Abstract:
A first interface includes a packet switched network connector mechanism. A second interface includes a public switched telephone network connector mechanism. A digital signaling mechanism and an analog signaling mechanism accompany the first and second interfaces.

Description:
BACKGROUND INFORMATION 
     It has long been known for a telephone customer&#39;s premises to be connected to a public switched telephone network (PSTN) via analog lines, e.g., a copper wire loop, etc. A customer premises may be associated with one or more telephone numbers, and one or more telephones may be installed at the customer premises. However, a telephone generally cannot be moved from one customer premises to another, i.e., from one location to another, without becoming associated with a new telephone number, or without associating a telephone number with a new customer premises. Accordingly, at present, telephone customers generally may only make and receive calls using a telephone associated with a telephone number within a particular customer premises, and to avail themselves of services associated with the telephone number, when physically present in the customer premises. This is unfortunate, because it means that a telephone customer is unable to take advantage of subscribed for services when not present within a customer premises, such as a flat rate long-distance plan, a flat rate local calling plan, call forwarding, call waiting, voice mail, etc. 
     It is also known to transfer voice communications, e.g., Voice over Internet Protocol (VoIP) calls, from a packet switched network to a PSTN. However, the conversion of a call from digital to analog, or vice versa, and the accompanying translation of signaling protocols respectively used for digital and analog calls, generally takes place at a softswitch or gateway between the PSTN and the packet switched network. At present, telephones or other voice communications devices within customer premises may be attached to a packet switched network or a PSTN, but not both. It is presently not possible for a user having a telephone line connected to the PSTN to use this telephone number, and any subscribed-for functionality associated with the telephone number, by accessing a packet switched network. Accordingly, it is not presently possible for a user of a packet switched network to take advantage of the user&#39;s line to a PSTN other than by using a telephone or other device plugged into the line, nor is possible for a user subscribing to a PSTN line to take advantage of the portability and flexibility of being able to access the PSTN line from anywhere there is a connection to a packet switched network such as the Internet. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1A  illustrates an exemplary system for using a handset to place calls through a telephone base unit over a public switched telephone network (PSTN). 
         FIG. 1B  illustrates an exemplary system for using VoIP handset to place calls through a packet switched network and a telephone base unit over a PSTN. 
         FIG. 2  illustrates an exemplary call flow for registering a handset with a base unit according to Session Initiation Protocol (SIP). 
         FIG. 3  illustrates an exemplary call flow for placing a call from a handset through a base unit. 
     
    
    
     DETAILED DESCRIPTION OF PREFERRED EMBODIMENTS 
       FIG. 1A  illustrates an exemplary system  100  for using a handset  105  to place calls through a telephone base unit  120  over a public switched telephone network (PSTN)  130 . 
     Handset  105  is generally a handheld computing device including a processor and a memory that is capable of storing program instructions and of sending and receiving communications with a router  112 , e.g., via an Ethernet connection or the like. Such an Ethernet connection may be accomplished through a variety of known mechanisms, such as a Cat5 cable, or a radiofrequency connection according to known protocols such as IEEE 802.11, etc. Handset  105  transmits voice communications to router  112 , e.g., according to some known form of Voice over Internet Protocol (VoIP). 
     Router  112  may be any one of a number of known routers that connect network devices such as handset  105  to other network devices, such as base unit  120 . 
     Base unit  120  also includes a processor and a memory and is also capable of digital communications with router  112  via a variety of known mechanisms such as those described above. It is possible that base unit  120  could be incorporated into a private branch exchange (PBX) or the like, thereby making the functionality of the PBX accessible through the base unit  120 . 
     Base unit  120  advantageously includes signaling connections for, and the capability for making and receiving, both digital and analog calls. Base unit  120  also includes a mechanism that may be hardware and/or software to provide for analog voice signals to digital voice signals and vice versa, and also for converting analog signaling protocols to digital signaling protocols and vice versa. Base unit  120  further generally includes connector mechanisms that allows it to connect to, and communicate with, both to router  112  and central office  125 . For example, as illustrated in  FIGS. 1A and 1B , base unit  120  may include an interface for an RJ11 jack or the like, as well as an interface for a modular connectors such as an RJ45 jack, a wireless connection to router  112 , etc. A public switched telephone network connector mechanism such as an RJ11 jack allows base unit  120  to connect to central office  125  via a conventional wire loop. At the same time, a packet switched network connector mechanism such as RJ45 jack allows base unit  120  to connect to router  112  and/or packet switched network  115  (illustrated in  FIG. 1B  and discussed below). 
