via a ConferenceTalk interface 40211 with the VideoServer MCS 40215 or via another vendor-zss specific interface 40210 with some Other MCU vendors' MCU 40214.
The Recording and Playback system 40105 communicates via DDE, OLE or DLL interfaces 40109, 40203 with both the Storage and Retrieval system s 40205 and the Video Store and Forward system 40204. The Storage and Retrieval system 40205 and Video Store and Forward system 40204 communicate via another DDE, OLE or DLL interface 40207 with the Call Control System 40209. The Call Control System 40209 communicates via another DDE, OLE or DLL interface 40212 with a uni-directional H.320 to recorder 40116 and a uni-directional H.320 playback device 4011?.
Conference calls recorded by transmitting the digitized audio and video signals from the Video Operator Console 40101 through the Storage and Retrieval system 40205 and Call Control System 40209 to the H.320 recorder 40116. Conference calls are played back by retrieving a 1 s previously recorded conference call from disk storage and transmitting the audio and video signals from the H.320 playback device 40117 through the Call Control System 40209 and Storage and Retrieval system 40205 to the Video Operator Console 40101. The Video Store and Forward system 40204 operates in a manner similar to the Storage and Retrieval system 20 40205, communicating between the Recording and Playback system 40105 and the Call Control System 40209.
The call monitor 40202 monitors the state of calls and connections by regularly polling the Call System Interface 40106 within the Video Operator 2s Console Software System 40103. The Cail System Interface 40106 communicates via a DDE, OLE or DLL interface 40207 with the Call Control System 40209 to manage call data, including switching functions such as dial, hold, etc., translating between the Video Operator Console 40101 internal data structures and the Call Control System 40209 data.
3o The Call Control System, in turn, manages either the Incite MCP 40113 or Other programs with call control interfaces 40216.
The Media Control system 4010? communicates via a DDE, OLE or DLL
interface with the Call Control System 40209, which communicates via a z,r6 . ..-_~..~.....~-_._. . _ . .. , , , WO 98/47298 PCTlUS98I07927 DDE interface 40110 with the Incite MCP application 40113 or with Other programs with call control interfaces 40216. The Incite MCP application 40113 provides all necessary call setup features and multimedia features such as video window placement and audio control either directly through a DDE interface 40110 to the internal Media Control system 40102 or via the Call Control System 40209. If Other programs with call control interfaces 40216 are used to provide call setup and multimedia features, they communicated with the Media Control system 40107 via the Call Control System 40209.
to C. Video Conference Call Flow Figure 99 shows how a video conference call initiated by the video operator is connected through the system pictured in Figure 96. In the first step, illustrated by call flow path 40301, the video operator initiates a call from is the Video Operator Terminal 40001 through the MM Hub 40003 to the BONDING Server 40006, where the BONDING Server 40006 converts the call to a BONDING call. In the second step, illustrated by call flow path 40302, the BONDING Server 40006 transmits the BONDING call through the MM Hub 40003 once again, through a WAN Hub 40004, through MCI
zo 40008, and to the Client Terminal 40007. This step is repeated for each Client Terminal 40007 that will participate in the video conference. In the third step, illustrated by call flow path 40303, the video operator initiates a call from the Video Operator Terminal 40001 through the MM Hub 40003, through a WAN Hub 40004, through MCI 40008, and to the MCU 40005.
25 In the fourth step, illustrated by call flow path 40304, the video operator uses the Video Operator Terminal 40001 to bridge the connections to the Client Terminal 40007 and MCU 40005. Each time the video operator calls a conference call client at its Client Terminal 40007, the MCU's ANI
for the particular conference site is passed in the Calling Party Field to 3o identify each client participating in the conference call with the correct conference site. When the MCU is called, the clients' ANI are passed. The MCU can then identify the correct conference site for each call.
In an alternate embodiment, the client initiates a BONDING call from the Client Terminal 40007 through MCI 40005, through a WAN Hub 40004, through the MM Hub 40003, through the BONDING Server 40006, and through the MM Hub 40003 once again to the Video Operator Terminal 40001. The video operator then places a call to the MCU as illustrated in s call flow path 40303 and finally bridges the two calls as illustrated in call flow path 40304. To determine the correct conference site for the client-initiated call, the initiating client's ANI is passed to the MCU when the connection is made by the video operator.
1o While a conference call is in progress, the video operator monitors each of the calls from the Video Operator Terminal 40001. Functions of the video operator include monitoring which calls remain connected, reconnecting disconnected calls, adding new clients to the conference, or joining the conference to inform the clients regarding conference status.
is All calls are disconnected to end a conference, and the video operator shared database j40214 in Figure 98] reflects an updated conference schedule.
2o D. Video Operator Software System 1. Class Hierarchy.
Figure 100 shows the class hierarchy for video operator software system classes. In one embodiment using the Visual C++ programming language, the VOObject 40401 class is extended from the Visual C++ base class 2s CObject. VOObject 40401 is a Superclass to all classes of objects in the internal software system for the video operator console system, such that all objects in the internal software system inherit attributes from VOObject 40401.
3o VOOperator 40402 is an assembly class associated with one VOSchedule 40403 Part-1 Class object and one VOUserPreferences 40404 Part-2 Class object, such that exactly one VOSchedule 40403 object and exactly one VOUserPreferences 40404 object are associated with each VOOperator 40402 object. VOSchedule 40403, in turn, is an Assembly Class a~-~;

associated with zero or more VOSchedulable 40405 Part-1 Class objects, such that any number of VOSchedulable 40405 objects may be associated with each VOSchedule 40403 object.
VOSchedulable 40405 is a Superclass to the VOConference 40406 Subclass-1 and the VOPlaybackSession 40407 Subclass-2, such that the VOConference 40406 object and the VOPlaybackSession 40407 object inherit attributes from the VOScheduiable 40405 object. VOConference 40406 is an Assembly Class associated with two or more VOConnection Io 40412 Part-1 Class objects and zero or one VOPlaybackCall 40415 Part-2 Class objects, such that at least two VOConnection 40412 objects and possibly one VOPlaybackCall 40415 object are associated with each VOConference 40406 object. VOPlaybackSession 4040? is an Assembly Class associated with one VOPlaybackCall 40415 Part-1 Class object, such Is that exactly one VOPlaybackCall 40415 object is associated with each VOPlaybackSession 4040? object.
VOCaIIObjMgr 40408 is an Assembly Class for zero or more VOCaII 40410 Part-1 Class objects, such that any number of VOCaII 40410 objects may 2o be associated with each VOCalIObjMgr 40408 object. Similarly, VOConnObjMgr 40409 is an Assembly Class for zero or more VOConnection 40412 Part-1 Class objects, such that any number of VOConnection 40412 objects may be associated with each VOConnObjMgr 40409 object. VOConnection 40412 is an Assembly class for two VOCaIl 2s 40410 Part-1 Class objects, such that exactly two VOCaII 40410 objects are associated with each VOConnection 40412 object. VOCaII 40410 is a Superclass to the VOPlaybackCall 40415 Subclass-1, such that VOPlaybackCall 40415 objects inherit attributes from the VOCaII 40410 object. VOCaIl 40410 is also an Assembly Class associated with two 3o VOSite 40413 Part-1 Class objects, such that exactly two VOSite 40413 objects are associated with each VOCaIl 40410 object. Finally, the VOCaII
40410 class object uses the VORecorder 40411 class object.
VOSite 40413 is a Superclass to the VOMcuPortSite 4041? Subclass-1, the z.s-~

