WO2004049655A1

WO2004049655A1 - System and method for voice over ip communication

Info

Publication number: WO2004049655A1
Application number: PCT/IL2003/000698
Authority: WO
Inventors: Shimon Ben David
Original assignee: Shimon Ben David
Priority date: 2002-11-26
Filing date: 2003-08-25
Publication date: 2004-06-10
Also published as: AU2003253238A1

Abstract

A communication system and method which bridges between digital and analog facilities, offering the full utilization of digital communication services over standard PSTN lines (10) and analog fax machines (17) and telephone sets (16). Voice and Fax over Internet Protocol (F/VoIP), SMS messages (19) and textual and voice messages may be initiated and received via any phone (6) or fax machine (7) equipped with an adapter (8). A base unit (11) provides the interface between the IP and the analog communication means. The system also allows Internet providers (3) to offer advanced storage capabilities for incoming analog and digital messages.

Description

System And Method For Voice Over IP Communication

BACKGROUND OF THE INVENTION

The present invention relates to the field of communications. More particularly, the present invention is directed to a systems that connects a Voice over Internet Protocol (VoIP) communications network, to standard analog telephone sets.

The widening availability of higher bandwidth and low-latency performance capabilities of broad-band network connections had made the possibilities of utilizing the communication possibilities of the personal computer (PC) at home or in SOHO (Small Office, Home Office) grow rapidly over the last few years.

In general, the appeal of communication via the Internet, compared to communication via the Public Switched Telephone Network (PSTN), is lower cost to the end user. Thus, in order to reduce costs, individuals and small businesses seek to shift more and more communication traffic from the PSTN to the Internet. A main player of this "traffic" is the VoIP which allows making telephone conversations via the Internet connection bypassing the costs of PSTN local, long-distance and international calls.

There are several solutions for utilizing VoIP services from home or in SOHO, mainly embodied in direct usage of the PC, through a microphone and speakers, or by similar solutions. Other solutions such as U.S. Pat. 6,167,043 allow the SOHO user to use FM channels to receive and transmit voice and control data between the gateway and the adapter. This invention depends on the operation of a private branch exchange (PBX) and it does not provide bridging abilities between the PSTN and the Internet Voice Gateway.

There is the need for a low-cost simple to install and maintain system that will allow users to bridge between the VoIP calls and other digital data communication means and the analog telephone lines. The intention of the present invention is to provide a simple low-cost device which enable switching capacities between these to modes of communication, facilitating the delivery of VoIP calls over the data network from and to the analog telephone sets at home or in SOHO environments, through an IP network, xDSL or cable TV network for example.

There is also the need for a low-cost, simple to install and maintain system that allows users to send and receive voice and textual messages over various digital data communication means. The present invention provides a device capable of receiving and displaying textual message and voice messages from cellular networks by commonplace analog devices. The present invention also allows storage and retrieval of said messages by remote servers, opening opportunities for service providers to enhance their service offerings.

SUMMARY OF THE INVENTION

It is therefore an object of the present invention to provide a system that enables establishing both digital voice communication over data communication (IP) network and analog voice communication over PSTN network over the same internal standard PSTN telephone system. F/VoIP calls can be initiated and received by standard analog telephone units, connected to the PSTN telephone line. The said system is comprised of any data communication device (modem) which is able to establish communication with the IP network; a processing unit, such as a PC computer, which can establish and manage IP communications; a base unit which connects to the computer and to the premises' PSTN network; and an adapter unit for each telephone extension nodes, connected to the twisted pair wiring system. The base unit identifies incoming VoIP calls, converts the digital signals to analog signals and vice versa. The adapter unit has a unique identifying code, enabling the base unit to distinguish between the existing adapter units on the premises, and to uniquely communicate with each of the adapters.

