JP2001503572A - Internet telecommunications system - Google Patents

Abstract

SUMMARY A method and apparatus for allowing a user (10) of a standard telephone system (12) to access a packet-based network such as the Internet (36). The first callover unit (32) converts the signal (14) from the standard telephone system (12) into a packet (34) formatted for Internet transmission. Embodiments of the present invention make voice telephone calls using the Internet (30) instead of the long distance carrier (22). The second call-over unit (40) converts the packet (38) again into a voice signal (24) that can be transmitted over a standard telephone network (20) to a standard voice telephone (28).

Description

DETAILED DESCRIPTION OF THE INVENTION                     Internet telecommunications system                                Background of the Invention   The present invention relates to voice and data telecommunications, and more particularly to standard telephones and Internet connection with an interface to reduce costs The present invention relates to a telecommunications system that provides a long-distance service via a network.   In order to provide a reliable communication means between two points, a conventional communication system as shown in FIG. Telephone systems have been developed for many years. In FIG. 1, two points are used for voice communication. Standard telephone, but a modem to enable computer communication Can be easily substituted.   Standard telephone systems include local exchanges that provide local telephone services (" LEC ") 12 requires a first telephone 10 to call. The first telephone 10 is a T carrier network using, for example, a digital transmission method based on T1. Designed to be used with telephone carrier networks such as networks Defined as a traditional telephone. Hereinafter, such a system will be referred to as a simple conventional telephone service. Service ("POTS").   In a long-distance call, the LEC 12 then turns to a long-distance carr. ier) 22 to connect to the regional switch 16 and transmit the voice signal 14 I do. The audio signal in this case is a numeric code, usually American numbering in. A dex ("ANI") code is directed to the recipient. Telephone exchange 16 Directs the audio signal 14 through the long-distance carrier trunk 18 to the receiving area. To the second telephone exchange 20. Next, the audio signal 24 is transmitted to the second telephone 28. Is transmitted to the LEC 26 in this area which is connected to Questions in such a system The problem is the cost of use. Each LEC 12 and 26, and long-distance carrier Billing for each call, usually per unit time, Long-distance calls are very expensive because they must support the cost of the work Can be.   One of the phones 10 and 28 has poor phone infrastructure In other countries, additional problems arise. In this case, each caller is limited Have to compete for the phone line, often because the phone line is unavailable There is also. The infrastructure needed to improve long-distance services is also It is costly and cannot improve the telephone network quickly.   In contrast, computer networks are growing rapidly around the world. The dissemination and retrieval of information via computers has become a pillar of commerce. Inter Is a network that integrates various computer networks into more cohesive integrated systems. Formed to stop. The Internet is available to governments, educational institutions, and businesses. A global computer that acts as a content server owned and maintained by Evolved into a system. These content servers operate and maintain independently The cost of using the Internet is usually a major There is only a small monthly fee required for a good transmission line or backbone. Obedience Internet restrictions can only be accessed by computer It is a point.   Transmission speed over the Internet can sometimes be a problem. Current Internet Technology could allow connection to about 20 times the number of computers that exist today, However, the backbone of the Internet has a line congestion commonly called a bottleneck. Problem. Bottleneck is throughput due to competition for IP path Restrict. This means that as more computers come online, transmission errors This means that transmissions are slowed down due to an increase in the number of retries. each Since only one address is assigned to a computer, redundant communications are Impossible. In addition, the maximum amount of data can be transmitted with minimal processing , The packet size usually corresponds to the maximum transmission rate of the network. Only And since the Internet is a network of multiple networks, Knowing the transmission rate is virtually impossible. Because the sender This is because the path to be finally taken is not known. The packet has only a small bandwidth If the packet is forwarded through a forwarding router, the Automatically split the packet into multiple packets. This process allows for an overall transmission speed And affect the transmission quality of real-time transmission.   In recent years, several commercial products that enable voice communication via the Internet have been registered. I've been playing. Each of these products is a personal computer ("PC") The sound card is used to convert voice and electronic signals. Next, the electronic signal is The Internet engine provided by the Internet subscription service Is packetized. The receiving terminal of the transmission also has a sound card and -You must have a PC subscribed to the Internet service. In this system Compatible with voice transmission quality, even with obvious hardware and subscription do not do. As mentioned above, we only access one Internet address , An error occurs in the voice transmission, which results in a retry. In addition, other professionals Tokor, usually point-to-point ("PPP") or continuous inline Protocol ("SLIP") is added for modem-to-modem transmission. Must be performed, thereby increasing throughput. In fact, this Delays and pauses can occur throughout speech, including in the middle of words and sentences. Such a delay and interval can be recognized on the receiving terminal side.   Therefore, POTS communication is integrated into the Internet, and the Internet is acceptable Used for long distance service and POTS interface with high quality You need to get it.   Thus, an object of the present invention is to reduce the cost of long-distance service.   Another object of the present invention is to provide an existing computer network infrastructure. To improve long-distance service overall.   Still another object of the present invention is to provide a content server arranged on the Internet. To enable POTS communication to   It is another object of the present invention to provide transmission speed, quality of transmission, and It is to raise reliability.   Another object of the present invention is to allow a user to purchase computer hardware or Internet for voice and data communications without the need to subscribe to the Internet To provide access to   These and other objects of the invention will become apparent hereinafter.                                