VOParticipantSite 40418 Subclass-2, and the VOOperatorSite 40419 Subclass-3, such that VOMcuPortSite 40417 objects, VOParticipantSite 40418 objects and VOOperatorSite 40419 objects inherit attributes from the VOSite 40413 object.
VOPlaybackCall 40415 is an Assembly Class associated with one VOMovie 40416, such that exactly one VOMovie 40416 object is associated with each VOPlaybackCall 40415 object. The VOPlaybackCall 40415 class object also uses the VOPlayer 40414 class object.
to VOMessage 40420 object has no associations other than inheriting the attributes of VOObject 40401, the Superclass to all objects in the internal software system.
is 2. Class and Object details.
a) VOObject All Internal Software System classes will inherit from the following base class. This base class is extended from the Visual C++ base class CObject.
2o Class VOObject Base Class CObject Inheritance Type public Friend Classes -25 ( 1 ) Data Types enum senderType_a { SENDER INTERNAL, SENDER SCHEDULE, SENDER_CONFERENCE, SENDER CONNECTION, SENDER CALL, SENDER TIMER ;;
3o enum messageType_e { MSG DEBUG, MSG _ERROR, MSG WARNING, MSG APPLICATION_ERROR, MSG _STATE UPDATE ~;
Delivery type flags: DELIVER_MESSAGE_QUEUE, DELIVER_LOG_FILE, DELIVER_MODAL_DIALOG, DELIVER_MODELESS DIALOG, 2E o DELIVER_CONSOLEOUTPUT
(2) Attributes Access Level Type Name Description s static VOOperatar* m_pV0 video operator pointer static VOSchedule* m_pSchedule scheduler pointer static VOCallObjMgr* m_pCa110M Call Object Manager pointer static VOConnectionObjMgr* m_pConnOM Connection Object Manager pointer io static VOCaIlSystem* m_pCallSys Call System Interface pointer (3) Methods (a) PostMessage virtual PostMessage (messageType_e type, int errCode, CString info="", ~s int delivery=(DELIVER MSG_QUEUE ~ DELIVER_LOG_FILE), senderType a senderType=SENDER INTERNAL, void*
sender=NULL);
(i) Parameters 2o type The type of message, as defined in the Data Types section errCode The error or warning code as defined in the application's resources.
Info Extra textual information to be passed as part of the message.
delivery Preferred method of message delivery. The delivery options are 2s shown in the Data Types section above. Default method of delivery is stored in the class member variable . m_delivery, which should be initialized to both DELIVER_MESSAGE_QUEUE and DELIVER LOG FILE
only.
3o senderType The message sender type, as defined in the Data Types section.
Sender A pointer to the object sending the message, i.e. this (ii) Description Use this function to create error, warning, debug, logging and notification z~ I

WO 98/47298 PCT/US98/0792?
messages. It will create a VOMessage object, which will then perform the appropriate actions as specified by the delivery flags.
(b) GetErrorString virtual CString GetErrorString (int errorCode);
Return Value: returns a CString object having the error string corresponding to the error code passed.
errorCode parameter: the error code for which you want the error string.
to Error strings are stored as resources.
This function is called to get a textual description corresponding to an error code.
b) Core Classes is (1) Class List Site Participant Site MCU Port Site 2o Video Operator Site Call Playback Call Movie Call Object Manager 25 Connection Connection Object Manager Message Video Operator 30 (2) Class Descriptions (a) Site This is a base class from which classes such as the Participant Site and MCU Port Site classes can be derived from. It's main purpose is to function as a data structure containing pertinent information about who or what is .2 6 -~.

WO 98/47298 PCT/tJS98/07927 taking part in a Call.
Class VOSite Base Class VOObject s Inheritance Type public Friend Classes (i) Data Types to enum Bandwidth_e { MULTIRATE, BONDING, AGGREGATED, HO j;
(ii) Attributes Access Level Type Name Description Cstring m name name of the site 15 ID t m_ID Unique site ID
ID_t m_locationID ID for physical location Cstring m timezone Time zone Cstring m dialNumber Numbers) to dial. See the Call System Interface section for multiple numbers format.
2o Bandwidth_e m_bandwidthUsage Bandwidth usage int m_maxNumChannels Maximum number of channels capable VOCaII* m_pCall pointer to Call object that this Site is a part of .
25 * Codec or Terminal Type (PictureTel, MCP, etc.) * Call Setup Type (dial-in, dial-out) (b) Participant Site 3o Inherits from VOSite base class.
All customers or conference participants will have their information stored in the VO shared database.
.~ ~ 3 Class VOParticipantSite Base Class VOSite Inheritance Type public Friend Classes -Attributes Access Level Type Name Description Cstring m_coordinatorName Site coordinator name 1o Cstring m coordinatorNbrSite coordinator telephone number ID_t m companyID ID of Company this Site belongs to VOMCUPortSite* m_pMCUPort MCU Port Site that is to be associated with in a Connection object (c) MCU Port Site Inherits from VOSite base class.
2o All conferences take place on an MCU. Each Participant Site needs to connect with a logical "port" on an MCU.
Class VOMcuPortSite Base Class VOSite 2s Inheritance Type public Friend Classes Attributes 3o Access Level Type Name Description ID_t m_mcuID ID to identify the MCU
VOParticipantSite* m_pParticipant Participant Site that is to be associated with in a Connection object .~ 6 ~

(dJ Video Operator Site Inherits from VOSite base class.
All calls will have the Video Operator Site as one of the sites in a point-to-s point call. This structure contains the real ANI of the video operator.
Class VOOperatorSite Base Class VOSite Inheritance Type public Friend Classes -Attributes Access Level Type Name Description is ID_t m operatorID Operator's ID
CString m voicePhone Operator's voice phone number ID_t m_groupID Operator's Group ID
ID_t m_superviserID Supervisor's ID
2o CObList m_Calls list of Call objects that this Site is a part of (ej Call A Call is defined as a full duplex H.320 stream between two sites. In all 2s Calls, the Video Operator Site will be one of the sites. A Joined pair of Calls is called a Connection.
Class VOCaII
Base Class VOObject 3o Inheritance Type public Friend Classes (i) Data apes ~6S' enum StateCall_a { ERROR, INACTIVE, INCOMING, DIALING, ACTIVE, DISCONNECTED, HELD, lastCallStates};
enum callOperation_e { ERROR, DIAL, ANSWER, HOLD, PICKUP, DISCONNECT, HANGUP, lastCallOperations }
(ii) Attributes Access Level Type Name Description ID t m ID call ID
VOSite* m_pSite other end of a call site (Participant, MCU
to Port or unknown) VOOperatorSite* m_pOperatorSite Operator site boolean m operatorInitiated TRUE if the call is initiated by the operator (default) CTime m startTime the actual time when the call t s became active boolean m_expectHangup flag that helps determine whether a Hangup is expected or not.
StateCall a m state state of the call StateCall a (nCallStatesj [nCallOperationsj m transitionTable state 2o transition table VORecorder* m_pRecorder recorder object for call VOConnection* m_pConnection pointer to Connection object this call belongs to.
25 (iii) Methods Disconnection(); is called when the other end of the line hangs up or the line goes dead. The member variable m expectHangup should be FALSE.
Otherwise, the Call Object Manager's Hangup() operation would have been called.
Reset(); resets the call state to an inactive state RecordingStart(); starts recording the H.320 input pipe of the Call.
~2 6G

