KR100509945B1 - The System and Method for providing the Internet Phone Service in the Home Gateway System - Google Patents

Abstract

The present invention relates to a H.323-based Internet phone system using a regular telephone in a home gateway and a service method thereof. The system provides a user interface for a telephone, an on / off hook event and a DTMF signal from a telephone. Control the ring, the user interface to switch the voice call network to the IP network or PSTN network, the call management unit for call signaling and control functions for call connection and disconnection, and when the connection is established, send and receive voice data Internet phone service is controlled by monitoring related information and controlling call manager and voice processor for H.323 support and user requirements that support H.323 protocol for voice processor, call manager and voice processor responsible for voice data processing. It characterized in that it comprises a system control unit to perform.

According to the present invention, in the home gateway, the user can make a voice call with the other party through the IP network using a general telephone. In addition, when there is a lot of traffic on the IP network, the user may change the type of connection network for voice transmission to PSTN. Therefore, the user has the effect of providing transparency to the other party's location or network.

Description

Home Gateway Internet Phone System and its service method

The present invention relates to an Internet phone service, and more particularly, to an H.323-based Internet phone system using a general telephone in a home gateway and a service method thereof.

Recently, VoIP (Voice over IP) technology, which is attracting attention from service providers, Internet service providers, industries, and general users, is emerging as the largest application service of the Internet. VoIP is a technology for transmitting voice using the IP layer of the Internet. The voice signal is converted into a digital signal and then composed of packets to transmit voice signals through the Internet, an IP network.

VoIP-based internet telephony (IP telephony) has been recognized as a service to supplement or replace the existing telephone service through public switched networks with the initial commercialization of VocalTec's InternetPHONE software in early 1995. . In addition, since the Internet phone uses the Internet network instead of a dedicated line for phone calls, international calls have a great advantage of being about 50% to 80% of the current rate. The company has already entered the market with great interest and is actively considering participation.

Internet telephony, which is recognized as a supplement to existing voice telephony on the public switched telephone network (PSTN) or as a future alternative, can be used to access corporate local area networks (WLANs), wide area networks (WANs), private intranets, or the public Internet. This includes sending standard circuit-switched voice conversations (phone-to-phone) over any form of packet-switched IP-based data network. In addition to phone-to-phone access, Internet telephony is a PC-to-phone that supports applications on IP networks such as relay bypass, real-time, cumulative delivery faxing, message processing integration, web-based call centers, and IP conferencing. PC, and PC-to-telephone connection. The core technologies of Internet telephony can be divided into speech coding and compression technology, real-time data transmission technology, lost packet recovery technology, and gateway technology. Here, the gateway technology refers to a technology that allows two or more different types or types of communication networks to be interconnected to exchange information between communication networks, and includes a LAN, a public data network (PDN), and a general telephone switching network (PSTN). Etc. are included.

On the other hand, the protocols of the public public telephone network and the Internet network have different characteristics. In other words, the real time response is very good because the two terminals are assigned part of the exchange for communication and occupy until the end of the communication.However, the Internet protocol is a packet-by-packet so that the packet can be delivered to the destination through multiple transmission networks. Because path selection is independent, packets sent first may arrive later at the receiver than packets sent later and may even be ignored or discarded during delivery. Sending connection-oriented public network signals that must be reliable to these connectionless-oriented Internets requires a set of protocols and devices.

The protocols commonly used for VoIP implementations include the Session Initiation Protocol (SIP) of H.323 and the Internet Engineering Task Force (IETF) proposed by the most widely used International Telecommunications Union-Telecommunications standardization sector (ITU-T). Media Gateway Control Protocol (MGCP), Media Gateway Control (MEGACO), Signaling Transport (SIGTRAN), and the like. Most equipment and service providers now support the H.323 protocol.

The H.323 protocol is a series of audio conferencing and video conferencing standards that can support multimedia over non-QoS, IP-based data networks, and is designed to provide real-time multimedia communications in data networks. will be.

