CA2180685C - Internet telephony gateway - Google Patents
Internet telephony gateway Download PDFInfo
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- CA2180685C CA2180685C CA 2180685 CA2180685A CA2180685C CA 2180685 C CA2180685 C CA 2180685C CA 2180685 CA2180685 CA 2180685 CA 2180685 A CA2180685 A CA 2180685A CA 2180685 C CA2180685 C CA 2180685C
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- switched network
- circuit
- telephone
- call
- packet switched
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Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M7/00—Arrangements for interconnection between switching centres
- H04M7/12—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
- H04M7/1205—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
- H04M7/1225—Details of core network interconnection arrangements
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6443—Network Node Interface, e.g. Routing, Path finding
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6472—Internet
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6475—N-ISDN, Public Switched Telephone Network [PSTN]
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L12/00—Data switching networks
- H04L12/64—Hybrid switching systems
- H04L12/6418—Hybrid transport
- H04L2012/6486—Signalling Protocols
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04M—TELEPHONIC COMMUNICATION
- H04M3/00—Automatic or semi-automatic exchanges
- H04M3/42—Systems providing special services or facilities to subscribers
- H04M3/428—Arrangements for placing incoming calls on hold
Abstract
An apparatus for enabling the establishment of telephone calls between a packet switched network carrying data packets and a circuit switched network carrying telephone signals, comprises a first interface unit for connection to the circuit switched network; a second interface unit for connection to the packet switched network; a device for receiving signals at said first interface and converting them to data packets for transmission over the packet switched network and vice versa; and a processor for determining destination information from incoming signals or data packets on one network and setting up a call to a destination on the other network in response to the destination information.
Description
~18Q88~
INTERNET TELEPHONY GATEWAY
This invention relates to an Internet Telephony Gateway, and more particularly to a gateway for permitting telephone calls to be made from an ordinary telephone to a subscriber on the Internet.
The PSTN (Public Switch Telephone Network) has enabled subscribers to make telephone calls over switch circuits for many years. The circuits are established at the time of call setup and remain dedicated to the communicating subscribers for the duration of the call.
More recently, the Internet has become a popular means of communication. The Internet consists of a multitude of interconnected networks each conforming to the TCP/IP
protocol suites, so that users on the network can communicate. Unlike the PSTN, the Internet is a packet switched network, i.e. one in which data is carried in individually addressed packets from the near end to the far end.
The Internet is convenient for non-time critical data, such as file transfer and E-mail, although recently it has become popular for real time applications. It has long been possible to communicate in real time over the Internet using "chat servers". In such an arrangement, users communicate by entering data on their computer keyboards in real time.
More recently, software has become available that in conjunction with an audio card permits audio to be exchanged in real time over the Internet. This enable subscribers to carry on "telephone conversations" in real time over the Internet.
The disadvantage of such software, however, is that it only enables communication between Internet subscribers. A
receiving subscriber must have his computer connected to the Internet and running the appropriate software in the receive mode. An Internet subscriber cannot communicate with an ordinary POTS (Plain Ordinary Telephone Service) subscriber on the PSTN.
An object of the invention is to alleviate this disadvantage.
According to the present invention there is provided a telephone system permitting the establishment of telephone calls in either direction between a packet switched network carrying data packets and a circuit switched network carrying telephone signals, comprising a circuit switched telephone network with a plurality of first subscribers with unique telephone numbers and a packet switched network with a plurality of second subscribers with unique addresses and a gateway between said circuit switched and packet switched networks, said gateway comprising a first interface unit for connection to the circuit switched network and exchanging signals over circuits therein with said first subscribers, a second interface unit for connection to the packet switched network and exchanging packets thereover with said second subscribers, means for receiving telephone signals at said first interface and converting them to data packets for transmission over the packet switched network using Internet telephony, means for receiving data packets at said second interface and converting them _ to telephone signals for transmission over the circuit switched network, and processing means for determining destination information from an incoming telephone call received from a first subscriber on said circuit switched network and sending call setup messages over said packet switched network to a second subscriber identified from said destination information to permit direct two-way conversation between said first and second subscribers, and for determining destination information from data packets received on said packet switched network from a second subscriber and setting up a call to a first subscriber identified from said destination information carried in said data packets over a circuit on said circuit-switched network to permit direct two-way conversation between said first and second subscribers and wherein said processing means sets up a call over the packet switched network based on the incoming circuit on said circuit switched network at said gateway, the addresses on said packet switched network being mapped one-for-one to circuits on said circuit switched network.
The invention thus provides a gateway through which, for example, Internet users can call subscribers on the public switch telephone network and vice ver-sa. In the case of a telephone subscriber wishing to call someone on the Internet, the subscriber calls the number of the gateway, which then establishes a virtual connection over the Internet with the user. In one embodiment each incoming circuit at the gateway i~s mapped to a user's TCP/IP address.
Alternatively, the gateway can determine call party information from incoming signals on a trunk.
- 2a -In the reverse direction, the computer user sends a message to the gateway requesting the establishment of a call to a telephone subscriber on the PSTN. The gateway sets up the call and the two can then communicate.
According to a second aspect of the invention there is provided a method of establishing calls between a packet switched network carrying data packets and a circuit switched network carrying telephone signals, comprising the steps of providing a gateway between the Packet switched network and the circuit switched network receiving a telephone call over an incoming circuit in said circuit switched network identifying destination information from said incoming telephone call, sending a call setup message to a subscriber on said packet switched network at an address determined from said destination information to set up a two-way telephone call with said incoming circuit using Internet telephony over said packet switched network, said destination addresses being mapped one-for-one to incoming circuits, and said destination information being determined by the particular circuit on which an incoming call is received at said gateway, and converting in real time signals from said circuit switched network to data packets for transmission over said packet switched network and vice versa while the call is in progress.