     In an exemplary embodiment, base unit  120  includes a digital signal processor (DSP). As is known, a DSP may be used to process digital signals, such as digital voice signals, and provide analog output representing the digital signals. Similarly, in an exemplary embodiment, base unit  120  includes an analog-to-digital converter (ADC) such as is known for converting analog signals such as analog audio signals, e.g., voice signals, to a digital representation of the analog signals. 
     Further, in an exemplary embodiment, base unit  120  includes program instructions for communicating with handset  105  using a signaling protocol such as Session Initiation Protocol (SIP). SIP is well known and is described in J. Rosenberg et al., SIP: Session Initiation Protocol, RFC 3261, published in June 2002 by The Internet Society of Reston, Va. In this embodiment, base unit  120  further includes program instructions for converting SIP messages to analog call signals, and vice versa. 
     Base unit  120  communicates with a central office  125 . Central office  125  is well known for including switches that connect a telephone unit such as base unit  120  with a PSTN  130 . 
     A conventional telephone  135  may be connected to PSTN  130 , as is known, e.g., through a central office  134 . Accordingly, VoIP handset  105  may place and receive calls to and from telephone  135  through base unit  120  and PSTN  130 . Of course, although not illustrated in  FIG. 1A  or  1 B, handset  105  could also place and receive calls to and from other VoIP phones and/or through base unit  120  and PSTN  130 , inasmuch as it is known for VoIP phones to access PSTN  130  using softswitches, gateways, and the like. Central offices  125  and  134  are sometimes thought of as being included within PSTN  130 . 
       FIG. 1B  illustrates an exemplary system  101  for using VoIP handset  105  to place calls through a packet switched network  115  and a telephone base unit  120  over a PSTN  130 . It may be noted that system  101  is described with certain elements in common with system  100  illustrated in  FIG. 1A . However, whereas system  100  illustrated in  FIG. 1A  may be suitable for allowing handset  105  to be used through a local area network (LAN) connection in a customer premises, system  101  illustrated in  FIG. 1B  may be suitable for allowing handset  105  to be used through a wide area network (WAN) such as the Internet or the like. Accordingly, system  101  may allow a user to use handset  105  to make and receive calls, and access other services, via base unit  120 , while at an unlimited number of locations around the world. 
     In system  101 , handset  105  communicates with base unit  120  through packet switched network  115  and routers  110  and  111 . Packet switched network  115  may be an Internet protocol (IP) network or the like such as is known for transporting digital data packets. Similar to router  112 , routers  110  and  111  may be any one of a number of known routers for routing digital packets, and for connecting network devices to each other and/or to packet switched network  115 . 
     Computing devices such as handset  105 , base unit  120 , etc. may employ any of a number of computer operating systems known to those skilled in the art, including, but by no means limited to, known versions and/or varieties of the Microsoft Windows® operating system, the Unix operating system (e.g., the Solaris® operating system distributed by Sun Microsystems of Menlo Park, Calif.), the AIX UNIX operating system distributed by International Business Machines of Armonk, N.Y., and the Linux operating system. While handset  105  generally is a handheld computing device and base unit  120  is generally a desktop unit, it is to be understood that computing devices including handset  105  and base unit  120  may include any one of a number of computing devices known to those skilled in the art, including, without limitation, a computer workstation, a desktop, notebook, laptop, or handheld computer, or some other computing device known to those skilled in the art. 
     Computing devices such as handset  105 , base unit  120 , etc. generally each include instructions executable by one or more computing devices such as those listed above. Computer-executable instructions may be compiled or interpreted from computer programs created using a variety of programming languages and/or technologies known to those skilled in the art, including, without limitation, and either alone or in combination, Java™, C, C++, Visual Basic, Java Script, Perl, etc. In general, a processor (e.g., a microprocessor) receives instructions, e.g., from a memory, a computer-readable medium, etc., and executes these instructions, thereby performing one or more processes, including one or more of the processes described herein. Such instructions and other data may be stored and transmitted using a variety of known computer-readable media. 