The method for establishing an outgoing call between an analog telephone set on the premises and a VoIP remote destination is comprised of the following steps: a dialup analog signal is detected, the destination number is identified as a VoIP destination and communication with a voice gateway is established and maintained through the IP connection. The method of establishing an incoming VoIP call from a remote caller is comprised of identifying the call as a VoIP, establishing and maintaining communication with the voice gateway, generating a ring signal voltage which will cause the telephone sets to ring and detecting the pickup signal when the user answers the call. In both incoming and outgoing calls analog voice signals are converted to digital signals and visa versa, and the call is switched to the appropriate adaptor by the base unit. This is all done in an on- hook state; the PSTN switcher is "unaware" of this procedure.

BRIEF DESCRIPTION OF THE DRAWINGS

The accompanying drawings, which are incorporated in and constitute a part of this specification, illustrate one possible embodiment of the invention and, together with the description, explain the invention. In the drawings,

Figure 1 illustrates an embodiment of the present invention integrated into a telecommunication system;

Figure 2 is a flow chart of the present invention's procedure of receiving an incoming voice call;

Figure 3 is a flow chart of the present invention's procedure of operating an outgoing voice call;

Figure 4 s a flow chart of the present invention's procedure of receiving a SMS message;

Figure 5 is a block diagram illustrating the components of the base unit;

Figure 6 is a block diagram illustrating the components of the adapter unit;

Figure 7 is a flow chart of the present invention's storage services;

Figure 8 is a flow chart of the present invention's procedure of retrieving voice and fax messages form an external IP services provider's messages server. DETAILED DESCRIPTION OF THE INVENTION

The present invention proposes a new system and method for solving existing problems and limitations in the growing field of private and small scale business usage of digital communication methods over the Internet. The invention offers a simple and low cost solution for the integration of these methods of communication with the existing household analog methods of communication.

The system connects a data communications network, which has the ability to deliver Voice over the Internet Protocol (VoIP) telephone calls from any Internet connection (IP network, xDSL or cable TV network for example), to multiple standard analog telephone sets which operate via a Public Switched Telephone Network (PSTN). The system enables the user to receive VoIP services over analog telephone sets, over existing telephone wiring at user's premises.

Analog or digital calls can be placed over the same twisted pair wiring, existing at user's premises, without any installation changes. The system does not interfere with the analog telephone operation and services, including advanced services such as call waiting and caller ID. The system also generates normal ring signal voltage in the case of an incoming digital phone call, and emulates the operation of the call waiting function for incoming digital phone calls. The system can work with any analog telephone set, corded or cordless and all system functions (voice transport, power transmission and control) are carried out while telephone line is kept in on hook state.

The system consists of at least two module units: a single Management module, which in the example embodiment is a Base Unit, which connects to the computer, to the Internet connection and to the telephone line, and one or more Telephone Set Interface modules, which in the example embodiment are Adapter Units which connect to individual analog telephone sets. All signaling between system modules are carried out over the twisted-pair telephone wires in on hook state.

The Base Unit identifies the incoming or outgoing VoIP call, performs the analog-to-digital (A/D) and digital-to-analog (D/A) signal conversion and directs the call to the appropriate analog telephone set via the Adapter Unit. The Base Unit is powered from an external supply and may support more then one simultaneous voice/data communication channel.

The Adapter Unit provides an interface for a standard analog telephone set; it does not intervene with the transmission of the voice signals between the analog telephone set and the Base Unit. It provides a means of unique identification of a specific analog telephone set, and the means of generating a normal ring signal voltage in the case of an incoming digital phone call. The Adapter Unit can be powered by rechargeable batteries, charging from the telephone line or by an external power source.

Figure 1 illustrates the implementation of the present invention in the environment of a PSTN and data communication networks. The preferred embodiment of the invention includes the Base Unit 11 and several Adapter Units 8.

On the user's premises, standard analog telephones 6 or fax machines 7 may each be connected to the Adapter 8. The Adapter 8 is also connected to the house standard internal telephone wires 9. Several analog telephones or fax machines may be connected (optionally through said Adapters) to the same telephone wires 9. The said telephone wires are connected to the PSTN 4 via a traditional telephone line 10.