RecordingStop(); stops the recording of the Call.
setState(callOperation_a operation);
_ operation parameter: indicates an operation that has been performed - s which will result in a change of state Operations that affect the state of the Call should call the setState function after the operation has been performed. This function will change the state of the Call by referencing the current state and the operation in the state-transition table. A VOMessage object will be created, with a type of STATUS_UPDATE and sent to the application queue. The GUI and any other component that reads the application queue will therefore be informed of the status update.
15 (~ Playback Call Inherits from VOCaII base class.
In this special case of a Call, the Video Operator audio and video output is replaced with the H.32~ stream from the playback of a movie by the Video 2o Operator Storage and Playback external system component.
Class VOPlaybackCall Base Class VOCaII
Inheritance Type public 2s Friend Classes -(i) Attributes Access Level Type Name Description 3o VOMovie* m_pMovie the movie object that will be played VOPlayer* m_pPlayer Player object that performs the playback (ii) Methods _z 6 PlaybackStart(); starts playback PlaybackStop(); stops playback (g) Movie A Movie is a recording of an H.320 Call. For Phase 1, the Video Operator Storage and Playback System manages files and H.320 data streams for recording and playback of movies, as well as storage and retrieval.
Class VOMovie Base Class VOObject Inheritance Type public Friend Classes -Attributes Access Level Type Name Description public ID_t m_movieID movie ID
public CString m_description movie description (h) Call Object Manager By having a Call Object Manager to perform the construction and destruction of Call objects, a list of all calls on the video operator's machine can be maintained. This includes calls that are not part of any Conference zs or Playback Sessions, including incoming calls and general purpose dial-out calls. Operations that affect a Call but do not create or destroy it can be performed by the Call object itself.
Class VOCallObjManager 3o Base Class VOObject Inheritance Type public Friend Classes -z6~

(i) Attributes Access Level Type Name Description int m_numChannels total number of unused channels int m numActive total number of active channels s CMapStringToOb m callList list of calls (ii) Methods Dial();
Dial(VOCaII* pCailing);
pCalling parameter: If not NULL, this pointer will be used for the Call object. This is necessary when creating or re-using a Call object that is in an inactive or disconnected state.
is Dial performs dial out. The numbers) to Dial are in the m_pSite Call member structure.
Answer( );
Answer(VOCaII* pIncoming);
2o plncoming parameter: If not NULL, this pointer will be used for the Call object. This is necessary when creating or re-using a Call object that is in an inactive or disconnected state.
Answer answers an incoming call.
Hangup(VOCaII* pCall);
pCall parameter: pointer to the call Hangup hangs up the call pointed to by pCall Hold(VOCaII* pCall);
pCall parameter: pointer to the call Hold puts the call pointed to on hold.
.?.6 ~

VOCaIl* CallCreate();
VOCalI* CallCreate creates a Call object.
s VOPlaybackCall* PlaybackCallCreate();
VOPlaybackCall* PlaybackCalICreate() creates a Playback Call object.
VOCall* GetCallPtr(ID_t idCallj;
idCall parameter: call ID
VOCaII* GetCallPtr gets the pointer to the call object identified by idCall (i) Connection A Connection is defined as a pair of Call objects that maintain a Join state, and each Call has the Video Operator Site as a common point for the Join to be implemented.
Class VOConnection Base Class VOObject 2o Inheritance Type public Friend Classes (i) Data Types enum StateConnection_a { ERROR, UNJOINED, JOINED, BROKEN, lastConnectionStates };
enum ConnectionOperation_e { ERROR, JOIN, UNJOIN, BREAK, RESET, lastConnectionOperations };
(ii) Attributes Access Level Type Name Description VOCalI* m_pParticipantCall pointer to the Participant Call VOCaII* m_pMCUPortCall pointer to the MCU Part Call ado VOParticipantSite* m_pParticipantSite pointer to the Participant Site VOMCUSite* m_pMCUPortSite pointer to the MCU Port Site s CTime m,joinTime time of join VOMovie* m_pMovie movie pointer for recording or playback boolean m_expectBreak flag that helps determine whether a Break is expected or not.
1 o StateConnection_a m_state state of the connection StateConnection_e [nConnectionStates] [nConnectionOps]
m_transitionTable state transition table VOConference* m_pConference pointer to the Conference that this Connection is a part of.
(iii) Methods Join(); joins the Participant and MCU Port Calls.
Unjoin(); unjoins the Participant and MCU Port Calls.
SetParticipantCall(VOCaII* participantCall);
participantCall parameter: pointer to a Call object SetParticipantCall sets the Call to be the Participant Call. This is useful zs when managing unknown incoming calls or for last minute participant substitution.
SetMCUPortCall(VOCaII* mcuPortCall);
- mcuPortCall parameter: pointer to a Call SetMCUPortCall sets the Call to be the MCU Port Call. This is useful when managing unknown incoming calls or for last minute call site substitution.
DoParticipantCall( j; calls the Participant Site and sets it as the Participant Call.
DoMCUPortCall( ); calls the MCU Port Site and sets it as the MCU Port Call.
setState ( ConnectionOperation_a operation;
operation parameter: the operation that has been performed which will result in a change of state.
to Operations that affect the state of the Connection should call the setState function after the operation has been performed. This function will change the state of the Connection by referencing the current state and the operation in the state-transition table. A VOMessage object will be created, with a type of STATUS_UPDATE and sent to the application queue. The GUI
and any other component that reads the application queue wrill therefore be informed of the status update.
protected Break( ); is called when a Joined Connection becomes Un joined. If the member variable m expectBreak is FALSE then one of the 2o Calls must have unexpectedly been disconnected. Otherwise, the Connection's Unjoin() operation would have been called.
protected Reset ( ); resets the state of the Connection to UNJOINED.
z5 (j) Connection Object Manager Similarly with the Call Object Manager, a list of all Connections in operation on the video operator's machine must be maintained. All operations that result in the creation or deletion of a Connection must use the Connection Object Manager.
Class VOConnectionObjMgr Base Class VOObject Inheritance Type public Friend Classes -~z ~z (i) Attributes Access Level Type Name Description > CMapStringToOb m_connectionsList list of all connections ' int m_numJoined number of joined connections to (ii) Methods VOConnection* Create();
Return Value: pointer to Connection object VOConnection* Create creates a new Connection object and adds it to the 15 llSt.
Remove (VOConnection* oldConnection);
oldConnection parameter: connection object to be removed 2o Return Value: returns TRUE if operation successful.
Remove deletes a Connection object and removes it from the list.
VOConnection* GetConnectionPtr( ID t idConnection);
25 Return Value: a pointer to the connection object idConnection parameter: ID of the Connection VOConnection* GetConnectionPtr returns the pointer to a Connection 30 object identified by its ID.
(k) Message All one-way communication from the Internal System Software to the rest of the Video Operator application, i.e. the Graphical User Interface, is sent as ~~3 messages that get placed on the Application Queue. The function to create and post a Message is in the base class VOObject, which all Internal System Software classes inherit from. All run-time errors or debugging information is put into a Message object, and posted to the application s queue so that an appropriate object will process it according to its type and severity. Therefore all class functions that do not return a specific type will post a Message if something goes wrong, e.g. out of memory, or debugging information to be displayed by the GUI or logged to a file.
to Class VOMessage Base Class VOObject Inheritance Type public Friend Classes -(i) Data Types enum senderType_a { INTERNAL, SCHEDULE, CONFERENCE, CONNECTION, CALL, TIMER };
2o enum message'I~pe_e i DEBUG, ERROR, WARNING, APPLICATION_ERROR, STATE_UPDATE };
Delivery type flags: DELIVER_MESSAGE_QUEUE, DELIVER LOG_FILE, DELIVER MODAL DIALOG, DELIVER_MODELESS DIALOG, DELIVER CONSOLEOUTPUT
{ii) Attributes 3o Access Level Type Name Description int m errorCode error code int m_delivery flags for preferred message delivery when posting.
sender'I~pe_e m_senderType sender type VOObject* m_pObject pointer to the sender messageType_a m messageType type of the message CString m_info message info * priority of message or error s * severity of message or error (iii) Methods Post(); posts a message to the application message queue to private static AppendLog();
Return Value: returns TRUE if the operation is successful.
This method is called by VOObject::PostMessage() when the flag for DELIVER_LOG_FILE is set.
is (1) Video Operator Generally there will be only one Video Operator per machine. Each Video Operator has a Schedule, and a list of customer Participant Sites to manage. The Call Object Manager and Connection Object Manager are also zo part of the Video Operator.
Class VOOperator Base Class VOObject Inheritance 'I~pe public 2s Friend Classes -(i) Attributes Access Level ape Name Description 3o ID_t m operatorID operatorID
VOSchedulem_schedule schedule for the current operator CObList m_MCUlist list of MCU objects CObList m_operatorSites Operator's sites) static VOUserPreferences m userPreferences default application user preferences (ii) Methods protected ScheduleStart(); initiates the schedule for the video operator.
protected Ca110bjMgrStart(); initiates the call object manager.
protected ConnectionObjMgrStart(); initiates the connection object manager.
~o protected CallSystemInterfaceStart(); initiates the Call System Interface.
(mJ User Preferences The Video Operator Console application will have a set of default application preferences which may be modified and saved. The values of these variables are taken from the following sources, in order of increasing preference: hard-coded default values, saved VO.INI file, command-line invocation arguments, GUI entry and run-time modifications saved to VO.INI file.
Class VOUserPreferences Base Class VOObject Inheritance Type public Friend Classes -(i) Attributes Access Level Type Name Description ID_t m_operatorID default operatorID
(ii) Methods SavePrefs(); saves all values to VO.INI.
LoadPrefs(); loads all values from VO.INI.
a ~~

