Patent Publication Number: US-2006007915-A1

Title: Connecting a VOIP phone call using a shared POTS line

Description:
BACKGROUND OF THE INVENTION  
      1. Field of the Invention  
      This invention relates generally to voice over Internet protocol (VOIP) technology, and particularly to connecting (i.e., terminating, making, or completing) a VOIP phone call using a shared plain old telephone system (POTS) line.  
      2. Description of the Background Art  
      In a public switched telephone network (PSTN), an initiating phone connects to a circuit switch and the PSTN via a first POTS line. A destination phone connects to the circuit switch and the PSTN via a second POTS line. The circuit switch electrically connects the initiating phone to the destination phone over the PSTN. The electrical connection is maintained for the entire duration of a phone call between the initiating phone and the destination phone. The electrical connection in the PSTN is commonly referred to as “circuit switched.” A problem with the PSTN is that because much of a conversation is silence, maintaining the electrical connection for the duration of the phone call wastes available bandwidth in the circuit switch.  
      VOIP is a technology that permits phone calls to be carried over the Internet as opposed to over the PSTN. In VOIP, a device known as an analog telephone adapter (ATA) or media gateway serves as an interface between an analog phone and the packet-based Internet. The ATA may be a standalone device or may be incorporated into another device such as a cordless phone base station or broadband modem. An initiating ATA converts analog signals from an initiating phone into packets using a voice codec (coder/decoder) algorithm. At the destination phone, to receive an incoming VOIP phone call, a destination ATA receives packets into a buffer and uses the same codec algorithm to convert the packets back into analog signals.  
      Typically, ATAs provide VOIP functionality via a connection to a broadband modem, such as through a cable Internet or a digital subscriber line (DSL) connection to the Internet. Typically, ATAs provide a foreign exchange subscriber (FXS) port to connect to an analog phone. As a backup for the VOIP service and for cases such as 911, some ATAs provide a Foreign Exchange Office (FXO) interface to connect to the PSTN. To provide VOIP service to a destination phone not directly connected to the Internet via a dedicated ATA, VOIP service providers (i.e., Internet telephony service providers or ITSPs such as Level 3 Communications, Inc.) offer connectivity from IP to the PSTN. ITSPs typically utilize large, centralized, multiple-port IP-to-PSTN media gateways.  
      A caller at the initiating phone with an ATA incorporating an FXO port must choose whether to use either VOIP to place the phone call over the Internet, or the FXO interface to place the phone call through the PSTN. For example, the caller may press the “#” key before dialing a phone number of the destination phone to choose VOIP. Accordingly, a problem with VOIP is that the initiating caller must dial additional digits or characters to choose between using VOIP or the PSTN.  
      A problem with the caller using some ITSPs is that the caller must first dial a special number to call in to one of the ATAs of the VOIP service provider, and the caller must then enter the phone number of the destination phone. This is referred to as two-stage dialing. A further problem with VOIP through an ITSP is that because the ITSP must pay service fees to PSTN operators, the ITSP typically charges significant termination and access fees to the caller at the initiating phone for service to destination phones via the PSTN.  
      Therefore, a need exists in industry to address the aforementioned deficiencies and inadequacies.  
     SUMMARY OF THE INVENTION  
      A system for connecting a voice over Internet protocol (VOIP) phone call using a shared plain old telephone system (POTS) line comprises an initiating analog telephone adapter (ATA), a VOIP control system, and a destination ATA. The initiating ATA includes an FXS port for connecting to an initiating telephone and for receiving a phone number of a destination phone therefrom, an FXO port for interfacing to a public switched telephone network (PSTN) via the POTS line, and a port for connecting to the Internet. The FXO port is configured to connect either the initiating phone or an incoming VOIP phone call from another ATA to the PSTN. The POTS line coupled to the FXO port is thus shared by the initiating phone and the incoming VOIP phone call. The VOIP control system is connected to the Internet, and includes a call manager server and a proxy server. The proxy server is configured to receive a signaling message including the phone number of the destination phone from the initiating ATA, and communicate the phone number to the call manager server. The call manager server is configured to select the destination ATA that is in the same local calling area of the destination phone by comparing the phone number of the destination phone to a database and communicate routing information for the destination ATA to the proxy server. The proxy server is further configured to communicate the routing information for the destination ATA to the initiating ATA.  
