1. Field of the Invention
The present invention relates to telecommunications, and is more particularly related to recording transaction data in a telecommunications network.
2. Technical Background
There are many factors driving the move toward converged networks such as deregulation, new sources of competition, substantial growth of the Internet, and the growth and importance of data in customers' enterprise networks. The popularity and convenience of the Internet has resulted in the reinvention of traditional telephony services. IP (Internet Protocol) telephony, which is also referred to as Voice-over-IP (VoIP), involves the conversion of voice information into data packets that are subsequently transmitted over an IP network. IP telephony over the Internet is often offered at minimal, or no cost to the users. Thus, IP telephony has found significant success, particularly in the long distance market.
Users also have turned to IP telephony as a matter of convenience. Both voice and data services are often accessible through a single piece of equipment, e.g., the personal computer. Furthermore, traditional DTMF (Dual-Tone Multi-Frequency) phones can enjoy the benefits of VoIP technology through the use of network adapters. The continual integration of voice and data services further fuels this demand for IP telephony applications.
The primary incentives for customers to adopt a converged solution are cost and the promise of new and expanded capabilities. However, if IP telephony is to be fully accepted in the marketplace, VoIP must be interoperable with the Public Switched Telephone Network (PSTN) and have a comparable Quality of Service (QoS). Therefore, to ensure the highest success rate with the customers, the service providers need to build a network that provides call quality, service reliability, and security that is at minimum, on par with the PSTN. It is essential that IP Telephony solutions meet customer demands of high-quality, ease of use, superior customer service, and lower cost. Since the public Internet can only provide “best-efforts”service, managed IP networks are required to support VoIP traffic with the call quality, service reliability, and security that users are accustomed to.
One approach that is being considered in providing VoIP with the call quality, service reliability, and security that users are accustomed to, involves the Session Initiation Protocol (SIP). SIP is an application-layer signaling protocol that has been developed to create, modify, and terminate sessions with one or more users. These sessions include Internet telephone calls, multi-media conferences, and multi-media distribution. SIP functionality is typically resident on application servers. Sip servers are configured to provide telephony services, and process call event information. Because vendors have developed their own custom SIP application programs, call events and telephony services are processed by each vendor's application server in a proprietary way. Unfortunately, when a network includes network elements provided by a multiplicity of vendors, it becomes necessary to accommodate a variety of proprietary interfaces that enable the devices to transmit and receive network transaction data. By way of example, transaction data may include call event information, billing information, monitoring information, error data, fraud prevention data, timeout data and any other data that must be tracked by the network.
What is needed is a platform independent method for capturing transaction data in a uniform manner. Preferably, the system and method will be extensible, providing embedded information that will enable a receiving computer to read the generic, uniformly formatted records without needing to accommodate any proprietary interface.
The present invention relates to a platform independent method for capturing transaction data and other information in a uniform manner. The method and system of the present invention is extensible, producing generic, uniformly formatted records that can be read by a receiving computer without needing a special proprietary interface.
One aspect of the present invention is a method for recording transactions in a telecommunications network. The method includes creating an XML transaction detail file. At least one transaction detail record is stored in the XML transaction detail file in response to a telecommunications transaction. The at least one transaction detail record includes transaction data corresponding to the telecommunications transaction.
In another aspect, the present invention includes a computer-readable medium having stored thereon a data structure for recording transactions in a telecommunications network. The data structure includes: an XML declaration field, the XML declaration field defining the data structure as an XML file; a server identification field, the server identification field including an IP address of a server generating the XML file; and a transaction detail section including at least one transaction detail record, the at least one transaction detail record being stored in the data structure in response to a telecommunications transaction, the at least one transaction detail record including transaction data corresponding to the telecommunications transaction.
In another aspect, the present invention includes a telecommunications network. The network includes at least one telecommunications apparatus configured to perform a telecommunications transaction. At least one SIP-server is coupled to the at least one telecommunications apparatus. The at least one SIP-server is configured to create an XML transaction detail file, process the telecommunications transaction, and store at least one transaction detail record in the XML transaction detail file. The at least one transaction detail record includes transaction data corresponding to the telecommunications transaction.