(nJ MCU
All MCU Port Sites correspond to a particular MCU. This class is used for MCU Port Site storage only. For Phase 2, MCU specific operations and interfaces would be implemented here.
Class VOMCU
Base Class VOObject Inheritance Type public Friend Classes -to (i) Attributes Access Level Type Name Description ID_t m_mcuID ID of the MCU
~5 CObList m_portList List of MCU Port Site objects (ii) Methods VOMCUPortSite* GetPortPtr( ID t idPort);
Return Value: a pointer to the MCU Port Site object.
zo IdPort parameter: ID of the MCU Port Site VOMCUPortSite* GetPortPtr returns the pointer to a MCU Port Site object identified by its ID.
25 VOMCUPortSite* CreatePort();
Return Value: a pointer to a new MCU Port Site object VOMCUPortSite* CreatePort returns the pointer to a newly created MCU
Port Site object identified by its ID.
(3} State Variable Transition Diagrams for Core Classes Figure 101 shows a state transition diagram illustrating the state changes that may occur in the VOCaIl object's m_state variable ("state variable").

The state variable starts 40501 in Inactive 40502 state.
If the VOCaII object receives a Dial 40503 input while in Inactive 40502 state, the state variable changes to Dialing 40504 state. In the Dialing 40504 state, the state variable changes to Inactive 40502 state upon receiving a Busy 40505 input or to Active 4050? state upon receiving an Answer 40506 input. In the Active 4050? state, the state variable changes to Held 40510 state upon receiving a Hold 40509 input, to Disconnected 40515 state upon receiving a Disconnection 40514 input, or to Inactive 40502 state upon receiving a Hangup 40508 input. In the Held 40510 state, the state variable changes to Active 4050? state upon receiving a Pickup 40511 input, to Disconnected 40515 state upon receiving a Disconnection 40513 input, or to Inactive 40502 state upon receiving a Hangup 40512 input. In the Disconnected 40515 state, the state variable 1s changes to Inactive 40502 state upon receiving a Reset 40516 input.
If the VOCaII object receives an Incoming Call 40517 input while in Inactive 40502 state, the state variable changes to Incoming 40518 state.
In the Incoming 40518 state, the state variable changes to Inactive 40502 2o state upon receiving a Reject 40520 input or to Active 40507 state upon receiving an Answer 40519 input.
Figure 102 shows a state transition diagram illustrating the state changes that may occur in the VOConnection object's m_state variable ("state 25 variable"). The state variable starts 40601 in Unjoined 40602 state. In the Unjoined 40602 state, the state variable changes to Joined 40604 state upon receiving a Join 40603 input. In the Joined 40604 state, the state variable changes to Unjoined 40602 state upon receiving an Unjoin 40605 input or to Broken 40607 state upon receiving a Break 40606 input. In 3o the Broken 4060? state, the state variable changes to Joined 40604 state upon receiving a Join 40608 input.
c) Scheduling System Classes ( 1 ) Class List Playback Session Conference Schedule Schedulable (2) Class Descriptions (aJ Playback Session Like Conferences, Playback Sessions need to be scheduled. A Call is made with a Participant Site and the Video Operator Site. The Video Operator Storage and Playback external component system will playback a scheduled and pre-selected movie, replacing the AV output to the Participant Site. No to MCU is used for a Playback Session, and only one Participant Site is involved in one embodiment.
Class VOPlaybackSession Base Class VOSchedulable ~5 Inheritance Type public Friend Classes (i) Data Types 2o enum StatePlaybackSession_e { ERROR, INACTIVE, SETUP, ACTIVE, ENDING, FINISHED, lastPBSessionStates ;;
enum playbackSessionOperation_e { ERROR, PREPARE, START, CLOSE, FINISH, lastPBSessionOperations};
z5 (ii) Attributes Access Level Type Name Description public ID_t m ID ID assigned when a reservation is made for the session public CString m_name a short name for the session 3o public CString m_description a brief description public CTime m_startTime start time public CTimeSpan m duration the duration of the playback session public int m xferRate The data transfer rate (number of av9 channels) protected VOPlaybackCall* m_playbackCall the playback call object protected StatePlaybackSession_a m_state state of playback session protected StatePlaybackSession_e (lastPBSessionStates]
jlastPBSessionOps] m_transitionTableThe state transition table to (iii) Methods public boolean Setup();
Return Value: returns TRUE if operation successful.
public boolean Setup() sets up the Playback Call by calling the Participant 1s Site and initialize a VOPlayer object. This function may be called by the Scheduler.
Public boolean Start();
Return Value: returns TRUE if operation successful.
zo Public boolean Start starts the Player to play to the Playback Call. This function may be called by the Scheduler.
Public boolean Close();
2s Return Value: returns TRUE if operation successful.
Public boolean Close sends messages to the Video Operator and maybe the Participant that the Playback Session will end soon.
3o Public boolean Finish();
Return Value: returns TRUE if operation successful.
Public boolean Finish stops the Player and Hangup the Playback Call. This function may be called by the Scheduler.
a ~o public StatePlaybackSession_e StateGet();
Return Value: returns the playback session's state.
Use the public StatePlaybackSession a StateGet; function to find out the state of the Playback Session.
protected boolean StateSet(playbackSessionOperation_a operation);
Return Value: returns TRUE if operation successful.
to operation parameter: the operation that has been performed which will result in a change of state Operations that affect the state of the Playback Session should call the protected boolean StateSet function after the operation has been performed. This function will change the state of the Playback Session by referencing the current state and the operation in the state-transition table.
A VOMessage object will be created, with a type of STATUS_UPDATE and sent to the application queue. The GUI and any other component that reads 2o the application queue will therefore be informed of the status update.
(b) Conference The main function of the Video Operator is to manage conferences. The scheduler system creates the Conference objects, which in turn create a list of Connections (or Participant-MCU Port Site Call pairs). In the special case of a movie being played back to a conference, an extra call is made to an MCU Port and the movie is played back to the MCU in a similar way as a Playback Session. This of course requires an extra MCU Port site to be available, and must be scheduled before the start of the conference.
Class VOConference Base Class VOSchedulable Inheritance Type public Friend Classes -(i) Data Types enum conferenceMode_a { CONTINUOUS_PRESENCE, VOICE ACTIVATED, LECTURE, DIRECTOR CONTROL };
enum StateConference_a { ERROR, INACTIVE, SETUP, ACTIVE, ENDING, FINISHED, lastConferenceStates};
enum conferenceOperation a { ERROR, PREPARE, START, CLOSE, FINISH, lastConferenceOperations};
to (ii) Attributes Access Level Type Name Description ID t m ID Conference ID given when the reservation is made CString m name name for conference CString m description brief description CString m_timeZonetime zone CTime m startTime start time of the conference 2o CTimeSpan m duration duration of the conference int m transferRate transfer rate int m numActiveConns number of active connections conferenceMode a m mode conference mode boolean m recordingScheduled TRUE if this conference is to be recorded CObList m connectionsList List to store the connection objects CMapStringToObj m_participantSiteList List of participant sites 3o VOPlaybackCall m_playbackCall If there is a playback in the conference, this is valid StateConference a m state current state of conference StateConference_e (lastConferenceStates] [lastConferenceOps]
, ,, m_transitionTable state transition table *Call Setup Type *Audio Protocol *Video Protocol *Multi MCU Conference *H.243 Chair Control & password (iii} Methods to public boolean Setup();
Return Value: returns TRUE if operation successful.
public boolean Setup sets up each Connection in the connection list (and the Playback CaII if required} by calling each Participant Site and MCU Port 1 s Site as appropriate, and perform the Join operations to create the Connections. This function may be called by the Scheduler.
Public boolean Start();
Return Value: returns TRUE if operation successful.
Public boolean Start starts the Conference. This function may be called by the Scheduler.
Public boolean End/);
Zs Return Value: returns TRUE if operation successful.
Public boolean End starts tearing down the Connections in the conference or issues warnings that the conference will end soon. This function may be called by the Scheduler.
Public boolean Finish();
Return Value: returns TRUE if operation successful.
Public boolean Finish stops the Conference and hangs up all Calls in the Conference. This function may be called by the Scheduler.
public StateConference a StateGet(~;
Return Value: returns the Conference state s Use the public StateConference_e StateGet function to find out the state of the Conference.
protected boolean StateSet(conferenceOperation_a operation);
to Return Value: returns TRUE if operation successful.
operation parameter: the operation that has been performed which will result in a change of state Operations that affect the state of the Conference should call the protected is boolean StateSet function after the operation has been performed. This function will change the state of the Conference by referencing the current state and the operation in the state-transition table. A VOMessage object will be created, with a type of STATUS UPDATE and sent to the application queue. The GUI and any other component that reads the application queue 2o will therefore be informed of the status update.
(c) Schedule The Scheduling System maintains a list of Conferences and Playback Sessions. Each Conference and Playback Session is created at a particular z5 time interval before its starting time. The Schedule in memory and the Schedule stored in the Video Operator Shared Database for the current Video Operator should always be synchronized.
Class VOSchedule 3o Base Class VOObject Inheritance Type public Friend Classes ~85~