     A computer-readable medium includes any medium that participates in providing data (e.g., instructions), which may be read by a computer. Such a medium may take many forms, including, but not limited to, non-volatile media, volatile media, and transmission media. Non-volatile media include, for example, optical or magnetic disks and other persistent memory. Volatile media include dynamic random access memory (DRAM), which typically constitutes a main memory. Transmission media include coaxial cables, copper wire and fiber optics, including the wires that comprise a system bus coupled to the processor. Transmission media may include or convey acoustic waves, light waves and electromagnetic emissions, such as those generated during radio frequency (RF) and infrared (IR) data communications. Common forms of computer-readable media include, for example, a floppy disk, a flexible disk, hard disk, magnetic tape, any other magnetic medium, a CD-ROM, DVD, any other optical medium, punch cards, paper tape, any other physical medium with patterns of holes, a RAM, a PROM, an EPROM, a FLASH-EEPROM, any other memory chip or cartridge, a carrier wave as described hereinafter, or any other medium from which a computer can read. 
       FIG. 2  illustrates an exemplary call flow  200  for registering handset  105  with base unit  120  according to Session Initiation Protocol (SIP). However, other protocols could be used to register handset  105  with base unit  120 . It is to be noted that  FIG. 2  is in the format of a standard call flow diagram. Messages shown in the column under handset  105  originate in handset  105  and are directed to base unit  120 , while messages shown in the column under base unit  120  originate in base unit  120  and are directed to handset  105 . 
     At  205 , handset  105  sends a SIP REGISTER message to base unit  120 . Handset  105  generally includes in memory a network address, e.g., an Internet protocol (IP) address for base unit  120 , to which handset  105  is programmed to send the registration message of block  205 . 
     Next, at  210 , base unit  120  responds to the message sent in  205  with a challenge and request for authentication. Authentication in the context of SIP generally involves a known process for the exchange of encrypted keys, which known process is consistent with, and is generally part of, process  200 , including blocks  210 -_ 220 . 
     Next, at  215 , handset  105  sends a SIP REGISTER message to base unit  120  including authentication credentials. 
     Then, at  220 , base unit  120  responds to the registration message sent in step  215  with a message indicating acceptance of the authentication credentials provided in step  215 . Of course, although not illustrated in  FIG. 2 , it is possible for the authentication attempt of block  215  to fail, in which case call flow  200  is terminated. 
     In any event, following block_ 220 , call flow  200  ends. However, assuming that the authentication attempt succeeds, handset  105  may proceed to place and receive calls, or access other services available through base unit  120 , an example of which is described next with reference to  FIG. 3 . In some embodiments, handset  105  may send a status message to base unit  120  at predetermined intervals, whereupon base unit  120  requires handset  105  to re-authenticate if a predetermined interval expires without a status message from handset  105 . 
       FIG. 3  illustrates an exemplary call flow  300  for placing a call from handset  105  through base unit  120 . It is to be understood that call flow  200  is generally a prerequisite for call flow  300 . As with  FIG. 2 , it is to be noted that  FIG. 3  is in the format of a standard call flow diagram. Messages shown in the column under handset  105  originate in handset  105  and are directed to base unit  120 , while messages shown in the column under base unit  120  originate in base unit  120  and are directed to handset  105  or central office  125 . Messages shown in the column under central office  125  originate in central office  125  and are directed to base unit  120 . 
     Next, at  305 , handset  105  sends a SIP INVITE message to base unit  120 . Such a message may be initiated, for example, by a user dialing a telephone number in handset  105 , e.g., a number of telephone  135 , and pressing a send button or the like to initiate a call. 
     Next, at  310 , program instructions in base unit  120  parse the INVITE message sent in block  305 , and cause an “off hook” signal to be sent to central office  125 . Such a signal is well-known with respect to conventional analog telephones for indicating that the telephone is ready to dial a call. 
     In response to the “off hook” signal sent at  310 , at  315 , central office  125  next sends a dial tone to base unit  120 . 
     Next, at  320 , base unit  120  sends a message to handset  105  indicating that base unit  120  is trying to place the call requested by handset  105  at block  305 . Program instructions in base unit  120  generally determine that a dial tone has been received as described above with respect to block  315 , and then create a SIP “trying” message to be sent to handset  105 . 