In another aspect, the present invention includes a computer-readable medium having stored thereon computer-executable instructions for performing a method for recording transactions in a telecommunications network. The method includes creating an XML transaction detail file. The XML transaction detail file is active for a predetermined period of time. At least one transaction detail record is stored in the XML transaction detail file in response to a telecommunications transaction. The at least one transaction detail record includes transaction data corresponding to the telecommunications transaction.
Additional features and advantages of the invention will be set forth in the detailed description which follows, and in part will be readily apparent to those skilled in the art from that description or recognized by practicing the invention as described herein, including the detailed description which follows, the claims, as well as the appended drawings.
It is to be understood that both the foregoing general description and the following detailed description are merely exemplary of the invention, and are intended to provide an overview or framework for understanding the nature and character of the invention as it is claimed. The accompanying drawings are included to provide a further understanding of the invention, and are incorporated in and constitute a part of this specification. The drawings illustrate various embodiments of the invention, and together with the description serve to explain the principles and operation of the invention.
Reference will now be made in detail to the present exemplary embodiments of the invention, examples of which are illustrated in the accompanying drawings. Wherever possible, the same reference numbers will be used throughout the drawings to refer to the same or like parts. An exemplary embodiment of the network of the present invention is shown in
In accordance with the invention, the present invention includes a method for recording transactions in a telecommunications network. The method includes creating an XML transaction detail file. At least one transaction detail record is stored in the XML transaction detail file in response to a telecommunications transaction. The at least one transaction detail record includes transaction data corresponding to the telecommunications transaction. The present invention provides a platform independent method for capturing transaction data and other information in a uniform manner. The system and method of the present invention is extensible, providing embedded information in generic, uniformly formatted transaction detail records that can be read by a receiving computer without needing a special proprietary interface.
As embodied herein, and depicted in
IP network 100 may be of any suitable type, but there is shown by way of example a network having a layered architecture. The layered nature of the architecture provides protocol separation and independence, whereby one protocol can be exchanged or modified without affecting the other higher layer or lower layer protocols. It is advantageous that the development of these protocols can occur concurrently and independently. The foundation of the layered architecture is the Internet Protocol (IP) layer. The IP layer provides a connectionless data delivery service that operates on a “best effort” basis; that is, no guarantees of packet delivery are made. A TCP (Transmission Control Protocol) layer is disposed above the IP layer. The TCP layer provides a connection-oriented protocol that ensures reliable delivery of the IP packets, in part, by performing sequencing functions. This sequencing function reorders any IP packets that arrive out of sequence. In another embodiment, the UDP (User Datagram Protocol) is employed instead of TCP. The User Datagram Protocol provides a connectionless service that utilizes the IP protocol to send a data unit, known as a datagram. Unlike TCP, UDP does not provide sequencing of packets. It relies on the higher layer protocols to sort the arriving packets. UDP is preferable over TCP when the data units are small, which saves processing time because of the minimal reassembly time. One of ordinary skill in the art will recognize that embodiments of the present invention can be practiced using either TCP or UDP, as well as other equivalent protocols.
A telephony application layer is disposed over the TCP layer. In one embodiment, the Session Initiation Protocol (SIP) is employed. SIP is an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. SIP is also a client-server protocol wherein servers respond to requests generated by clients. A detailed discussion of SIP and its call control services are described in IETF RFC 2543 and IETF Internet Draft “SIP Call Control Services”, Jun. 17, 1999. Both of these documents are incorporated herein by reference as though fully set forth in their entirety. Those of ordinary skill in the art will recognize that application-layer protocols other than SIP may be employed, including the 11.323 protocol.
Finally, the Session Description Protocol (SDP) is disposed above SIP in the layered architecture. SDP provides information about media streams in the multimedia sessions, permitting the recipients of the session description to participate in the session.
IP network backbone 120 may be of any suitable type, but there is shown by way of example a network that includes a nationwide high speed network that operates at 622 MB/sec (OC-12). Backbone 104 employs advanced packet switching technology commonly known as the Asynchronous Transfer Mode (ATM). Backbone 120 also utilizes a fiber-optic transmission technology referred to as the Synchronous Optical Network (SONET). The combination of ATM and SONET enables high speed, high capacity voice, data, and video signals to be combined and transmitted on demand. The high speed of backbone 120 is achieved by connecting Internet Protocol through the ATM switching matrix, and running this combination on the SONET network.
