KR100679806B1 - IP voice service system for voice over IP and method thereof - Google Patents

Abstract

The present invention provides IP voice for voice service over Internet Protocol (IP), which enables telephone service by transmitting voice using data packets in an IP network (eg, Internet, intranet, etc.) instead of providing voice service in a telephone network. The present invention relates to a service apparatus and a method, and has an advantage of providing a user with a low cost efficient voice service by enabling a voice service using a packet over an IP network using VoIP technology.

Description

IP voice service system for voice over IP and method

1 is a system configuration using the VoIP service according to the present invention.

2 is a hardware configuration diagram according to the present invention.

Figure 3 is a system protocol stack and hierarchy diagram according to the present invention.

4 is a detailed configuration diagram of a voice processing unit of FIG. 3.

5 is a software structure diagram according to the present invention.

6 is a diagram illustrating a process for providing an IP voice service according to the present invention.

<Description of Symbols for Major Parts of Drawings>

1: home 2: local office

3: voice gateway 4: gatekeeper

10: user interface unit 11: the keypad

12: sound indicator 13: display

14 serial interface 15 microphone

20: voice interface unit 21: pulse code modulation codec

30: user interface processing unit

31: user interface logic 32: serial port device

40: processor core 41: digital signal processor

42: micro control device 43: flash memory

44: Romans 45: Ram

50: network interface unit 51: Ethernet

52: Ethernet Controller 53: RJ-45 Ethernet Connector

60: related logic device

61: oscilloscope and logic device 70: power supply

71: power management unit 80: user interface function

81: display driver 82: keypad driver

83: Sound Indicator Driver

84: user interface processing module 90: voice processing unit

91: PCM interface module 911: tone generation module

92: line echo canceller module 93: tone detection module

94: voice activity detection module 95: voice coding module

96: packet playout module 97: packet protocol encapsulation module

971: voice encryption module 98: control module

99: network interface driver

100: telephony signal gateway unit 101: call processing module

102: address translation module 103: network signal processing module

103a: H.323 protocol 103b: H.225 protocol

103c: H.245 protocol

103d: Registration, Admission, and Status (RAS) protocol / Real-time Transport Control Protocol (RTCP) protocol

103e: SGCP / MGCP protocol

110: network interface protocol unit 111: transmission control protocol module

112: user datagram protocol module 113: IP module

114: MAC / ARP module 115: Ethernet driver module

120: network management unit 121: network management agent module

122: web server module 123: SNMP module

124: Trivial File Transport Protocol (TFTP) module

130: system service department

131: Startup / Initialization Module

132 Power-On Self-Test (POST) Module

133: Real-Time Operating System (RTOS) module

134: Board Support Package (BSP) Module

135: Watch Dog Timer Driver Module (WDT)

136: flash memory manager module 140: DSP interface manager

The present invention relates to an IP voice service apparatus for voice service over Internet Protocol (IP), and instead of providing a voice service in a telephone network, voice is transmitted using a data packet in an IP network (eg, the Internet, an intranet, etc.). The present invention relates to an apparatus and a method for enabling telephone service.

Recently, as the interest for transmitting voice in IP networks such as the Internet, corporate intranets and extranets has increased, many studies have been conducted and actual initial services have been provided. Voice services over IP networks (hereinafter referred to as VoIP) are expected to provide full-scale services in the future.

The biggest advantage of the VoIP service using the present invention is the implementation of low usage fee. If the cost of service is low when the service fee is set, the service fee can of course be set low. The reason why VoIP charges can be as low as that is the key to the low cost of packet transmission. have.

Existing telephones can be used for telephone and fax transmissions using line transmission. In line transmission, both callers occupy a single telephone line exclusively to transmit and receive voice and fax data. This occupied line cannot be shared by a third party.

With tens of thousands of people talking at the same time all over the world, limited phone lines are forced to raise the line's usage fees, and these high line charges are immediately reflected in the phone bill. In addition, since the line transmission method is transmitted and received while maintaining the data size of the phone as it is 64Kbyte, the price per unit capacity in a certain amount of line capacity is also high.

On the other hand, the packet transmission method does not occupy a specific line but throws a chunk of data called a packet into a line that transmits data. It is not something to be shared.

And because packets are data that can be compressed as you want, you can save money on circuits.