The main component of the invention is a Base Unit 11 , which is connected to a telephone line 9 and to a computer 12, via digital connections

13. The computer 12 may be a stand-alone device, such as a PC, or an embedded component of the Base Unit 11 or the data communications device

14. Alternatively, the Base Unit 11 can be embedded in the computer 12. The data communications device 14 may be a peripheral unit of the computer 12 or an embedded part of the Base Unit 11 or the computer.

The Wide Area communication Network (WAN) 1 is accessed by the voice gateway 2 and the Internet Service Provider Point-of-Presence (ISP POP) 3, enabling communications of data and digital audio between the voice gateway 2 and the ISP POP 3. The voice gateway 2 can establish and receive telephone calls on the Public Switched Telephone Network (PSTN) 4 via a local telephone line 5.

The computer 12 can access the WAN 1 via a data communications device 14 and a data communication link 15 to the ISP POP 3. The data communications link 15 may be any broadband data connection, such as xDSL or Cable TV connection. For the invention's operation a constant data communication link has to be established.

Any other PSTN subscriber may have a standard analog telephone set 16 or fax machine 17 connected to the PSTN 4 via a local telephone line 18.

Using telephone set 16 or fax machine 17, a digital phone call destined to one of the said telephones 6 or fax machines 7 may be placed through the voice gateway 2.

All communications between the units, including voice transmission during digital calls, takes place while the telephone line is kept in on hook state. Control signals flowing between the units are in the 0-4KHz frequency range, for example Dual Tone Multi-Frequency (DTMF) tones. Since the telephone line is kept in on hook state, the telephone company's exchange does not react to tones.

The sequence of operations for establishing and managing a VoIP call which is initiated by a standard phone unit 16 is described in a flow chart in figure 2:

An Analog voice signal flow from telephone set 16 to the voice gateway 2 through telephone line 18 and the PSTN 4 utilizing an IVR ("Interactive Voice Response") system. The caller is enabled to interact with the voice gateway 2 using common telephone interface (of DTMF dialing) by selecting an option of voice over IP in an IVR menu, and entering a specific destination address to be placed over the WAN 1, which may be the respective IP address, for example. The Voice Gateway 2 communicates the request to the computer 12 over the WAN 1 , using industry-standards protocols (such as SIP).

The Computer 12 communicates the request to the Base Unit 11 over the Digital Link 13. The Base Unit 11 transmits control signals over the telephone line 9 to all the Adapters 8 connected to said telephone line. It is important to note that all telephone sets connected to telephone line 9 are in on-hook state and do not draw any current from the PSTN 4, therefore, no part of the PSTN is "aware" of these operations or effected by them.

The Base Unit 11 stores in its internal memory a list of the unique addresses of all Adapter Units operating on the premises. The said control signals include an encoded specific unique address, identifying the destination Adapter 8. Each Adapter 8 has its corresponding unique address stored in its internal memory. All Adapters on telephone line 9 decode the address from the said control signals, but only the designated Adapter with the address that matches the sent address in the control signals responds to the signals. All other Adapters on telephone line 9 return to idle state.

The said destination Adapter 8 sends ring signals to the attached telephone set 6 for a limited period. If the said telephone set 6 is answered during the timeout period, the Adapter 8 detects this event and sends an acknowledge signal to the Base Unit 11 over the telephone line 9. The said Adapter isolates the attached telephone set 6 or from the PSTN 4 and prevents it from drawing any current from it. Therefore, no part of the PSTN is affected by the system during its operation. If the said attached telephone set 6 is not picked up during the timeout period, the destination Adapter 8 sends a negative acknowledge signal to the Base Unit 11 over the telephone line 9. The Base Unit 11 detects the signal and communicates it digitally to the computer 12 over the digital link 13. The computer 12 then transmits a terminating signal through the data communication link 15 to the voice gateway 2 over the WAN 1.

If the destination Adapter 8 did send an acknowledge signal to the Base Unit 11 , the Base Unit detects it and passes it digitally to the computer 12 over digital link 13. The computer 12 communicates with the voice gateway 2 over the WAN 1 to setup the call, using industry-standard protocols.