INCP 116 is an Intelligent Network Control Point that is accessed by RS 104 to obtain dial plan information for existing private network customers. INCP 116 is an additional database that may be queried by RS 104 to route specific private calls. INCP 116 may also be accessed by SPS 102.
PSTN 160 is, by way of example, a circuit switched network employing Signaling System No. 7 (SS7). Plain-Old-Telephone-Service (POTS) telephone 22 may be any suitable telephone set currently in use or on the market.
Enterprise gateway 118 may be of any suitable type. In the example depicted in
DAL gateway 112 is a system configured to support private traffic between IP locations and non-IP locations. DAL gateway 112 may be optionally employed in network 100.
Network gateway 114 is an SS7 (Signaling System 7)/C7-to-SIP Gateway. This provides users with the ability to place calls between locations within IP network 100 and locations within PSTN 20. For example, network gateway 114 is configured to provide in access to a voice switch (not shown), such as a Class 3 switch for domestic call processing, or a Class 5 switch for long-haul and/or international connections.
SIP phones 50, and SIP-client devices 52, may be of any suitable type provided that they conform to the standards promulgated in IETF 2543. SIP phones 50 have a 10-key dial pad similar to traditional phones. SIP URLs, which include PSTN 20 numbers, can be entered using the keypad or retrieved from a speed dial directory. To place a call, digits are entered using the dial pad. The entered digits are collected by the phone. When the “Dial” button is pressed, the call is initiated. SIP phones 50 ring similar to traditional phones when receiving an incoming call. SIP phones 50 may take the form of standalone devices—e.g., a SIP phone may be configured to function and appear like a Plain Old Telephone Service (POTS) telephone set. On the other hand, SIP client 52, is a software client that runs, for example, on a personal computer (PC) or laptop. From a signaling perspective, SIP-devices 50 and 52 are very similar, if not identical, in some cases.
At this point, the various SIP-servers disposed in network 100 will be described in more detail. Each type of server in network 100 has a critical role in recording and managing the various transactions supported by network 100.
Referring back to
Voice mail server (VMS) 108 is a SIP-server that provides voice mail services. Users of the IP network 100 are provided with the capability to integrate voicemail services based upon SIP. Calls are routed to the voice mail system 108 by SPS 102 and RS 104 for certain calls, such as those that indicate a Busy or Ring No Answer condition. Calls to voice mail can also occur as a Find-Me/Follow-Me termination option, or as an Unconditional Call Forward option selected by the user. Calls by the user to log in and retrieve messages are routed to VMS 108 as a SIP endpoint. A voice mail address can be entered for any destination address in RS 104. For instance, the Call Forwarding Unconditional address or Find-Me address, etc., can be the SIP URL of a voice mail account. SIP enabled VMS 108 supports all alphanumeric SIP URLs, Headers, Request, Methods and Status codes (e.g., per IETF RFC 2543). VMS 108 supports SUBSCRIBE, NOTIFY, and Message Waiting Indicator (MWI) messages. VMS 108 may restrict access to the system through a variety of ways. Access may be secured through private access code. The access code may be supplied in the SIP INVITE message or through. DTMF. VMS 108 may reject messages based on the IP address of the originating server. In other words, if the message is coming from a server that is not trusted, then VMS 108 may reject the message. VMS 108 is also configured to create a transaction detail file in XML format to thereby record transaction data corresponding to all network transactions processed by VMS 108. Because the format of the VMS XML transaction detail file is very similar to the SPS 102 XML transaction detail file, it will not be repeated here.
SIP conferencing server (SCS) 106 is a centralized SIP-conference server configured to provide audio conferencing capabilities. SCS 106 support G.711 (RTP/AVT 0), as well as other codecs. SCS 106 may specify two modes of operation. Under a Reserved mode, the users are required to reserve a bridge ahead of time. An Instant Conferencing mode refers to the ability to set-up a conference as needed without any need for advance reservation, allowing ad-hoc set-up of conferences as well permitting client based conferences to migrate to a bridge. Conference access is secured through an access code. Participants joining the bridge can send their access code via the SIP Invite message. POTS telephone users can enter through DTMF depending on the support for DTMF at the gateway. An audible tone may be played to announce each participant as they join the bridge. The system supports a coordinator/operator initiated conference, wherein the operator dials-out to each of the conference participants and brings them into the conference. The conference operator can enter and announce the name of the participants into the conference. The conference coordinator can notify the participants of the time and date for the call. The operators may be able to put parties On and Off Hold. Music On Hold is supported, whereby the parties on Hold are provided with music.