      A method for connecting the VOIP phone call using the shared POTS line comprises initiating the VOIP phone call to the destination phone in the initiating ATA, transmitting the phone number of the destination phone to the VOIP control system, selecting the destination ATA that is in the same local calling area of the destination phone from a plurality of ATAs, communicating routing information for the destination ATA to the initiating ATA, placing a call to the phone number of the destination phone over the shared POTS line in the destination ATA, and connecting the VOIP phone call from the initiating ATA to the destination ATA.  
      A decentralized architecture allows the POTS line to be shared for use. Routing logic is handled by the VOIP control system and by logic in the ATAs. The architecture enables lower cost routing (i.e., termination) than currently available. Selecting the destination ATA based upon the phone number of the destination phone and communicating routing information for the destination ATA to the initiating ATA relieves a caller at the initiating phone from having to select whether to use VOIP or the PSTN, and relieves the caller from dialing special phone numbers. The caller at the initiating phone is not required to choose between making an Internet VOIP phone call or a PSTN phone call at the initiating ATA connected to the initiating phone.  
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS  
       FIG. 1  illustrates a telecommunications system spanning the Internet and the PSTN, in one embodiment of the invention;  
       FIG. 2  illustrates an exemplary architecture of a VOIP control system for connecting a VOIP phone call using a shared POTS line, in one embodiment of the invention;  
       FIG. 3  illustrates an exemplary user database for connecting a VOIP phone call using a shared POTS line, in one embodiment of the invention;  
       FIG. 4  illustrates an exemplary ATA for connecting a VOIP phone call using a shared POTS line, in one embodiment of the invention;  
       FIG. 5  illustrates an exemplary method for connecting PSTN and VOIP phone calls using a shared POTS line, in one embodiment of the invention; and  
       FIG. 6  illustrates an exemplary method for registering an ATA, in one embodiment of the invention.  
    
    
     DETAILED DESCRIPTION  
      The invention provides systems and methods for connection and routing logic to automatically decide whether to use the Internet or a public switched telephone network (PSTN) to connect a phone call to a destination phone over a shared plain old telephone system (POTS) line. As further detailed below, a benefit of the shared POTS line is that termination of a VOIP phone call to a destination analog telephone adapter (ATA) in the same local calling area as the destination phone is the least expensive manner to route a VOIP phone call to the destination phone.  
       FIG. 1  illustrates a telecommunications system spanning the Internet and the PSTN, in one embodiment of the invention. Although only one initiating ATA  100  and one destination ATA  105  are depicted in  FIG. 1  for clarity, a VOIP control system  110  can be coupled to numerous such ATAs. Additionally, while the following discussion refers to the initiating ATA  100  and the destination ATA  105  for ease of description, it will be appreciated that the initiating ATA  100  can also receive phone calls, and the destination ATA  105  can also initiate phone calls. The ATAs  100  and  105  are each coupled to the Internet and at least one PSTN  115  via standard POTS lines (e.g., POTS lines  151  and  152 ). The PSTN  115  comprises any level of the global PSTN, including, for example, a local transport network, a regional transport network, or an international transport network. The initiating ATA  100  is also coupled to an initiating phone  120 . As referred to herein, a phone is any device capable of placing a phone call to a destination phone  125 , and includes, for example, an analog phone, a facsimile machine, and a cellular phone. The initiating phone  120  is configured to communicate a phone number of the destination phone  125  to the VOIP control system  110  utilizing a signaling message (i.e., control message) transmitted over a signaling pathway  130 .  