During the call session, the computer 12 communicates with the voice gateway 2 according to standard VoIP protocol mode of operation. From the computer 12, the digital data flows to the Base Unit 11 over the digital link 13. The Base Unit 11 converts the digital data back to an analog signal and transmits it over the telephone line 9 to all Adapters Units 8. Only the destined Adapter Unit 8 allows the analog signal to flow through to the attached telephone set 6.

It is important to note that all telephones sets connected to telephone line 9, including said destination Adapter 8, do not draw any current from the PSTN 4. Therefore, no part of the PSTN is "aware" of these operations or is affected by them.

Simultaneously, analog voice signals generated at the telephone set 6 pass unchanged through Adapter 8 over the telephone line 9 to the Base Unit 11. The Base Unit 11 digitizes the analog signal and delivers it in digital form over the digital link 13 to the computer 12.

A digital phone call can also originate from telephone 6 to telephone set 16. The sequence of operations for such a call is described in a flow chart in figure 3: User dials a phone number at telephone set 6. The attached Adapter Unit 8 detects the digits being dialed and compares the number against a list of predefined phone numbers of VoIP destinations. The list may be stored in a storage facility, such as in said Adapter's 8 internal memory, in Base Unit 11 memory, in the computer 12 memory or in the call services server 47, for example. If a match is found, said Adapter 8 will continue handling the call as described below. Otherwise, said Adapter 8 will return to an idle state and the call will proceed as an uninterrupted traditional PSTN call.

If a digital destination number is identified and the said Adapter 8 is handling the call, it isolates the attached telephone set 6 from the PSTN 4, so it does not draw any current from the PSTN. Therefore, no part of the PSTN 4 will be "aware" of the system's operations.

Said Adapter 8 then sends control signals to the Base Unit 11 over the telephone line 9, encoding a request to make the call. The Base Unit 11 decodes the signal and communicates the request to the computer 12 over the digital link 13. The computer then communicates the said call request to the voice gateway 2 over the WAN 1 , using industry-standards protocols (such as SIP). The voice gateway 2 places a PSTN call over telephone line 5 through PSTN 4 and telephone line 18 to telephone set 16. When the call is answered, the said gateway establishes the digital call with the computer 12 using industry-standard protocols.

During the call session, the bi-directional operation of the system is identical to the first scenario of an incoming call.

Analog fax call can be operated as FolP calls in the same manner that VoIP call operate, as described above, provided that the Adapter Unit 8 is connected to the analog fax machine 7 and telephone line 9. Outgoing fax calls operate exactly as an outgoing voice calls, the same is true for incoming analog fax calls coming from the PSTN 4. Incoming digital fax call may be recognized as fax calls by the Base Unit 11 and directed to the Adapter Unit 8 of the fax machine 7 according to the list of unique addresses stored in its internal memory.

Operating F/VoIP calls via analog telephone sets and fax machines, as aforesaid, are only one of the facilities that this invention offers.

According to alternative embodiment of the present invention, the said Base Unit 11, which serves as bridging apparatus between the analog and the digital lines, is enabled to support advance storage services for both analog and digital communications by IP service providers. These services may be implemented in the Base Unit 11 , in the computer or by an external IP services provider as described in figure 7, depending on the nature and the extent of the services. Implementing the storage application within external IP server allows IP service providers to offer advanced operator storage services to user premises, which are currently offered by the PSTN service provider, for both the PSTN and the Internet Voice Gateway lines. Such services may include voice message storage facilities which can integrate the incoming messages from the PSTN and from the Internet Voice Gateway lines in a single voice mailbox. In this case, the manner of operating the retrieval of incoming messages is illustrated in Figure 8. Having new messages stored on the server will be indicated by a message tone, which substitutes for the standard dial tone. The user then chooses whether to retrieve the messages. If the user gives the request to retrieve the messages, the request is forwarded to the call service server, which sends the messages one at a time to the said system. The Base Unit 11 then distinguishes between voice and fax messages, and forwards them to the appropriate adapters.

