MXPA98002804A - Method and apparatus for transmitting and routing voice telephone calls on a computer network connected in paque - Google Patents

Abstract

A method and system for routing and transmitting voice conversations is provided through a packetized computer network (200) and a public network of circuit switched telephones (300). The conversion between the protocols of the network of packet-switched computers and the protocols of the network of switched telephones in circuits is carried out by means of one or more telephone switches (600) which are coupled to the network of packet switched computers (200) and the network of switched telephones in circuits (300). The routing of voice conversations between multiple telephone switches coupled to the network of packet switched computers (200) is carried out by one or more routing servers (500) coupled to the network of packet switched computers (200), or to a computer user's local (10

Description

METHOD AND APPARATUS FOR TRANSMITTING AND ROUTING TELEPHONE CALLS OF VOICE ON A NETWORK OF COMPUTERS SWITCHED IN PACKAGES This application claims priority to the request of Patent of the United States of America with Number of Series: 08 / 542,641, filed on October 13, 1995, which is hereby incorporated by reference in its entirety.

TECHNICAL FIELD This invention relates to a method and architecture for the transmission and routing of voice signals over a packet switched network and more particularly to a method and system for routing and converting voice signals between a public switched telephone network in circuits. ("circuit switched telephone network") and a network of packet switched computers.

PREVIOUS TECHNIQUE The advantages of transmitting voice information in a packet form have been recognized for some time. Packet switching provides an easy solution to problems where the voice information to be transmitted is presented in bursts, with significant pauses between bursts. The application of compression techniques for digitized voice transmissions often results in these characteristic transmissions. The traditional telephone service, the so-called Old Ordinary Telephone Service (POTS), is provided over a switched network in circuits which dedicates a sequence of physical links through nodes of the network of switched telephones between ordinary old telephone service stations . At each node, the incoming voice signals are routed to the appropriate output channel without delay. Circuit switched networks typically dedicate a multiplexed communication path, in space and / or time division multiplexing, between the caller and the calling party that lasts the entire duration of the call. In contrast, in packet switched networks, which are typically associated with "data" transmissions instead of voice conversations, it is not necessary to dedicate the transmission capacity along a sequence of physical links through the network. net. Instead, the data is sent in packets which pass from node to node through the network. Each data packet typically consists of several elements that include the address of the data source, the destination address of the data, error verification information, as well as the actual data sent. Each node briefly stores and analyzes the packet and then transmits it to the next node. Current technologies allow a voice signal to be digitized and compressed. When several digitized compressed voice conversations are transmitted over a network, significant savings in bandwidth can be realized through packet switched transmissions of voice conversations. As mentioned above, traditional circuit switched networks require a constant allocation of bandwidth for each voice channel in the network. Statistically, this results in an inefficient use of bandwidth due to the large amount of time in which relatively little voice information is being transmitted. For example, for many voice conversations, only one channel at a time is sufficient for a large portion of the conversation. Compression techniques are available which reduce the total voice data that is being transmitted, however, these techniques often result in bursts of data of limited duration. To accommodate these potential bursts of data transmissions, circuit switched networks must assign a constant bandwidth for each voice channel which is large enough to transmit the "widest" burst of data possible. Thus, although compression techniques can make enormous savings in terms of the total data transmitted, they nevertheless require a relatively inefficient allocation of bandwidth in a switched network in circuits. Instead, packet switched transmission of voice information, can reduce the total bandwidth of the system, and results in a lower cost system, multiplexing several simultaneous conversations so as to take advantage of the statistical characteristics of voice data digital tablets. Personal computers equipped with available audio signal processing boards allow a user's voice to be digitized and transmitted to a second personal computer. This second personal computer will then convert the digitized transmission back to an analog audio signal and amplify the signal for an audio output, reproducing the first user voice. Typically a pair of modems is used to transmit the digitized information. In one mode of operation, the digitized voice information is transmitted directly over a telephone network switched in circuits to the second personal computer. In a second mode of operation, the digitized voice information is transmitted via a packet switched network. Typically, the network switched into packets will be the Internet ("Internet"). The Internet PhoneMR, available from VocalTech Inc., Northvale, New Jersey, and the Personal Internet Companion Kit ™ available from Camelot Corp., Dallas, Texas, uses this second method of operation to communicate between two computers equipped with audio coupled to the Internet. The transmission of digitized voice conversations through this second mode of operation over long distances allows the user to save significant amounts of money. This reduced cost is partially a result of the efficiency of packet switched networks over switched networks in circuits. Additionally, user savings are also a result of the fact that packet switched networks typically charge the user based on either the amount of information transmitted or the user's connection time, rather than a function of distance to the user. which voice conversation travels, which is typical in circuit switched telephone networks. Although the transmission of voice conversations through a packet switched network may result in some aspects of lower quality sound, due to the occasional delays introduced in the system nodes or data loss, many users can accept these delays as a transaction in order to make significant cost savings. However, the addressing protocols and mechanisms used in switched telephone networks in circuits and the Internet are not compatible, and therefore do not allow a user to easily establish a voice conversation over the Internet, which either originates or terminates at an ordinary old telephone service station. There is a need, therefore, for a method and system for establishing a voice conversation between an ordinary old telephone service station coupled to a circuit switched telephone network and a computer with audio connected to a network of packet switched computers , like the Internet. Moreover, because this system ideally uses a plurality of doors, or access points, to gain access to the circuit switchboard telephone network in a plurality of geographic locations, there is also a need for a method and system for using a plurality of doors to route voice calls between a network of switched telephones in circuits and a network of packet switched computers. There is also a need for a method and system to authorize these calls. Users of ordinary old telephone service may also wish to use the Internet, or a network of computers switched in similar packages, to save money on voice conversations between ordinary old telephone service stations. There is also a need H. , therefore, for a method and system of transmitting a voice conversation between two ordinary old telephone service stations where at least a portion of the speech conversation path between the two ordinary old telephone service stations is transmitted through a public network of packet switched computers, generally accessible, such as the Internet.