SUMMARY OF THE INVENTION In view of the conventional requirements as described above, the present invention provides a telephone service by transmitting voice using a data packet in an IP network (for example, the Internet, an intranet, etc.) instead of providing a voice service in a telephone network. An object of the present invention is to implement an IP voice service apparatus and method.

An IP voice service apparatus for voice service on Internet Protocol (IP) of the present invention for achieving the above object includes a user interface unit for providing a user interface function of a public telephone network (PSTN) telephone, and the user interface unit. A user interface processor for controlling an operation function of each of the interface devices operated in the network, a voice interface unit for converting a user analog voice input through the user interface unit into digital samples, a local area network (LAN), and a digital subscriber line (xDSL). And a processor core unit for establishing an Internet connection through a UDP / IP network based on a cable modem, and performing voice signal processing, call processing, and protocol processing output from the voice interface unit. Characterized in that.

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In addition, the IP voice service method for the voice service on the Internet Protocol (IP) of the present invention for achieving the above object, the first process of processing the voice signal input through the user interface unit into a digital signal packetized; And a second process of establishing an Internet connection through a local area network (LAN), a digital subscriber line (xDSL), and a cable modem based user datagram protocol / internet protocol (UDP) network for transmission of the voice signal. And a third process of transmitting and receiving the packetized voice signal through the Internet connection, and a network of IP voice service means performing the first to third processes, and using a web server function. And a fourth process of providing a network management function of the network and a fifth process of monitoring a system service for monitoring an error operation of each process.

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The above objects, features and advantages will become more apparent from the following detailed description taken in conjunction with the accompanying drawings. Hereinafter, embodiments of the present invention will be described in detail with reference to the accompanying drawings.

 The VoIP service may be provided in various forms, and may be used as a voice service providing terminal such as an analog telephone and a PC. However, in the present invention, a VoIP service may be used simply and efficiently by implementing a dedicated IP voice service terminal for the VoIP service. do.

The system configuration using the VoIP service according to the present invention is shown in FIG.

The service provided by the present invention may be divided into two aspects as follows. The first is to provide a VoIP service using the present invention in a private business network. Since the home (1) or the local office (2) in the distant place is already connected to the network such as Ethernet and intranet for data service, it is possible to receive low cost and high quality voice service using the present invention using VoIP. .

The second is to provide a VoIP service using the present invention in the public network (for example, network operators). In this case, it is possible to use the VoIP service according to the present invention by using a voice gateway 3 of an internet telephony service provider in the public network. It is expected that many Internet telephony service providers will emerge as full-fledged VoIP services are provided in the future.

2 is a block diagram showing a hardware configuration of an IP voice service apparatus according to the present invention, wherein the IP voice service terminal is largely comprised of a user interface 10, a voice interface 20, a user interface processor 30, and a processor core unit ( processor core 40, a network interface unit 50, an associated logic device unit 60, and a power supply unit 70.

The user interface unit 10 provides the typical user interface functions of a conventional PSTN telephone. Basically there is a keypad 11 for dialing numbers (0-9, *, #), and an audible indicator 12 for notifying the user of an incoming call. There are keys with features such as dial, hold, transfer, and conferencing. A display (LCD) 13 is provided for displaying user prompts, dialed numbers, Caller ID information for incoming calls, and the like. Optionally, a serial interface 14 may be provided to connect a device such as 'PDA (Personal Digital Assistants') to allow synchronization of IP voice service terminal information and to use automatic dialing.

The voice interface unit 20 converts analog voice into a digital sample. The audio signal from the microphone 15 is sampled at 8 KHz and passed through a Pulse Coded Modulation (PCM) codec 21 to digitize a 64 kbps data stream to the processor. Similarly, the processor delivers a 64 kbps data stream in the direction towards the speaker via a PCM CODEC 21 for converting digital samples to speech.

The user interface processor 30 includes a user interface logic 31 and a serial port device 32 that provides control devices and device connection functions such as 'PDA'.

The processor core 40 performs voice processing, call processing, protocol processing, and network management software functions of the IP voice service terminal. As shown in FIG. 2, the processor core 40 includes a digital signal processor (DSP) 41 for voice related functions, a micro control device 42 for the remaining call processing and protocol processing functions, and software. Flash memory 43 for upgrade, ROM 44 and RAM 45 are provided.