It includes several encoding / decoding algorithms that are essential for circuit-packet conversion and speech compression processing at the Internet telephony gateway. Several major vendors have already adopted H.323 as their Internet telephony standard, including companies leading personal computer client applications such as Microsoft, Netscape, and Intel. In addition, the Voice over IP (VoIP) Forum ) And its parent organization, the International Multimedia Teleconferencing Consortium (IMTC), use the ITU Recommendation G.723.1 algorithm to compress and restore VoIP. The problem is that many Internet telephony gateway vendors adopt their own encoding / decoding algorithms because their proprietary technology can achieve better gateway voice quality and transmission delay performance.

The H.323 standard, adopted by the ITU in 1996, comprehensively covers audio, video and data transmitted over packet-based networks such as local area networks, intranets, extranets and the Internet. Multimedia products and applications from numerous manufacturers have been developed to interoperate. In particular, the H.323 standard has had a profound impact on shaping new areas of cooperative LAN-based applications for multimedia communications, including point-to-point and multipoint video conferencing. It also includes stand-alone equipment and PC technology.

The most important thing in providing such Internet telephony service is call voice quality. The voice quality of Internet telephony was significantly poorer than that of early Public Switched Telephone Network (PSTN) telephones, but has been significantly improved due to recent advances in Internet telephony software development and improvements in computer speed and compression methods.

However, since voice data is transmitted through the IP network, quality deterioration due to factors such as jitter, delay, and packet loss remains a problem to be solved. In other words, jitter is a distribution of packet arrival interval times, which exists only in packet-based networks. When jitter occurs on a VoIP-enabled phone, when a receiver plays a packet, a packet that arrives late cannot be played. Delays are not a problem when making calls on the PSTN, but delays such as propagation delays, serialization delays, handling delays, and access delays are inevitable in the Internet, a packet network. In addition, packet loss on the network is common and easily predicted. Most protocols use packet loss to reduce the number of packets by knowing the state of the network. Loss of voice packet in VoIP, especially if packet loss occurs continuously or packet loss occurs frequently, call quality drops drastically.

In addition, from the user's point of view, the purchase price of the Internet phone terminal supporting the VoIP service is another economic burden for the consumer, and the process of initializing the terminal is relatively easy. Therefore, user convenience is recognized as an essential element for providing services.

SUMMARY OF THE INVENTION The present invention has been made to solve the above problems, and provides a H.323-based Internet phone service system and an Internet phone service method using a general telephone in a home gateway.

In order to achieve the above technical problem, the home gateway internet phone system according to the present invention provides a user interface for a general telephone, controls an On / Off hook event and DTMF signaling from a telephone, and connects a voice call network to an IP network. Or a user interface unit for switching to a PSTN network, a call management unit for call signaling and control functions for call connection and disconnection, and when a connection is established through the call management unit, transmitting and receiving voice data and monitoring information related thereto. The call manager and the call manager and the call manager and the call manager and the call manager and the call manager to System that performs internet phone service function by controlling voice processing unit It characterized in that it comprises a control unit.

The Internet phone system further includes a RAS management unit for a function for communicating with a gatekeeper connected to the Internet, wherein the H.323 support unit is one of VoIP protocols for the call management unit, voice processing unit, and RAS management unit. 323 protocol is supported, and the system controller controls the call manager, the voice processor, and the RAS manager according to the user's requirements input through the user interface.

The user interface unit may include a user event unit for processing an action from a user and a network event unit for processing an action from a network. The user event unit may be a ringtone if the user is a receiver when the user action is Off-Hook. The off-hook processing unit for generating a dial tone when the sender is stopped, the on-hook processing unit for going to the initialization state when the user action is On-Hook, and the DTMF value of the input keypad when the user action is pressing the keypad. According to the connection to the PSTN network, through the gatekeeper and the keypad to determine the connection to the IP network is made, the network event unit is a call setup, if not On-Hook If the user is busy system A call to notify the control and, if on-hook, to send a ringtone to the user and act as a receiver A setup processing unit; And in the case of call reject / release, it comprises a call release processing unit for sending a busy tone to the user.

The RAS management unit requests communication with the gatekeeper and receives a permission to register with the gatekeeper. If the registration is registered through the registration request, the RAS management unit notifies the system controller of the registration completion and the call connection procedure is performed by the system controller. Is implemented, the event receiving unit waiting for the event and the event is the call connection start, call release, bandwidth change, if the registration of any one of the event to send a corresponding message to terminate the session or wait for the event again do.