The invention will now be described in more detail, by way of exam~le.only, with reference to the accompanying drawings, in which:-Figure 1 is a general block diagram of a system employing a gateway in accordance with the invention;
Figure 2 is a block diagram showing the establishment of a call from a computer user to a telephone subscriber;
INTERNET TELEPHONY GATEWAY
This invention relates to an Internet Telephony Gateway, and more particularly to a gateway for permitting telephone calls to be made from an ordinary telephone to a subscriber on the Internet.
The PSTN (Public Switch Telephone Network) has enabled subscribers to make telephone calls over switch circuits for many years. The circuits are established at the time of call setup and remain dedicated to the communicating subscribers for the duration of the call.
More recently, the Internet has become a popular means of communication. The Internet consists of a multitude of interconnected networks each conforming to the TCP/IP
protocol suites, so that users on the network can communicate. Unlike the PSTN, the Internet is a packet switched network, i.e. one in which data is carried in individually addressed packets from the near end to the far end.
The Internet is convenient for non-time critical data, such as file transfer and E-mail, although recently it has become popular for real time applications. It has long been possible to communicate in real time over the Internet using "chat servers". In such an arrangement, users communicate by entering data on their computer keyboards in real time.
More recently, software has become available that in conjunction with an audio card permits audio to be exchanged in real time over the Internet. This enable subscribers to carry on "telephone conversations" in real time over the Internet.
The disadvantage of such software, however, is that it only enables communication between Internet subscribers. A
receiving subscriber must have his computer connected to the Internet and running the appropriate software in the receive mode. An Internet subscriber cannot communicate with an ordinary POTS (Plain Ordinary Telephone Service) subscriber on the PSTN.
An object of the invention is to alleviate this disadvantage.
According to the present invention there is provided a telephone system permitting the establishment of telephone calls in either direction between a packet switched network carrying data packets and a circuit switched network carrying telephone signals, comprising a circuit switched telephone network with a plurality of first subscribers with unique telephone numbers and a packet switched network with a plurality of second subscribers with unique addresses and a gateway between said circuit switched and packet switched networks, said gateway comprising a first interface unit for connection to the circuit switched network and exchanging signals over circuits therein with said first subscribers, a second interface unit for connection to the packet switched network and exchanging packets thereover with said second subscribers, means for receiving telephone signals at said first interface and converting them to data packets for transmission over the packet switched network using Internet telephony, means for receiving data packets at said second interface and converting them _ to telephone signals for transmission over the circuit switched network, and processing means for determining destination information from an incoming telephone call received from a first subscriber on said circuit switched network and sending call setup messages over said packet switched network to a second subscriber identified from said destination information to permit direct two-way conversation between said first and second subscribers, and for determining destination information from data packets received on said packet switched network from a second subscriber and setting up a call to a first subscriber identified from said destination information carried in said data packets over a circuit on said circuit-switched network to permit direct two-way conversation between said first and second subscribers and wherein said processing means sets up a call over the packet switched network based on the incoming circuit on said circuit switched network at said gateway, the addresses on said packet switched network being mapped one-for-one to circuits on said circuit switched network.
The invention thus provides a gateway through which, for example, Internet users can call subscribers on the public switch telephone network and vice ver-sa. In the case of a telephone subscriber wishing to call someone on the Internet, the subscriber calls the number of the gateway, which then establishes a virtual connection over the Internet with the user. In one embodiment each incoming circuit at the gateway i~s mapped to a user's TCP/IP address.
Alternatively, the gateway can determine call party information from incoming signals on a trunk.
- 2a -In the reverse direction, the computer user sends a message to the gateway requesting the establishment of a call to a telephone subscriber on the PSTN. The gateway sets up the call and the two can then communicate.
According to a second aspect of the invention there is provided a method of establishing calls between a packet switched network carrying data packets and a circuit switched network carrying telephone signals, comprising the steps of providing a gateway between the Packet switched network and the circuit switched network receiving a telephone call over an incoming circuit in said circuit switched network identifying destination information from said incoming telephone call, sending a call setup message to a subscriber on said packet switched network at an address determined from said destination information to set up a two-way telephone call with said incoming circuit using Internet telephony over said packet switched network, said destination addresses being mapped one-for-one to incoming circuits, and said destination information being determined by the particular circuit on which an incoming call is received at said gateway, and converting in real time signals from said circuit switched network to data packets for transmission over said packet switched network and vice versa while the call is in progress.
The invention will now be described in more detail, by way of exam~le.only, with reference to the accompanying drawings, in which:-Figure 1 is a general block diagram of a system employing a gateway in accordance with the invention;
Figure 2 is a block diagram showing the establishment of a call from a computer user to a telephone subscriber;
Figure 3 is a block diagram showing the establishment of a call from a telephone subscriber to a computer user;
Figure 4 is a block diagram showing the speech paths;
- 3a -r ~18068~
Figure 5 is a functional block diagram of a gateway in accordance with the invention;
Figure 6 is a more detailed block diagram of a TCP/IP
module;
Figure 7 is a block diagram of a telephone interface;
Figure 8 is a top level system flow chart:
Figure 9 is a flow chart showing the handling of an incoming call; and Figure 10 is a flow chart showing the handling of an outgoing call to a PSTN.
The system shown in Figure I comprises a computer node 1 connected through the Internet 2 to a telephony gateway 3.
Gateway 3 is connected through the public switched telephone network (PSTN) 4 to a individual telephone subscriber 5.
Computer user at node 1 and telephone subscriber at telephone 5 can establish two-way voice communications in a manner that will described in more detail.