The network interface unit 50 is responsible for transmitting and receiving analog voice from the voice service terminal. In the case of an enterprise LAN, the network interface unit 50 includes '10BaseT' or '100BaseT' Ethernet 51 of the TCP / IP protocol and an Ethernet controller 52 for controlling the same. The IP voice service terminal provides a separate RJ-45 Ethernet connector 53 for plug-in of a PC to share one connection at a wall jack. Using a cable modem or xDSL service allows for permanent high-speed Internet IP access, allowing not only LAN but also IP network access using a cable modem or xDSL.

The associated logic device unit 60 includes a LCD 13 of the user interface unit 10, a logic device and an oscilloscope 61 related to the operation of the keypad 11 and the sound indicator 12 to generate a clock. do.

The power supply unit 70 includes a power management unit 71 that manages the total power of the IP voice service apparatus.

FIG. 3 is a block diagram illustrating a protocol stack and a hierarchy required for operation of an IP voice service device, and illustrates a protocol stack and a hierarchy based on the ITU-T H.323 standard for VoIP.

The software structure of the present invention includes a user interface function unit 80, a voice processing unit 90, a telephony signaling gateway 100, a network interface protocol unit 110, a network management unit 120, and a system service unit. 130 and the DSP interface manager 140.

The user interface function unit 80 provides software for the function of the user interface processor 30. The user interface function unit 80 provides software for processing an interface for a user of an IP voice service terminal and is configured as follows.

A display driver 81 for controlling hardware for generating / displaying characters on the display 13;

A keypad driver 82 which performs scanning on the keypad 11 and processes keys input by a user;

A sound indicator driver 83 for controlling hardware for generating a ring to a user;

And a user interface processing module 84 for controlling the information displayed by the display driver 81 and processing the user key input to convert it into primitives for call processing.

The voice processing unit 90 is a unit for providing software for the voice interface unit 20 function, and is composed of a software module as shown in FIG.

The PCM interface module 91 receives Pulse Code Modulation (PCM) samples from the user interface unit 10 and sends them to a suitable DSP (Digital Signal Processing) 41 software module for processing. In addition, the processed PCM samples are sent to the user interface unit 10.

The tone generator module 911 generates a call progress tone to the user and in-band dual tone multi-frequency in the network based on a key input relayed from the user interface unit 10. (Dual Tone Multi-Frequency: hereinafter referred to as DTMF) A digit is generated. Since the compression algorithm of the voice codec (CODEC) 21 does not faithfully transmit DTMF tones, the software for that algorithm (eg G.723.1) must be remote IP voice service terminal (or gateway) to reproduce the DTMF tones. Create an in-band message on the network used by.

The line echo canceller module 92 performs an ITU-T G.168 compliant echo cancellation function on the sampled full-duplex voice signal. In a telephone network, echo is due to signal echo generated by a hybrid circuit that switches between a 4-wire circuit and a 2-wire circuit.

The tone detection module 93 detects the DTMF signal of the telephone output through the line echo canceller module 92.

Voice activity detector module 94 activates or deactivates the transmission of packets to detect voice activity and optimize bandwidth. When no activity is detected, the output of the encoder is not sent over the network. The module also measures the 'Idle' noise of the interface and reports this information to the packet voice protocol in the voice coding module for sending to the remote IP voice service terminal or gateway. The 'Idle' noise is reproduced remotely when it is not voice active so that the remote user does not feel that line is broken.

The speech encoding module 95 performs packetization of the 64 kbps data stream received from the user. There are various compression algorithms with different performance characteristics: G.711 PCM (no compression) running at 64kbps, G.723.1 running at 5.3kbps or 6.3kbps, and G.729 running at 8kbps. Typically, speech compression that performs greater compression requires much more processing power.

The packet playout module 96 performs compensation for network delays, network jitter, and discarded packets. Several techniques are used because there is currently no standard for packet playout.

The packet protocol encapsulation module 97 performs encapsulation of packet voice data output to the network interface driver 99. For VoIP, this encapsulation is done per Real-time Transport Protocol (RTP), which runs just above UDP, and provides encryption of voice packet data prior to transmission over the network to ensure privacy. Voice encryption module 971.

The control module 98 manages the exchange of monitor information and control information between the voice processing unit 90, the telephony signal gateway unit 100, and the network management unit 120. The information exchanged includes software downloads, configuration data, signal information, and status reports.