The voice processing unit appropriately stores the voice and surplus information contained in the packet transmitted from the network in the playback buffer, and controls the flow of the voice data in the system and the voice data processing unit for transferring the voice data input from the user to the system. And it characterized in that it comprises a voice data management unit to manage.

In order to achieve the above technical problem, the home gateway internet phone network system according to the present invention is connected to a general telephone, provides a user interface for the general telephone, and connects a home network and an external network to a voice call connection service to a user. It is a home gateway that provides a phone, a VoIP gateway that connects a general telephone network to which the phone is connected to the Internet, and exchanges information with each other, and a gatekeeper that changes the call connection network or provides RAS functions according to a user's selection. Characterized by including.

Home gateway Internet phone service method according to the present invention in order to achieve the above another technical problem, the step of waiting for an event for a voice call from the user or the network, by performing the H.323 protocol procedure to establish a voice call channel connection between the voice A channel connection step of enabling a call, a step of making a voice call, and a channel release step of performing a voice call end procedure when the voice call ends, and the channel connection step includes a user when the event is Off-Hook. If the receiver is called, the ringtone is stopped, and if it is the sender, and if the user action is to press the keypad, the connection to the PSTN network, the connection through the gatekeeper, and the IP network are performed according to the DTMF value of the input keypad. The channel disconnecting step, the user puts down the handset. Detects On-Hook, notifies that the call is terminated, waits for the initial mode, and receives the end of the call, generates a busy tone to the other party, and notifies the On-Hook event when the other party puts down the handset. After the shutdown is completed, including the initial standby mode.

In the channel connection step, the connection through the gatekeeper requests communication with the gatekeeper and requests registration with the gatekeeper when the permission is granted. When the registration request is registered, the system control module is notified of the completion of registration. Waiting for an event when a call connection procedure is executed by the control module, and sending a corresponding message and ending a session or waiting for an event again if the event is any one of start call release, call release, bandwidth change, or deregistration. It is characterized by including.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings. 1 is a block diagram of an Internet phone service network system for providing an Internet phone service in a home gateway according to the present invention. The home gateway 100 is connected to the general telephone 110 and provides a user interface for the general telephone, and provides a voice call connection service to the user by interconnecting the home network and the external network, in particular, the Internet 120.

The VoIP gateway 130 exchanges information with each other by connecting the general telephone network 140 and the Internet 120 to which the telephone is connected.

In particular, the gate keeper (Gate Keeper, 150) is connected to the Internet network 120 to change the call connection network or provide the RAS function by the user's selection, the home gateway 100, VoIP gateway 130 and VoIP application terminal ( 160) and it is in charge of address translation, user authentication and bandwidth control through the exchange of information with them. Referring to FIG. 1, a user may make a voice call with a user in another home gateway 170 using the general telephone 110 connected to the home gateway 100. In addition, users connected to the home gateway can also make voice calls with PSTN users.

FIG. 2 is a block diagram showing the configuration of an Internet Phone Service (VoIP) system in a home gateway according to an exemplary embodiment of the present invention. The user interface unit 241, the call manager 221, and voice are shown in FIG. Voice Data Processing (251), Voice Data Manager (VoiceData Manager, 231), H.323 Support (H.323 Protocol, 261), RAS Manager (RAS 211) and System Control (System Control, 201) It is made to include.

The user interface unit 241 provides a user interface for a general telephone, controls on / off hook events and DTMF signaling from a telephone, and switches a voice call connection network to an IP network or a PSTN network.

3 shows the user interface unit 241 in more detail as a functional block, and includes a user event unit 300 for processing an action from a user and a network event unit 350 for processing an action from a network. It is composed.

The user event unit 300 includes an off-hook processor 310, an on-hook processor 320, and a keypad processor 330. When the user action is Off-Hook, the off-hook processor 310 stops the ring tone when the user is a receiver and generates a dial tone when the sender is the sender. The on-hook processing unit 320 goes to the initialization state when the user action is on-hook. When the user action is to press the keypad, the keypad processing unit 330 determines the connection to the PSTN network, the connection through the gatekeeper, and the connection to the IP network according to the DTMF value of the input keypad.