Summary of the Invention   The above and other objects of the present invention are electrically connected to a telephone network. An information signal is bi-directionally communicated between a first communication device and a second communication device. This is accomplished by the present invention which provides a telecommunications system for operating. One In an embodiment of the invention, the telecommunications system connects the first communication device to a telephone network. Long distance call by connecting to the first interface means through the work Forming system. For example, a first communication device may communicate via a local telephone exchange. Phone that communicates with the interface means.   Called by the first interface means to use the long distance telephone system It is. Next, the first interface means appropriately determines the best quality and low cost method. And complete the telephone call. Available methods include tradition through network means Long distance service or various transmission methods. Conventional long-distance carry The operation of the hook is substantially constant and known. Therefore, a comparative decision is made This is done by regularly testing the network tools.   The network means comprises a plurality of computers or Is a computer network. Communication is on a specific computer Separate packets of digital information addressed to a It is performed using. An example of such a network is the Internet You.   Interface to achieve high quality communication for real-time transmission of voice and other The source means transmits redundantly over a plurality of IP streams. Such transmission Therefore, it is highly likely that the data will be received without delay in one trial.   Similarly, the second interface means connected to the interface means Receives individual packets and sends them to other phones through the second phone network Assemble this packet so that it can be First and second interfaces The means operates in two modes, a transmission mode and a reception mode, so that bidirectional communication is possible. Shin is accomplished.   In the transmission mode, the interface means transmits the information signal from the exchange means to the digital signal. And converts this individual packet to network means. I believe. In the receiving mode, the interface means receives from the network means The individual packet thus converted is converted into an information signal, and the information signal is transmitted to the exchange means. This is repeated in both directions until the call is completed.   In a second embodiment, the telecommunications system is directly associated with the network means. Used to interact. For example, if you have a modem, but usually Computers without Internet access can access the Internet. POTS phones access content servers over the Internet and Leave an email or interact with the menu system. Complete these tasks To complete, the first communication device establishes a telephone network, eg, POTS And contact the first interface means as before.   Next, the interface separates the information sent from the first communication device into individual The packet is converted into a packet, and the packet is transmitted via the network means. Second If the communication device is a content server on a network means, The server responds with its own individual packet. Next, Internet means Translates the individual packets of the content server into information for the first communication device Establish two-way communication.   In another aspect, the present invention provides a method according to the above apparatus. On the present invention The description and other aspects will be apparent from the following drawings and description.                             BRIEF DESCRIPTION OF THE FIGURES   The above and other objects, various features, and the invention itself of the present invention are as follows. It will be better understood by reading the description together with the accompanying drawings U.   In the drawing,   FIG. 1 is a block diagram of a prior art long distance communication system;   FIG. 2 is a block diagram of a long-distance communication system according to the present invention;   FIG. 3 is a block diagram of a call-over unit as part of the long-distance communication system of FIG. Lock diagram;   FIG. 4 is a flowchart of the callover unit of FIG. 3;   FIG. 5 shows a charging and management server user with the long distance communication system of FIG. Block diagram;   FIG. 6 shows a POTS telephone and a computer on the long distance communication system shown in FIG. Block diagram showing a computer communicating with the data;   FIG. 7 shows that the Internet usually does not have Internet access. FIG. 2 is a block diagram showing one computer communicating with another computer;   FIG. 8 is a graphic card for software necessary for the computer shown in FIG. User interface;   FIG. 9 shows a computer using the long-distance communication system of FIG. 2 on the Internet. FIG. 1 is a block diagram illustrating a standard telephone communicating with a computer.                                Detailed description   The present invention uses the long distance carrier 22 shown in FIG. The above goals are achieved by replacing the communication subsystem 30. FIG. Shows the telephone set 10 calling the LEC 12 through which the audio signal 14 passes. Phone 10 Generically means any of a variety of telecommunications devices that communicate over telephone lines It will be appreciated by those skilled in the art. For example, phones are typically used for voice communications Type 500 telephone, fax machine modem or other similar modem, E telephones, pay telephones, pagers, mobile telephones and the like.   The audio signal 14 is then replaced by the call-over unit instead of the long-range carrier 22. Transmitted by the client 32. The call-over unit 32 packetizes the audio signal 14 The packet is time-division multiplexed and transmitted through the Internet 36. Sent. When sent to the appropriate area through the Internet 36, the packet 38 is reconfigured in a second callover unit 40. Second call o The bar unit 40 reconstructs the packet into the audio signal 24 and converts the audio signal 24 to L The data is transmitted to the second telephone set 28 through the EC 20. As described above, the second telephone 2 8 may be any of a variety of telephone devices that operate normally over telephone lines; It cannot be restricted to ordinary voice telephones.   When the telephone 10 initiates a call, as is well known in the art, the LEC 12 Gives a dial tone to the telephone. The geography covered by LEC12 Ringing, busy signal, etc. LEC12. Uses conventional long-distance service as shown in FIG. These notifications are provided by the recipient LEC 20 and the Automatically returned through 2. However, in the present invention, this feedback is Specific configuration of the call, call progress, and other POTS signaling Must be treated. Therefore, the feedback from the destination LEC 20 to the second It is sent back to the callover unit, where it is packaged and And transmitted to the first callover unit 32 through the port 36. 1st call o The bar unit 32 then reconstructs the packet and passes the telephone 10 through the LEC 12. To form a signal that can be transmitted to From the user's perspective, these signals are traditional A sound that is substantially equivalent to the signal transmitted through the long distance service.   Although only one signal direction has been described, the system is bidirectional and the signal is It will be appreciated by those skilled in the art that traveling back along a path that is substantially the same as the path in one direction. Let's do it.   