The telephony signaling gateway 100 is a unit that performs a function of setting up, managing, and terminating a call. The telephony signaling gateway 100 is configured as follows.

The call processing module 101 performs 'state machine' processing for call setup, call management, and call disconnection.

Address translation and parsing module 102 performs digit collection and parsing to determine when a complete number is dialed and to make that number capable of address translation.

Network signaling module 103: performs a signaling function for establishment, management and termination of calls on an IP network. Two standards are implemented: H.323 and SGCP / MGCP (Simple Gateway Control Protocol / Multimedia Gateway Control Protocol).

The H.323 protocol (103a) is a standard that describes how multimedia communication occurs between user terminals, network devices, and classified services over local and wide area IP networks. The following H.323 standards are used for VoIP services in IP voice service terminals.

H.225 protocol 103b is a call signaling protocol that performs signaling for establishing and tearing down call connections based on Q.931.

H.245 protocol 103c is a control protocol that provides capability negotiation between two end-points.

RAS (Registration, Admission, and Status) protocol 103d provides registration, subscription, bandwidth change, and status messages between IP voice service terminals and servers called gatekeepers that provide address translation and access control. Is used to transmit.

The Real-time Transport Control Protocol (RTCP) protocol 103d provides statistical information for monitoring the quality of service of a voice call.

In the SGCP / MGCP protocol 103e, SGCP is a standard for describing a master / slave protocol for establishing a VoIP call. The slave side or client is the gateway (or IP voice service terminal), and the master side is an entity called a call agent. SGCP was adopted by cable modem companies as part of the DOCCIS (Data Over Cable Systems Interface Specification) standard and developed into MGCP.

The network interface protocol unit 110 provides software for the function of the network interface unit 50 and includes a module as follows.

The transport control protocol (TCP) module 111 provides reliable transmission through data retransmission and flow control, and is used for web query and call signaling functions.

User Datagram Protocol (UDP) module 112 provides for efficient but unreliable data transfer. Since retransmission of real-time data introduces too much delay in voice conversations, UDP is used for the transmission of real-time data. UDP is also used for SNMP and TFTP network management traffic.

IP module 113 provides a standardized encapsulation of data transmitted over the network and includes destination and source addresses for routing.

The MAC / ARP module 114 performs a media access control (MAC) management function and processes an address resolution protocol (ARP).

The Ethernet driver module 115 configures and controls the hardware of the Ethernet controller 52, including the setting of Direct Memory Access (DMA) operation.

The network manager 120 includes the following modules.

The network management agent module 121 performs network management functions of the IP voice service terminal, including status monitoring and alarm reporting, and collection of statistical data on SNMP queries from the network management system.

The web server module 122 supports operation management using a standard web browser. A web page is presented to the user to set up an IP voice service terminal and collect statistical data.

The SNMP module 123 performs a Simple Network Management Protocol (SNMP) function to process the Gets and Sets (MIs) of the Management Information Base (MIB) and to set an alarm function.

The Trivial File Transport Protocol (TFTP) module 124 is used to update software with flash memory.

The system service unit 130 includes the following modules.

Startup / Initialization Module 131 performs startup and initialization of hardware and software modules.

The power-on self test (POST) module 132 provides a self test function when the IP voice service terminal is powered on.

The Real-Time Operating System (RTOS) module 133 provides functions such as task management, memory management, and task synchronization. In the present invention, VRTX, which is a real-time OS, is used.

The Board Support Package (BSP) module 134 provides a hardware interface driver, an interrupt vector, and the like for allowing a real-time OS to operate on a target hardware platform, and is provided with the VRTX.

The watch dog timer driver module (WDT) 135 controls the hardware watchdog timer as a control mechanism for preventing the IP voice service terminal from being stopped due to a software or hardware error.

The flash memory manager module 136 provides a function for reading / writing from the flash memory 43.

The DSP interface manager 140 provides a driver for exchanging information between the micro control device 42 and the DSP 41, including software download, voice packet and network management functions.

Referring to the operation of the IP voice service terminal configured as described above, when a user makes a call, the call is input to the user interface 10 to the voice interface 20.

In this case, the user interface logic 31 of the user interface processor 30 may display the LCD 13, the keypad 11, and the sound indicator 12 on the terminal used by the user through various driver software provided by the user interface function unit 80. ) To control the operation.