INDUSTRIAL APPLICABILITY The object of the present invention is to provide a system for establishing a voice conversation fa computer equipped with audio connected to a network of packet switched computers, such as the Internet, to an ordinary old telephone service station coupled to a network of switched telephones in circuits. It is another object of the present invention to provide a method and system for transmitting a voice conversation between two ordinary old telephone service stations where the speech conversation path between the two stations is routed through a network of switched telephones in circuits. public and a network of packet-switched computers, such as the Internet. The present invention is directed to a method and system for routing and transmitting voice conversations between a computer equipped with audio and an ordinary old telephone system station through a network of packet switched computers such as the Internet. The present invention further provides a method and system for routing and transmitting a voice conversation between two ordinary old telephone service stations which at least are partially transmitted over a network of packet switched computers. Ordinary old telephone service stations are coupled to the system through one or more circuit switched telephone networks. A route server is provided for routing calls between multiple destinations in the network of packet switched computers. A telephone switch is provided for converting protocols fa network of packet-switched computers to a network of circuit-switched telephones.

BRIEF DESCRIPTION OF THE DRAWINGS For a more complete understanding of the present invention, reference is made to the following Detailed Description taken in conjunction with the accompanying drawings in which: Figure 1 is a high-level block diagram of an architectural system of according to the present invention; Figure 2A is a functional block diagram of a system architecture for supporting a voice conversation between a personal computer equipped with sound and an ordinary old telephone service station in accordance with the present invention; Figure 2B is a functional block diagram of a system architecture for supporting a voice conversation between two ordinary old telephone services through a network of packet switched computers according to the present invention; Figure 3 is a block diagram of a personal computer system in which the client software of the present invention can be incorporated; Figure 4A is a flowchart illustrating a method of implementing a telephone switch for bridging voice conversations between the network of packet switched computers and the circuit switched telephone network in accordance with the present invention; Figure 4B is a functional block diagram of a telephone switch according to the present invention; Figure 5 is a flow chart illustrating a method for registering users with the system according to the present invention. Figure 6 is a functional block diagram illustrating database models in accordance with the present invention; and Figure 7 is a schematic representation of a data packet for transmitting speech and / or control information in accordance with the present invention.