Further, by connecting the billing unit 42, billing can be performed on the Internet. Can be realized through The billing unit 42 is a call-over unit A server on the Internet that can be addressed by 32, 40, More accurate billing is assured.   In one implementation, a telephone call is placed from telephone 10 . The telephone dials the local telephone number within the LEC12 area and calls over Connect to unit 32. The callover unit 32 answers the call and Provides a menu for selection. Choices include internet service or Includes but is not limited to services by various long-haul carriers. less than It is assumed that Internet services will be selected unless otherwise specified. Set.   Upon request, the telephone number of the receiving party is keyed in via the keypad of the telephone 10 Is done. The callover unit 32 then searches the internal database and receives Determining a remote callover unit 20 that is geographically close to the destination telephone 28 . Preferably, this is in the region of the LEC 20. If you are not within the LEC20 area In that case, a long distance carrier is used to bridge the distance between them.   To understand how the present invention provides cost savings, and to To fully understand the body, first connect to an IP network such as the Internet. Must understand. As mentioned earlier, the Internet is essentially a sub A network between networks. Each subnetwork is connected to another subnetwork. To communicate with the network, a certain set of protocols must be used. The two most basic of these protocols are the Internet Protocol ( IP) and Transmission Control Protocol (TCP). These two protocols are Along with the supporting protocols of the TCP / IP Internet Protocol Suite (" TCP / IP "). By using TCP / IP, data A foundation for the Internet based on the transmission of packets is provided. Data packet Is also generally important for cost savings in data communications. Data path Each wire is dedicated to one set of communicating computers by communication via the bracket. It does not need to be used for Can be. What each packet actually contains depends on the network, but each packet Is usually the final decision on which computer will receive the packet Contains addressing information. All computers on the Internet are Each packet goes through the Internet because it has a unique address Sent properly. Furthermore, TCP / IP is an ordered protocol Packets are received on a first-in first-out ("FIFO") basis, Transmitters need to number each packet in the order they were sent for proper reassembly There is no.   FIG. 3 will be described based on the above basic background. Figure 3 shows a functionally disassembled 2 shows a rollover unit 32. When discussing functionality, hardware Specific examples are also described. Specific examples of these hardware are provided in the preferred implementation. Should be regarded as illustrative of the form and restrict in any way Not something.   The audio signal 14 is sent to the line interface 50 of the callover unit 32. To reach. The line interface is preferably a T1 card from Digilogic, Inc. Multi-port digital card such as a card, but with analog-to-digital conversion Any of analog cards, ISDN cards, or various standards in telephony It may be another card for adapting to the game.   Regardless of the input format, the output of the line interface 50 is To the switch 52. The matrix switch converts the audio signal 14 into the internal Send to IVR54. The matrix switch 52 in the preferred embodiment A split multiplex bus, which is incorporated in an IVR card described later.   Additional calls such as voice and data bridging for multiple calls One of ordinary skill in the art will appreciate that features can also be incorporated into the callover unit. Let's go.   The IVR 54 may use the interactive voice response described above to provide menu options to the user. ("IVR") module. In the preferred embodiment, the touchtone telephone system Dual tone multi-frequency ("DTFM") signaling is used for the stem You. Analog systems may also be used for pulse dial systems. Depending on the above choice of long distance communication method, IVR 54 may be a credit card or Provide options, such as billing for a third institution. If you choose to bill, Proceed to Kim Manager 66. This will be described later.   Assuming that Internet service is selected, the audio signal 14 is digitized. The digitizer digitizes and compresses the audio signal 14. voice If the signal is already digital, digitizer 56 may use a known lossless signal in the art. Only compression of the audio signal 14 is performed using the compression technique. Specific examples of hardware In the following, the digitizer 56, the IVR module 54, And the matrix switch 52 are all D-24CC manufactured by Digilogic, Inc. -On a single IVR card, such as a T1 card.   Next, packetization is performed by the packetizer 58. Packetization is obtained Is performed to conform to the Internet Protocol (IP). Pa The packet also reduces size to ensure convenient transmission. Pake as described above Instead of matching the packet size to the maximum transmission rate of a known part of the system, The packetizer matches the packet size with the maximum transmission rate of the transmission path. next Reduced to virtually ensure transmission without parsing across virtually all routers The packet size is used. The reduced packet size is determined during the quality test. This will be described later.   Each packet contains a destination IP address, a sender IP address, and other management information. Includes a header containing the information. The sender IP address simply sends the packet This is the IP address of the callover unit 32 that is operating. Destination IP address Is determined as follows.   Use the recipient phone number keyed in during the menu provision step to Database is retrieved and the remote call Is determined. First, the area code of the phone number Used to determine the callover units that cover the code. This Exchange of information on which of these callover units A determination is made as to whether the selection is appropriate for trusted telephone 28. Of the receiving telephone 28 If there are many callover units in the area of LEC26 for First callover unit 32 finds out which one takes less time By doing a test to determine which one to use. In addition, connection Time is considered an important quality measure of quality, so call-over The knit 32 tests and maintains a given IP route by expectation, Speeds the availability of public recipients.   In a preferred embodiment, the internal database is relational. data The base identifies each callover unit on the Internet with an area code and exchange. Associate with exchange. In addition, each callover unit corresponds to itself It has a list of assigned IP addresses. In a preferred embodiment, There are 255 IP addresses per router box, these are the addresses Are stored in the table as an array. Different callers have the same IP address Use of the same IP stream, voice communications and data communications are garbled. The database stores which IP streams are already in use. And keep track of which IP stream is the next available IP stream . An IP stream has a specific IP address that carries asynchronous data repeatedly Is defined as a transmission path.   