The voice interface unit 20 removes the echo sound from the voice 92 through various voice processing functions provided by the voice processor 90, encodes the voice to be transmitted 95, and then encapsulates 97 to transmit it to the network. I make it possible.

When the user signal subjected to the voice processing is input to the processor core 40, the processor core 40 performs digital signal processing on the signal transmitted from the voice interface 20 through the digital signal processor 41. In the microcontroller device 42, network signal processing is performed through a function provided by the telephony signal gateway unit 102, and an address translation is performed for the gateway of the destination through the address translation module 102.

As such, when the network state to the destination is completely set, the user signal is transmitted to the network interface unit 50, and the network interface unit 50 is based on the protocol provided by the network interface protocol unit 110. Or transmit user signal through IP network connection using xDSL.

On the other hand, the system service 130 performs the startup and initialization of the IP voice service terminal so that the problem does not occur in the above operation, the watchdog timer driver module 135, the hardware watchdog timer to control the error operation of the terminal To control.

5 is a diagram illustrating a software structure according to the present invention. Each software task is implemented based on VRTX, which is a real-time operating system, and is required for implementing a transmission and reception queue and an IP voice service terminal for communication between tasks. It includes a device driver for controlling each hardware device.

The user interface task A1 sets up drivers for controlling user interface devices and processes user input signals from them. It also passes the processed data to the Telephony Signal Gateway task. On the contrary, the received call information and signal are output to the user interface.

The system service task A2 is responsible for starting the system, initializing each hardware and software module, and performing a self test.

Voice processing task A3: sets and controls device drivers related to the voice processing unit such as the CODEC 21 and the DSP 41, and transmits and receives the user's voice data through the network interface according to the call processing result of the telephony signal gateway task.

The telephony signaling gateway task (A4) handles call processing and address resolution functions for IP voice services, and is under the control of the call processing task (A5) and the call processing task (A5) for call processing. SGCP / MGCP and H.323 A signal protocol processing task A6 for processing the protocol.

The call processing task A5 is responsible for the overall call processing of the IP voice service terminal, and calls the signaling protocol processing task to process the transmission and reception signal protocol.

The signal protocol processing task A6 processes the SGCP / MGCP and H.323 protocols.

Network management task (A7) performs network management function using SNMP and provides convenient web-based network management function using web server function.

There is a device driver that provides drivers for devices related to the user interface, the voice processing unit, and the network interface unit, and sets an operation state of the corresponding device and provides hardware access functions to tasks.

The operation of the IP voice service terminal providing the IP voice service through the operation of the hardware and the software embedded in the hardware will be described with reference to FIG. 6.

The VoIP service using the present invention can be provided in various ways and forms, and can be provided using various networks such as a LAN, an intranet, an extranet, an ATM network, a frame relay network, and the like as a backbone network.

For example, first of all, a gatekeeper 4 and a gateway 3 system exist on the network to communicate with an existing PSTN analog phone using an IP voice service terminal. . The gatekeeper 4 controls network access and provides address translation for H.323 terminals and gateways. The gateway 3 is a system that enables bidirectional communication between an H.323 terminal on a packet network and an analog terminal of a circuit switched network.

First, the IP voice service terminal and the gateway 3 register with the gatekeeper 4 via the H.323 RAS protocol 103d in the telephony signal gateway 102 (B1).

In this state, when an IP voice service terminal attempts a voice call, the call requests the gatekeeper 4 to translate the address to the gateway 3 of the destination via the user interface 10 and the gatekeeper 4. ) Transmits the translated address to the IP voice service terminal (B2), and then performs a call setup procedure between the gateway 3 and the IP voice service terminal using the H.225 protocol of H.323 (B3). ).

The gateway 3 connects the line call with the analog terminal subscribed to the PSTN (B4). After all connections are made, a voice call between the IP voice service terminal and the PSTN terminal is enabled (B5).

Secondly, in case of making a voice call using IP voice service terminal between two points which are separated from each other, IP voice service terminals are registered in the gatekeeper 4 in advance (C1), and the call is established using the H.323 protocol between the two systems. After the normal connection is made (C2), the voice call between the IP voice service terminal is possible (C3).