Figure 2 shows the initiation of a call from a computer user 10 to a telephone subscriber 1I. Firstly, the computer node 1 formats and sends a UDP (User Datagram Protocol) message at 11 to the ITG 3. This UDP contains information about the number of the called party on the PSTN. The ITG 3 decodes this message, goes off hook at 12 and calls the user 11 by dialling digits on the PSTN as shown in block 13. At block 14, the PSRN rings the telephone 5, which if not busy, is answered by the user 11. From this point, two-way conversation can be established through the ITG as indicated by block 15. Of course, if the telephone 5 is busy, a ring back flash busy indication is sent from the PSTN 4 via the 2~8~168~
ITG 3 to the user 10. From this point on, the system on the Internet side operates in the manner of a conventional Internet telephone, and on the PSTN side, operates in the manner of a conventional PSTN telephone. From the Internet side, the computer uses voices digitized and sent in packets to the ITG 3, where it is depacketized and forwarded over the PSTN to the telephone user 11, either in the form of analog or digital signals, for example PCM.
The establishment of a call from the telephone subscriber to the computer user is represented in Figure 3.
As shown at block 20, the telephone dials the number of the ITG 3 and the call-is routed by the PSTN 4. Based on the incoming circuit, or called party information contained within the incoming signals, the ITG 3, as shown at block 21, sends a packetized UDP call setup message which is received by the computer node 1 at block 22 and, in accordance with the computer node's software, "rings" the computer if the user is free (block 23). If the computer user is unavailable, a message is sent back through the system, which is picked up by the PSTN. As indicated by block 23, this sends a busy signal back to the user (block 24).
If-the computer user answers the call (block 25), computer node 1 sends a UDP answer message to IDTG 3, which in turn sends an off hook signal to the PSTN, which is received by the PSTN 4 (block 26). Two-way voice connection is then established as represented by block 27.
Figure 4 is a representation of the system once two-way voice communication has been established. Block 31 shows the computer user inputting voice, which is digitized at the computer node at block 32 and send in UDP packets to the ITG
3 which converts the packets into an analog voice data stream (although it could be digital) and transmits it to _ 5 _ the PSTN 4 which is then outputted at 33 to the telephone user. In the reverse direction user input 34 is transmitted through the PSTN to the ITD 3, which packetizes the voice input and sends it as UDP voice packets (block 35) to the computer node which outputs it as voice output 36.
Referring now to Figure 5, the ITG (Internet Telephony Gateway) comprises a main processor complex 40, for example a personal computer, connected to a system bus 41. The system bus is also connected to a switching matrix 42 and a messaging system 43. The switching matrix interconnects telephone line interface modules 44 and TCP/IP into phase modules 45. The switching matrix 42 is connected to the module for PCM (Pulse Code Modulator) links 46. The messaging system 43 is connected to the interface modules over message links 47.
The main processor 40 is responsible for the high level control of the ITG. It issues instructions over the system bus to the messaging system 43, which in turn sends control message to the interface modules 44 and 45 to direct the operations of these modules in order to setup and tear down calls.
The interface modules 44, 45, serve as slaves to the main processor40. Changes in call state (for example, originations, disconnects) are reported to the main processor 40 over the message links 47.
Calls originating on the Internet are routed to TCP/IP
interface module 45. The incoming data packets are depacketized and routed through the switching matrix to the appropriate outgoing telephone line module 44, where they are sent over the PSTN as either analog or digital signals.
The system works in a similar manner in reverse.
~18a~8~
The TCP/IP module is shown in more detail in Figure 6.
This consists of a high speed physical interface, for example T1, ISDN, PRI/BRI, etc. interface 50 to the TCP/IP
network, which would normally be the Internet. This is connected to a TCP/IP-to-PCM converter 51, the entire unit being controlled by a microprocessor 52. The microprocessor includes local RAM and runs software supporting TCP/IP and the connection protocols to the network. As shown in Figure 7, the telephone interface unit comprises a line interface 60, which can be conventional telephone line interface, and a line termination logic unit &1, which is a standard interface unit that detects ringing and the like.
The operation of the system will be understood better by referring to the flow charts shown in Figure 8 to 10.
Referring to Figure 8, after a start 70 the system initializes and downloads software and data tables to all modules 71. The system then scans for call request 72.
Block 73 detects a ring on the telephone line interface and if a ring is detected control is handed to block 74, which will be described in more detail with reference to Figure 9.
If no call is detected on the line interface, the system looks for a call setup request from the TCP/IP interface at 75, and if a call is found, control is handed over to block 76 (which will be described in more detail with reference to Figure 10). If no call request is detected, the system loops back to block 72, and the cycle repeats.
Block 74 is shown in more detail in Figure 9. After start-up 80, the main processor 40 uses the circuit idea of the originating telephone line circuit to determine the TCP/IP address of the call destination, and forwards call setup information to the TCP/IP interface as represented by block 81. The TCP/IP.interface module formats and sends a UDP {User Datagram Protocol) call setup message to the call _ 7 _ ~18~68J
destination computer 82_ The establishment of a call across the Internet in this manner is per se known.
The ITG then waits for a UDP answer message from the computer node 83. Decision unit 84 then determines whether an answer is received. If not, control is passed to unit 85 which determines whether ringing should be stopped (based on the time period). If no, the system loops back to block 83.
If yes, the call setup mode terminates as shown at 85. If an answer is received, decision unit 84 passes control to the main processor as represented by block 86, which then sets up a circuit switch path through the switching matrix 42 (as shown by block 86) and the TCP module begins TCP/IP
packet to PCM conversion at shown at 87. As shown at 88, the telephone line interface module sends an off hook signal to the subscriber telephone, and two-way connection is established. The system then goes into a wait mode for teardown as shown at 89. The system loops through 90 to determine whether a UDP disconnect message is received from the computer user.- If yes the system terminates as shown at block 85.
