Source: http://www.google.com/patents/US20060165068?dq=6480844
Timestamp: 2015-10-10 11:41:28
Document Index: 530244783

Matched Legal Cases: ['arty 120', 'arty 130', 'arty 120', 'arty 130', 'arty 130', 'arty 130', 'arty 120', 'arty 120']

Patent US20060165068 - Method and system for securely authorized VoIP Interconnections between ... - Google PatentsSearch Images Maps Play YouTube News Gmail Drive More »Sign inAdvanced Patent SearchPatentsA peering authority or settlement clearinghouse can be used to control access, collect session accounting information, and provide financial settlement of interconnect or session fees among anonymous Internet Protocol (IP) peers or networks. The addition of peering policy criteria, such as price and...http://www.google.com/patents/US20060165068?utm_source=gb-gplus-sharePatent US20060165068 - Method and system for securely authorized VoIP Interconnections between anonymous peers of VoIP networksAdvanced Patent SearchPublication numberUS20060165068 A1Publication typeApplicationApplication numberUS 11/301,637Publication dateJul 27, 2006Filing dateDec 13, 2005Priority dateDec 13, 2004Also published asUS7457283, WO2006065789A2, WO2006065789A3Publication number11301637, 301637, US 2006/0165068 A1, US 2006/165068 A1, US 20060165068 A1, US 20060165068A1, US 2006165068 A1, US 2006165068A1, US-A1-20060165068, US-A1-2006165068, US2006/0165068A1, US2006/165068A1, US20060165068 A1, US20060165068A1, US2006165068 A1, US2006165068A1InventorsJames Dalton, Dmitry IsakbayevOriginal AssigneeDalton James P Jr, Dmitry IsakbayevExport CitationBiBTeX, EndNote, RefManPatent Citations (85), Referenced by (42), Classifications (29), Legal Events (3) External Links: USPTO, USPTO Assignment, EspacenetMethod and system for securely authorized VoIP Interconnections between anonymous peers of VoIP networks
US 20060165068 A1Abstract
A peering authority or settlement clearinghouse can be used to control access, collect session accounting information, and provide financial settlement of interconnect or session fees among anonymous Internet Protocol (IP) peers or networks. The addition of peering policy criteria, such as price and quality of service, to peer to peer route discovery mechanisms enable a trusted intermediary, such as the settlement clearinghouse, to authorize acceptable interconnection or peering sessions between anonymous IP peers. Any financial settlement transactions which result from the peering sessions may be subsequently executed by the settlement clearinghouse. Images(9) Claims(13)
1. A method for securely authorizing VoIP interconnection between peers of VoIP networks: determining if a VoIP call originating from a first peer computer network must be completed by one or more second peer computer networks that are separate from the first computer network; identifying the one or more second peer computer networks that are capable of completing the VoIP call; identifying peering criteria comprising at least one of bandwidth, network quality of service, and price for the VoIP call; sending the peering criteria to a peering authority; evaluating the peering criteria with the peering authority; and based on the evaluation of the peering criteria, generating one or more tokens corresponding to the one or more second peer computer networks; and completing the VoIP call with a token. 2. The method of claim 1, further comprising: sending the one or more tokens to the first peer computer network (Step 044); and sending a call setup request comprising a token from the first peer computer network to the second computer network. 3. The method of claim 1, wherein identifying one or more second peer computer networks further comprises reviewing available second peer computer networks stored in a routing table. 4. The method of claim 1, wherein identifying one or more second peer computer networks further comprises discovering second peer computer networks using route discovery protocols. 5. The method of claim 1, wherein evaluating the peering criteria further comprises identifying a type of service requested in the peering criteria. 6. The method of claim 1, wherein evaluating the peering criteria further comprises determining if pricing associated with the VoIP call are acceptable by one or more second peer computer networks. 7. The method of claim 1, wherein evaluating the peering criteria further comprises determining if quality of service associated with the peering criteria are acceptable by one or more second peer computer networks. 8. The method of claim 1, wherein evaluating the peering criteria further comprises comparing historical quality of service of the second computer network to the quality of service requested in the peering criteria. 9. A system for securely authorizing VoIP interconnection between devices of VoIP networks: a first call point control device of a first computer network, for determining if a VoIP call originating from a first telephone to a second telephone must be completed by one or more second computer networks that are separate from the first computer network, the call point control device identifying one or more second peer computer networks that are capable of completing the VoIP call to the second telephone; the call point control device determining peering criteria comprising at least one of bandwidth, network quality of service, and price for the VoIP call; and a peering authority coupled to the call point control device, for receiving the peering criteria and evaluating the peering criteria, for generating one or more tokens corresponding to the one or more second peer computer networks based on the evaluation of the peering criteria, the first call point control device selecting a token and contacting a second call point device on a second computer network associated with the selected token for completing the VoIP call with the token. 10. The system of claim 9, wherein the peering authority comprises a settlement clearing house. 11. The system of claim 9, wherein the peering authority and first and second call point control devices comprise computer servers with IP addresses. 12. The system of claim 9, wherein the second call point control device receives the selected token and determines if the token is valid. 13. The system of claim 9, wherein the second call point device validates a digital signature of the token using a public key generated by the peering authority.
PRIORITY CLAIM TO PROVISIONAL AND NON-PROVISIONAL APPLICATIONS The present application claims priority to provisional patent application entitled, “Settlement Service for DUNDi Type VoIP Networks,” filed on Dec. 13, 2004 and assigned U.S. Application Ser. No. 60/635,621; the entire contents of which are hereby incorporated by reference.