BEST MODES FOR CARRYING OUT THE INVENTION Preferred embodiments of the present invention will now be described with continued reference to the drawings. 1. General view Figures 1 and 2A show an overview of the architecture of the system. The system consists of a personal computer 100 running client application software 101 and a system server 500. To establish a voice conversation fthe personal computer 100, the client application software 101 connects, over the computer network 200, with the router authentication server 500 and requests a voice connection with a specific telephone number. The system server 500 uses a specialized telephone switch 600 to dial the telephone number via the network of switched telephones on circuits 300. The preferred embodiment includes a plurality of telephone switches 600 (Figure 2A) in various places. Each of the switches 600 is coupled to both the computer network 200 and the network of switched telephones on circuits 300. The router authentication server 500 determines the optimum telephone switch 600 to route the call based on the cost of the telephone. connecting the called part to the telephone switch over the switched telephone network in circuits 300, as well as the traffic through the possible telephone switches 600. In an alternative embodiment of the present invention, multiple router authentication servers 500 can be coupled to the network of 200 packet switched computers in one or more geographic locations. The personal computer 100 then sends the call request, including any authentication data provided by the router authentication server 500, to the telephone switch 600. The telephone switch 600 verifies the authentication data, either through communication with the router authentication server 500, or through other security means such as a digital signature generated by the router authentication server 500. Telephone switch 600 sends a signal indicating disengagement to the network of switched telephones in circuits 300 and tones or pulses corresponding to the telephone number of the called party over the network of switched telephones in circuits 300. The telephone switch 600 then waits for a response signal from the network of switched telephones in circuits 300 indicating that the remote telephone 400 has been off-hooked and answered the call. After the remote telephone 400 answers and a call is established, the telephone switch 600 then converts the voice data received from the network of switched telephones into 300 circuits into a format convenient for the network of 200 packet switched computers and the software of application of the client 101 through any of several known conventional techniques to implement this gate between two networks. Similarly, the telephone switch 600 converts the voice data received from the packet switched PC network 200 into a convenient format for the switched telephone network in circuits 300 through conventional door techniques. The personal computer 100 is physically connected to a network service provider 220 via a communications link 221 and modem 150 as is well known in the art. The communication link 221 may be a network of circuit switched telephones, a dedicated connection, or any of several known means. The network service provider 220 provides the personal computer 100 access to the computer network 200. The computer network 200 is preferably the Internet. 2. PC-Client Phone System As shown in Figure 3, one aspect of the present invention can be incorporated into a personal computer 100 equipped with audio, which comprises a central processor 110, a main memory 111, a keyboard 112 , a pointing device 113, such as a mouse, slide control or the like, a display device 114, a mass storage device 115, such as a hard disk, and an internal clock 116. The personal computer 100 also includes a sound device 130, which includes a signal processing unit 120. The components of the personal computer system 100 are communicated through a bus 119 of the system. In a preferred embodiment the personal computer 100 is an IBM compatible personal computer which is commercially available in many sales centers. The preferred central processor 110 will be compatible with an Intel 80486 operating at 33 MHz, or greater and more preferably an Intel Pentium ™ that operates at 75 MHz or more. Other computer systems, such as the Macintosh ™, available from Apple Computer, or the Sun SPARC R Station from Sun Microsystems ™, and other processors, such as the Motorola 680x0, the Sun Microsystems SPARC1 ^, and the Power PCMR developed jointly by Apple Computer, IBM and Motorola, are also convenient. Additionally the personal computer 100 is preferably connected to an internal or external modem 150 or similar device for communication with the computer network 200. This modem is preferably capable of transmitting a minimum of 14.4 kbs, and more preferably transmits at 28.8 kbs or more. . Alternatively, the personal computer 100 can be connected via an ISDN adapter and an ISDN line for communications with the computer network 200 or via an Ethernet connection to a network connected to the Internet or any other type of network interface. In the preferred embodiment, the sound device 130 may be any of a variety of readily available sound cards, such as the SoundBlaste card: - ^, available from Creative Labs, Inc. or the SoundChoice 32M available from Spectrum Signal Processing. The sound device 130 is connected to one or more speakers 125 and a microphone 126. The sound device 130 may optionally include a standard RJ11 telephone plug for connection to a standard analog telephone. The personal computer 100 is preferably under the control of a multi-tasking operating system that includes a TCP / IP interface, such as that available under Microsoft Windows ™, MacOS ™, UNIX ™, NextStep ™, or OS / 2 ™. The personal computer can establish a connection to the network of packet switched computers 200 via a network service provider 220 (Figure 2A). Commercial network service providers include: Hackensack IDT, New Jersey, and Performance Systems International. The network service provider preferably provides a Serial In-Line Internet Protocol (SLIP) or Point-to-Point Protocol (PPP) connection to the 200-packet switched computer network. The user initiates a call request by entering a call. standard telephone number through the graphical user interface of the client application software 101. Alternatively, the graphical user interface will allow the user to enter the name of the called party or other information which will be translated by the execution of the software. application of the client 101 on the personal computer 100 to a standard telephone number based on the user's personalized database. The client application software 101 may further incite the user to enter login name and password, or credit card number, each time a call is established. Alternatively, the client application software 101 may store the access name information and the user's password (or credit card) when the user first configures or uses the software 101 and automatically sends the login name and password (or credit card) to the router authentication server 500. The client application software 101 creates a call connection request packet containing the called party's telephone number and the user's access information, such as card information. credit or the user's access number and password. The called party number can be determined through an optional local or online directory. The call connection request packet is sent from the personal computer 100 to the router authentication server 500 (Fig. 2A). After receiving the call connection request packet, the router authentication server 500 verifies the caller's access name and password and determines the appropriate telephone switch 600 to route the direct call based on several factors, which include the traffic load