The said system can also provide users the bi-directional access to digital text messages, such as e-mails and short messages service (SMS), via the analog communication facilities on the premises. Equipping the Adapter Units with a built-in display enables them to display incoming text messages. The telephone's keypad can be utilized to compose outgoing messages.

Following is a further embodiment as suggested by the present invention enabling the transference of SMS messages. The flow of this process is illustrated in a flow chart in figure 4: The SMS server 19 is a computer connected to the WAN 1 and to the cellular network 20. The SMS server 19 may receive from the cellular network 20 an SMS message destined to one of the Adapters 8 on the user's premises. The SMS server 19 then forwards the said SMS message to the computer 12 over the WAN 1 , through the ISP POP 3, the data communications link 15 and the data communication device 14.

The computer 12 then forwards the said SMS message to the Base Unit 11 over the digital link 13. Base Unit 11 detects the specific adapter unique address and transmits control signals encoding the said SMS message text over the telephone line 9 to the specific Adapter 8 connected to said telephone line. It is important to note that all telephones or fax machines connected to telephone line 9 are in on-hook state and do not draw any current from the PSTN 4. Therefore, no part of the PSTN is "aware" of these operations or effected by them.

A specific address uniquely identifying the destination Adapter 8 is also encoded in the said control signals. All Adapters on telephone line 9 decode the address from the said control signals, but only the said destination Adapter finds it matching to a unique address stored in its internal memory, therefore only the said Adapter 8 decodes the said SMS message text from the signal. The SMS message text then appears on a display unit built into the Adapter 8. All other Adapters on telephone line 9 ignore the said control signal and return to idle state.

Following is the detailed description of the embodiment of the Base Unit as shown in Figure 5:

The telephone line 9 is connected to the Base Unit 11 through the Tip and Ring ports 21. The Isolation and Gain Control circuits 22 control the voltage surges and match the signal level on the telephone line 9. The Hybrid and Filters circuit 23, which operate by instructions of the Controller 25, separates the incoming and outgoing signals on the two wires of the telephone line 9 to distinct signal paths and perform echo cancellation and other needed signal filtration functions.

During a digital call, the Codec 24 periodically performs the analog-to- digital (A/D) and digital-to-analog (D/A) signal conversion functions, under instructions of the Controller 25. The voice signals flowing from the Hybrid and Filters circuit 23 is sampled by the Codec 24 and transferred to the computer 12 through a digital link 13. In the opposite direction, digital signals flowing from the computer 12 through the digital link 13 are converted to analog signals by the Codec 24. The signals bi-directional flow occurs simultaneously.

An Off-Hook Detector 26 is connected to the telephone line 9 ports. When a digital call request is communicated to the Base Unit 11 , the Controller 25 checks the state of the Off-Hook Detector 26 that senses whether the telephone line 9 is being used. If said line is not being used, the Controller 25 instructs the Tone Generator 27 to transmit control signals, encoding a ring command, over the telephone line 9 to all Adapters 8. The said control signals are encoded in the 0-4KHz frequency band, for example by industry-standard DTMF tones. The Controller 25 then activates the Tone Detector 28 and waits for an incoming acknowledge signals from a specific Adapter 8. Said control signals are also coded in the 0-4KHz frequency band. During this time, the Controller 25 switches on the High-Pass Filter circuit

29 to reflect high DC impedance to the PSTN line 5 and prevent the Base Unit

11 from drawing DC current from the PSTN 4. Therefore, the PSTN is

"unaware" of the Base Unit 11 existence on the PSTN line 5 and "unaware" of the said signals being transmitted by the Base Unit 11.

If the said telephone line 9 was found to be in use, the Controller 25 emulates a call waiting by instructs the Tone Generator 27 to transmit a periodic audible tone to the telephone line 9, similar to a call waiting alert tone, to prompt the user to answer the incoming digital call. If the user performs hook flash to answer the incoming call during a pre-defined timeout period, the Controller 25 detects this through the Off-Hook Detector 26. The said Controller acknowledges the computer 12, which establishes the digital call with the voice gateway 2. The Controller 25 also instructs the Codec 24 to start converting the data in both directions simultaneously.