Figure 10 illustrates block 76 in more detail, which handles a call setup request from the TCP/IP interface_ The system starts at 91 and uses the calling party address contained in the UDP call setup message to determine the telephone interface that it is desired to go off hook on 92.
The called party number is obtained from the call setup packet 93, and the main processor sets up a PCM circuit switch path from the TCP/IP module to the selected telephone interface circuit at block 94. The main processor attaches a tone generator to the circuit switch path at 95. The telephone interface.circuit then goes off hook at 96, and two-way speech is established when the called party answers at 97. The system then goes into a wait for teardowm _ g _ 2~8~685 routine at 98. When a UDP disconnect message is received, decision unit 99 terminates the call at 100.
In the embodiment described, there is a one-to-one mapping between each telephone line interface circuit and a computer user's TCP/IP address. In other words, each computer user is allocated a telephone line at the ITG. The computer user can thus give out his ITG-telephone number, and any party calling that number will be connected to the computer user associated with that number. In the reverse direction, however, the computer user will be connected to the associated telephone line, but the ITG will dial any number requested by the computer user.
In an alternative embodiment, the ITG can be connected to a trunk, in which case call party information received on the trunk can be used to determine the TCP/IP address of the destination computer user.
It is assumed in the above description that the computer has a software sub-system that implements some form of Internet telephony capability. Such software is currently conventionally available and translates the actions of the computer user into UDP messages. The User Datagram Protocol is a convenient method of exchanging messages, but a similar low overhead transmission technique (such as XTP/Express Transfer Protocol), can be employed instead.
In the above, if the above system is described in a situation where the computer user wishes to terminate the conversation, the computer node sends a UDP disconnect message to the ITG 3. If the telephone user terminates the conversation, the telephone network terminates the connection and the computer user just hears the dial tone coming from the network. The computer user can then _ g _ disconnect himself from the network or initiate another call.
In effect, the system allows the Internet to serve as a means of permitting a computer user to establish two-way communication with an assigned remote telephone line interface. For example, conceivably, a computer user in Ottawa, Canada who has reason to make a substantial number of calls in the Los Angeles area could subscribe to an ITG
service in the Los Angeles area. This would in effect give the computer user in Ottawa access to a local line for Los Angeles over the Internet. The computer user could dial out from his line in Ottawa as if he were physically present with a telephone in Los Angeles.
It has been mentioned that it is possible to substitute a digital trunk interface for the line interface described above. This approach has the advantage that the conversion of digital voice data to analog voice data is no longer necessary, and it is also no longer necessary to have a dedicated line interface per computer user_ On the other hand, the call setup procedures are different and require the ITG to generate ring and busy tones for calls going to the computer user, depending on the state of the user. On a trunk interface, the ITG must also be prepared to received dial digits and must be able to send calling party directory numbers to the public network. For a call from a telephone subscriber to a computer, the ITG
has to do the following (assuming a trunk interface). The ITG receives call party digits over the trunk and must translate these into the associated computer user TCP/IP
address. He must then attempt to establish voice communication with the computer user and place a busy or ring back tone on the trunk depending on whether the computer user is free or already involved in another call..
21$~68~
Disconnect procedures are essentially the same as described above. The computer user signals a disconnect to the ITG
using a UDP and the ITG forwards a disconnect instruction to the PSTN.
For a call from the computer user to the telephone subscriber, the computer user sends the called party's telephone number to the ITG in a UDP setup message. The ITG
initiates the call over the trunk to the public telephone network using standard trunk procedures_ This involves sending the calling party's directory number to the network so that the correct billing procedures can be followed. A
voice pass is established at the same time to the network so that the computer user can hear the call progress tones (ring/busy) being supplied by the network. An end-to-end path is set up by the telephone network when the subscriber answers the calls.
The described system thus offers significant advantages over the prior art, in that it allows effective voice communication between computer users on the Internet and conventional telephone users on the PSTN.
The system has been described with reference to voice signals, although of course it would work with any signals capable of being carried over the PSTN, such as fax or even data signals.
Figure 4 is a block diagram showing the speech paths;
- 3a -r ~18068~
Figure 5 is a functional block diagram of a gateway in accordance with the invention;
Figure 6 is a more detailed block diagram of a TCP/IP
module;
Figure 7 is a block diagram of a telephone interface;
Figure 8 is a top level system flow chart:
Figure 9 is a flow chart showing the handling of an incoming call; and Figure 10 is a flow chart showing the handling of an outgoing call to a PSTN.
The system shown in Figure I comprises a computer node 1 connected through the Internet 2 to a telephony gateway 3.
Gateway 3 is connected through the public switched telephone network (PSTN) 4 to a individual telephone subscriber 5.
Computer user at node 1 and telephone subscriber at telephone 5 can establish two-way voice communications in a manner that will described in more detail.
Figure 2 shows the initiation of a call from a computer user 10 to a telephone subscriber 1I. Firstly, the computer node 1 formats and sends a UDP (User Datagram Protocol) message at 11 to the ITG 3. This UDP contains information about the number of the called party on the PSTN. The ITG 3 decodes this message, goes off hook at 12 and calls the user 11 by dialling digits on the PSTN as shown in block 13. At block 14, the PSRN rings the telephone 5, which if not busy, is answered by the user 11. From this point, two-way conversation can be established through the ITG as indicated by block 15. Of course, if the telephone 5 is busy, a ring back flash busy indication is sent from the PSTN 4 via the 2~8~168~
ITG 3 to the user 10. From this point on, the system on the Internet side operates in the manner of a conventional Internet telephone, and on the PSTN side, operates in the manner of a conventional PSTN telephone. From the Internet side, the computer uses voices digitized and sent in packets to the ITG 3, where it is depacketized and forwarded over the PSTN to the telephone user 11, either in the form of analog or digital signals, for example PCM.