TECHNICAL FIELD The invention relates to video, voice, data communications and application services. More particularly, the invention relates to a system and method for securely authorizing VoIP interconnection access control between anonymous peers of VoIP networks. BACKGROUND OF THE INVENTION In the traditional telephone carrier operating model, calls between Local Exchange Carriers (LECs), or Retail Service Providers (RSPs) are transported by an Inter-Exchange Carrier (IXC). The RSP provides retail telephone services to its end user subscribers on its network. When a RSP end user subscriber calls a telephone number which is not in the RSP's network, the RSP will switch that call to an IXC that will transport the call to the RSP serving the called number to complete the telephone call to the receiving party. The business model for this call scenario starts with the source RSP that switches the call to the IXC. The RSP pays the IXC a fee to transport the call to the destination RSP. The IXC transports the call to the destination RSP which completes the call to the receiving party. The IXC pays the destination RSP a fee to complete the telephone call. An important operating value added by the IXC is route discovery. The IXC manages a central routing table that enables routing among a multitude of RSPs to any telephone number on the global Public Switched Network (PSTN). This action simplifies operations for the RSP operator whom needs to route to only one IXC to obtain termination to any telephone number in the PSTN. In this document, the operating model described above is referred to as the IXC operating model. This common telephony business model for the operating model described above is referred to as the Calling Party Pays model. The end user of the source RSP pays a retail service fee to the source RSP. The source RSP pays the IXC a fee to locate and transmit the call to the destination RSP. The IXC pays the destination RSP a termination fee to complete the call. An important aspect of this business model is the role of the IXC as the central routing and billing intermediary among many RSPs. Source and destination RSPs do not have commercial interconnect agreements with one another. An important commercial value added by the IXC is the clearing of calls (routing and access control) between RSPs, accounting of interconnected calls and settlement of interconnect fees to ensure the destination RSP receives a share of the revenue compensation as expected in the Calling Party Pays business model. Each RSP has a single bilateral interconnect agreement with the IXC which eliminates the costly need for commercial bilateral agreements with every other RSP. Relative to the conventional IXCs, a new communications model has evolved: The increasing use of Voice over IP (VoIP) communications has made possible a new operating model referred to as the Peer To Peer operating model. The Peer To Peer operating model differs from the IXC operating model because end to end routing and signaling for telephone calls is achieved directly from the source RSP (peer) to the destination RSP (peer) without the need for a central intermediary such as an IXC. Two examples of the Peer To Peer operating model are DUNDi and ENUM. DUNDi (Distributed Universal Number Discovery, www.dundi.com) enables source networks (peers) to discover routes to destination networks (peers) without the need for a central routing directory or intermediary signaling point. ENUM is the Internet Engineering Task Force (www.itef.org) protocol (RFC 2916) which defines how a source peer may resolve telephone numbers into IP addresses in order to route and signal a VoIP call directly to the destination network (peer). In other words, ENUM is a standard adopted by the Internet Engineering Task Force (IETF) that uses the domain name system (DNS) to map telephone numbers to Web addresses or uniform resource locators (URL). The goal of the ENUM standard is to provide a single number to replace the multiple numbers and addresses for an individual's home phone, business phone, fax, cell phone, and e-mail. However, while IP technology has enabled the Peer To Peer operating model, there is no scalable mechanism to implement the Calling Party Pays business model with a Peer To Peer operating model. With the Peer To Peer operating model, the Calling Party Pays business model can only be implemented if every RSP (peer) has a bilateral commercial interconnect agreement with every other RSP (peer). Bilateral agreements among RSPs is not practical because the number of commercial peering agreements for all RSPs increases by the square of the number of RSPs (peers) [n*(n−1)/2 where n=number of peers], making large scale peer to peer networks using the Calling Party Pays business model virtually impossible. Referring now to FIG. 1 a, this figure illustrates a VoIP call within a RSP's network. Circle 100 in FIG. 1 a represents the RSP network. The RSP network could be a private IP network or a subset of public Internet. The call control point 110 controls calls between the calling and receiving parties by providing calling party authentication, additional service features such as call forwarding, call signaling to the receiving party and generating called detail records to account for the call transaction. One of ordinary skill in the art who is familiar with VoIP technology will recognize that the Call Control Point could be either an H.323 gatekeeper, H.323 IP to IP gateway, SIP proxy, SIP back to back user agent, softswitch, session border controller or any other device which controls routing or signaling between source and destination VoIP devices. Two end user subscribers of the RSP Network are represented by a first telephone 120 with number 14045266060 and second telephone 130 with number 14045724600. FIG. 1 a represents a call scenario where the calling party 120 calls a receiving party 130. The call from the calling party 120 is initiated with a call setup message 400, such as a SIP Invite message to the Call Control Point 110. The Call Control Point determines if, and how, the call should be routed to the receiving party 130. To complete the call to the receiving party 130, the Call Control Point 110, sends a message 410 to the receiving party 130 to complete the call between the calling and called parties. When the RSP provides service to both the calling and called parties, the call can be completed within the RSP's network 100 without the use of facilities provided by another VoIP service provider. In FIG. 1 a, inter-IP network peering does not occur. Referring now to FIG. 1 b, this Figure illustrates a VoIP call that requires inter-IP network peering. FIG. 1 b includes the Source RSP Network 100, and with elements 110, 120 and 130 that are similar those described in FIG. 1 a. Destination RSP Network 200 with Call Control Point 210 is introduced in FIG. 1 b. Two end user subscribers of the Destination RSP Network 200 are represented by a third telephone 220 with number 17033089726 and fourth telephone 230 with number 17036054283. The calling party 120 places a call to telephone number 17036054283. The call starts with a call setup message 400 from the calling party 120 to the Call Control Point 110