The said analog signals present on the telephone line 9 are in the 0-

4KHz frequency band, therefore a traditional PSTN 4 call can not be performed at the same time. Since the High-Pass Filter 29 reflects a high DC impedance and low AC impedance to the telephone line 9, the Base Unit 11 does not draw any DC current from the PSTN 4 and the PSTN is not "aware" of the said voice signals on the telephone line 9.

At any time during the digital call, a PSTN ring signal may be transmitted by the PSTN 4 on the telephone line 9. The Controller 25 periodically samples the state of the Ring Detector circuit 30 to detect said situation. Once the Controller 25 determines presence of said ring signal, it switches the High- Pass Filter circuit 29 to reflect high AC and DC impedance on the PSTN 4.

A single Base Unit 11 may have more then one telephone line 9 port. In such configuration, Voice Channel 1 31 in Figure 5 is duplicated by the number of telephone lines supported by the Base Unit 11.

Base Unit 11 is powered by external DC power source through the DC- DC Unit 33, which feeds the Power Feeding Unit 32.

Figure 6 shows an embodiment of the Adapter Unit; following is the detailed description of the structure of this embodiment:

A telephone set 6 or fax machine 7 is connected to Adapter 8 using a standard two-wire interface 34. An Off-Hook Detector 35 is also connected to the said interface 34. The Off-Hook Detector 35 senses when user picks up the attached telephone set 6 or fax machine 7 and notifies the Controller 36. A Ringer 37 is also connected to the said interface 34. By command of the Controller 36, the said Ringer 37 transmits a PSTN-standard telephone ringing signal to the attached telephone set 6 or fax machine 7, making it sound an audible ring alerting user of the incoming digital call. A Tone Detector 38 is also connected to the said interface 34.

When the Controller 36 is notified by the Off-Hook Detector 35 that a user had picked up the telephone set 6 or fax machine 7, it activates the Tone

Detector 38 which decodes the DTMF digits dialed by the user. If the destination telephone number is identified as a F/VolP destination, it switches on the High-Pass Filter circuit 39. The said High-Pass Filter circuit 39 reflects a high DC impedance and low AC impedance to the PSTN line 5, allowing voice signals to be transmitted and received over the telephone line 9 but preventing the Adapter 8 from drawing any DC current from the PSTN 4. Therefore, the PSTN is "unaware" of the Adapter 8 existence on the PSTN line 5 and of any voice signals present on telephone line 9.

Telephone line 9 is connected to Adapter 8 through another Tip and Ring interface 40. The Isolation and Gain Control circuits 41 control the voltage surges and match the signal level on the telephone line 9. The Filters circuit 42 performs analog signal filtration.

The Tone Generator 43 is used to transmit, by command of the Controller 36, tones in the 0-4KHz frequency band over the telephone line 9 to all Adapters 8 and Base Units 11. The Tone Detector 38 decodes, by request from the Controller 36, incoming tones in the 0-4KHz frequency band, including DTMF tones on the telephone line 9, and transmits them digitally to the Controller 36.

A Ring Detector 38 is connected to the telephone line interface 40 and to the Controller 36. At any time during the digital call, a PSTN ring signal may be transmitted by the PSTN 4 on the telephone line 9. The Controller 36 periodically monitors the state of the Ring Detector circuit 38 to detect such a situation. Once the Controller 36 determines presence of said ring signal, it emulates the operation of a call waiting in the following manner: Controller 36 opens Line Switch 46, isolating the Adapter 8 and the telephone set 6 or fax machine 7 from the telephone line 9 and PSTN 4. The said Controller 36 then instructs the Tone Generator 43 to transmit a periodic audible tone to the attached telephone set 6 or fax machine 7, similar to a call waiting alert tone, to prompt the user to answer the incoming PSTN call.