      As described in more detail below, the VOIP control system  110  is coupled to a database  135  for call routing. The database  135  contains information on appropriate cost centers for call routing, including numbering plan area (NPA) codes (i.e., area codes) and dialing prefixes (i.e., dedicated exchanges NXX). The database  135  in some embodiments comprises a LERG™ Routing Guide database available from Telecordia Technologies, Inc. The VOIP control system  110  is any combination of hardware and software configured to select the appropriate destination ATA  105  in the same local calling area as the destination phone  125  from among many such ATAs based upon the phone number of the destination phone  125 . The VOIP control system  110  is further configured to verify availability of the destination ATA  105  utilizing a signaling message transmitted over a signaling pathway  140  and communicate routing information for the destination ATA  105  to the initiating ATA  100  utilizing the signaling pathway  130 .  
      Once connected, a VOIP phone call over the Internet from the initiating phone  120  to the destination phone  125  connected to a POTS line  145  comprises a real time protocol (RTP) media stream  150  between the initiating ATA  100  and the destination ATA  105 . As discussed below, a shared POTS line  151  between the destination ATA  105  and the PSTN  115  is available to be shared between a phone  160  (e.g., for a PSTN phone call from the phone  160  to the destination phone  125 ) and the initiating ATA  100  (e.g., for a VOIP phone call from the initiating phone  120  to the destination phone  125 ). In one embodiment the phone  160  may receive a phone call originating from the PSTN  115  even if the POTS line  151  is in use by another ATA. A user database  155  coupled to the VOIP control system  110  will be discussed in conjunction with  FIG. 3  below.  
       FIG. 2  illustrates a VOIP control system  110  for connecting a VOIP phone call using a shared POTS line, in one embodiment of the invention. Although the embodiment of  FIG. 2  depicts individual servers  201  through  211 , it will be appreciated that the servers  201 - 211  can be combined or distributed in various ways. For example, the servers  201 - 211  can be implemented in a decentralized peer-to-peer network rather than in a client-server configuration. Similarly, the separate databases  221 - 224  can also be combined or distributed. It will also be appreciated that the servers  201 - 211  and the databases  221 - 224  can comprise any combination of hardware and software elements.  
      A registration server  201  is configured to register ATAs coupled to the Internet. In one embodiment, the registration server  201  registers information from an ATA after the ATA is initially coupled to the Internet. In other embodiments, the registration server  201  receives asynchronous event notification messages from an ATA after the ATA changes state, for example, from an idle state to an in use (i.e., currently busy) state or vice-versa. In other embodiments, the registered ATAs transmit periodic status information to the registration server  201 . The registration server  201  updates a registration database  221  to reflect registration and status information for each registered ATA. The registration server  201  is further configured to communicate update messages to a call manager server (i.e., a routing engine)  202  and set flags in a routing tables database  222  to reflect which ATAs are available for a phone call. For example, if a particular ATA is in use, the registration server  201  can send a message to the call manager server  202  and set flags in the routing tables database  222  to flag that the particular ATA is not available as a valid route and is unavailable to receive a phone call from another ATA.  
      A proxy server  203  is configured to communicate signaling messages with ATAs across the Internet. To enable call routing between ATAs, the proxy server  203  of one embodiment issues requests for call routing information from the call manager server  202 . The call manager server  202  can be configured to determine appropriate call routing for connecting a VOIP phone call based upon the routing tables database  222  and the database  135  ( FIG. 1 ). In one embodiment, routing logic of the call manager server  202  is decentralized among a number of servers or processors (not shown).  
      In some embodiments, an applications server  204  and a features server  205  provide advanced call features, such as, for example, “follow-me” services, services that require media stream intervention, teleconferencing services, call-forwarding, 3-way calling, and/or call waiting. Advanced call features can comprise both standard session initiation protocol (SIP) features and custom-developed features. An announcement server  206  in some embodiments provides one or more sound files, such as “.wav” files, stored in an announcements database  223  to simulate a voice response. For example, the announcement server  206  may issue a .wav file containing “your call cannot be completed” for a phone call that cannot be placed.  
      A trivial file transfer protocol (TFTP) server  207  of one embodiment is configured to download routing tables, usage profiles, and firmware updates to the registered ATAs. Routing tables are described in more detail in conjunction with  FIG. 5 , below. The TFTP server  207  also provides provisioning features as described below. A benefit of the TFTP server  207  is that software or firmware in the ATAs can be updated periodically or as needed, to maintain up-to-date configuration.  