The establishment of a call from the telephone subscriber to the computer user is represented in Figure 3.
As shown at block 20, the telephone dials the number of the ITG 3 and the call-is routed by the PSTN 4. Based on the incoming circuit, or called party information contained within the incoming signals, the ITG 3, as shown at block 21, sends a packetized UDP call setup message which is received by the computer node 1 at block 22 and, in accordance with the computer node's software, "rings" the computer if the user is free (block 23). If the computer user is unavailable, a message is sent back through the system, which is picked up by the PSTN. As indicated by block 23, this sends a busy signal back to the user (block 24).
If-the computer user answers the call (block 25), computer node 1 sends a UDP answer message to IDTG 3, which in turn sends an off hook signal to the PSTN, which is received by the PSTN 4 (block 26). Two-way voice connection is then established as represented by block 27.
Figure 4 is a representation of the system once two-way voice communication has been established. Block 31 shows the computer user inputting voice, which is digitized at the computer node at block 32 and send in UDP packets to the ITG
3 which converts the packets into an analog voice data stream (although it could be digital) and transmits it to _ 5 _ the PSTN 4 which is then outputted at 33 to the telephone user. In the reverse direction user input 34 is transmitted through the PSTN to the ITD 3, which packetizes the voice input and sends it as UDP voice packets (block 35) to the computer node which outputs it as voice output 36.
Referring now to Figure 5, the ITG (Internet Telephony Gateway) comprises a main processor complex 40, for example a personal computer, connected to a system bus 41. The system bus is also connected to a switching matrix 42 and a messaging system 43. The switching matrix interconnects telephone line interface modules 44 and TCP/IP into phase modules 45. The switching matrix 42 is connected to the module for PCM (Pulse Code Modulator) links 46. The messaging system 43 is connected to the interface modules over message links 47.
The main processor 40 is responsible for the high level control of the ITG. It issues instructions over the system bus to the messaging system 43, which in turn sends control message to the interface modules 44 and 45 to direct the operations of these modules in order to setup and tear down calls.
The interface modules 44, 45, serve as slaves to the main processor40. Changes in call state (for example, originations, disconnects) are reported to the main processor 40 over the message links 47.
Calls originating on the Internet are routed to TCP/IP
interface module 45. The incoming data packets are depacketized and routed through the switching matrix to the appropriate outgoing telephone line module 44, where they are sent over the PSTN as either analog or digital signals.
The system works in a similar manner in reverse.
~18a~8~
The TCP/IP module is shown in more detail in Figure 6.
This consists of a high speed physical interface, for example T1, ISDN, PRI/BRI, etc. interface 50 to the TCP/IP
network, which would normally be the Internet. This is connected to a TCP/IP-to-PCM converter 51, the entire unit being controlled by a microprocessor 52. The microprocessor includes local RAM and runs software supporting TCP/IP and the connection protocols to the network. As shown in Figure 7, the telephone interface unit comprises a line interface 60, which can be conventional telephone line interface, and a line termination logic unit &1, which is a standard interface unit that detects ringing and the like.
The operation of the system will be understood better by referring to the flow charts shown in Figure 8 to 10.
Referring to Figure 8, after a start 70 the system initializes and downloads software and data tables to all modules 71. The system then scans for call request 72.
Block 73 detects a ring on the telephone line interface and if a ring is detected control is handed to block 74, which will be described in more detail with reference to Figure 9.
If no call is detected on the line interface, the system looks for a call setup request from the TCP/IP interface at 75, and if a call is found, control is handed over to block 76 (which will be described in more detail with reference to Figure 10). If no call request is detected, the system loops back to block 72, and the cycle repeats.
Block 74 is shown in more detail in Figure 9. After start-up 80, the main processor 40 uses the circuit idea of the originating telephone line circuit to determine the TCP/IP address of the call destination, and forwards call setup information to the TCP/IP interface as represented by block 81. The TCP/IP.interface module formats and sends a UDP {User Datagram Protocol) call setup message to the call _ 7 _ ~18~68J
destination computer 82_ The establishment of a call across the Internet in this manner is per se known.
The ITG then waits for a UDP answer message from the computer node 83. Decision unit 84 then determines whether an answer is received. If not, control is passed to unit 85 which determines whether ringing should be stopped (based on the time period). If no, the system loops back to block 83.
If yes, the call setup mode terminates as shown at 85. If an answer is received, decision unit 84 passes control to the main processor as represented by block 86, which then sets up a circuit switch path through the switching matrix 42 (as shown by block 86) and the TCP module begins TCP/IP
packet to PCM conversion at shown at 87. As shown at 88, the telephone line interface module sends an off hook signal to the subscriber telephone, and two-way connection is established. The system then goes into a wait mode for teardown as shown at 89. The system loops through 90 to determine whether a UDP disconnect message is received from the computer user.- If yes the system terminates as shown at block 85.
Figure 10 illustrates block 76 in more detail, which handles a call setup request from the TCP/IP interface_ The system starts at 91 and uses the calling party address contained in the UDP call setup message to determine the telephone interface that it is desired to go off hook on 92.