(i) Attributes . Access Level Type Name Description ID_t m_operatorID responsible operator ID
CMapStringToObj m_schedItems list of schedulable objects ' a (Conferences and Playback Sessions) CMapWordToOb m_schedAlarms list of alarms currently set for operations on schedulable objects (construction and deletion) (ii) Methods to SynchWithDb(); synchronizes with the VO shared database for the schedule.
AddSchedulable(VOSchedulable* pSchedulable);
pSchedulable parameter: pointer to schedulable object to be added to list AddSchedulable adds a Schedulable object to the list DeleteSchedulable(ID_t aSchedulable);
2o aSchedulable parameter: schedulable object to be removed from list DeleteSchedulable deletes a Schedulable object and remove from list.
(d) Schedulable Items or Objects that are schedulable in Phase 1 are Conferences and Playback Sessions. This class allows us to create a schedule for any type of event.
Class VOSchedulable 3o Base Class VOObject Inheritance Type public Friend Classes -.~ 85-(i) Attributes Access Level Type Name Description ID_t m_requestor ID of requestor Ctime m_startTime scheduled starting time CTimeSpan m duration scheduled duration of event Ctime m endTime scheduled end time of event MMRESULTm_aiarmID ID of alarm currently set (ii) Methods io public SetAlarm(Ctime time, LPTIMECALLBACK func );
time parameter: time for alarm to be triggered func parameter: pointer to callback function when alarm is triggered Return Value: returns TRUE if operation successful.
~5 public SetAlarm sets an alarm to be triggered at a specified time. When the alarm is triggered, the callback function will be called. This is useful for time dependant events such as 15 minutes before a Conference starts, 5 minutes before a Conference ends, and 30 minutes after a Conference has finished.
public KillAlarm();
Return Value: returns TRUE if operation successful.
public KillAlarm kills the last alarm that has been set by SetAlarm(). This would be used in the case of aborting a Conference, etc.
(3) State Variable Transition Diagram for Schedule System Classes Figure 103 shows a state transition diagram illustrating the state changes 3o that may occur in the VOConference object's m_state variable ("state variable"). The state variable starts 40?01 in Inactive 40702 state. In the Inactive 40?02 state, the state variable changes to ConnectionSetup 40?04 state upon receiving a " 15 minutes before scheduled time" 40703 input. In the ConnectionSetup 40?04 state, the state variable changes to Active ~ ~6 40?06 state upon receiving a Start Conference 40705 input. In the Active 40?06 state, the state variable remains in Active 40706 state upon receiving an Extend Conference 40707 input or changes to Ending 40707 state upon receiving a CloseConference (Proper Termination) 40708 input.
s In the Ending 4070? state, the state variable changes to Finished 40711 state upon receiving a Finish 40?10 input.
d~ Recording and Playback Glasses ( 1 ) Class List to Recorder Playe r (2) Class Details (a) Recorder A recorder communicates with whatever external components performs the ~ s actual movie creation and recording of the input pipe of a Call. This external component is known as the Video Operator Storage and Playback system.
Class VORecorder 2o Base Class VOObject Inheritance Type public Friend Classes -2s (i) Data Types enum StateRecorder_e { ERROR, IDLE, RECORDING, PAUSED, FINISHED, lastRecorderStates};
enum recorderOperation_a { ERROR, BEGIN, PAUSE, RESUME, STOP, lastRecorderOps }
(ii) Attributes Access level Type Name Description VOMovie* m_movie Movie VOCaiI* m_pCall Call pointer (for recording) Cstring m info Participant and Conference Names Ctime m startTime Start Time Ctime m endTime End time CtimeSpan m_duration Total recorded time StateRecorder a m state State StateRecorder_e [lastRecorderStatesJ [lastRecorderOps]
m_transitionTable state transition table *VSF Object *Recording Mode io (iii) Methods InitMovie(~; VOSP initializes a recording. This will tell the VOSP to prepare to record.
start(); VOSP starts a recording.
stop(); VOSP stops a recording.
setState(recorderOperation_e operation);
operation parameter: the operation that has been performed which will result in a change of state.
Operations that affect the state of the Recorder should call the setState z5 function after the operation has been performed. This function will change the state of the Recorder by referencing the current state and the operation in the state-transition table. A VOMessage object will be created, with a type of STATUS UPDATE and sent to the application queue. The GUI and any other component that reads the application queue will therefore be 3o informed of the status update.
(b) Player A Player communicates with whatever external component performs the actual playback of a movie to the output pipe of a Call. For Phase 1, this a ~g . ,, external component is known as the Video Operator Storage and Playback system.
Class VOPlayer Base Class VOObject Inheritance Type public Friend Classes -(i) Data Types to enum StatePlayer_e { ERROR, IDLE, PLAYING, PAUSED, FINISHED, nPlayerStates};
enum playerOperation_e { ERROR, BEGIN, PAUSE, RESUME, STOP, RESET, nPlayerOps }
(ii) Attributes Access level Type Name Description VOMovie* m_pMovie Movie VOCaII* m_pCall Call pointer (for playback) Cstring m_info Participant and Conference Names 2o Ctime Ctime m_startTime m_endTime Start and End Time CTimeSpan m_duration Total playback time StatePlayer_a m_state State StatePlayer_a (nPlayerStatesJ [nPlayerOps] m_transitionTable state transition table *VSF Object *Playback Mode (iii) Methods 3o public InitMovie();
Return Value: returns TRUE if operation successful.
public InitMovie VOSP initializes playback. This will tell the VOSP to prepare for playback.
z~~