      A web server  208  of one embodiment comprises the customer-facing part of the VOIP control system  110 . The web server  208  is configured to provide web pages for the application servers  201 - 211  that interface to customers. The web server  208  of one embodiment provides a customer support server  209  with web pages by which the customer can communicate with the customer support server  209 . The customer support server  209  of one embodiment is configured to control customer account and customer access information. The customer support server  209  supports functions such as customer relationship management (CRM) and/or online trouble resolution. For example, a customer having difficulty setting up an ATA may log into a smart knowledge base hosted on the customer support server  209  and use online chat with a technician for trouble resolution. Further, the customer support server  209  can provide a provisioning application for initial setup and service activation.  
      A call log server  210  of some embodiments is configured to track signaling messages and other events associated with PSTN and VOIP phone calls. For example, the call log server  210  can track a time a call is initiated and a time a call is ended. The call log server  210  can also compute a call duration. In one embodiment, each registered ATA collects such call detail information and passes the call detail information to the call log server  210 . The call log server  210  can maintain the call detail information, for example, in a call detail record (CDR) file. In some embodiments, the call log server  210  advantageously supports Communications Assistance for Law Enforcement Act (CALEA) requirements, provides call logs to customers, and supports customer billing.  
      An operation support system (OSS) server  211  is configured to provide internal support of the servers  201 - 210  and the databases  221 - 224 . In one embodiment, the OSS server  211  provides for connections between databases and servers via simple network management protocol (SNMP). A customer profile/accounting (i.e., billing account management) database  224  of one embodiment contains records detailing customer premises equipment (CPE), a summary of CDR data from the call log server  210 , billing records, and customer trouble history.  
       FIG. 3  illustrates a user database  155  for connecting a VOIP phone call using a shared POTS line, in one embodiment of the invention. The user database  155  of this embodiment comprises two user databases tied to the registration database  221  of  FIG. 2 . An account management database  301  contains records of customer information, for example residence address, telephone number, billing address, trouble history, credit history, bill presentation, and subscribed features/services. A network database  302  comprises a customer profile database linked by customer ID. The network database  302  includes information such as user ID, registration information, a list of support services, personal identification numbers (PIN), and currently registered numbering plan area (NPA) codes (i.e., an area code), and central office (NXX) codes (i.e., a local number prefix), as described below in conjunction with  FIG. 5 .  
       FIG. 4  illustrates an ATA  400  for connecting a VOIP phone call using a shared POTS line, in one embodiment of the invention. The ATA  400  of this embodiment (e.g., the initiating ATA  100  or the destination ATA  105 ) comprises CPU/firmware  401 , input/output (I/O) ports  402  through  405 , and status indicators (e.g., LEDs)  407 . The CPU/firmware  401  of one embodiment includes memory (not shown) to store a routing table downloaded from the TFTP server  207  of  FIG. 2 . An FXO port  402  is configured to interface to the PSTN (e.g., PSTN  115  of  FIG. 1 ) via a POTS line (e.g., the POTS line  152  of  FIG. 1 ). In one embodiment, the ATA  400  includes multiple FXO ports  402 . A foreign exchange subscriber (FXS) port  403  is configured to interface to an analog phone (e.g., the initiating phone  120  of  FIG. 1 ). In one embodiment, the FXS port  403  is configured to interface to multiple analog phones, so that a single ATA can interface to all phones in a residence, for example. A first RJ-45 connector  404  provides for a wide area network (WAN) (e.g., Ethernet) interface to the Internet to communicate signaling messages with the VOIP control system  110  ( FIG. 1 ) and media streams with other ATAs. A second RJ-45 connector  405  can optionally be provided for a local area network (LAN) interface to a personal computer, router, and/or hub. In one embodiment, the ATA  400  includes router circuitry (not shown) to provide router functions if both the WAN interface and the LAN interface are connected to external networks. Status indicators  407  provide, for example, indications of activity, power on, voicemail, and line-in use. The ATA  400  includes circuitry (not shown) to convert analog signals into packets, and a buffer and voice coder/decoder (not shown) to convert packets into analog signals. In one embodiment, high performance coder/decoder circuits (not shown) in the ATA  400  support high fidelity media transmission. In another embodiment, encryption circuitry (not shown) is included in the ATA  400  to encrypt the transmitted media stream. In other embodiments, DSP circuitry (not shown) is included in the ATA  400  so that two phone calls can be maintained simultaneously in a single ATA  400 . In this embodiment, for example with respect to  FIG. 1 , the phone  160  may connect and transmit a media stream over the Internet to the ATA  100  and phone  120  while simultaneously, the ATA  105  receives a media stream from another ATA  400  and couples the received media stream to the POTS line  151  and the local PSTN  115 .  