If the user performs hook flash on the attached telephone set 6 or fax machine 7 during a timeout period, the Controller 36 detects this via the Off- Hook Detector 35. Then Controller 36 instructs the Tone Generator 43 to transmit a control signal to the Base Unit 11 , informing it to put the digital call on hold. The said Controller 36 then switches off the High-Pass Filter circuit 39, reconnecting the Adapter 8 and the attached telephone set 6 or fax machine 7 to the telephone line 34 and the PSTN 4.

An Off-Hook Detector 44 is also connected to the telephone line 34 interface. It is used by the Controller 36 to sense the state of said telephone line 34 and disable the Adapter's 8 outgoing call functionality until the said telephone line 34 is back in on-hook state.

Display Device 47 is connected to the Controller 36, used to display SMS messages received from the Base Unit 11 over the telephone line 9. The said

SMS message text is encoded by signals in the 0-4KHz frequency band, detected by Tone Detector 38 and decoded by the Controller 36. The said

Controller 36 operates the Display Device 47 to make the text visible on it.

The power source for the Adapter 8 circuit may be external, connected through the DC-DC Unit 48, or internal from Rechargeable Battery 49. In case said battery exists, it is charged by Charger 50 feeding off the telephone line 9 power when said line is in off-hook state. The Controller 36 periodically monitors Off-Hook Detector 44 and switches the said Charger 50 on when said line is in off-hook state and off when said line gets back to on-hook state. While the above description contains many specifities, these should not be construed as limitations on the scope of the invention, but rather as exemplifications of the preferred embodiments. Those skilled in the art will envision other possible variations that are within its scope. Accordingly, the scope of the invention should be determined not by the embodiment illustrated, but by the appended claims and their legal equivalents.

Claims

What is claimed is:

1. a communication system for enabling an internal telephone system which uses a single phone line to establish both digital voice communication over data communication (IP) network through voice gateway and analog voice communication over PSTN network, wherein the digital voice communication can be established using any standard telephone unit connected to the PSTN network through internal phone wiring, said digital communication not interfering with the PSTN network communication , said system comprised of following modules:

- interface communication module, enabling communication with the IP network voice gateway;

- processing module, enabling to establish and manage digital communication with the IP network;

- Management module, enabling to identify VoIP incoming call, perform signal conversion of digital to analog or analog to digital, and to enable communication with any system functions through the same internal twisted-pair phone wiring, using standard signaling range, () wherein an on-hook state signal is transmitted to the PSTN network throughout the digital call interval.

- telephone set interface module each having a unique ID, enabling to identify and divert internal or external call wherein an on-hook state signal is transmitted to the PSTN network throughout a digital call interval.

2. The communication system of claim 1 wherein an IVR system implemented within the PSTN environment is utilized for enabling to initiate a digital voice call by a standard telephone unit and wherein the voice gateway identifies the destined IP address of the incoming analog call and converts the analog call signal to digital data.

3. The communication system of claim 1 wherein the telephone set interface module identifies the destined IP address of the initiated analog call of the telephone system and wherein the voice gateway initiates an analog call signal to the PSTN system and converts received analog call signals to digital data.

4. The communication system of claim 1 wherein the e-mail message are exchanged between any IP terminal node and the MANAGEMENT module over the IP network.

5. The communication system of claim 1 wherein the IP network has communication with an SMS sever enabling communication with cellular phone by SMS text messages through IP network

6. The communication system of claim 1 further enables direct text messages between the modules wherein the TELEPHONE SET INTERFACE module includes a display enabling the user to write text through DTMF tones typed on the phone keypad.

7. The communication system of claim 1 wherein the text message are email message transmitted through the data communication network.

8. The communication system of claim 1 wherein the text messages are SMS messages transmitted through the data communication network and respective cellular network.

9. The communication system of claim 1 wherein the Telephone Set Interface module keeps the phone extension units in on hook state during VoIP call session for isolating the phone extension unit from the PSTN network exchange

10. The communication system of claim 1 wherein the Telephone Set Interface module is programmed to convert an analog dialup number to an IP address according to predefined list.