Next, the packet is sent to the Internet engine 60. Internet The engine 60 typically has a transmission control program along with other administrative tasks. This is a software package that handles protocol (TCP). In this embodiment The Internet engine 60 sends packets and returns packets. Dedicated (p) to perform tasks such as monitoring whether the roprietary) UNIX based package. If such reception is at a predetermined time If not received within an interval (eg, 3 seconds), the packet is retransmitted. Inter -Another task performed in the net engine 60 is, among other things, It is to disregard the repeated packet.   Next, the Internet engine 60 sends the packet to the IP channelization 62. You. IP channelization 62 indicates how many virtual IP streams are used for transmission. Or decide. Using the method described below, the packet group is transmitted to the Internet 36. A plurality of I Sent on P stream. In a preferred embodiment, the Internet is also The interface 64 is a TI card.   By using a plurality of IP streams, the call-over unit 32 Maximize reliability while minimizing average latency in data transmission It becomes possible. Latency is the delay between the origin of the transmission and the reception of the transmission . For voice telephone calls, the waiting time is the delay between when a person talks and when you hear it. It is. The callover unit 32 decomposes the voice transmission into parts and Is transmitted on multiple IP streams almost simultaneously, reducing latency. Let Reliability is that the same part is sent multiple times, and that part is Increased by increasing the likelihood of being received at callover unit 40 . In the previous example, the lost part of the transmission would be skipped as one word or part of a sentence It will sound like it has been skipped.   The table below shows that all bytes of voice and data are always accountable. That lossless compression is used in order to use multiple IP streams Explains how tied to packet size and therefore can reduce latency are doing.   FIG. 4 illustrates the decision process performed in the callover unit 32 and how Explains how the waiting time is reduced. The process starts at 80 That is, when the user calls the callover unit 32, the user's preference (use r preferences) are determined at (82). The most basic of these preferences Is to use the long distance carrier 84 or the Internet 86 . When the long-distance carrier is selected (84), the call is made to the selected long-distance vendor 11 0 is connected.   When the Internet is selected, the next question is: Is transmission sensitive to latency? (88 ). That is, must transmission occur in real time? Is . The answer "NO (90)" indicates that the transmission is not latency sensitive. That's it Examples of such transmissions are facsimile, computer-to-computer, etc. It is pure data transmission.   Latency-sensitive transmissions are, for example, voice. Let two people communicate by phone Then, it is unacceptable that there is a delay in transmission. Therefore, the transmission is delayed If it is determined to be sensitive (92), the waiting time is minimized.   First, the callover unit 32 has enough IP streams available (94) is determined. The actual number of IP streams depends on the implementation . In the present embodiment, the minimum number is two; The call does not continue (96) and the call is transferred to conventional long-distance carrier service 110 .   If there are enough IP streams (98), determine the average waiting time for transmission. Perform a test to determine Average waiting time, ie average transmission delay T_delayIs It is defined algebraically as follows: T_delay= T_c+ T_a+ T_s+ T_d+ T_p In the above formula,   T_c   --- Time for collecting audio data   T_a   --- Time to assemble into packets   T_s   --- Time to send data on the Internet   T_d   --- Time to decompose   T_p   --- Time for digital-to-analog conversion   As can be seen from the above equation, most of the latency variables are Under the control of. T_c, T_a, T_d, And T_pAre each in a preferred embodiment Is hardware dependent, so by using faster hardware Latency is reduced. But T_sDepends on the Internet. the Internet Has a backbone that is the trunk of communication. Backbone is traffic The waiting time T_sIncreases. Therefore, the next step is Method minimizes latency in the current state of the Internet backbone Is to decide what to do.   Generally, the waiting time T_sMust be minimized, but at the expense of data reliability Don't be. The goal of reliability is up to 500 ms per transmit direction This is to provide almost error-free transmission with voice delay. Multiple calls to a single call Rely on time-consuming packet retries by using IP streams Transmission capacity can be averaged to a fixed high level.   In a preferred embodiment, separate IP streams are defined for sending and receiving. The waiting time T_sIs further minimized. As a result, IP streams Systems are not isochronous, so unidirectional communication along an IP stream is unidirectional. Communication improves performance because it can be better measured and maintained. Sa Furthermore, by using a small packet size, the Reconstruct packets into a single coherent voice signal faster .   Waiting time T_cData using a controlled buffer size to minimize And then use the least latent transmission method as much as possible. To send the buffer. This buffer size is equal to the maximum It is adjusted according to the transmission rate.   Various considerations must be considered to determine the best transmission method. Empirically, the size of the buffer is the audio channel rate, ie Must be changed in proportion to the packet size as described above._c Is the total transmission delay T_delayShould not be larger than the value that exceeds 20% of No. Buffer period for some bytes of data is 20% of transmission delay The larger the transmission delay T_sIs lower, increase the number of IP streams. The size of the buffer period is determined so that it is close to the packet size.   Further, the number of IP streams for one call is generally the sum of the IP streams. Count (minus the IP stream reserved for system use) ) Is the number of voice call requests and the number of calls in progress or the average over that time. It is determined by dividing by the sum of the larger deviation.   Latency control is thus dynamic and adaptive to the current state of the Internet It is done on a regular basis.   Internet latency used in the preferred embodiment An example method for determining is shown. This example shows four paths AD and five I Use P stream. Those skilled in the art will recognize that more or less paths and IP streams can be used, and these quantities are as adaptive as above. Will understand.   At 8000 bytes / second, every 100 millisecond period, And test the stream.   At 8000 bytes / second, every 200 millisecond period, And test the stream.   At 4000 bytes / sec, every 100 millisecond period, And test the stream.   At 8000 bytes / second, every 100 millisecond period, And test the stream. This test uses the same data on multiple IP streams. Data part is transmitted at the same time. This is very redundant, so at most n -2 (n is the number of IP streams) the result of the retry for any IP stream , The IP stream is terminated as unusable.   This four-pass method is repeated periodically, thereby producing a minimum latency Is updated. Such tests are based on communications (real-time (Including communication), without significant disruption to the quality of the communication. In the preferred embodiment, this test is repeated every 3 seconds.   