       FIG. 5  illustrates a method for connecting PSTN and VOIP phone calls using a shared POTS line, in one embodiment of the invention. The method begins in step  500  by transmitting a phone number of the destination phone  125  ( FIG. 1 ) from the initiating phone  120  ( FIG. 1 ) to the initiating ATA  100  ( FIG. 1 ). The phone number comprises an International Telecommunications Union Recommendation E.164 compliant number (i.e., a ten-digit number in the U.S. PSTN). The initiating phone  120  sends a dual tone multiple frequency (DTMF) tone to the initiating ATA  100  for each digit of the phone number dialed into the initiating phone  120 .  
      At step  505 , the initiating ATA  100  determines whether the phone number comprises an emergency code, for example “911.” in the U.S. and Canada. If so, then at step  510  the initiating ATA  100  determines whether the FXO port (e.g., FXO port  402  of  FIG. 4 ) is available. The FXO port  402  may not be available, for example, because the initiating ATA  100  is connected to the PSTN  115  ( FIG. 1 ) in cooperation with another ATA (e.g., to place a VOIP phone call from the phone  160  of  FIG. 1 , in which the initiating ATA  100  performs receiving functions as described with respect to the destination ATA  105 ). In one embodiment, if the FXO port  402  is not available, at step  515 , the initiating ATA  100  disconnects an existing call using the FXO port  402 . At step  520 , the initiating ATA  100  seizes the available FXO port  402  and places a “911” call to the PSTN  115 . In another embodiment, at step  520 , rather than disconnecting the existing call if the FXO port  402  is not available, the initiating ATA  100  reroutes the existing call to another ATA. In other embodiments, if the FXO port  402  is not available, the initiating ATA  100  and the VOIP control system  110  cooperate to reroute the existing call to another ATA. It will be appreciated that although rerouting existing calls is described with respect to “911” phone calls, rerouting may occur for any condition in which the FXO port  402  is not available.  
      The following steps  525 - 550  describe a “lowest cost route” call routing algorithm. A local PSTN route, for example, one in which both the initiating phone  120  and the destination phone  125  are within the local calling area, and the FXO port  402  of the initiating ATA  100  is not busy, is the lowest cost routing. Referring to  FIG. 1 , an example of a local PSTN route is shown from the initiating phone  120 , across the initiating ATA  100 , the POTS line  152 , the PSTN  115 , the POTS line  145 , and to the destination phone  125 . A more expensive call route is a remote PSTN route using the shared POTS line (e.g., the POTS line  151  of  FIG. 1 ). An example of a remote PSTN route is shown in  FIG. 1  from initiating phone  120 , across the initiating ATA  100  to the destination ATA  105  via media stream  150 , the shared POTS line  151 , the PSTN  115 , and to the destination phone  125 . The remote PSTN route may be necessary, for example, if the FXO port  402  of the initiating ATA  100  is busy (e.g., connected to the PSTN  115  via the POTS line  152  in conjunction with reception of a VOIP phone call from another initiating ATA not shown in  FIG. 1 ). If the FXO port  402  of the initiating ATA  100  is in use, the phone call from the initiating phone  120  to the destination ATA  105  and the destination phone  125  may be treated as a VOIP phone call. The highest cost route comprises a path (not shown) from the initiating phone  120  to a commercial ITSP through the Internet.  
