KR20030062616A - IP-PBX for Supporting Multi-Channel by Using SC-BUS - Google Patents

Abstract

The present invention relates to an Internet protocol private exchange for transmitting time-division-multiplexed analog voice signals inputted from extension telephones and trunk lines to multiple channels using SC-BUS.

An Internet Protocol Private Exchange (IP-PBX), which is connected to an extension telephone, a trunk line telephone, and a computer, enables voice calls and data communication, and includes a control unit, an extension telephone, which controls the overall operation of the Internet Protocol Private Exchange and forms a call channel. An extension matching board (SSM) unit for receiving an analog voice signal from an extension telephone and transmitting an analog voice signal to a counterpart telephone or converting an analog voice signal into multiple time-division digital data; Data from a trunk line matching board (PSM) section, a station matching board section, and a trunk line matching board section for receiving an analog voice signal and transmitting the analog voice signal to the counterpart phone or converting the analog voice signal into multiple time-division digital data. By receiving Digital data processing board (VSM) section and digital matching board section, trunk line matching board section and digital signal processing section for transmitting data received from the counter computer or receiving digital data transmitted from the counterpart computer and the counterpart computer. The present invention provides an IP-PBX that supports multiple channels using an SC-BUS, which includes an SC-BUS functioning as a channel to which the transmitted digital data moves.

Description

IP-PBX for Supporting Multi-Channel by Using SC-BUS}

The present invention relates to an internet protocol private exchange for supporting multiple channels. More specifically, the number of circuits limited by the number of ports of the conventional Internet protocol private exchange is provided in the Internet protocol private exchange having a chip and a bus supporting time division multiple access in the Internet protocol private exchange. It is about.

In modern society, landline telephones are prevalent in almost all homes. In particular, large buildings have numerous landline telephones. A private branch exchange (PBX) is a device that connects a number of landline telephones used in such a large building to a public switched telephone network (PSTN). That is, the private exchange performs exchange and connection between the extension telephones used in the premises of government offices, companies, factories, hotels and the like, or between the extension telephones and the trunk telephones connected to the public switched telephone network.

On the other hand, recently, a technology for making a call using a computer connected to the Internet has been developed. A method of making a call using a computer connected to the Internet includes installing a call program on a computer and then executing a call program to talk to the other party or to provide an Internet call service, for example, a dial pad ( Call www.dialpad.co.kr) or wowcall (www.wowcall.co.kr) to talk with the other party.

1 is a block diagram schematically showing a system for providing an Internet telephone service using a conventional telecommunication telephone.

The conventional Internet telephony service providing system is based on the Internet 100 as a network for transmitting and receiving data through a wired network by a plurality of computers located in remote locations.

The client computer 120 executes an Internet phone call program that enables a call using the Internet 100 or accesses a specific website that provides Internet phone call service using the Internet 100. This function allows you to talk to the user. The client computer 120 includes a microphone (not shown) for inputting a voice for a call and a headset (not shown) for listening to the voice or a headset 122 capable of inputting and listening to the voice.

The premises communication service unit 130 is connected to the Internet through a Voice Over Internet Protocol (VoIP) Gateway A (140) to enable a call using the Internet. The premises communication service unit 130 is connected to a plurality of premises telephones 132 and 133 for the call and the premises telephones 132 and 133 to exchange data with the communication device of the other party through the Internet 100. ).

Here, the VoIP gateway A 140 is a device that allows the local telephone 132 to talk with other communication terminals using the Internet. That is, VoIP gateway A 140 enables the exchange of voice information between callers using the Internet 102 after a call is connected between the local telephone and another communication terminal.

VoIP is a call technology on the Internet that uses Internet Protocol to transfer not only data but also voice. The advantage of VoIP technology is that telephone services can be utilized while utilizing existing Internet protocol networks, greatly reducing the line cost of conventional telephone calls or fax transmissions, so that telephone users can receive low-cost Internet and intranet services in the Internet or intranet environment. It makes the phone service available. With the advent of VoIP technology, the utilization of local area networks is maximized and network service quality is further enhanced by tying telephone and data networks into local area networks (LANs) using a single Internet protocol. Recently, the trend of communication technology is progressing to make this VoIP function essential.

A gateway is a device used to connect a local area network with other networks. In more detail, a gateway is used when connecting two different local area networks or connecting a local area network with an external long distance network. In general, various types of communication networks are used in a communication environment, and each of them uses its own data format and communication protocol. Therefore, since data cannot be directly transmitted from one communication network to another, a gateway is used to enable communication between these communication networks.