Once the best average latency is determined (100), the best mode of transmission is determined (102). All four paths in the above example provide acceptable latency If not, or if the data error rate is too high, the best mode is determined Is determined (104), and this transmission is shifted to the long distance carrier (11). 0).   Once the best mode of transmission has been determined (106), the call is sent to the geographic location of the receiving telephone 28. The route is routed via the local callover unit (108). Or Best mode if only some part of the transmission path has unacceptably high latency. Mode may be via a call-over unit outside its geographic region Use the distance service to bypass the slow part of the transmission path. This is true The details will be described later in the detailed description.   As the call continues, the Internet is repeatedly tested, thereby waiting. Time and data errors are minimized. Additional IP streams used When possible, call-over unit 32 is used to send telephone calls. The number of IP streams to be transmitted can be increased. Similarly, the callover unit 32 The higher the demand for it, that is, the more telephone calls it makes, the more IP Streams are removed from the current telephone call and increase with the increase in telephone calls Can adapt to the load.   When the telephone call is completed (112), the resources are released and the next caller can These resources can be used.   FIG. 5 shows billing for tracking telephone calls made on the Internet. And a management system. In this illustration, the LEC is a general It has been replaced by the Public Switched Telephone Network ("PSTN") 120. PSTN1 20 may be an LEC, or may include a long distance carrier, or a foreign It may be equivalent. In any of these cases, PSTN 120 It is associated with a three party billing system 122.   Third party billing system 122 is used by the telephone company to charge telephone calls. It is one of many billing systems used. Due to the billing capability of the present invention, call Over unit 32 may selectively interact with third party billing system 122. Obedience Internet processes such as the long-distance service described above are It can be charged directly to the stem. For example, if the user When calling 32, one of the menu options is to charge a third party. The third party billing system 1 can be charged via the PSTN 120. Sent back to 22.   Once a call is made by the callover unit 32 via the Internet 36 Once established to the recipient, charges begin to accumulate. Charges are temporarily internal data It is stored in the callover unit 32 of the base 124. When the call ends, When the charge accumulation is completed, the final amount temporarily stored in the internal database 124 Is sent over the Internet 36 to a billing and management server 42, where Stored in a larger database. Larger databases are available on the Internet. Create call detail records ("CDR") invoices for calls made on the communication system Finally used to.   The billing and management server 42 also has a database in the callover unit. Is provided. For example, every time a new callover unit enters service, Or as the exchanges associated with each callover unit change , The accounting and management server 42 stores the database included in each call-over unit. Update source. This is a new call over unit on the internet , Area code, and the exchange served by that callover unit. Download updated database containing IP address information Is achieved. The database is also encoded before transmission for security .   Another function performed by the billing and management server 42 is that each callover Update the waiting time information in the knit. Callover unit Each database in the database is related to a transmission route, that is, an Internet segment. Including waiting time information. This latency information, in a preferred embodiment, is Stored in individual callover units that represent calls made during the last 24 hours You. Periodically, the billing and management server 42 waits from each callover unit. Upload time values, thereby updating its own database . Each time a segment of the transmission path deviates from the stored latency value by a predetermined amount In addition, the billing and management server 42 assigns a new latency value to each callover. Download to unit.   Knowing the latency value allows the callover unit to address known problems. It is possible to avoid. For example, the internet between London and Paris the previous day Suppose the transmission on the packet was very slow. Boston Caller Calls Paris Attempt to call someone in the country. The Boston area callover unit , Knowing this wait problem, so instead of the Paris callover unit To send calls to the London Callover Unit over the Internet Wear. London's callover unit, in turn, Access to conventional long-haul carriers, thus avoiding the problematic delay .   In addition to the billing function, the billing and management server 42 provides a symmetric network management It has a protocol ("SNMP") management console 126. SNMP console 1 26 monitors system failures and monitors system resources.   Backup accounting and management server 128 and backup SNMP console Rule 130 is also shown. The first billing and management server 42 for some reason If a failure occurs, the backup continues to operate. Therefore, billing and operating information Collect duplicate information.   FIG. 6 illustrates another embodiment of the present invention, in which the computer 140 accesses the LEC 12 and via the Internet provider 142 Connect to the Internet. As described earlier with reference to this figure, the interface Nets are well known in the art. Computer users often Internet provider via the LEC12 from the modem in the computer 140 Dial 142 to connect to the Internet. Internet The provider 142 has a modem that communicates with a modem in the computer 140. To provide the computer 140 with a gateway to the Internet 36. You. This allows computer 140 to interface using a single IP stream. The content server 144 on the net 36 can be accessed.   In the present invention, the callover unit 40 It can be connected by the computer 140. This allows the computer to Via the callover unit 40 via the callover unit 40 A certain LEC 26 is called through the telephone line, and then the telephone 28 is called directly. This is useful in many situations. For example, computer 140 is located Geographic area has a local call over unit for any reason If not, the computer itself can act as a telephone. In addition, computer 1 40 calls telephone 2 to access information on one or more content servers 144. 8 with the user. This, of course, is when the computer 140 receives a voice response. This is the case. Another embodiment is for marketing or billing, A telephone call made by a computer.   In the latter embodiment, the content server 144 is connected via the Internet 36. Then, a telephone call to the callover unit 40 is started. Commercial software Means that the content server 144 dials a telephone number using a modem. It allows you to perform this task and transmit messages to telephone 28. I However, many modems require a dial-to , Ring indication and other status messages. This For this reason, the call-over unit 40 normally transmits the Play these status messages artificially.   