The called party number is obtained from the call setup packet 93, and the main processor sets up a PCM circuit switch path from the TCP/IP module to the selected telephone interface circuit at block 94. The main processor attaches a tone generator to the circuit switch path at 95. The telephone interface.circuit then goes off hook at 96, and two-way speech is established when the called party answers at 97. The system then goes into a wait for teardowm _ g _ 2~8~685 routine at 98. When a UDP disconnect message is received, decision unit 99 terminates the call at 100.
In the embodiment described, there is a one-to-one mapping between each telephone line interface circuit and a computer user's TCP/IP address. In other words, each computer user is allocated a telephone line at the ITG. The computer user can thus give out his ITG-telephone number, and any party calling that number will be connected to the computer user associated with that number. In the reverse direction, however, the computer user will be connected to the associated telephone line, but the ITG will dial any number requested by the computer user.
In an alternative embodiment, the ITG can be connected to a trunk, in which case call party information received on the trunk can be used to determine the TCP/IP address of the destination computer user.
It is assumed in the above description that the computer has a software sub-system that implements some form of Internet telephony capability. Such software is currently conventionally available and translates the actions of the computer user into UDP messages. The User Datagram Protocol is a convenient method of exchanging messages, but a similar low overhead transmission technique (such as XTP/Express Transfer Protocol), can be employed instead.
In the above, if the above system is described in a situation where the computer user wishes to terminate the conversation, the computer node sends a UDP disconnect message to the ITG 3. If the telephone user terminates the conversation, the telephone network terminates the connection and the computer user just hears the dial tone coming from the network. The computer user can then _ g _ disconnect himself from the network or initiate another call.
In effect, the system allows the Internet to serve as a means of permitting a computer user to establish two-way communication with an assigned remote telephone line interface. For example, conceivably, a computer user in Ottawa, Canada who has reason to make a substantial number of calls in the Los Angeles area could subscribe to an ITG
service in the Los Angeles area. This would in effect give the computer user in Ottawa access to a local line for Los Angeles over the Internet. The computer user could dial out from his line in Ottawa as if he were physically present with a telephone in Los Angeles.
It has been mentioned that it is possible to substitute a digital trunk interface for the line interface described above. This approach has the advantage that the conversion of digital voice data to analog voice data is no longer necessary, and it is also no longer necessary to have a dedicated line interface per computer user_ On the other hand, the call setup procedures are different and require the ITG to generate ring and busy tones for calls going to the computer user, depending on the state of the user. On a trunk interface, the ITG must also be prepared to received dial digits and must be able to send calling party directory numbers to the public network. For a call from a telephone subscriber to a computer, the ITG
has to do the following (assuming a trunk interface). The ITG receives call party digits over the trunk and must translate these into the associated computer user TCP/IP
address. He must then attempt to establish voice communication with the computer user and place a busy or ring back tone on the trunk depending on whether the computer user is free or already involved in another call..
21$~68~
Disconnect procedures are essentially the same as described above. The computer user signals a disconnect to the ITG
using a UDP and the ITG forwards a disconnect instruction to the PSTN.
For a call from the computer user to the telephone subscriber, the computer user sends the called party's telephone number to the ITG in a UDP setup message. The ITG
initiates the call over the trunk to the public telephone network using standard trunk procedures_ This involves sending the calling party's directory number to the network so that the correct billing procedures can be followed. A
voice pass is established at the same time to the network so that the computer user can hear the call progress tones (ring/busy) being supplied by the network. An end-to-end path is set up by the telephone network when the subscriber answers the calls.
The described system thus offers significant advantages over the prior art, in that it allows effective voice communication between computer users on the Internet and conventional telephone users on the PSTN.
The system has been described with reference to voice signals, although of course it would work with any signals capable of being carried over the PSTN, such as fax or even data signals.
Claims (13)
1. A telephone system permitting the establishment of telephone calls in either direction between a packet switched network carrying data packets and a circuit switched network carrying telephone signals, comprising:
a circuit switched telephone network with a plurality of first subscribers with unique telephone numbers; and a packet switched network with a plurality of second subscribers with unique addresses; and a gateway between said circuit switched and packet switched networks, said gateway comprising:
a first interface unit for connection to the circuit switched network and exchanging signals over circuits therein with said first subscribers;
a second interface unit for connection to the packet switched network and exchanging packets thereover with said second subscribers;
means for receiving telephone signals at said first interface and converting them to data packets for transmission over the packet switched network using Internet telephony;
means for receiving data packets at said second interface and converting them to telephone signals for transmission river the circuit switched network; and processing means for determining destination information from an incoming telephone call received from a first subscriber on said circuit switched network and sending call setup messages over said packet switched network to a second subscriber identified from said destination information to permit direct two-way conversation between said first and second subscribers, and for determining destination information from data packets received on said packet switched network from a second subscriber and setting up a call to a first subscriber identified from said destination information carried in said data packets over a circuit on said circuit-switched network to permit direct two-way conversation between said first and second subscribers; and wherein said processing means sets up a call over the packet switched network based on the incoming circuit on said circuit switched network at said gateway, the addresses on said packet switched network being mapped one-for-one to circuits on said circuit switched network.
a circuit switched telephone network with a plurality of first subscribers with unique telephone numbers; and a packet switched network with a plurality of second subscribers with unique addresses; and a gateway between said circuit switched and packet switched networks, said gateway comprising:
a first interface unit for connection to the circuit switched network and exchanging signals over circuits therein with said first subscribers;
a second interface unit for connection to the packet switched network and exchanging packets thereover with said second subscribers;
means for receiving telephone signals at said first interface and converting them to data packets for transmission over the packet switched network using Internet telephony;
means for receiving data packets at said second interface and converting them to telephone signals for transmission river the circuit switched network; and processing means for determining destination information from an incoming telephone call received from a first subscriber on said circuit switched network and sending call setup messages over said packet switched network to a second subscriber identified from said destination information to permit direct two-way conversation between said first and second subscribers, and for determining destination information from data packets received on said packet switched network from a second subscriber and setting up a call to a first subscriber identified from said destination information carried in said data packets over a circuit on said circuit-switched network to permit direct two-way conversation between said first and second subscribers; and wherein said processing means sets up a call over the packet switched network based on the incoming circuit on said circuit switched network at said gateway, the addresses on said packet switched network being mapped one-for-one to circuits on said circuit switched network.