in each of the telephone switches 600, and the cost of transmitting the voice conversation from the potential telephone switches 600 to the called party over the network of switched telephones in circuits 300. An alternative mode of the present invention does not use a router authentication server. Instead, the same application software of the client 101 selects a telephone switch 600. The telephone switch 600 will verify, itself, the access name or password of the caller or credit card information. The client application software 101 may use any of several techniques to select the 600 telephone switch, including an internal database that maps destination area codes and exchanges from the central office to the 600 telephone switches. This database can be uploaded and updated periodically through the network of packet-switched computers 200 according to the telephone switches, they are moved, added, deleted or temporarily out of service. The process for converting between an analog signal such as user voice input or audio output, and digitized packets suitable for transmission over packet switched PC network 200 is well known in the art. Several sound devices, such as the SoundBlaster ™ card, are available to convert between digital and analog audio signals. When converting from an audio input to digitized packet data, the audio input is first sampled or digitized. These sampled data are compressed using any of several speech compression algorithms such as GSM. In the preferred embodiment, the speech will be compressed to be transmitted at a rate of approximately 10 kilobytes / second (kbs) in order to use 14.4 kbs modem, leaving approximately 30 percent of the bandwidth available for the information of control. In the preferred embodiment, this algorithm is also capable of achieving such compression in a personal computer using an Intel 80486SX operating at 33 MHz at less than 1/2 full load. The client application software 101 is preferably installed via a self-extracting file.

The installation code determines whether the necessary hardware and software resources reside on the personal computer. This will include checking disk space and the presence of a sound device, and installing the necessary drives, such as sound drivers and the Windows plug-in interface ("winsock"). The installation process may also require the user to register with the user registration server 550 (Figure 2A). 3. Computer network The computer network 700 preferably is the World-Wide Internet ("Internet"). The Internet is a worldwide network that connects thousands of computers ("hubs") and computer networks. The Internet is organized as a multi-level hierarchy that contains local networks connected to several mid-level, regional networks. Each of the regional networks is connected to a central structure network. The dominant protocol used to transmit information between computers on the Internet is the Network Protocol: Transmission Control Protocol / Internet Protocol (TCP / IP). Computers typically connect to the Internet through a local network of telephones that connects the computer to an Internet service provider. Internet addresses are the addressing system used in communications of the transmission control protocol / Internet protocol to specify a particular network or computer in the network with which to communicate. Computers can either directly use the numeric Internet address or, alternatively, a host name plus a domain name. Domain and domain names are then translated to Internet addresses through a resolver process. 4. User registration server and billing server now referring to Figure 5, we describe user registration server 550 and billing server 560. The system preferably includes at least one user registration server 550, which stores user information, including access name, password, and billing information. The user can register either manually or through interaction with the client application software 101. The database is available for the other components of the system, such as the router authentication server 600 and the billing server 560. The server Billing 560 (Figure 2A) maintains a call history database for each call established through the system. Billing server 560 will invoice the user, either immediately or monthly. The charge can be submitted directly to the user's credit card. 5. Telephone Switch Referring now to Figure 4B, the telephone switch 600 acts to convert between the packet data transmitted over the network of packet switched computers 200 and the information transmitted over the network of switched telephones into 300 circuits. transmitted over the switched telephone network in circuits 300 may be of any variety of formats (also known as "protocols"), as described below, including analog or digital transmissions. The telephone switch 600 further performs the functions of data interference separation 611 and data injection 612 to smooth out delays by using windows of various data interferences separators that initially contain data representing silence and stamped time stamp input packets. The interference separation technique is used to smooth the delays due to the transmission of packets. The telephone switch 600 further performs compression and decompression 613 through any of several known techniques. The telephone switch 600 is logically divided into two portions, a portion of routing to send and receiving data on the network of packet switched computers 200 and a portion of voice processing card for linking to the network of switched telephones on circuits 300. The two portions preferably communicate through a data bus. The routing portion performs the function of routing multiple connections over the network of packet switched computers 200. The voice processing card portion of the telephone switch 600 consists of one or more voice processing cards, also known as telephone interface cards, which are typically inserted into input / output slots on telephone switch 600. Voice processing cards handle call control, including sending or detecting the appropriate signals to pick up, dial telephone numbers, ringer detection, answer detection, busy detection, and disconnection detection and signaling. The voice processing cards also perform analog to digital (A / D) and digital to analog (D / A) conversion where the network interface of switched telephones in circuits is in analog format or protocol. Alternatively, the voice processing cards perform the necessary protocol conversion when the network interface of switched telephones in circuits is digital, such as an IT connection. These conversions are typically transparent to the routing portion of the telephone switch 600. Additionally, the voice processing cards perform data compression and decompression as described below. Voice processing cards and associated software drivers are available from many manufacturers, including Dialogic, Rhetorex, or National Microsystems. Each voice processing card preferably provides a multi-channel interface to handle several simultaneous telephone conversations. Referring now to Figure 4A, the establishment of the call and routing from the telephone switch 600 to the circuit switched telephone network is described. The telephone switch 600 typically must respond to the following events and perform the following functions: Establish new calls by receiving an authorized call connection request packet. The telephone switch 600 should verify the connection request packet, dial the called party telephone number (633) over the switched telephone network in 300 circuits, wait for the called party to answer (638, 639, 620), and update the connection database (621). Disconnect the existing call arrangements (634) after receiving a disconnect signal in a disposition channel for the switched telephone network in circuits or a disconnect packet through the network of packet switched computers.