      At step  525  of  FIG. 5 , if the phone number of the destination phone  125  does not comprise “911,” the initiating ATA  100  determines whether the phone number represents a destination phone  125  within the local calling area (i.e., local calling radius or zone). The determination can be made, for example, by the CPU/firmware  401  ( FIG. 4 ) using a routing table that defines the local calling area previously downloaded from the TFTP server  207  ( FIG. 2 ) to the initiating ATA  100 . In one embodiment, the routing table comprises a subset of the database  135  of  FIG. 1 . For example, in the U.S. PSTN, the routing table comprises NPA and NXX codes in the local calling area of the initiating ATA  100 , toll-free NPAs such as 800, 877, or 866, pay numbers such as 900 or 976, three-digit sequences such as 511, 411, or 911, and operator, international and carrier-access codes such as 0, 0+, 01, 00, 011, or 1010.  
      At step  530  of  FIG. 5 , the initiating ATA  100  determines whether the FXO port (e.g., the FXO port  402  of  FIG. 4 ) is available. If the FXO port is available, at step  520 , the initiating ATA  100  dials the phone number of the destination phone  125  via the PSTN  115 . In one embodiment, the initiating ATA  100  sends an asynchronous event notification message to the registration server  201  ( FIG. 2 ) to reflect that the initiating ATA  100  has changed status from an idle state to an in use state. In another embodiment, throughout the duration of the PSTN phone call, the initiating ATA  100  periodically sends status messages to the registration server  201 . While the initiating ATA  100  is in use, the VOIP control system  110  ( FIG. 1 ) of one embodiment flags the initiating ATA  100  as unavailable to accept a VOIP phone call from another ATA in the routing tables database  222  ( FIG. 2 ).  
      If the destination phone number does not comprise a phone call within the local calling area, or if the FXO port  402  is unavailable, at step  535 , the initiating ATA  100  queries the VOIP control system  110  for a destination ATA  105  in the local calling area of the destination phone  125 , to establish a remote PSTN route. The initiating ATA  100  sends a SIP message via the signaling pathway  130  ( FIG. 1 ) to communicate the phone number of the destination phone  125  to the VOIP control system  110 . In some embodiments, to prevent unregistered (e.g., unsubscribed or stolen) ATAs from placing calls, the VOIP control system  110  searches the user database  155  ( FIG. 1 ) to verify that the initiating ATA  100  is registered and properly authenticated.  
      At step  540 , the VOIP control system  110  compares the phone number of the destination phone  125  to the database  135  ( FIG. 1 ) to select the best available destination ATA  105  ( FIG. 1 ) from the registered ATAs. The VOIP control system  110  determines, based upon one or more of the dialed NPA, NXX, or ten-digit phone numbers, which of the registered ATAs is available to connect the VOIP phone call. Specifically, in one embodiment, the call manager server  202  ( FIG. 2 ) of the VOIP control system  110  determines from the routing tables database  222  which ATAs are available, and selects the available ATA as the destination ATA  105  in the same local calling area as the destination phone  125 . The call manager server  202  selects the destination ATA  105  in the same local calling area as the destination phone  125  to minimize the cost of the call through the PSTN  115  to the destination phone  125 . The call manager server  202  then supplies an identification of the selected destination ATA  105  to the proxy server  203  ( FIG. 2 ). In one embodiment, the proxy server  203  of the VOIP control system  110  sends an invite message to the selected destination ATA  105  using a signaling message. The destination ATA  105  acknowledges the VOIP control system  110 , and the VOIP control system  110  responds with an acknowledge message to the initiating ATA  100 . If the destination ATA  105  responds that it is available, the proxy server  203  replies to the initiating ATA  100  with routing information for the destination ATA  105 .  
