Note: Descriptions are shown in the official language in which they were submitted.
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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
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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,
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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.
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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;
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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;
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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
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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.
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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
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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
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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
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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..
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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.
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