In each of the other embodiments where computer 140 initiates the call, the computer The internet engine used by 140 is callover unit 4 Must be applied to interface with 0. The application is generally Software add-in that handles communication and billing by the failover unit 40 It is.   If the user of computer 140 does not have an Internet provider In this case, the call-over unit 32 connects to the Internet 36 as shown in FIG. It can also be used to establish a connection. Computer 140 is simply an ISDN Internet access devices such as communication devices, Ethernet, modem, etc. Make a call using the chair. Then, the Internet access device becomes the LEC It communicates with the callover unit 32 via 12. Callover unit 32 Is a transmission of a data type whose transmission is not latency sensitive. And provides a connection to the Internet 36. Computer 140 Later, as if computer 140 were using an Internet provider , Can access a content server 144 on the Internet 36, Can use a system that bills per unit time rather than by subscription become.   The software add-in in the preferred embodiment is a graphical add-in shown in FIG. User interface ("GUI"). Software is available on the Internet Access to the world's telephone lines connected to your computer and all computers Provide a viewer to make it possible. This software also provides Internet services Unserviced computers are interfaced via the internet facility. -Enables communication with computers receiving Internet services. View Core also provides a system for billing third parties with Internet access. Debit card billing system from callover unit, not limited to computer System or other third party billing system.   The GUI shown is a WINDOW header 160, as can be seen. Designed for use in OWS. WINDOWS is MicrosoftCorpora is a trademark owned by tion. The GUI provides a telephone-like entry of a phone number Provide a dialing dialog box 162 with a keypad 166 . When the number is keyed in, the number is displayed in calling window 164. Thus, the user may have various options presented by the radio buttons next to the keypad 166. Options. By clicking dial button 168, the number is Can be dialed. By dialing, via the Internet, as shown in FIG. Connection to a telephone as shown or connection to another computer as shown in FIG. Is made.   When the call is completed, hang up the call to free resources for the next caller Hang-up button 170 may be used.   Attach button is another software that links to the running data stream Used to pre-configure applications. As a result, the terminal Third party software applications such as emulators or program managers Action is performed via the viewer to establish peer-to-peer data communication. And become possible. This allows, for example, a computer operator to Remote control application linked via Internet Can be used to remotely operate software running on another computer. Can be.   The buttons shown below the keypad 166 are used once the call has been established. Hold button 176 temporarily suspends the call, and “conf” button 174 Enables conference call with hands. The L1 and L2 buttons are It is used for telephone line 1 and telephone line 2 when calling a number of parties. The desktop has sufficient bandwidth and / or sound card resources Is assumed.   Various statuses are presented to the user during the connection and conversation. status Window 180 presents a text message to the user and A window 190 indicates the number dialed. Call time window 192 Informs the user how long the call has been connected and gives the call type Window 194 presents the type of communication, such as fax, voice, data, etc. . The connection status 196 also indicates that the handset is off-hook, busy, idling Indicates the current connection characteristics, such as medium.   Use the audio level dialog box 182 to control the audio level. You can also control. This dialog box allows the user to Audio volume in peaker system and volume of audio in transmission Allows you to adjust the system.   If the email is sent over the Internet, the message composer button By pressing the button 186, the message composer dialog box is It is. This allows you to send messages written over the Internet. Access to the file editor.   Pressing the IVR script editor button 188 allows the user to A dialog box is retrieved that allows you to edit the script. IV R script allows users to compose IVR applications To The user simply receives any key on the keypad using DTMF Menu structure to respond to the message being sent or sent Key in. The sent message indicates that the computer has sound capabilities Can be recorded and saved to a file, or use a standard computer disk Can be typed and read using the option. Of many applications of this menu system One is multi-user, where each user listens only to messages intended for that user It is to have an answering machine.   In the preferred embodiment, the clock indicating the time of the day is also indicated by 184. Is done.   FIG. 9 shows a user who is initiating a call on a standard telephone 10 with an interactive voice message. Indicates that it can be used to provide a response ("IVR"). The call is PSTN The callover unit 32 is reached via 120. The call is Data over the Internet 36 to another callover unit 150 Reach.   What happens next depends on the implementation. In one embodiment The call-over unit then converts the packaged voice message into a traditional voice Then, the voice message is transmitted to the PSTN 152. Boy The message is directed from the PSTN 152 to the content server 154. However However, the content server 154 is not on the Internet 36, Stand-alone computer accessible by standard telephone service (POTS) Or PSTN.   In one embodiment, the callover unit is a The message is converted to a conventional voice message and the voice message is converted to the PSTN 15 Transmit to 2. Voice messages are sent from the PSTN 152 to the content server 15. Pointed at 4. However, the content server 154 is Not a standard telephone service (POTS) It is a door-loan type computer or PSTN.   In another embodiment, the content server Part. In the preferred embodiment, the callover unit 150 Content server 154 simply calls on callover unit 150 It can be a standalone application that runs.   To access the content server 152, the content server 152 Software application running on the Must be written as follows. Application is World Wide Web The software used to perform the translation for the IVR application Hypertext markup language (“HTML”), a software language Can be used. HTML is for hypermedia documents on the Internet. A computer language used to identify content. Or, Application may receive a fax via HTML reception or storage. No. The IVR system relies on pre-recorded messages to touchtone phones. Communication can be incorporated and used. Possible of such a system Applies to modem dial-in service for serial applications, voiceme And text-to-speech features.   The present invention may be in other specific forms without departing from its spirit or essential characteristics. It can also be embodied in a state. Accordingly, embodiments of the present invention are illustrative in every aspect. It is not intended to limit the present invention. The scope of the present invention is above It is shown not by the description but by the appended claims. Therefore, the claims All changes within the meaning and range of equivalents are included within the scope of the present invention.