11.The communication system of claim Error! Reference source not found. wherein the modules are capable of performing voice calls with each other wherein all communications between the modules are carried out through the twisted pair wiring of the internal phone system, wherein no current is being drawn from the PSTN (so the PSTN in "unaware" of this communication)not interfering with the PSTN network communication

12. The communication system of claim 1 wherein all modules enable waiting call services for voice communication message.

13. The communication system of claim 1 wherein the internal telephone system further includes a fax unit including or connected to respective telephone set interface module enabling to receive and send fax through the digital data communication network.

14. The communication system of claim 1 for enabling to provide an advanced storage services to the internal telephone system, further comprised of Storage unit enabling to manage data or audio digital mailbox.

15. The communication system of claim 14 wherein the storage unit is implemented within any of the systems' modules.

16. The communication system of claim 14 wherein the storage unit is implemented within a device on the data network.

17. The communication system of claim 14 wherein the storage unit is implemented within the voice gateway. 18. A communication system for enabling to provide an advanced storage services to local telephone system including a single phone line over IP network through voice gateway and analog voice communication over

PSTN network, said system comprised of:

- Interface Communication Module enabling to communicate with the IP network voice gateway;

- processing module enabling to establish and manage digital communication with the IP network.

- a Manamgemnet Module enabling to identify type of communication, perform signal conversion of digital to analog or analog to digital according to identified communication type;

9. A communication method for enabling an digital voice communication over data communication (IP) network through voice gateway, between two analog telephone units which are connected to the PSTN network, each of them using a single phone line, wherein at least one telephone unit is connected to Interface Communication Module, processing module Management Module and Telephone Set Interface Module, said method comprising the steps of:

- detecting dialup analog signal;

- identifying destination IP address of the dialed analog number;

- converting digital signal to analog signal

- establishing and managing digital communication through an IP network using voice gateway;

- identify VoIP incoming call at the destination side;

- performing signal conversion of digital to analog (voice)

- diverting external or internal call between system modules through the same twisted-pair wiring which is connected to the PSTN, using standard signaling range

- transmitting on-hook state signals to the PSTN network

20. The method of claim 19 wherein the telephone unit initiates the digital voice communication the destination IP address is identified according to pre-defined list and Management module establishes the IP call using the IP communication interface.

21. The method of claim 19 wherein the standard telephone unit initiates the digital voice communication the destination IP address is entered by the user utilizing an IVR system which enable to divert the call from the PSTN network to the voice gateway.

22. The method of claim 19 wherein the telephone unit is further connected to at least one telephone unit through twisted pair wiring wherein each additional telephone unit can be connected to a Telephone Set Interface module , each having a unique identification, said method further comprised the steps of: diverting the call to and from the telephone units according to their unique identification, creating ring tone at the destination telephone unit, connecting said telephone unit to the IP call once the hook on operation is detected and keeping the telephone units on hook state through the digital voice communication

23. The method of claim 19 further comprises the steps of: creating a new call waiting signal while a digital voice communication takes place and switching between two call when identifying hook flash operation.

24. The method of claim 19 wherein during the voice communication session the high DC impedance is reflected to the PSTN line preventing the Interface Modules from drawing current from the PSTN line.

25. The method of claim 19 wherein at least one fax unit can be connected to the Telephone Set Interface module through the twisted pair wiring , said method further comprises the steps of identifying fax signal and diverting the call the fax unit according to the unique identification of the fax Telephone Set Interface module wherein the fax data is transmitted through the data network.

26. The method of claim 19 wherein the Telephone Set Interface module is equipped with a screen, said method further includes the steps of: creating an SMS message by typing on the phone unit or displaying received SMS message on the Telephone Set Interface module screen and diverting the message through cellular network to destination.

27. The method of claim 19 wherein the Telephone Set Interface module is equipped with a screen, said method further includes the steps of: creating an E-mail message by typing on the phone unit or displaying received E- mail message on the Telephone Set Interface module screen and diverting the message through an IP network to destination.

28. The method of claim 19 further comprises the steps of: storing voice messages received through the data communication network, managing a voice mailbox for users and retrieving the messages upon request.