Unzip data from digital packets of the packet switched network after receiving a voice packet, and convert it to a convenient format ("protocol") for the circuit switched telephone network. - Digitize and compress voice data received from the network of switched telephones in circuits and convert them to packaged protocol for the switched packet network. Perform separation of audio interference. - Perform database updates for billing purposes on the establishment and disconnection of voice conversations. 6. Network and communication protocols. The general mechanism and protocols for communicating via packet switched computer networks, such as the Internet, and the circuit switched telephone network, are known in the art. See, for example, Stallings, W., Data and Computer Communications, Second Edition, Macmillan Publishing Co. (1988). Communication over the network of packet switched computers is preferably implemented through a set of standardized application layer protocols. The most preferred mode of the packet switched computer network uses the Transport Control Protocol (TCP) and the Internet Protocol (IP), or alternatively, the OSI layer model, which is also well known in the art. See, for example, Martin J., TCP / IP Networking-, PTR Prentice Hall (1994). The telephone switch 600 that preferably adapts to a variety of telephone network interfaces, however, more preferably supports connection to a digital IT line. In the typical ordinary old telephone service, the analog telephone cables extend from an ordinary old telephone service of the fixed user to a telephone company central station which converts the telephone signals into digital signals by sampling. Typically, in-band signaling is used to transmit call control information. Analog signals are typically sampled at 8,000 samples per second using 8 bits per sample. The resulting digital signals are commonly combined over a four-wire line commonly called an IT line. Each T line multiplexes 24 voice channels using well-known multiplexing techniques, in accordance with the standards established by the International Standards Organization (ISO). See, in general, Stallings, Data and computer Communications, (chapter 6). The Modification of the telephone switch 600 to support other protocols, including the Committee Consultateur International de Téléphonie et de Télégraphie (CCITT) The lines, or other protocols of digital or analog transmission, would be obvious to a technician with ordinary experience in the field. Technicians with experience in the field also know methods for establishing telephone calls from the telephone switch 600 through the telephone network interface. In order to reduce package overhead, and because the errors detected by the TCP protocol can introduce excessive delays not convenient for the voice conversation, the system will preferably use an off-line transport layer protocol for the transmission of information of voice over the network of switched telephones in circuits. These protocols in connection do not provide recovery error and do not guarantee delivery of sequenced data. The most preferred system will use the User Datagram Protocol (UDP), which is well known to technicians in the field. See, for example, Martin J., TCP / IP Networking (Chapter 8). However, certain control information, such as call connection requests and database information, will preferably use the TCP protocol (Figure 4B). Referring now to Figure 7, the contents of the packets transmitted on the network of packet switched computers will now be described. Each package will have a command, followed by a connection identification (Connld), followed by the data for the type of command. The connection identification is used to determine the highest level connection, and optionally to demultiplex many connections from a single central computer. Packet data can be encrypted for security reasons and to protect the user's privacy. The different types of commands supported by the system include: Registration request Command - Connection ID User name Password Credit card information Authorization / Routing request - Command Connection identification Destination telephone number User name Password - Connection request Telephone Command Connection ID Destination telephone number Server key - Compression schemes Voice data package Command Connection identification Voice data - Telephone disconnection request Command Connection identification Log response packet Command - Connection identification Result data Authorization Routing Response Package Command Connection Identification - Status Server Password Telephone Connection Response Package Command Connection Identification - Result Data Error Package Command Connection Identification Reason Referring now to Figure 2B, a system for connecting two apparatuses of the ordinary old telephone service will be described, wherein at least a portion of the path of the call connection passes through a network of packet switched computers. A first user picks up a first ordinary old telephone service device 401 and has access to a first telephone switch 650 via a first network of switched telephones on circuits 300. The user then enters touch tone data, including billing information and the called station number. The tone detectors in the first telephone switch 650 capture this information. The first telephone switch 650 then generates a call connection request which is sent by the network of packet switched computers 200 to the router authentication server 500. The router authentication server 500 selects a destination telephone switch 600 and returns the address of the target telephone switch network 600. The first telephone switch 650 then has access to the destination telephone switch 500 and the calls are processed as described above for ordinary old telephone service calls. 