2. A system as claimed in claim 1, wherein said second interface unit comprises a TCP/IP interface unit for sending and receiving data packets over the packet switched network and said first interface unit is a telephone line interface for connection to the public switched telephone network.
3. A system as claimed in claim 2, further comprising a message system for sending and receiving control messages to said interface units under the control of said processing means.
4. A system as claimed in claim 3, further comprising a circuit switching matrix connected between said first and second interface units.
5. A system as claimed in claim 1, wherein said processing means sets up a call over the packet switched network based on the incoming circuit on said circuit switched network, the addresses on said packet switched network being mapped to one-for one to circuits on said circuit switched network.
6. A system as claimed in claim 1, wherein said processing means extracts the destination information from the called party information carried on the incoming signals aver the circuit switched network.
7. A system as claimed in claim 2, wherein said signals are PCM (pulse coded modulation) signals.
8. A system as claimed in claim 7, wherein said second interface unit is a primary rate TDM interface.
9. A method of establishing calls between a packet switched network carrying data packets and a circuit switched network carrying telephone signals, comprising the steps of:
providing a gateway between the packet switched network and the circuit switched network;
receiving a telephone call over an incoming circuit in said circuit switched network;
identifying destination information from said incoming telephone call;
sending a call setup message to a subscriber on said packet switched network at an address determined from said destination information to set up a two-way telephone call with said incoming circuit using Internet telephony over said packet switched network, said destination addresses being mapped one-for-one to incoming circuits, and said destination information being determined by the particular circuit on which an incoming call is received at said gateway; and converting in real time signals from said circuit switched network to data packets for transmission over said packet switched network and vice versa while the call is in progress.
providing a gateway between the packet switched network and the circuit switched network;
receiving a telephone call over an incoming circuit in said circuit switched network;
identifying destination information from said incoming telephone call;
sending a call setup message to a subscriber on said packet switched network at an address determined from said destination information to set up a two-way telephone call with said incoming circuit using Internet telephony over said packet switched network, said destination addresses being mapped one-for-one to incoming circuits, and said destination information being determined by the particular circuit on which an incoming call is received at said gateway; and converting in real time signals from said circuit switched network to data packets for transmission over said packet switched network and vice versa while the call is in progress.
10. A method as claimed in claim 9, wherein said packets are transmitted over said packet switched network using a TCP/IP protocol.
11. A method as claimed in claim 9, wherein the call is setup messages are controlled by a main processor directing operations.
12. A method as claimed in claim 9, wherein said signals are PCM signals.
13. A method as claimed in claim 9, wherein said destination addresses are mapped one-for-one to incoming circuits, and said destination information is determined by the particular circuit on which an incoming call is received.
Priority Applications (4)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA 2180685 CA2180685C (en) | 1996-07-08 | 1996-07-08 | Internet telephony gateway |
| GB9714058A GB2315190A (en) | 1996-07-08 | 1997-07-04 | An internet telephony gateway for providing an interface between a circuit switched and packet switched network |
| DE1997128878 DE19728878A1 (en) | 1996-07-08 | 1997-07-07 | Internet phone gateway |
| FR9708918A FR2754124A1 (en) | 1996-07-08 | 1997-07-08 | TELEPHONY GATEWAY ON THE INTERNET |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CA 2180685 CA2180685C (en) | 1996-07-08 | 1996-07-08 | Internet telephony gateway |
Publications (2)
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| CA2180685A1 CA2180685A1 (en) | 1998-01-09 |
| CA2180685C true CA2180685C (en) | 2003-09-09 |
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|---|---|---|---|
| CA 2180685 Expired - Fee Related CA2180685C (en) | 1996-07-08 | 1996-07-08 | Internet telephony gateway |
Country Status (4)
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| DE (1) | DE19728878A1 (en) |
| FR (1) | FR2754124A1 (en) |
| GB (1) | GB2315190A (en) |
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| US5910946A (en) * | 1997-01-13 | 1999-06-08 | Samsung Electronics Co., Ltd. | Wireless internet network architecture for voice and data communications |
| US6104802A (en) | 1997-02-10 | 2000-08-15 | Genesys Telecommunications Laboratories, Inc. | In-band signaling for routing |
| US6480600B1 (en) | 1997-02-10 | 2002-11-12 | Genesys Telecommunications Laboratories, Inc. | Call and data correspondence in a call-in center employing virtual restructuring for computer telephony integrated functionality |
| US7031442B1 (en) | 1997-02-10 | 2006-04-18 | Genesys Telecommunications Laboratories, Inc. | Methods and apparatus for personal routing in computer-simulated telephony |