public Start();
Return Value: returns TRUE if operation successful.
s public Start VOSP starts playback.
public Stop();
Return Value: returns TRUE if operation successful.
~o public Stop VOSP stops playback.
setstate(playerOperation_a operationj;
Return Value: returns TRUE if operation successful.
~5 operation parameter: the operation that has been performed which will result in a change of state.
Operations that affect the state of the Player should call the setstate function after the operation has been performed. This function will change 2o the state of the Player by referencing the current state and the operation in the state-transition table. A VOMessage object will be created, with a type of STATUS_UPDATE and sent to the application queue. The GUI and any other component that reads the application queue will therefore be informed of the status update.
(3) State Transition Diagrams for Recording and Playback Classes Figure 104 shows a state transition diagram illustrating the state changes that may occur in the VORecorder object's m_state variable ("state 3o variable"). The state variable starts 40801 in Idle 40802 state. In the Idle 40802 state, the state variable changes to Recording 40804 state upon receiving a Begin Recording 40803 input. In the Recording 40804 state, the state variable changes to Paused 40806 state upon receiving a Pause 40805 input or to Finished 40810 state upon receiving a Stop 40808 a 90 ,.

input. In the Paused 40806 state, the state variable changes to Recording 40804 state upon receiving a Resume 40807 input or to Finished 40810 state upon receiving a Stop 40809 input.
Figure 105 shows a state transition diagram illustrating the state changes that may occur in the VOPlayer object's m_state variable ("state variable"}.
The state variable starts 40901 in Idle 40902 state. In the Idle 40902 state, the state variable changes to Playing 40904 state upon receiving a Begin Playing 40903 input. In the Playing 40904 state, the state variable changes to Paused 40906 state upon receiving a Pause 40905 input or to Finished 40910 state upon receiving a Stop 40908 input. In the Paused 40906 state, the state variable changes to Playing 40904 state upon receiving a Resume 4090? input or to Finished 40910 state upon receiving a Stop 40909 input. In the Finished 40910 state, the state variable is changes to Playing 40904 state upon receiving a Replay 40911 input.
e) Call System Interface Class Description The Call Control System will manage all calls that a Video Operator can manage. This includes incoming and outgoing H.320 call management and 20 low level operations on a call, such as recording and playback. The Video Operator Application uses its Call System Interface to communicate with the Call Control System external component which manages all calls in a uniform way. This allows the video operator to manage calls that require different external programs, adding an extra codec to the machine, or even 2s managing calls on a remote machine.
Class VOCaIISys Base Class VOObject Inheritance Type public 3o Friend Classes ( 1 ) Data Types enum Bandwidth_e { MULTIRATE, BONDING, AGGREGATED, HO }
~9~

Q.931 UserInfo for a call using BONDING:
0x00 0x01 0x07 0x44 0x79 0x00 0x00 Bonded, 1 number, 7 digits long, 447-9000 Q.931 UserInfo for Aggregation:
0x01 0x02 0x07 0x44 0x79 0x00 0x00 OxFF OxOI
1 2 7 447-9000 , 1 Aggregated, 2 numbers, 7 digits long, 447-9000, 447-9001 to (2) Attributes Access Level Type Name Description public int m_numCalls total number of calls available public int m numConnections total number of connections t 5 available (3) Methods public Dial(Bandwidth_e calltype, CString destination);
2o public Dial(Bandwidth_a calltype, CString destination, CString origination);
Return Value: returns TRUE if operation successful.
calltype parameter: specifies the type of call to make.
destination parameter: specifies the destination number to be dialed.
25 origination parameter: specifies an origination number to be used, instead of the real number of the operator's console.
public Dial dials out.
3o public Answer(ID_t call);
call parameter: The Call ID of a Call waiting to be answered.
public Answer answers an incoming call.
~9~
r.