The network event unit 350 includes a call setup processor 360 and a call release processor 370. The call set-up processing unit 360 transmits a ring tone to the user and functions as a receiver if the call set-up is On-Hook, and notifies the system controller that the user is Busy if not On-Hook. The call release processing unit 370 sends a busy tone to the user in case of call reject / release. A more detailed operation description of the user interface unit 241 shown in FIG. 3 will be described below with reference to FIGS. 7A and 7B.

The call manager 221 is responsible for call signaling and control functions for call connection and disconnection.

The voice data processor 251 and the voice data manager 231 constitute a voice processor. When the connection is established through the call manager 221, the voice data processor 251 transmits and receives voice data (253, 254). It monitors information (252) and is responsible for processing voice data.

The voice data processing unit 251 suitably stores the voice and surplus information contained in the packet transmitted from the network in the reproduction buffer, and delivers the voice data input from the user to the system.

The voice data manager 231 controls and manages the flow of voice data in the system.

The H.323 support unit 261 supports the H.323 protocol, which is one of VoIP protocols, for the call manager 221, the voice data manager 231, and the RAS manager 211.

The RAS management unit 211 is responsible for the communication with the gatekeeper connected to the Internet. 4 illustrates the RAS management unit 211 in more detail as a functional block, and includes a registration request unit 400, an event receiver 410, and an event processor 420. The registration request unit 400 requests communication with the gatekeeper and requests registration with the gatekeeper when the permission is received. When the event receiving unit 410 is registered through the registration request, the event controller notifies the completion of registration, and waits for an event when a call connection procedure is executed by the system controller. The event processing unit 420 sends a corresponding message and terminates the session or waits for the event again when the event is any one of a call connection start, call release, bandwidth change, and registration cancellation. A more detailed operation description of the RAS manager 211 shown in FIG. 4 will be described below with reference to FIGS. 8A and 8B.

The system control unit 201 is responsible for the overall control of the Internet phone service function. That is, the call manager 221, the voice data manager 231, and the RAS manager 211 perform the Internet phone service function on the user's requirements input through the user interface 241.

5 is a flowchart illustrating a voice call connection procedure between home gateway users according to an exemplary embodiment of the present invention. First, when user 1 picks up the handset to make a call (500), the user interface (UserInteraction) in the HG-VoIP to which user 1 is connected detects an off-hook and sends a ring back tone to the user's handset (502). The user 1 inputs the IP address of the user 2 using the keypad. (504) In the user interface of the user 1, the system control unit converts the DTMF (Dual Tone MultiFrequency) signal input from the keypad into an IP string. In step 506, using this address, System Control establishes a TCP connection for H.225.0-Q.931 message exchange. After the TCP connection is established, the H.225.0-Q.931 signaling procedure is performed. User 1's System Control sends a Setup message containing CID = N and conferenceGoal = create information. (508) System 2's SystemControl receives User 1's System Control. Alerting or Call Proceeding messages are sent to the user (512). At this time, User 2's phone rings (510), and when User 2 answers the phone (514), it sends a Connect message to User 1 (516). User 1 who receives the ezTCP establishes a TCP connection for H.245 message exchange with User 2 using the IP address and port number included in the Connect message, and then performs the H.245 control procedure.

First, TerminalCapabilitySet and TerminalCapabilitySetAck messages are exchanged for capability exchange negotiation. It will open a real-time transport protocol / RTP control protocol (RTP) channel for the host (556).

6 illustrates a voice call release procedure between home gateway users according to an exemplary embodiment of the present invention. When the user ends the call, the user hangs up the handset. Referring to FIG. 6, when user 1 puts down the handset (600), the system control unit (SystemControl) detects an on-hook (605) and sends a CloseLogicalChannel message (610). Upon receiving the message, it detects that the user 1 has terminated the call, and transmits the CloseLogicalChannelAck message to the HG-VoIP of the user 1 (625) and notifies the user interface of the user 2 of the user 2 that the call connection is terminated. 615) User 2's user interface (UserInteraction) generates a busy tone to the user 2's phone (620), and when the user 2 puts down the handset, and notifies the System Control (SystemControl) On-Hook event (630). After receiving the response message, SystemControl of User 1 sends the EndSessionCommand message again (635) and after receiving the response (EndSessionCommandAck) (640). The protocol waits in the initial mode by sending the ReleaseComplete message, which is the protocol release message. (645) In addition, after receiving the message, the system control unit of User 2 also terminates all connection states and is in the initial standby mode again.