Claims (1)

[Claims] 1. A first communication device connected to the first telephone network and a second communication device; A telecommunications system for performing bidirectional communication of information signals with a device, The first communication device transmits, as at least one of the information signals, the first telephone network. A two-way communication by providing the network with the identifier of the second communication device. Starting, the telecommunications system comprises:   A network for communicating individual packets of digital information to the second communication device. Network means;   In electrical communication with the first telephone network and the network means (in  electrical communication) first interface means; Pacing means for converting the information signal into individual packets of the digital information, A transmission mode for sending a packet to the network means; Converting the received individual packet into an information signal, and converting the information signal into the first telephone network; A receiving mode for sending to the first communication device via a network. Face means for receiving the identifier from the first telephone network; And adapted to transform the at least one of the information signals including When one communication device starts the two-way communication, the interface means: The two-way communication is performed using the identifier received from the first telephone network. A telecommunications system for directing to a second communication device. 2. The first communication device communicates via the first telephone network 2. The telecommunications device of claim 1, wherein the computer has a modem adapted to: Communications system. 3. The telecommunications system of claim 1, wherein the first communication device is a telephone. M 4. The second communication device is a communication device electrically connected to the network means. The telecommunications system of claim 1, wherein the telecommunications system is a computer. 5. The second communication device comprises the network means and a second telephone network. And second interface means for electrically communicating with the workpiece. Means for converting the information signal into individual packets of the digital information; A transmission mode for sending a packet to the network means; Converting the separated individual packet into an information signal, and converting the information signal into the second telephone network; A telecommunications system according to claim 1, comprising a receiving mode for sending over a network. Tem. 6. The second communication device communicates with the second communication device via the second telephone network. The telecommunications device of claim 5, further comprising a telephone in communication with said interface means. Shin system. 7. The second communication device communicates with the second communication device via the second telephone network. Further comprising a modem for communicating with the interface means of the computer A telecommunications system according to claim 5. 8. The second communication device communicates from the second telephone network with the network. Communicating a status signal to said first interface means via a communication means. The telecommunications system of claim 5, wherein the telecommunications system is adapted to: 9. The interface means may provide the least average before transmission of the individual packet. The telecommunications of claim 1, further comprising management means for determining a waiting time. system. 10. The management unit is configured to separate individual information from the information signal in a plurality of IP streams. The telecommunications system according to claim 9, wherein the packet is transmitted. 11. The management means duplicates each of the individual packets from the information signal many times The telecommunications system according to claim 9, wherein the transmission is performed. 12. The interface means compresses the information signal so that the information signal is compressed. 2. The remote distance of claim 1, further comprising compression means for reducing the size of the compressed signal. Communications system. 13. 2. The method according to claim 1, wherein the compression unit compresses information without reducing a data amount. Telecommunications system. 14． The interface means interactively determines a communication path for the information signal. 2. The telecommunications system of claim 1, further comprising voice response means for determining. 15. The interface means responds to the voice response means, and To send to a long distance carrier Item 15. A telecommunications system according to Item 14. 16. The interface means is electrically connected to the network means. 2. The method of claim 1, further comprising: billing means for generating time and billing information associated with the billing information. A telecommunications system as described. 17． The interface means, wherein communication is performed through the network means 17. The method according to claim 16, wherein the device is distributed away from the billing means. Telecommunications system. 18. The interface means temporarily stores the time and billing information, At the end of the communication between the first communication device and the second communication device. 18. The telecommunications system of claim 17, wherein inter- and billing information is sent to the billing means. . 19. The interface means provides an interface with a third party billing system. The telecommunications system of claim 1, wherein the telecommunications system is adapted to: 20. The third party billing system is electrically connected via the network means. 20. The telecommunications system of claim 19, wherein the telecommunications system is adapted to communicate. 21. The third party billing system communicates via the network means. 20. The telecommunications system of claim 19, wherein the telecommunications system is not adapted to: 22. The first communication device communicates with the second communication device via the network means. Computer with a user interface that allows interaction with communication devices The telecommunications system according to claim 1. 23. The second communication device is electrically connected to the network means 23. The telecommunications system of claim 22, which is a computer. 24. The second communication device is a telephone, and the telecommunications system comprises:   Second interface means for electrically communicating with the network means. And the interface means converts the information signal into a separate packet of the digital information. A transmission mode for converting the individual packets to the network means and transmitting the individual packets to the network means; Converting the individual packet received from the network means into a reassembled information signal And a receiving mode for conversion.   Electrically communicating with the second communication device, and communicating the reassembled information via a telephone line. Further comprising a second local exchange means for communicating the broadcast signal. 23. The telecommunications system according to claim 22. 25. The first interface means is provided by the second interface means. 25. The system of claim 24, further comprising a database having a list of telephone exchanges performed. A telecommunications system according to claim 1. 