     Next, at  325 , base unit  120  dials a number requested by handset  105  at block  305  and sends the requested digits to central office  125 . Program instructions and base unit  120  may parse a telephone number provided by handset  105  at block  305 , and then dial the telephone number by providing to central office  125  a dual-tone multi-frequency (DTMF) signal or set of signals representing the telephone number, e.g., of telephone  135 . 
     Next, at  330 , central office  125  sends a “ring back” message to base unit  120 , e.g., an audible indication that telephone  135  is ringing. 
     Next, at  335 , base unit  120 , according to program instructions for handling a “ring back” message, sends a “ringing” message to handset  105 , thereby making it possible for handset  105 , e.g., according to program instructions, to indicate to a user that telephone  135  is ringing. 
     Next, at  340 , the call placed at block  305  is answered, at telephone  135 , and central office  125  signals a message so indicating to base unit  120 . 
     Next, at  345 , base unit  120  sends an “OK” message to handset  105 , according to program instructions for handling an answered call. 
     Next, add  350 , handset  105  response to the “OK” message sent in  345  by sending an “ACK” message to base unit  120 . 
     Next, media path  355  represents the transmission of call data, e.g., voice data, between handset  105  and telephone  135 . As mentioned above, base unit  120  may include a DSP, and ADC, etc. for converting digital signals to analog and vice-versa. 
     Next, at  360 , the call ends and handset  105  accordingly sends a SIP BYE message to base unit  120 . 
     Next, at  365 , according to program instructions for handling a BYE message, base unit  120  signals central office  125  that it is now “on hook.” 
     Next, at  370 , base unit  120  sends a SIP OK message to hand said  105 . 
     Call flow  300  ends following block  370 . 
     Call flow  300  is intended to be exemplary, and not limiting. Besides placing a call using handset  105 , numerous other calling scenarios are possible by using handset  105  and base unit  120  in systems  100  or  101 . Certain examples, also not intended to be limiting, are provided in the following paragraphs. 
     In one example, when a telephone call is received at base unit  120 , e.g., from telephone  135 , it is possible, e.g., using SIP, to too send an INVITE message to handset  105 , and to establish a call between handset  105  and telephone  135 . Moreover, because SIP supports call forwarding, it is further possible that handset  105  could forward calls received via base unit  120  to some other destination. 
     Further, SIP or some other protocol could be used in handset  105  to access conference calling functionality available through base unit  120 . SIP messages for establishing a conference call are known. Base unit  120  could be configured, e.g., could include program instructions, to translate such SIP messages to analog messages that would establish a conference call, e.g., a three-way call over PSTN  130 . 
     Also, SIP includes the ability to set a “do not disturb” state. Accordingly, handset  105  may send a message indicating a do not disturb state to base unit  120 , which may then send an “off hook” message upon receiving a call or may forward calls to voicemail. 
     Again, the foregoing scenarios are exemplary, and it is to be understood that possible embodiments include virtually any scenario in which a digital signaling protocols such as SIP may be converted to analog signals, and vice versa. 
     CONCLUSION 
     With regard to the processes, systems, methods, heuristics, etc. described herein, it should be understood that, although the steps of such processes, etc. have been described as occurring according to a certain ordered sequence, such processes could be practiced with the described steps performed in an order other than the order described herein. It further should be understood that certain steps could be performed simultaneously, that other steps could be added, or that certain steps described herein could be omitted. In other words, the descriptions of processes herein are provided for the purpose of illustrating certain embodiments, and should in no way be construed so as to limit the claimed invention. 
     Accordingly, it is to be understood that the above description is intended to be illustrative and not restrictive. Many embodiments and applications other than the examples provided would be apparent to those of skill in the art upon reading the above description. The scope of the invention should be determined, not with reference to the above description, but should instead be determined with reference to the appended claims, along with the full scope of equivalents to which such claims are entitled. It is anticipated and intended that future developments will occur in the arts discussed herein, and that the disclosed systems and methods will be incorporated into such future embodiments. In sum, it should be understood that the invention is capable of modification and variation and is limited only by the following claims. 
     All terms used in the claims are intended to be given their broadest reasonable constructions and their ordinary meanings as understood by those skilled in the art unless an explicit indication to the contrary in made herein. In particular, use of the singular articles such as “a,” “the,” “said,” etc. should be read to recite one or more of the indicated elements unless a claim recites an explicit limitation to the contrary.