SCS 106 also permits private conferencing (i.e., sub-conferencing), wherein designated conference callers may confer privately within a conference call and then be returned to the main call. Calls from PSTN 20 may be forwarded to SCS 106 by network gateway 114. From the perspective of SCS 106, a SIP originated call is not processed differently than a non-SIP call because network gateway 114 is able to translate the called number to the conference URL. However, SCS 106 is able to validate the caller by prompting for passwords and validating the password entered as DTMF digits. As an alternate to password collection through DTMF, SCS 106 may support authentication using SIP. In this scenario, the SIP INVITE message carries additional user parameters, such as username/password combination that may be used by SCS 106 to validate the user. Further, conferencing system 106 supports web based provisioning by the users. SCS 106 interfaces with the OSS 110 for provisioning, alarming and reporting. The provisioning and reporting interface of the OSS 110 assists with a number of conferencing functionalities, such as the capability to Setup, Modify and Delete conferences. The administrator or moderator of the conference is able to specify the number of attendees to a conference, as well as specify duration of the conference, date and time-by-time zone, and name of reserved conference.
SCS 106 is configured to create a transaction detail file in XML format to thereby record transaction data corresponding to all the above described transactions processed by conferencing server 106. Because the format of the SCS 106 XML transaction detail file is similar to the SPS 102 XML transaction detail file, it will not be repeated.
RS 104 is a SIP redirect server that conforms with SIP standards detailed per IETF RFC 2543. RS 104 accepts SIP messages, maps the address into one or more new addresses, and returns these addresses to the client, which could be SPS 102. RS 104 does not initiate its own SIP requests, and RS 104 does not accept calls. RS 104 is essentially, a location server wherein information about possible terminating locations can be obtained, RS 104 also serves as a repository for end user information enabling address validation, feature status, and real-time subscriber feature configuration. RS 104 may also be used to store configuration information.
RS 104 is also configured to create a transaction detail file in XML format to thereby record transaction data corresponding to all SIP transactions, timeouts and errors processed by RS 104. The transaction detail file includes transaction detail records used to record network transactions processed by RS 104. RS 104 includes an XML processor module that is called by RS 104 application software module to create the XML transaction detail file. The XML processor module may also be called to read an XML file. Because RS 104 has a different function in the management of network 100, its XML transaction detail file is substantially different than the SPS XML transaction detail file. The format of the RS XML transaction detail file is shown in detail in Table II.
Referring back to
In one embodiment, the OSS computing system is based on technology provided by SUN Microsystems, the databases employed by the computing system are based on technology provided by ORACLE. OSS 110 provides and controls access to customer accounts. Users may utilize a web page to monitor service, login to their account, and manage certain elements permitted by user profiles. The account management system allows network personnel to establish, maintain, or deactivate customer accounts. In one embodiment, customer information is viewed via a web interface. The billing system processes customer event records, the customer pricing plan data, adjustments, taxation and other data in the preparation of customer invoices. The network facilities provisioning system provides the information required by network engineers to ensure that the appropriate hardware and software is in place to provide service. This may involve the creation of a customer profile, and the reconfiguration of SPS 102, RS 104, or other network elements. Network provisioning may also require the placement of hardware plug-in devices used in backbone 120.
A process management/work flow system serves as the core of OSS 110. The software is a Common Object Request Broker Architecture (CORBA) based publish-and-subscribe messaging middleware that provides graphical process automation, data transformation, event management and flexible connectors to transact with interfacing applications. This middleware architecture software fulfills the function of integrating all OSS 110 components and may provide hooks to non-OSS components using designated standard interfaces.
As embodied herein, and depicted in
Once the timer has elapsed, the server transmits the XML file to OSS 110. After the XML file is transmitted, a new file is created and the process repeats. If the timer has not elapsed, the server waits for additional transactions to process. In step 216, the server may suspend operations for any number of reasons. For example, if the server requires maintenance and is off-line, it is unnecessary to continue to monitor and record network transactions.