26. The database is used by the second interface means The telecommunications system of claim 25, further comprising a list of IP addresses. 27. The database stores the average waiting time over the segment of the network means. And further comprising waiting time information about the time, the second operating in the transmission mode. One interface means uses multiple transmission methods for one transmission. So avoid problematic segments determined by the average latency 26. The telecommunications system of claim 25, wherein the telecommunications system is adapted to: 28. The second communication device is a command line for electrically communicating with the network means. A content server, wherein the content server interacts with the first communication device. 2. The remote distance of claim 1 having a user interface adapted to: Communications system. 29. The first interface means is configured to dial in the content server. 29. The telecommunications system of claim 28, which simulates tone. 30. The receiving mode is via a first set of one or more IP streams. Communicate, said transmission mode comprising a second set of one or more IP streams. The telecommunications system according to claim 1, wherein the telecommunications is performed via a communication. 31. 31. The method of claim 30, wherein the first set and the second set are mutually exclusive. Telecommunications system. 32. The individual packet is transmitted on the maximum transmission route of the network means. 2. The telecommunications system of claim 1, wherein the telecommunications system is scaled to approximately match the communication rate. Stem. 33. Said individual packets are formatted according to the Internet Protocol The telecommunications system according to claim 1, wherein: 34. Said network means is accessible by subscription , Said first interface means accessing said network means without said subscription. The telecommunications system of claim 1, wherein the telecommunications system is capable of wireless communication. 35. A first communication device connected to a first telephone network; A second communication device connected to the second telephone network via the data network. A long-distance communication system for performing bidirectional communication with the first communication device. A chair is sent as part of the electronic signal through the first telephone network; Providing an identifier representing the second communication device to the first telephone network. Therefore, two-way communication starts, and the telecommunications system   An electronic signal from the first telephone network is converted into the electronic signal from the computer network. Information packets sent from the computer network To the first telephone network to be sent on the first telephone network Having a telephone connection and a network connection to the computer network One callover unit,   An electronic signal from the second telephone network is converted into the electronic signal from the computer network. Information packets sent from the computer network Sent over the second telephone network, thereby providing access to the computer network. Bidirectional communication between the first communication device and the second communication device via a network A telephone connection to the second telephone network and the computer A second callover unit having a network connection to the network; The first callover unit includes the first call over unit from the first telephone network. Receiving an identifier and selectively converting and transmitting a portion of the electronic signal including the identifier. Sent by the first communication device when the two-way communication is started Utilizing the identifier, directing the two-way communication to the second communication device, Shin system. 36. The first callover unit establishes a communication path for the information signal. The telecommunications system of claim 35, further comprising voice response means for interactively determining. Stem. 37. The first call-over unit responds to the voice response means, Transmitting to said second call-over unit via a computer network 37. The telecommunications of claim 36, wherein the telecommunications are performed or transmitted to a long distance carrier. Shin system. 38. Electrically connected to the computer network, wherein the first call Billing means to generate time and billing information related to the use of 36. The telecommunications system of claim 35, comprising: 39. The first callover unit connects the computer network 39. The method of claim 38, wherein said time and billing information is transmitted to said billing means via Telecommunications system. 40. Manages packet transmission in telecommunications systems that transmit individual packets of information A method comprising:   Determine if the packet transmission is latency-sensitive Steps to   Minimizing the latency if the packet transmission is sensitive to the latency. Including, methods. 41. The step of minimizing the waiting time comprises the step of: Determining the time further comprising determining the average waiting time In, the test packet is sent via a plurality of IP streams, and the transmission time is Measured for each of the streams, resulting in the fastest IP stream 41. The method of claim 40, wherein 42. The step of minimizing the waiting time comprises breaking the test packet into pieces. Further comprising splitting and distributing the strips into the plurality of IP streams. 42. The method of claim 41, wherein 43. The step of minimizing the latency is performed in a number of passes, 42. The method of claim 40, wherein each transmits the individual packet according to a predetermined method. the method of. 44. The predetermined method includes changing a size of the individual packet. 44. The method of claim 43, wherein 45. The predetermined method includes changing a compression rate of the individual packet. 44. The method of claim 43, comprising. 46. The predetermined method may include a step of transmitting the information transmitted in the individual packet. 44. The method of claim 43, further comprising the step of varying the amount of 47. Said step of minimizing said waiting time comprises the steps of: Transmit over the second set of IP streams and send other individual packets to the second set of IP streams. 41. The method of claim 40, further comprising receiving via a stream. 48. The step of minimizing the waiting time may be performed periodically during the transmission of the packet. 41. The method of claim 40, wherein the method is repeated. 49. A first communication device connected to a first telephone network; Long-distance communication for two-way communication with computers connected to A telecommunications system, wherein the telecommunications system comprises: Electronic signals are converted and sent over the computer network, Information packets from the telecommunications network are translated onto the first telephone network. Telephone connection to the first telephone network and the computer as sent A callover unit having a network connection to a network. Telecommunications system. 50. The callover unit uses a separate packet of information to 50. The telecommunications system of claim 49, wherein the telecommunications system communicates via a computer network. Tem. 51. 50. The computer of claim 49, wherein the computer network is the Internet. Telecommunications system.