7. Database machine. Referring to Figures 5 and 6, the database 580 will be described. The database 570 stores the routing, registration, authentication and billing data and can be distributed or centralized as is known to those skilled in the art. Several vendors provide tools to build these databases, including Sybase and Oracle. The database 570 includes data related to the user of billing information and server routing information. The database 570 will include a register 582 for each telephone switch 600 that includes the internet protocol address of the telephone switch and the port number, as well as its physical location. The telephone switch registers 582 will be mapped to a set of area code registers 583, such that the system can easily determine all area codes served by the telephone switch 600. The area code register 583 will also be mapped back to the telephone switch register 582 to facilitate determining which telephone switch to route a given call to. Each user will be represented by a user record 581 which will contain the user's name, address and telephone number. Each user record 581 will be mapped to several other fields or records, which include: the user's credit card registration 584; an authentication information record 585, which includes the user's password; and a set of telephone call registers 586 for each call the user has made in a certain time frame. Each call record will include the start time of the call, the termination time and the billing rate. It is understood that those skilled in the art will be apparent, and will readily be able to make other modifications, without departing from the scope and spirit of the present invention. For example, it will be apparent to those skilled in the art to substitute digital or other telephone sets, or other user telephone systems, such as the PBX (private branch exchange), instead of the ordinary old telephone service apparatuses described. In accordance with the foregoing, it is not intended that the scope of the claims be limited to the description or illustrations presented herein, without the claims being construed as encompassing all patentable novelty features that reside in the present invention, including all the characteristics that would be treated as equivalent by technicians with experience in the field.

Claims (19)

1. CLAIMS 1. A system for routing and transmitting voice conversations, said system comprising: a network of switched telephones in circuits that supports at least one voice protocol for routing and transmitting voice conversations; a plurality of telephone sets coupled to the network of switched telephones in circuits, each of the plurality of telephone sets has a unique telephone number for access through the network of switched telephones in circuits; a network of packet switched computers that supports a digital data packet protocol; a computer equipped with audio coupled to the network of packet switched computers, said computer equipped with audio to convert analog voice signals into digital data packet protocol and to convert the digital data received from the network of packet switched computers into signals analog, said computer equipped with audio that generates and sends from the user command, via the network of packet switched computers, a packaged call connection request comprising a telephone number called, - and at least one telephone switch having a network address in the network switched in packet and coupled with said network of switched telephones in circuits, the telephone switch to establish a connection to a telephone device identified through its unique telephone number through the network of switched telephones in circuits and to convert the voice information and the information of c control between said digital data packet protocol and the, at least one, voice protocol, by means of which the audio equipped computer establishes a voice connection by sending a call request containing a unique telephone number to the telephone switch establishing a voice connection with the called telephone device and converts the protocols between the network of switched telephones into circuits and the packet switched network of computers.
2. 2. The system for routing and transmitting voice conversations of claim 1, wherein the computer equipped with audio further comprises: a database for mapping telephone area codes and exchanges with the, at least one, telephone switch; and a selection element for selecting one of said, at least one, telephone switch based on the database mapping.
3. 3. The system for routing and transmitting voice conversations of claim 1, wherein the packaged call connection request further comprises user payment information; further comprising said system for routing and transmitting voice conversations an authentication element for verifying the user's payment information.
4. 4. The system for routing and transmitting voice conversations of claim 3, wherein the user's payment information comprises a user password.
5. 5. The system for routing and transmitting voice conversations of claim 3, wherein the user's payment information comprises credit card information.
6. 6. The system for routing and transmitting voice conversations of claim 1, wherein the packet switched computer network is the Internet.