Those of ordinary skill in the art will recognize that the use of XML transaction detail files in accordance with the present invention can be employed for any events occurring within network 10. Calls placed between all or any combinations of SIP-phones, enterprise gateways, network gateways, DAL gateways, INCP gateways, SIP-voicemail servers, and SIP conferencing servers may employ the present invention. Those of ordinary skill in the art will also recognize that the present invention can be employed using any suitable type of transport network. Further, the present invention is applicable to any type of session that may be established including, but not limited to, telephony, video, audio, instant messaging, and etc. It is also contemplated that the present invention may be employed for billing, monitoring, management, or for any of a wide variety of services performed by the network.
It will be apparent to those skilled in the art that various modifications and variations can be made to the present invention without departing from the spirit and scope of the invention. Thus, it is intended that the present invention cover the modifications and variations of this invention provided they come within the scope of the appended claims and their equivalents.
This application is a continuation application of U.S. patent application Ser. No. 10/099,323, filed Mar. 15, 2002, which claims priority under 35 U.S.C. §119(e) based on U.S. Provisional Patent Application Ser. No. 60/276,923, filed Mar. 20, 2001, U.S. Provisional Patent Application Ser. No. 60/276,953, filed Mar. 20, 2001, U.S. Provisional Patent Application Ser. No. 60/276,954, filed Mar. 20, 2001, and U.S. Provisional Patent Application Ser. No. 60/276,955, filed Mar. 20, 2001, the contents of which are relied upon and incorporated herein by reference in their entirety.
Number | Name | Date | Kind |
---|---|---|---|
4979207 | Baum et al. | Dec 1990 | A |
5027388 | Bradshaw et al. | Jun 1991 | A |
5565316 | Kershaw et al. | Oct 1996 | A |
5579379 | D'Amico et al. | Nov 1996 | A |
5812668 | Weber | Sep 1998 | A |
5827070 | Kershaw et al. | Oct 1998 | A |
5867495 | Elliott et al. | Feb 1999 | A |
6016343 | Hogan et al. | Jan 2000 | A |
6122359 | Otto et al. | Sep 2000 | A |
H1897 | Fletcher et al. | Oct 2000 | H |
6134307 | Brouckman et al. | Oct 2000 | A |
6151624 | Teare et al. | Nov 2000 | A |
6233248 | Sautter et al. | May 2001 | B1 |
6282193 | Hluchyj et al. | Aug 2001 | B1 |
6311186 | Melampy et al. | Oct 2001 | B1 |
6377672 | Busuioc | Apr 2002 | B1 |
6377939 | Young | Apr 2002 | B1 |
6418467 | Schweitzer et al. | Jul 2002 | B1 |
6466971 | Humpleman et al. | Oct 2002 | B1 |
6490564 | Dodrill et al. | Dec 2002 | B1 |
6499054 | Hesselink et al. | Dec 2002 | B1 |
6577718 | Kalmanek et al. | Jun 2003 | B1 |
6611818 | Mersky et al. | Aug 2003 | B1 |
6631185 | Fleming | Oct 2003 | B1 |
6631186 | Adams et al. | Oct 2003 | B1 |
6639975 | O'Neal et al. | Oct 2003 | B1 |
6707915 | Jobst et al. | Mar 2004 | B1 |
6714992 | Kanojia et al. | Mar 2004 | B1 |
6718023 | Zolotov | Apr 2004 | B1 |
6751652 | Thomas | Jun 2004 | B1 |
6768722 | Katseff et al. | Jul 2004 | B1 |
6865681 | Nuutinen | Mar 2005 | B2 |
6870845 | Bellovin et al. | Mar 2005 | B1 |
6895438 | Ulrich | May 2005 | B1 |
6907032 | Eastman | Jun 2005 | B2 |
6952800 | Danner et al. | Oct 2005 | B1 |
6980526 | Jang et al. | Dec 2005 | B2 |