As described in detail above, the present invention has an advantage of providing a user with a low cost efficient voice service by enabling a voice service using a packet over an IP network using VoIP technology, and when a full-fledged VoIP service is provided in the future. Can be used as the primary subscriber device needed.

In addition, preferred embodiments of the present invention are disclosed for the purpose of illustration, those skilled in the art will be able to various modifications, changes, additions, etc. within the spirit and scope of the present invention, such modifications and modifications belong to the following claims You will have to look.

Claims (7)

A user interface unit providing a user interface function of a public telephone network (PSTN) telephone; A user interface processor for controlling an operation function of each interface device operated in the user interface unit; A voice interface unit for converting a user analog voice input through the user interface unit into a digital sample; And It establishes Internet connection through LAN (local area network), xDSL (digital subscriber line), cable modem based UDP / IP (user datagram protocol / internet protocol) based network, and processes voice signal and call output from voice interface unit. And a processor core for performing processing, and protocol processing. IP voice service apparatus for voice service over Internet Protocol (IP). The method of claim 1, wherein the voice interface unit, A tone generating module for generating a call progress tone to a user and generating an in-band dual tone multi-frequency (DTMF) digit in the network based on a key input relayed from the user interface; A line echo canceller module for performing echo cancellation on the sampled full-duplex speech signal; A tone detection module for detecting a DTMF signal of a telephone output through the line echo canceller module; Enables or disables the transmission of packets to detect voice activity and optimize bandwidth, and measures the idle noise of the interface and sends it to the packet voice protocol in the voice encoding module for transmission to a remote IP voice service device or gateway. Reporting voice activity detection module; A speech encoding module for packetizing the data stream received from the user; A packet playout module that compensates for network delay, network jitter and discarded packets; A packet protocol encapsulation module comprising a voice encryption module for providing encryption of voice packet data before transmission to a network to ensure user privacy, the packet protocol encapsulation module for encapsulating the voice packet data; And And a control module for managing the exchange of control information with the voice interface unit, the processor core unit, and an IP voice service device for voice service over an Internet protocol (IP). The processor core of claim 1, wherein the processor core unit comprises: A telephony signal gateway that performs a signaling function for setting up, managing, and terminating calls on the UDP / IP network; A system service unit which executes startup and initialization of the IP voice service device and controls a watchdog timer for malfunction control; And And a network manager configured to monitor and manage a network state of the IP voice service device. The method of claim 3, wherein the telephony signal gateway unit,  A call processing module that performs processing for call setup, call management, and call cutting; An address translation module that performs digit collection and decomposition to determine when a complete number is dialed and to enable address translation for the number; And Voice service over Internet Protocol (IP), comprising a network signal processing module for performing call processing with various protocols for performing signal functions for establishing, managing and terminating calls on the UDP / IP network. IP voice service device. The method of claim 3, wherein the network management unit, A network management agent module for collecting status monitoring, alarm reporting, and statistical data to perform network management functions of the IP voice service device; Web server module that supports operation management using standard web browsers' A network management protocol (SNMP) module for processing reception and transmission of the management information base and performing a protocol function for setting an alarm function; And An IP voice service device for voice service over Internet Protocol (IP), comprising a file transfer protocol (TFTP) module for updating software with flash memory. The method of claim 3, wherein the system service unit, A startup and initialization module for starting and initializing the IP voice service device; A power self test module for providing a self test function when the IP voice service device is powered on; A real-time operating system module that provides task management, memory management, and task synchronization functions; A board support packet module for providing a hardware interface driver and an interrupt vector for allowing a real-time OS to operate on a target hardware platform; A watchdog timer driver module configured to control a watchdog timer to prevent an operation from being stopped due to an error of the IP voice service device; And An IP voice service apparatus for voice service over Internet Protocol (IP), comprising a flash memory manager module providing a function for reading / writing from flash memory. A first step of processing a voice signal input through the user interface unit into a digital signal and packetizing the voice signal; A second step of establishing an internet connection through a local area network (LAN), a digital subscriber line (xDSL), and a cable modem based user datagram protocol / internet protocol (UDP) network for transmission of the voice signal; A third step of transmitting and receiving the packetized voice signal through the Internet connection; A fourth step of managing a network of an IP voice service means for performing the first to third steps and providing a web-based network management function using a web server function; And And a fifth process of monitoring a system service for monitoring an error operation of each process.

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