7. 7. A method for establishing and transmitting a voice conversation between a computer equipped with audio to a network of packet-switched computers and a telephone device coupled with a network of switched telephones in circuits, using the method a telephone switch coupled to the network of switched telephones in circuits and the network of packet switched computers, the method comprising the steps of: (a) transmitting a call connection request packet containing a telephone number identifying the telephone equipment of the computer equipped with audio to the Telephone switch; (b) establishing a voice connection between the telephone switch and the telephone device through the network of switched telephones in circuits; (c) transmit, in a digital packet protocol format, voice input received by the computer equipped with audio during the voice conversation to the telephone switch via the network of packet switched computers; (d) transmitting, in a telephone voice protocol and control information format, voice input received by the telephone apparatus during the voice conversation to the telephone switch via the switched telephone network in circuits; (e) converting the voice input formatted into a digital packet in the telephone switch to a telephone voice protocol and control information; (f) transmitting the converted information from step (e) to the telephone apparatus via the switched telephone network in circuits; (g) converting the formatted speech voice telephone input and control information received in the telephone switch to a digital packet protocol; (h) transmitting the converted information from step (g) to a computer equipped with audio via the network of packet switched computers; and (i) reconstructing the digital packet information received by the computer equipped with audio in an analog signal, whereby the telephone switch is used to bridge the voice conversation between the network protocol of switched telephones in circuits and the protocol of computer network switched in package.
8. The method for establishing and transmitting a voice conversation of claim 7 wherein steps (c) and (g) further comprise the step of compressing speech input prior to transmission through the network of switched computers in package; and steps (e) and (i) further comprise the step of decompressing the compressed speech input.
9. The method for establishing and transmitting voice conversations of claim 7 further comprising the steps of: selecting the telephone switch from a plurality of telephone switches coupled to the packet switched network, based on the selection in telephone numbers that match the database for telephone switches.
10. The method of establishing and transmitting a voice conversation of claim 7 further comprising the steps of: transmitting user payment information within the call connection request; and verifying user payment information before establishing the voice connection of step (b).
11. 11. A system for routing and transmitting voice conversations, the system comprising: a network of switched telephones in circuits that supports at least one voice protocol for routing and transmitting voice conversations; a telephone set coupled to the network of switched telephones in circuits; a network of packet switched computers that supports a digital data packet protocol; a computer equipped with audio coupled to the network of packet switched computers, said computer equipped with audio to convert analog voice signals into digital data packet protocol and to convert digital data received from the network of packet switched computers into signals analog, said audio-equipped computer generating a call connection request packaged from the user command; at least one telephone switch having a network address in the packet switched network and coupled with said network of switched telephones in circuits, the telephone switch to establish a connection to a telephone device identified through its unique telephone number through the network of switched telephones in circuits and for converting the voice information and control information between said digital data packet protocol and the, at least one, voice protocol; and a routing server coupled with the network of packet switched computers, to select, the routing server, a telephone switch selected from, at least one, telephone switch upon receiving the call connection request packaged from the computer equipped with audio, the routing server returning the network address of the selected telephone switch to the computer equipped with audio, whereby the audio-equipped computer establishes a voice connection by sending a call request containing a single telephone number to the telephone switch that establishes a voice connection with the called telephone device and converts the protocols between the network of switched telephones into circuits and the network of packet switched computers.
12. The system for routing and transmitting voice conversations of claim 11, wherein the packaged call connection request further comprises a user password; said system for routing and transmitting voice conversations also comprises an authentication element for verifying the user password with a system database.
13. 13. The system for routing and transmitting voice conversations of claim 11, wherein the digital data packet protocol includes a transport layer protocol without connection, said transmission of the digitized voice signals over the network of packet switched computers using the transport layer protocol without connection.
14. The system for routing and transmitting voice conversations of claim 13, wherein the offline transport layer protocol is the User Datagram Protocol.
15. 15. A method for establishing and transmitting a voice conversation between a computer equipped with audio coupled to a network of packet-switched computers and a telephone device coupled to a network of switched telephones in circuits, using the method a router server coupled to the network of packet switched computers, and a plurality of telephone switches coupled to the circuit switched telephone network and to the network of packet switched computers, the method comprising the steps of: (a) transmitting a call connection request packet which contains a telephone number that identifies the telephone equipment of the computer equipped with audio to the router server; (b) selecting a telephone switch from the plurality of telephone switches after receipt of the call connection request packet from the computer equipped with audio; (c) transmitting an authorized call connection request packet containing the network address of the selected telephone switch from the router to the computer equipped with audio; (d) transmitting the authorized call connection request packet to the selected telephone switch of the computer equipped with audio; (e) establishing a voice connection between the selected telephone switch and the telephone apparatus through the switched telephone network in circuits; (f) transmitting, in a digital packet protocol format, the speech input received by the audio-equipped computer during the voice conversation to the selected telephone switch via the network of packet switched computers; (g) transmitting, in a telephone voice protocol and control information format, the voice input received by the telephone apparatus during the voice conversation to the telephone switch via the switched telephone network in circuits; (h) converting the voice input formatted into a digital packet in the telephone switch to a telephone voice protocol and control information; (i) transmitting the converted information from step (h) to the telephone apparatus via the switched telephone network in circuits; (j) converting the formatted speech voice telephone input and control information received in the telephone switch to a digital packet protocol; and (k) transmitting the converted information from step (j) to a computer equipped with audio via the network of packet switched computers; whereby the telephone switch is used to bridge the voice conversation between the network protocol of switched telephones in circuits and the network protocol of packet switched computers.
