CA 02286132 1999-10-14 '
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CA 02286132 1999-10-14
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
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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
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"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.
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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.
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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;
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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;
6
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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
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WO 98/47298 PCT/US98/07927
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;
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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
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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
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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
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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
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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
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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;
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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'
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
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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
64
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.
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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
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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
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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
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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
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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
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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;
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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
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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
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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.
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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
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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~
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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
W3
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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.
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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
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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
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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
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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
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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
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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
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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
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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.
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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
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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(
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~ 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~
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~ 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.
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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;
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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.
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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
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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.
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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.
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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.
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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~
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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.
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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
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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~
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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~
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~ 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
2200
(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
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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
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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... ..,.
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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).
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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,
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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):
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~ 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
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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~
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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;
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~ 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
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~ 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.
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~ 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
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~ 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
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~ 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
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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
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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
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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.
.~ . , ,
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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--
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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:
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( 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.
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~ 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.
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~ 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,~
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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
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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.
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(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
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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
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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~
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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;
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~ 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
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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.
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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
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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.
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~ 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
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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.
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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.
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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
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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;
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~ 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
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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
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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-
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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.
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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
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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
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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.
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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
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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
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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.
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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.
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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.
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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.
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~ 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
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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:
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~ 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;
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~ 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.
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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.
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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);
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~ 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);
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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.
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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
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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.
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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.
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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.
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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.
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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
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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.
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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
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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,
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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 .
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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
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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;
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~ 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
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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
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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.
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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
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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
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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
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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
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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.
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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
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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
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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:
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? 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
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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
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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
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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
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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
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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.
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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
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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
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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
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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.
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(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
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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
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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.
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(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.
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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
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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
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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
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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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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
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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
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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
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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
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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
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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
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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
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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.
1
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.
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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
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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
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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
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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.
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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
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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
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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
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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
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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,
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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
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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
CA 02286132 1999-10-14
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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
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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
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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
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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:
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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
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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
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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.
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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,
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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
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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.
I5
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
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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
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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
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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
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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.
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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.
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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
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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
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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.
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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
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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
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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
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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
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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
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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
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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
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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
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(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
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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,
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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
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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
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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
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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
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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
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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.
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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
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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
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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.
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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.
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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.
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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
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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
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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)
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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
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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
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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
... ......._...~.~.~.". ....-.....- ,.._..... ..... .. . ~, ~. ... . ...
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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
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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
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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
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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.
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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
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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
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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.
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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
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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
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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
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("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-/
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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
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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
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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.
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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
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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
. ..-_~..~.....~-_._. . _ . .. , , ,
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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
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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
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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-~
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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
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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
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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
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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
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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
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(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
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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.
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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
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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 -
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(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.
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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
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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
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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 -
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(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
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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
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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
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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.
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(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").
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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
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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
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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.
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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 -
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(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]
, ,,
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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
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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
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(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 -
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(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
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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)
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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
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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.
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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
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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 }
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Q.931 UserInfo for a call using BONDING:
0x00 0x01 0x07 0x44 0x79 0x00 0x00
0 1 7 447-9000
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.
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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
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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
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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
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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
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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
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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
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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.
.~c.~ c?
r. ~
CA 02286132 1999-10-14 '
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'. 5.._' \''A
... ~,?\ ..m.~.w.