7058704 | Mangipudi et al. | Jun 2006 | B1 |
7076040 | Carson et al. | Jul 2006 | B2 |
7136467 | Brockman et al. | Nov 2006 | B2 |
7197560 | Caslin et al. | Mar 2007 | B2 |
7305081 | Kalmanek et al. | Dec 2007 | B1 |
7406306 | Gallant et al. | Jul 2008 | B2 |
20010012346 | Terry | Aug 2001 | A1 |
20010027420 | Boublik et al. | Oct 2001 | A1 |
20010032197 | Chandra et al. | Oct 2001 | A1 |
20010040886 | Jimenez et al. | Nov 2001 | A1 |
20010050984 | Jordan | Dec 2001 | A1 |
20010051962 | Plotkin | Dec 2001 | A1 |
20020010798 | Ben-Shaul et al. | Jan 2002 | A1 |
20020064149 | Elliott et al. | May 2002 | A1 |
20020075880 | Dolinar et al. | Jun 2002 | A1 |
20020090071 | Book et al. | Jul 2002 | A1 |
20020103898 | Moyer et al. | Aug 2002 | A1 |
20020112187 | Dalton et al. | Aug 2002 | A1 |
20020124100 | Adams | Sep 2002 | A1 |
20020126654 | Preston et al. | Sep 2002 | A1 |
20020127995 | Faccinn et al. | Sep 2002 | A1 |
20020129093 | Donovan et al. | Sep 2002 | A1 |
20020160810 | Glitho et al. | Oct 2002 | A1 |
20020188712 | Caslin et al. | Dec 2002 | A1 |
20030074313 | McConnell et al. | Apr 2003 | A1 |
20030079223 | Galloway | Apr 2003 | A1 |
20030126257 | Vijay | Jul 2003 | A1 |
20040078349 | Syrjala et al. | Apr 2004 | A1 |
20070116232 | Sprokkereef | May 2007 | A1 |
20070206576 | Radulovic | Sep 2007 | A1 |
20080013531 | Elliott et al. | Jan 2008 | A1 |
20080025295 | Elliott et al. | Jan 2008 | A1 |
Number | Date | Country |
---|---|---|
1202528 | May 2002 | EP |
WO 0052916 | Sep 2000 | WO |
WO 02075559 | Sep 2002 | WO |
Entry |
---|
Overview of the Session Initiation Protocol, Copyright 1992 Cisco System [retrieved on Sep. 11, 2002] Retrieved from the internet: <http://www.cisco.com/univered/c/td/doc/product/voice/sipsols/biggulp/bisipov.htm>. |
Sterman, Real-Time Billing in SIP, Copyright 2002, Deltahree [retrieved on 2002-09-171. Retrieved from the Internet: <http://www.sipcenter.com/files/SIPrealtimebilling.pdf>. |
Lennox et al., “Implementing Intelligent Network Services with the Session Initiation Protocol,” Columbia University Technical Report CUCS-002-99, Jan. 1999. |
Polyzois et al., “From POTS to PANS—A Commentary on the Evolution to Internet Telephony,” Mar. 26, 1999. |
Handley et al., “RFC 2543—SIP: Session Initiation Protocol,” downloaded from www.ietf.org/rfc/rfc2543.txt, Mar. 1999. |
Aboba et al., “The Accounting Data Interchange Format (ADIF),” ROAMOPS Working Group, Apr. 25, 2000. |
Brownlee et al., “RFC 2924—Accounting Attributes and Record Formats,” Sep. 2000. |
Schulzrinne et al., “Signaling for Internet Telephony,” Columbia University, Department of Computer Science Technical Report CUCS-005-98, Jan. 31, 1998. |
Cisco Systems, “Release Note for Cisco MC3810—Software Requirement for Analog Personality Modules,” Document No. 78-6053-01, 1998. |
Usdin et al., “XML: Not a Silver Bullet, but a Great Pipe Wrench,” StandardView, Sep. 1998, vol. 6, No. 3, pp. 125. |
Schulzrinne et al., “SIP Call Control Services”, IETF Internet Draft, Jun. 17, 1999, 34 pages. |
Kausar, et al., “A Charging Model for Sessions on the Internet,” Proceedings of the Fourth IEEE Symposium on Computers and Communications, pp. 32-38, Apr. 1999. |
Pan, et al., “Diameter: Policy and Accounting Extension for SIP (draft-pan-diameter-sip-01),” Internet Engineering Task Force (IETF), The Internet Society, 20 pages, Nov. 15, 1998. |
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