public Hangup(ID t call);
Return Value: returns TRUE if operation successful.
call parameter: the Call ID of a Call to Hangup public Hangup hangs up a call.
public Hold(ID t call);
Return Value: returns TRUE if operation successful.
call parameter: the Call ID of a Call to Hold io public Hold puts the call on hold.
public Join(ID t calll, ID t ca112);
Return Value: returns TRUE if operation successful.
is calll parameter: the Call ID of a Call.
ca112 parameter: the CaII ID of a Call.
public Join joins two Calls.
20 (ID t connection);
Return Value: returns TRUE if operation successful.
connection parameter: he ID of a Connection to Unjoin public Unjoin un joins the specified Connection.
public StateCall a CallStatus(ID t call);
. Return Value: returns the state of a Call connection parameter: the ID of a Connection to Unjoin 3o public StateCali a CallStatus reports status of the specified Call.
public StateConnection_a JoinStatus(ID t connection);
Return Value: returns the state of a Connection connection parameter: the ID of a Connection to Unjoin .~93 public StateConnection_e JoinStatus reports status of the specified Join.
protected LaunchMCP(~;
s Return Value: returns TRUE if operation successful.
protected LaunchMCP launches Incites MCP application.
E. Graphical User Interface Classes 1. Class Hierarchy.
Figure 106 shows the class hierarchy for the video operator graphics user interface {"GUI") classes. In general, the video conference operator will perform all the features of the video conferencing operator system described herein by interacting with the video operator console GUI ("console GUI").
~s The main components of the console GUI are the Main Console Window, Schedule and Connection List Windows, Conference and Connection Windows, a message area, audio and video controls, dialog boxes presenting timely information, and menu items for actions that may be performed infrequently. MCU operations and features will not be 2o implemented in the video operator console GUI, so as to allow different embodiments of the video operator system employing different MCU model types. Vendor-specific MCU operations will be performed by the vendor's software that comes with the MCU application. In one embodiment employing VideoServer's MCS, the MCS Workstation Software can be used 25 to implement features such as conference finish time extension, audio and video blocking, conference director control, etc. This software can run in parallel to the video operator GUI.
Described in object-oriented programming terms, the GUI has a main 3o application object which creates and maintains all the windows and views within. The main window is the VOMainFrame 41009 which is created by the VOConsoleApp 41008. This mainframe window creates the VOScheduleWnd 41016, VOAiertWnd 41015, VOConferenceVw 41014 and the VOVideoWatchVw 41013. The VOScheduleWnd 41016 and the a g 5~
, VOAIertWnd are dockable windows meaning that they can be attached to one of the sides of their parent window. In this case the parent window is the VOMainFrame 41009 window. The dockable windows can also be separated from the border by dragging them away. In such a situation they s will act like normal tool windows.
The function of each class of object can be summarized as follows.
VOConsoleApp 41008 is the main application class, and VOMainFrame 41009 is the main window which contains all the other windows.
to VOScheduleWnd 41016 is a window displaying the operator's schedule, and VOAiertWnd 41015 is a window where the error messages and alerts are displayed. VOChildFrame 41010 is a frame window for the multiple document interface ("MDI") windows. VOChildFrame 41010 will act like the mainframe window for each of the views. VOConferenceFrame 41018, is derived from the VOChildFrame 41010, is the frame window for the conference view, and VOConferenceVw 41014 is the window displaying the conference information. VOConferenceDoc 41012 is the document class corresponding to the VOConferenceVw 41014. VOVideoWatchFrame 4101?, derived from the VOChildFrame 41010, is the frame window for 2o the Video Watch view, and VOVideoWatchVw 41013 is the window displaying the video stream and controls for making calls.
VOVideoWatchDoc 41011 is the document class corresponding to the VideoWatch view.
2s In one embodiment using Visual C++ as the programming language, CWnd 41001 is a Superclass to the CMDIFrameWnd 41005 Subclass-1, CMDIChildWnd 41006 Subclass-2, CFromView 4100? Subclass-3, and CDialogBar 41002 Subclass-4, such that CMDIFrameWnd 41005 class objects, CMDIChildWnd 41006 class objects, CFromView 4100? class 30 objects, and CDialogBar 41002 class objects inherit attributes from the CWnd 41001 class. CMDIFrameWnd 41005 is a Superclass to VOMainFrame 41009 Subclass-1; CMDIChildWnd 41006 is a Superclass to VOChildFrame 41010 Subclass-1; CFromView 41007 is a Superclass to both VOVideoWatchVw 41013 Subclass-1 and VOConferenceVw 41014 2 9.5 Subclass-2; and CDialogBar 41002 is a Superclass to both VOAlertWnd 41015 Subclass-1 and VOScheduleWnd 41016 Subclass-2.
VOChildFrame 41010 is a Superclass to both VOVideoWatchFrame 4101?
Subclass- i and VOConferenceFrame 41018 Subclass-2. CWinApp 41003 s is a Superclass to VOConsoleApp 41008 Subclass-1, and CDocument 41004 is a Superclass to both VOVideoWatchDoc 41011 Subclass-1 and VOConferenceDoc 41012 Subclass-2.
VOConsoleApp 41008 is an Assembly Class associated with one to VOMainFrame 41009 Part-1 Class object, such that exactly one VOMainFrame 41009 object is associated with each VOConsoleApp 41008 object. VOMainFrame 41009 is an Assembly Class associated with one VOVideoWatchFrame 4101? Part-1 Class object, one VOConferenceFrame 41018 Part-2 Class object, one VOAIertV~lnd 41015 Part-3 Class object, is and one VOScheduleWnd 41016 Part-4 Class object, such that exactly one VOVideoWatchFrame 41017 object, exactly one VOConferenceFrame 41018 object, exactly one VOAIertWnd 41015 object, and exactly one VOScheduleWnd 41016 object are associated with each VOMainFrame 41009 object.
VOVideoWatchFrame 41017 is an Assembly Class associated with one VOVideoWatchDoc 41011 Part-1 Class object and one VOVideoWatchVw 41013 Part-2 Ciass object, such that exactly one VOVideoWatchDoc 41011 object and exactly one VOVideoWatchVw 41013 object are associated with each VOVideoWatchFrame 4101? object. Each VOVideoWatchDoc 41011 object, extended from the CDocument 41004 class object as discussed above, uses a VOVideoWatchVw 41013 object, extended from the CFormView 41007 class object.
3o Similarly, VOConferenceFrame 41018 is an Assembly Class associated with one VOConferenceDoc 41012 Part-1 Class object and one VOConferenceVw 41014 Part-2 Class object, such that exactly one VOConferenceDoc 41012 object and exactly one VOConferenceVw 41014 object are associated with each VOConferenceFrame 41018 object.
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VOConferenceDoc 41012 uses VOConferenceVw 41014.
2. Class and Object details.
a) User Interface Classes ( 1 ) Class List VOConsoleApp The main application class VOMainFrame The main window which has all the other windows VOScheduleWnd Window displaying the operator's schedule VOOutputWnd Window where the error messages and alerts are t o displayed VOChildFrame Frame window for the MDI windows. This will act like the mainframe window for each of the views.
VOConferenceFrame The frame window for the conference view. This is derived from the VOChildFrame VOConferenceVwThe window displaying the conference information VOConferenceDoc The document class corresponding to the VOConferenceVw VOVideoWatchFrame The frame window for the Video Watch view. This is derived from the VOChildFrame 2o VOVideoWatchVw The window displaying the video stream and controls for making calls.
VOVideoWatchDoc Document class corresponding to the VideoWatch view.
2s (2) Class Details (a) VOConsoleApp Class VOConsoleApp Base Class CWinApp ~ Inheritance Type public 3o Friend Classes (i) Attributes Access Level Type Name Description protected VOOperator* m_pOperator A pointer to the logged a9 ~

CA 02286132 1999-10-14 "

in video operator (ii) Methods Retcode CreateVideoOperator(CString login, CString password);
Return Value: returns a non-zero value if successful, zero otherwise.
login parameter: login id for the operator password parameter: operator's password to The Retcode CreateVideoOperator function is initially called during the application instantiation.
Retcode InitializeCallSystemComponents();
Return Value: returns a non-zero value if successful, zero otherwise The Retcode InitializeCallSystemComponents function is initially called during the application initiation, after the creation of the video operator, which makes a local copy of the pointers to the VOCaIISystemInterface, VOCaIIObjMgr and the VOConnectionObjMgr objects, initiated by the 2o internal software system.
void OnGetVOMessage(VOMsg voMsg);
voMsg parameter: the message object passed by the internal software system The void OnGetVOMessage function is called when the application receives a message from the internal software system to redirect the message to the appropriate windows. In the initial implementation, the message will be passed on to the VOMainFrame, which interprets the message. Depending on the type of the message it is either displayed in the VOOutputWnd, displayed in a message box, or passed on to the VOConferenceVw and the VOVideoWatch windows.
(b) VOMainFrame a y ~1 t.

Class VOMainFrame Base Class CFrameWnd Inheritance Type public Friend Classes -(i) Attributes Access Level Type Name Description protected VOOperator* m_pOperator A pointer to the logged in video operator to VOScheduleWnd* m_pScheduleWnd A pointer to the schedule window VOOutputWnd* m_pOutputWnd A pointer to the output window VOConferneceVw* m_pConfVw A pointer to the conference window.
This will be collection if we have multiple conference windows active at the same time.
VOVideoWatchVw* m_pVideoWatchVw Pointer to the video watch window.
(ii) Methods Retcode SynchWithDb(j;
Return Value: returns a non-zero value if successful. zero otherwize login parameter: login id for the operator password parameter: operator's password The Retcode SynchWithDb function is called if the schedule has changed and the needs to be synchronized with the database.
3o Retcode DisplayMessage(VOMsg voMsg);
Return Value: returns a non-zero successful, zero otherwise voMsg parameter: the VOMsg object received from the internal software system ~t9y The Retcode DisplayMessage function displays the content of the voMsg object in the output window. Based on the severity, an alert message box is also displayed.
void OnConferenceStatusChanged(VOConference* pConference);
pConference parameter: pointer to the conference object whose status has changed The void OnConferenceStatusChanged function is called when the status of 1o a particular conference has changed.
(c) VOScheduleWnd Class VOScheduleWnd Base Class CDialogBar ~s Inheritance Type public Friend Classes (i) Attributes Access Level Type Name Description zo protected VOMainFrame* m_pMainFrame A pointer to the Main Frame window VOSchedule* m_pSchedule pointer to the video operator's schedule 25 (ii) Methods Retcode DisplaySchedule(BOOL filter = O);
Return Value: returns a non-zero value if successful. zero otherwise filter parameter: the filter to be applied for display of the schedule. filter =
0 displays the entire schedule. filter = 1 displays only the active conferences 3o and playback calls The Retcode DisplaySchedule function is called to display the list of conferences and playback calls in the schedule window.
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Claims (45)

WE CLAIM:

1. A hybrid telecommunications system, which comprises:
a switched communications network;
a packet transmission network coupled to the switched communications network;
a call router coupled to the switched communications network and the packet transmission network; and a memory coupled to the call roister and having stored therein a call parameter database;
the call router being configured to route a call over the switched communications network and the packet transmission network based on at least one call parameter from the call parameter database.