      At step  545 , the initiating ATA  100  sends the phone number of the destination phone  125  to the destination ATA  105 . The destination ATA  105  accesses the PSTN  115  to dial the phone number of the destination phone  125 . After connecting to the PSTN  115 , the initiating ATA  100  transmits media stream  150  to the destination ATA  105 . In one embodiment, throughout the duration of the VOIP phone call, the initiating ATA  100  and the destination ATA  105  periodically send status messages to the registration server  201 . In another embodiment, when the initiating ATA  100  and the destination ATA  105  change state from idle to in use, the initiating ATA  100  and the destination ATA  105  send asynchronous event notification messages to the registration server  201 . The VOIP control system  110  flags, in the routing tables database  222 , the initiating ATA  100  and the destination ATA  105  as unavailable to accept a VOIP phone call from other ATAs.  
      In one embodiment, if no ATAs are available in the local calling area of the destination phone  125 , at step  550 , the VOIP phone call may connect to an ITSP. In one embodiment, if no routes are available to the destination ATA  105 , the call manager server  202  requests a .wav file from the announcement server  206  ( FIG. 2 ), such as for example, “customer not available” to provide to the initiating ATA  100 . In one embodiment, if the FXO port  402  of the initiating ATA  100  is busy, phone calls to 800, 900, 511, 411, 0+, and 1010 are not routed to the destination ATA  105 , but rather the initiating ATA  100  sends a busy signal to the initiating phone  120 .  
      Selecting the destination ATA  105  based upon the phone number of the destination phone  125  and communicating routing information for the destination ATA  105  to the initiating ATA  100  shares the POTS line  151  ( FIG. 1 ) between the initiating ATA  100  and the phone  160  connected to the destination ATA  105 . The POTS line  151  is shared because the destination ATA  105  can initiate a phone call from the phone  160  to the PSTN  115 , and can connect the phone call from the initiating ATA  100  to the destination phone  125 , using the a single POTS line  151 .  
      A benefit of sharing the POTS line  151  is that the phone call to the PSTN  115  from the destination ATA  105  may be a low-cost local call. Further, in the case where the shared POTS line  151  is a residential POTS line, the shared POTS line  151  is relatively unused by the phone  160  throughout the course of a day. Sharing the POTS line  151  reduces the cost of phone calls through increased utilization of existing POTS lines  151 , and by placement of calls to local PSTNs  115 .  
      Further, the caller at the initiating phone  120  is not required to choose whether to use VOIP or PSTN  115 . As connection and routing logic automatically determines whether to use the Internet or the PSTN  115  to connect the call to the destination phone  125 , selection of an appropriate ATA allows the VOIP phone call to be placed to the destination phone  125  that otherwise would not be VOIP-compatible, and without requiring the caller to dial additional digits or characters.  
       FIG. 6  illustrates a method for registering an ATA, in one embodiment of the invention. A benefit of registering ATAs is to assure that unsubscribed or stolen ATAs cannot be used for making free calls. A further benefit of registering the ATA is to dynamically determine the local calling area of the ATA. At step  601 , the ATA sends its media access control (MAC) address to the registration server  201  ( FIG. 2 ) by way of a signaling message. At step  602 , the registration server  201  provides a toll-free (e.g., “800-xxx-xxxx”) number to the ATA. At step  603 , the ATA dials the 800-number over the PSTN (e.g., PSTN  115  of  FIG. 1 ) and connects to the authentication server (not shown) of the VOIP control system of  FIG. 2 . At step  604 , the authentication server collects the caller ID from the ATA. In one embodiment, at step  605 , the authentication server checks for a match between the caller ID and the local phone number of the FXO. At step  606 , if the numbers match in step  605 , the ATA is registered.  
      In the foregoing specification, the present invention is described with reference to specific embodiments thereof, but those skilled in the art will recognize that the present invention is not limited thereto. Various features and aspects of the above-described present invention may be used individually or jointly. Further, the present invention can be utilized in any number of environments and applications beyond those described herein without departing from the broader spirit and scope of the specification. The specification and drawings are, accordingly, to be regarded as illustrative rather than restrictive. It will be recognized that the terms “comprising,” “including,” and “having,” as used herein, are specifically intended to be read as open-ended terms of art.