| GB2326055B (en) * | 1997-06-04 | 2002-03-13 | Northern Telecom Ltd | Communication system |
| EP0935861A2 (en) * | 1997-08-13 | 1999-08-18 | Koninklijke Philips Electronics N.V. | Method and system for requesting a dial up connection via a packet switched network |
| US6711611B2 (en) | 1998-09-11 | 2004-03-23 | Genesis Telecommunications Laboratories, Inc. | Method and apparatus for data-linking a mobile knowledge worker to home communication-center infrastructure |
| US6985943B2 (en) | 1998-09-11 | 2006-01-10 | Genesys Telecommunications Laboratories, Inc. | Method and apparatus for extended management of state and interaction of a remote knowledge worker from a contact center |
| USRE46528E1 (en) | 1997-11-14 | 2017-08-29 | Genesys Telecommunications Laboratories, Inc. | Implementation of call-center outbound dialing capability at a telephony network level |
| US7907598B2 (en) | 1998-02-17 | 2011-03-15 | Genesys Telecommunication Laboratories, Inc. | Method for implementing and executing communication center routing strategies represented in extensible markup language |
| US8130749B2 (en) | 1998-02-17 | 2012-03-06 | Genesys Telecommunications Laboratories Inc., A Corp of California | Telephone network interface bridge between data telephony networks and dedicated connection telephony networks |
| US6332154B2 (en) | 1998-09-11 | 2001-12-18 | Genesys Telecommunications Laboratories, Inc. | Method and apparatus for providing media-independent self-help modules within a multimedia communication-center customer interface |
| DE19811888B4 (en) * | 1998-03-18 | 2004-10-07 | Ncp Engineering Network Communications Products Engineering Gmbh | Method and device for telecommunications |
| US6424707B1 (en) * | 1998-04-20 | 2002-07-23 | Lucent Technologies Inc. | Point-to-point-to-point calling |
| DE19840329A1 (en) * | 1998-09-04 | 2000-03-09 | Alcatel Sa | Telecommunication system with switching device and data concentrator for access to the Internet |
| USRE46153E1 (en) | 1998-09-11 | 2016-09-20 | Genesys Telecommunications Laboratories, Inc. | Method and apparatus enabling voice-based management of state and interaction of a remote knowledge worker in a contact center environment |
| US6445697B1 (en) * | 1998-09-25 | 2002-09-03 | Cisco Technology, Inc. | Audio codec reselection for increased port density |
| US6442169B1 (en) | 1998-11-20 | 2002-08-27 | Level 3 Communications, Inc. | System and method for bypassing data from egress facilities |
| US6614781B1 (en) | 1998-11-20 | 2003-09-02 | Level 3 Communications, Inc. | Voice over data telecommunications network architecture |
| US6603762B1 (en) | 1999-03-12 | 2003-08-05 | Lextron Systems, Inc. | System for controlling processing of data passing through network gateway between two disparate communications network |
| GB2350521B (en) * | 1999-04-30 | 2001-07-11 | Nokia Corp | A gateway arrangement |
| DE19948433A1 (en) * | 1999-10-08 | 2001-04-19 | Alcatel Sa | Procedures to support the establishment of telephone connections |
| US7929978B2 (en) | 1999-12-01 | 2011-04-19 | Genesys Telecommunications Laboratories, Inc. | Method and apparatus for providing enhanced communication capability for mobile devices on a virtual private network |
| CA2328008A1 (en) * | 1999-12-20 | 2001-06-20 | Nortel Networks Limited | Private reuse of the public switched telephone network dial plan |
| EP1128682A1 (en) * | 2000-02-21 | 2001-08-29 | Alcatel | Voice-data integration architecture for PABX |
| AU2001242291B2 (en) * | 2000-03-06 | 2004-12-02 | Nokia Siemens Networks Gmbh & Co. Kg | Method and device for transmitting voice data in a mobile radio network |
| US7324635B2 (en) | 2000-05-04 | 2008-01-29 | Telemaze Llc | Branch calling and caller ID based call routing telephone features |
| DE10161509A1 (en) | 2001-12-14 | 2003-07-03 | Siemens Ag | Method and arrangement for transporting data packets of a data stream |
| EP1341356A3 (en) * | 2002-02-25 | 2008-10-22 | Alcatel Lucent | Two-way video gateway and method for establishing an audio and video communications link between dissimilar multimedia terminals |
| US7366198B2 (en) * | 2003-03-07 | 2008-04-29 | Lucent Technologies, Inc. | Method and system for packet and circuit telephony in a distributed telecommunications switching system |
| DE102005001351A1 (en) | 2005-01-11 | 2006-07-20 | Teles Ag Informationstechnologien | Method for transmitting data to at least one communication end system and communication device for carrying out the method |
| US9008075B2 (en) | 2005-12-22 | 2015-04-14 | Genesys Telecommunications Laboratories, Inc. | System and methods for improving interaction routing performance |
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| US4556972A (en) * | 1983-12-27 | 1985-12-03 | At&T Bell Laboratories | Arrangement for routing data packets through a circuit switch |
| DE3609985A1 (en) * | 1986-03-25 | 1987-10-01 | Bayer Ag | BASIC POLYCONDENSATES |
| JPH07170288A (en) * | 1993-12-15 | 1995-07-04 | Hitachi Ltd | Voice communication system and voice communication method |
| US7336649B1 (en) * | 1995-12-20 | 2008-02-26 | Verizon Business Global Llc | Hybrid packet-switched and circuit-switched telephony system |
-
1996
- 1996-07-08 CA CA 2180685 patent/CA2180685C/en not_active Expired - Fee Related
-
1997
- 1997-07-04 GB GB9714058A patent/GB2315190A/en not_active Withdrawn
- 1997-07-07 DE DE1997128878 patent/DE19728878A1/en not_active Ceased
- 1997-07-08 FR FR9708918A patent/FR2754124A1/en not_active Withdrawn
Also Published As
| Publication number | Publication date |
|---|---|
| FR2754124A1 (en) | 1998-04-03 |
| CA2180685A1 (en) | 1998-01-09 |
| DE19728878A1 (en) | 1998-01-15 |
| GB2315190A (en) | 1998-01-21 |
| GB9714058D0 (en) | 1997-09-10 |
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