2. The telecommunications system of claim 1 in which the call parameter database comprises profile information pertaining to a subscriber to the hybrid telecommunications system.

3. The telecommunications system of claim 1 in which the call parameter database comprises information pertaining to a call type.

4. The telecommunications system of claim 1 in which the call parameter database comprises information pertaining to usage of the switched communications network and the packet transmission network.

5. The telecommunications system of claim 1 in which the call parameter database comprises information pertaining to time of the call.

6. The telecommunications system of claim 1 in which the packet transmission network comprises an internet.

7. The telecommunications system of claim 1 in which the switched communications network comprises a public switched communications network.

8. The telecommunications system of claim 1 in which the public switched communications network is a telephone network.

9. A method for directing calls in a hybrid telecommunications system including a switched communications network and a packet transmission network, which comprises:
storing a call parameter database in a memory;
receiving a call on the hybrid telecommunications system;
accessing the call parameter database to determine at least one call parameter; and routing the call over the switched communications network and the packet transmission network based on the at least one call parameter.

10. The method of claim 9 in which the call parameter database comprises profile information pertaining to a subscriber to the hybrid telecommunications system.

11. The method of claim 9 in which the call parameter database comprises information pertaining to a call type.

12. The method of claim 9 in which the call parameter database comprises information pertaining to usage of the switched communications network and the packet transmission network.

13. The method of claim 9 in which the call parameter database comprises information pertaining to time of the call.

14. The method of claim 9 in which the packet transmission network comprises an internet.

15. The method of claim 14 in which the switched communications network comprises a public switched communications network.

16. The method of claim 15 in which the public switched communications network is a telephone network.

17. A computer program embodied on a computer-readable medium for directing calls in a hybrid telecommunications system including a switched communications network and a packet transmission network, which comprises:
first software that stores a call parameter database in a memory;
second software that accesses the call parameter database when the hybrid telecommunications system receives a call to determine at least one call parameter; and third software that routes the call over the switched communications network and the packet transmission network based on the at least one call parameter.

18. The computer program embodied on a computer-readable medium of claim 17 in which the call parameter database comprises profile information pertaining to a subscriber to the hybrid telecommunications system.

19. The computer program embodied on a computer-readable medium of claim 17 in which the call parameter database comprises information pertaining to a call type.

20. The computer program embodied on a computer-readable medium of claim 17 in which the call parameter database comprises information pertaining to usage of the switched communications network and the packet transmission network.

21. The computer program embodied on a computer-readable medium of claim 17 in which the call parameter database comprises information pertaining to time of the call.

22. The computer program embodied on a computer-readable medium of claim 17 in which the packet transmission network comprises an internet.

23. The computer program embodied on a computer-readable medium of claim 22 in which the switched communications network comprises a public switched communications network.

24. The computer program embodied on a computer-readable medium of claim 23 in which the public switched communications network is a telephone network.

25. A hybrid telecommunications system, which comprises:
a switched communications network;
a packet transmission network coupled to the switched communications network;
a call router coupled to the switched communications network and the packet transmission network;
a computer with an attached display in communication with the switched communications network and the packet transmission network; and the computer being configured to initiate remote management of the hybrid telecommunications system.

26. The hybrid telecommunications system of claim 25 further comprising callback logic capable of initiating tests of the hybrid telecommunications system.

27. The hybrid telecommunications system of claim 25 wherein the tests includes circuit analysis.

28. The hybrid telecommunications system of claim 25 further comprising a management device for selecting signaling states such as loop start, ground start or signaling such as dual tone multifrequency detection, multifrequency or dialpulse.

29. The hybrid telecommunications system of claim 25 wherein the hybrid telecommunications system includes support for internet telephony.

30. The hybrid telecommunications system of claim 25 comprising means for an operator to monitor the management of the hybrid network.

31. The hybrid telecommunications system of claim 25 further comprising an expert system regulating the Quality of Service of the hybrid telecommunications system.

32. A method for enabling communication on a hybrid telecommunications system, the hybrid telecommunications system including one or more switched communication networks coupled to one or more packet transmission networks, comprising the steps of:
coupling a call router to the switched communication network and the packet transmission network; and integrating a computer with an attached display to communicate with the packet transmission network, the computer being capable of initiating remote management of the hybrid telecommunications system.

33. The method for enabling communication on the hybrid telecommunications system as recited in claim 32, wherein callback logic is utilized to initiate tests of the hybrid telecommunications system.

34. The method for enabling communication on the hybrid telecommunications system as recited in claim 33, wherein the tests includes circuit analysis.

35. The method for enabling communication on the hybrid telecommunications system as recited in claim 32, further comprising managing the hybrid telecommunications system by selecting signaling states such as loop start, ground start, or detecting signals such as dual tone multifrequency, multifrequency or dialpulse.

36. The method for enabling communication on the hybrid telecommunications system as recited in claim 32, wherein the hybrid telecommunications system includes support for internet telephony.

37. The method for enabling communication on the hybrid telecommunications system as recited in claim 32, further comprising the step of an operator monitoring the management of the hybrid telecommunications system.

38. The method for enabling communication on the hybrid telecommunications system as recited in claim 32, further comprising the step of using an expert system to regulate the Quality of Service of the hybrid telecommunications system.

39. A computer program embodied on a computer-readable medium for enabling communication on a hybrid telecommunications system, the hybrid telecommunications system including one or more switched networks coupled to one or more packet transmission networks, comprising:
first software that couples a call router to the switched communications network and the packet transmission network; and second software that communicates with the packet transmission network, the second software initiating remote management of the hybrid communications system.

40. The computer program embodied on a computer-readable medium for enabling communication on the hybrid telecommunications system as recited in claim 39, further including callback logic to initiate tests of the hybrid telecommunications system.

41. The computer program embodied on a computer-readable medium for enabling communication on the hybrid telecommunications system as recited in claim 40, wherein the tests includes circuit analysis.

42. The computer program embodied on a computer-readable medium for enabling communication on the hybrid telecommunications system as recited in claim 39, further comprising management software for selecting signaling states such as loop start, ground start, or detecting signals such as dual tone multifrequency, multifrequency or dialpulse.

43. The computer program embodied on a computer-readable medium for enabling communication an the hybrid telecommunications system as recited in claim 39, wherein the hybrid telecommunications system includes support for Internet telephony.

44. The computer program embodied on a computer-readable medium for enabling communication on the hybrid telecommunications system as recited in claim 39, further comprising operator software facilitating an operator to monitor the management of the hybrid telecommunications system.

45. The computer program embodied on a computer-readable medium for enabling communication on the hybrid telecommunications system as recited in claim 39, wherein the hybrid telecommunications system includes an expert system regulating the Quality of Service of the hybrid telecommunications system.