7A and 7B are flowcharts illustrating operations of a user interface unit (UserInteraction) according to an exemplary embodiment of the present invention. This module manages and controls all events related to user interaction, and is responsible for sending and receiving voice data to and from the phone interface. This module starts when the home gateway internet phone service application is initialized. First, after initializing various file descriptors and pipe variables to be used and environment values for using a telephony interface (TI) device (702), an event from a user or a network is waited for (703).

When the event occurs from the user, when the user picks up the handset (off-hook), the user first determines whether the user is a sender or a receiver. (710) If the user is the sender, dialtone is sent to the handset (712). In case of the receiver, ringtone generation is stopped (711). The sender enters the address or telephone number of the other party using the keypad of the telephone, whether or not to use the Registration, Administration, and Status (RAS) function. If a dial tone occurs when using the keypad, stop it first and accept the keypad value (713) and finally press the (#) button to indicate that the input is completed (730). ) Converts the input DTMF signal to an ASCII code value (714,715) and checks the value (731). First, if the value is " 009 # ", the system control unit (SystemControl) is informed that the user has selected the call connection model through the gate keeper (734), and then goes back to the standby mode (7_3). If the value is "*** #", it means that the user wants to use the general telephone network. Therefore, the call connection mode is switched to the PSTN (732), and then terminated (733). After determining whether the input value is the correct value (735), if the correct value is passed to the TCPClient module (736), otherwise notifying the user with an error message (737), then waits for user input again.

If the detected event is from the network, first check the received message type (720). If the received message is Call Setup, if the telephone is not placed and is in use, the user is using the SystemControl. If the phone is placed, the phone generates a ring tone (721) and sets the user to receive mode (722). If the message is Call reject or Call release, the user generates a busy tone. 723, the user puts down the phone and enters the initial mode 7_1 again.

8A and 8B are flowcharts illustrating operations of a RAS manager according to an exemplary embodiment of the present invention.

This module is a module that operates when connecting through a gatekeeper during call connection with reference to FIGS. 5 and 6, and execution is determined by user selection as shown in FIGS. 7A and 7B (734). If not in operation, as shown in Figures 5 and 6, the call service is possible as a direct connection model between the callers. However, in this case, there is a limitation that the IP address of the other terminal or the VoIP gateway or home gateway to which the other party belongs is known in advance. Therefore, when operating this module, it has the advantage that the call can be connected to the other party's name or phone number. First, the RAS module sends a gatekeeper request (GRQ) message requesting communication with the gatekeeper to the corresponding gatekeeper set in advance (801). The response to the GRQ message is either a Gatekeeper Conform (GCF) that accepts the request or a Gatekeeper Reject (GRJ) that rejects the request.

If the GRJ is received or the response waiting time is exceeded, the corresponding error message is sent and then terminated (805, 806). Otherwise, if the received message is a GCF, it sends a RRQ (Register Request) message for registration to the gatekeeper (803) and waits for a response again. The response to the RRQ (Register Request) message is either a RegistrationConform (RCF) that permits registration or a RegistrationReject (RRJ) that refuses registration. If the RRJ is received or the response waiting time is exceeded, the corresponding error message is sent and then terminated (808 and 809). If not, that is, if the received message is the RCF, the system control unit (SystemControl) notifies the completion of registration (807).

If a procedure for call connection is implemented by the SystermControl, it waits for an event for the next procedure (811). If the call setup event is received, an ARQ message is sent (812). DRQ (Disengage Request) 813 for an event for call release, BRQ (Bandwidth Request) 814 for bandwidth change (Bandwidth_Info), and URQ (Unregistration) for unregistration Create and send a message (815). After doing this, it waits for a response to each message (821). If a DCF (Disengage Confirm) or DRJ (Disengage Reject) is received from the response message, the session is terminated. Otherwise, the event is received from the system controller again. (822,823,824).