16. 16. A system for routing and transmitting a voice conversation between a first telephone set and a second telephone set over a network of packet switched computers that supports a digital data packet protocol including voice and call set-up information, comprising the system: a first network of switched telephones in circuits coupled with the first telephone apparatus, supporting the first network of switched telephones in circuits at least one voice protocol including voice and call disposition information; a second circuit switched telephone network coupled with the second telephone set, the second circuit switched telephone network supporting at least one voice protocol including voice and call set-up information; a first telephone switch coupled to the first network of switched telephones in circuits and a second telephone switch coupled to the network of switched telephones in circuits, each of the first and second telephone switches coupled with the network of packet switched computers and having each one a single network address in the network of packet switched computers, each of the first and second telephone switches to convert between voice and ready call information from the first and second switched telephone networks in circuits, respectively, and the digital data packet protocol, the first telephone switch also to generate and transmit a call connection request over the network of packet switched computers after receiving a touch tone request from the first telephone set, the second telephone switch in addition to establish a call disposition so The network of switched telephones in circuits to the second telephone apparatus after receiving the call connection request from the first telephone switch, by means of which a first user has access to the first telephone switch to generate a call request over the telephone network. computers switched in package with the second telephone switch, the second telephone switch then establishes a call to the second telephone set, the first and second telephone switches then convert and transmit voice information received between the telephone sets and the network of packet switched computers.
17. 17. A system for routing and transmitting a voice conversation between a first telephone set and a second telephone set over a network of packet switched computers that supports a digital data packet protocol including voice and call set-up information, comprising the system: a plurality of switched telephone networks in circuits each supporting at least one voice protocol that includes voice information and call disposition; a plurality of telephone devices coupled to the plurality of telephone networks switched in circuits; a plurality of telephone switches each coupled to the packet switched network and at least one of the switched telephone networks in circuits, each of the plurality of telephone switches having a unique network address in the network of packet switched computers , each of the plurality of telephone switches for converting voice and call disposition information between the, at least one, voice protocol and the digital data packet protocol, at least one originating telephone switch of the plurality of telephone switches capable of generating a call connection request including a telephone number called after receiving a touch tone request from one of the plurality of telephone sets; and a router server coupled to the network of packet switched computers, said router server to select a telephone switch selected from the plurality of telephone switches after receiving the call connection request from the originating telephone switch, returning the server from routing a network address of the selected telephone switch to the originating telephone switch, whereby a user has access to a first telephone switch through a first telephone set coupled to the first network of circuit switched telephones and enters a telephone number of destination using touch tone keys, the first telephone switch then transmits a call connection request containing the destination telephone number to the routing server which selects a second telephone switch based on routing considerations, the second telephone switch connects n second destination telephone apparatus a second network of switched telephones in circuits, the first and second telephone switches then communicate directly through the network of packet switched computers by coupling the first and second telephone sets.
18. 18. A method for routing and transmitting a voice conversation between a first telephone set and a second telephone set over a network of packet switched computers, using the method a router server coupled with the network of packet switched computers and a plurality of telephone switches coupled with the network of packet switched computers, the method comprising the steps of: (a) accessing a first telephone switch from the first telephone set; (b) generating dialing information corresponding to a telephone number of the second telephone set starting from the first telephone set; (c) detecting the first dialing switch the dialing information; (d) transmitting a call connection request packet containing the telephone number of the first telephone switch to the routing server; (e) selecting the routing server a telephone switch between the plurality of telephone switches after receiving the call connection request packet from the first telephone switch; (f) transmitting an authorized call connection request packet containing the address of the selected telephone switch network from the routing server to the first telephone switch, - (g) transmitting the authorized call connection request packet to telephone switch selected from the first telephone switch; (h) establishing a voice connection between the selected telephone switch and the second telephone set through a network of circuit switched telephones by coupling the selected telephone switch and the second telephone set; (i) convert the telephone voice and the formatted control voice and control information received in the first telephone switch and the selected telephone switch to a digital packet protocol and send the voice in converted digital packet and the control information between the first and the selected one of the telephone switches on the network of computers switched in package; and (j) transmitting the converted information of step (i) between the first telephone switch and the selected telephone switch via the network of packet switched computers, whereby the first telephone switch and the selected telephone switch are used to bridge the conversation of voice between the first telephone set and the second telephone set through the network of packet switched computers.
19. 19. The method for routing and transmitting voice conversations of claim 18 wherein the dialing information comprises touch tones.