9A is a flowchart illustrating an operation of a voice transmitter (VoiceSend module), which is one of voice processing units according to an exemplary embodiment of the present invention. As shown in FIG. 2 and FIG. 9A, the voice transmitter (VoiceSend module) is one of the sub-modules of the voice processor, and receives the encoded voice data from the telephony device driver (TI_DeviceDriver) (903) RTP packet. It performs the function of making and transmitting the data (904, 905, 906). At this time, the amount of surplus information is determined according to the reception status of the current receiver. If the reception status is bad, surplus information is created (907, 908) and is transmitted in the Extension field of the currently generated RTP packet (906). . This process is repeated until the session ends.

9B is a flowchart illustrating an operation of a voice receiver (VoiceReceive module), which is one of voice processing units according to a preferred embodiment of the present invention. As shown in FIG. 2 and FIG. 9B, the voice receiving unit (VoiceReceive module) is one of the sub-modules of the voice processing unit, and after copying the voice data 953 received from the RTP channel to the play buffer. In operation 963 and 962, the terminal transmits a function to the telephony device driver TI_DeviceDriver in operation 963. At this time, if the received packet is RTP, it checks whether there is surplus information, checks whether there is voice data corresponding to the information in the playback buffer, and if the buffer is empty, the surplus information is copied to the corresponding block of the playback buffer. . Otherwise, only the received RTP payload part is copied to the corresponding block of the playback buffer and transmitted to the telephony device driver. If the received packet is RTCP, the type of the RTCP packet is checked, and if the type is BYE, the session is terminated (972). Otherwise, information on the current reception state included in the packet is collected (971). ).

The invention can also be embodied as computer readable code on a computer readable recording medium. The computer-readable recording medium includes all kinds of recording devices in which data that can be read by a computer system is stored. Examples of computer-readable recording media include ROM, RAM, CD-ROM, magnetic tape, floppy disk, optical data storage, and the like, and may also be implemented in the form of a carrier wave (for example, transmission over the Internet). Include. The computer readable recording medium can also be distributed over network coupled computer systems so that the computer readable code is stored and executed in a distributed fashion.

The above detailed description and contents posted in the drawings are not intended to limit the present invention, and various modifications or changes may be made by those skilled in the art without departing from the spirit of the present invention. Possible is obvious.

According to the present invention described above, the user within the home gateway can make a voice call with the other party through the IP network using a normal telephone. At this time, the target of the call may be not only a user in another home gateway, but also a user of an Internet telephony application service terminal or a user of a PSTN network. In addition, since the call quality is poor when there is a lot of traffic on the IP network, in this case, the user can change the type of connection network for voice transmission to PSTN. Therefore, the user has the effect of providing transparency to the other party's location or network. In addition, the loss packet recovery method using buffering and surplus data is used to minimize the degradation of voice quality, which is a disadvantage of the existing VoIP technology, thereby providing a better call quality to the user.

1 is a block diagram of an Internet phone service network system for providing an Internet phone service in a home gateway according to the present invention.

2 is a block diagram illustrating a configuration of an Internet phone service (VoIP) system in a home gateway according to an exemplary embodiment of the present invention.

3 illustrates the user interface unit in more detail as a function-specific block.

4 shows the RAS management unit in more detail as a function-specific block.

5 is a flowchart illustrating a voice call connection procedure between home gateway users according to an exemplary embodiment of the present invention.

6 illustrates a voice call release procedure between home gateway users according to an exemplary embodiment of the present invention.

7A and 7B are flowcharts illustrating operations of a user interface unit according to an exemplary embodiment of the present invention.

8A and 8B are flowcharts illustrating operations of a RAS managing unit according to an exemplary embodiment of the present invention.

9A is a flowchart illustrating an operation of a voice transmitter, which is one of voice processors according to an exemplary embodiment of the present invention.

9B is a flowchart illustrating an operation of a voice receiver, which is one of voice processors according to an exemplary embodiment of the present invention.

Claims (8)

A user interface unit for providing a user interface for a general telephone, controlling on / off hook events and DTMF signaling from a telephone, and switching a voice call connection network to an IP network or a PSTN network; A call manager in charge of call signaling and control for call connection and disconnection; When a connection is established through the call manager, a voice processor for transmitting and receiving voice data, monitoring information related to the voice manager, and handling voice data; An H.323 support unit supporting the H.323 protocol, which is one of VoIP protocols, to the call manager and the voice processor; And And a system controller configured to control the call manager and the voice processor according to the user's requirements input through the user interface to perform an internet phone service function. The method of claim 1, Further provided with a RAS management unit for the function for communication with the gatekeeper connected to the Internet, The H.323 support unit Supports the H.323 protocol, one of the VoIP protocols for the call management unit, voice processing unit and RAS management unit, The system control unit Home gateway Internet phone system characterized in that to perform the Internet phone service function by controlling the call management unit, voice processing unit and RAS management unit for the user's requirements input through the user interface unit. The method of claim 2, wherein the user interface unit A user event unit which processes an action from a user; And Characterized in that it comprises a network event unit for handling actions from the network, The user event unit An off-hook processing unit for stopping a ring tone if the user is a receiver and generating a dial tone if the user is an off-hook; If the user action is On-Hook, On-Hook processing unit to go to the initialization state; And If the user action is to press the keypad, and comprises a keypad processing unit for determining the connection to the PSTN network, the connection through the gatekeeper and the IP network according to the DTMF value of the input keypad, The network event unit In the case of call setup, if not On-Hook call the system control unit that the user is busy state, if On-Hook call set-up processing unit to send a ringtone to the user to function as a receiver; And If the call reject / release, the home gateway Internet phone system comprising a call release processing unit for sending a busy tone to the user. The method of claim 2, wherein the RAS management unit Registration request section that requests registration with the gatekeeper if communication is requested with the gatekeeper: An event receiving unit notifying the completion of registration to the system controller when the registration is made through the registration request, and waiting for an event when a call connection procedure is executed by the system controller; And If the event is any one of a call connection start, call release, bandwidth change, deregistration, the home gateway Internet phone system comprising an event processing unit for sending a corresponding message and ending the session or waiting for the event again. The voice processing unit of claim 2, wherein the voice processing unit A voice data processing unit suitable for storing voice and surplus information contained in packets transmitted from the network in a reproducing buffer, and transferring voice data input from the user to the system; And Home gateway Internet phone system comprising a voice data management unit for controlling and managing the flow of voice data in the system. A home gateway connected to a general telephone and providing a user interface for the general telephone, and providing a voice call connection service to a user by interconnecting a home network and an external network; A VoIP gateway interconnecting a general telephone network to which telephones are connected and the Internet to exchange information with each other; And A home gateway internet phone network system, comprising: a gatekeeper connected to the internet network to change a call network according to a user's selection or provide a RAS function. Waiting for an event for a voice call from a user or a network; A channel connection step of enabling a voice call between users by performing a voice call channel connection by performing an H.323 protocol procedure; Making a voice call; After the voice call is characterized in that it comprises a channel release step of performing a voice call termination procedure, Channel connection step If the event is Off-Hook, stopping the ringtone if the user is a receiver and generating a dialtone if the sender; And If the user action is to press the keypad, the step of connecting to the PSTN network, through the gatekeeper and to the IP network according to the DTMF value of the input keypad, The channel release step Detecting a On-Hook when the user puts down the handset, notifies that the call is terminated, and waits for the initial mode; And The home gateway Internet phone service method comprising the step of generating a busy tone to the other party when the end of the call connection is received, the On-hook event when the other party puts down the handset, and terminates all the connection state and goes to the initial standby mode. The method of claim 7, wherein the connection through the gatekeeper in the channel connection step is Requiring communication with the gatekeeper and requesting registration with the gatekeeper if allowed. Notifying the completion of registration to the system control module when the registration is registered in the registration request, and waiting for an event when a call connection procedure is executed by the system control module; And The home gateway Internet phone service method comprising the step of sending a corresponding message and ending the session or waiting for the event again if the event is any one of a call connection start, call release, bandwidth change, and deregistration.