The telephones 180 and 181 are connected to a public switched telephone network (PSTN) 170 to perform a call through the Internet 100. The PSTN 170 is connected to the Internet 100 through the VoIP gateway B 160 to provide a service for exchanging voice data between callers.

The Internet phone service unit 150 is connected to VoIP gateway devices through the Internet 100 to provide Internet phone service between the client computer 120, the local telephones 132 and 133, and the trunk line telephones 180 and 181. . That is, it receives a call request from the VoIP gateway A 140, the VoIP gateway B 160, and the client computer 120 through the Internet 100 and performs a function of making a call to the communication terminal of the other party.

However, the conventional private exchange described with reference to FIG. 1 has the disadvantage that only an analog voice call is possible between a plurality of telephones, and that a separate solution or a VoIP gateway is required to be connected to the Internet. Accordingly, a new private exchange called IP-PBX (Internet Protocol-Private Branch Exchange, hereinafter called IP-PBX) has been developed that overcomes the disadvantages of the conventional private exchange.

IP-PBX is capable of sending and receiving various digital data as well as analog voice information, so that not only calls between ordinary telephones, but also calls using regular telephones and computers connected to the Internet, voice mail (VMS) function, fax ( It provides various functions such as FAX) function. In other words, IP-PBX is a combination of a private exchange and a VoIP gateway into a single device.

However, one IP-PBX generally supports only 16 channels.

FIG. 2 is a block diagram showing an IP-PBX divided into a private exchange and a VoIP gateway.

The private exchange 210 is connected to a plurality of local telephones (not shown) by wire 212.

One side of the VoIP gateway 230 is connected to the private exchange 210, the other side is connected to the network and wired (232).

The connection between the private exchange 210 and the VoIP gateway 230 is a VoIP gateway 230 is assigned a specific port (Port) from the private exchange 210 directly connected to the bus 220. However, the number of channels supported by the VoIP gateway 230 is usually limited to about 16 channels. Therefore, since the line capable of making calls through the VoIP gateway 230 is composed of a transmission channel and a reception channel, only about eight lines, which are half of the total channels supported by the VoIP gateway 230, are provided. That is, in order to secure more call lines, the VoIP gateway 230 needs to be additionally installed. Therefore, the VoIP gateway 230 is required to be installed in a large amount of cost, and a great difficulty arises in maintaining and managing the VoIP gateway 230. There is a problem.

In order to solve the above problems, the present invention provides a chip and a bus that supports time-division multiple access inside an Internet protocol private exchange to limit the number of lines limited by the number of ports of the internet protocol private exchange. It relates to an internet protocol private exchange.

1 is a block diagram schematically showing a system for providing an Internet telephone service using a conventional telecommunication telephone;

2 is a block diagram showing a state in which an IP-PBX is divided into a private exchange and a VoIP gateway;

3 is a block diagram schematically showing an internal configuration of an IP-PBX according to an embodiment of the present invention;

4 is a block diagram showing the structure of a data frame in a time division multiple access scheme;

5 is a block diagram schematically illustrating a time division multiplexing process of an analog voice signal according to an embodiment of the present invention.

<Explanation of symbols for the main parts of the drawings>

130: premises communication service unit 132: premises telephone

134, 210: Private exchange 140, 160, 230: VoIP gateway

170: public switched telephone network 200: Internet protocol private exchange

220: bus 320: extension matching board portion

326, 336, 346: 328: extension matching unit

330: trunk line matching board section 337: codec section

338: trunk line matching unit 340: digital signal processing board

348: digital signal processor 350: SC-BUS

500: pulse code modulation unit 510: sampling unit

520: quantization unit 530: encoding unit

540: multiplexer 550: switching unit

To this end, the present invention, in the Internet Protocol Private Exchange (IP-PBX) that is connected to the extension telephone, trunk line telephone and computer to enable voice calls and data communication, controls the overall operation of the Internet Protocol Private Exchange and forms a call channel. Connected to the control unit and the extension telephone, an analog matching signal (SSM) unit for receiving an analog voice signal from the extension telephone and transmitting the analog voice signal to the counterpart telephone or converting the analog voice signal into multiple time-division digital data, connected to the trunk line telephone A trunk line matching board (PSM) unit, a station matching board unit, and a trunk line to receive the analog voice signal from a trunk line telephone and transmit the analog voice signal to the counterpart telephone or to convert the analog voice signal into multiple time-division digital data. Matching board part Digital signal processing board (VSM) unit and digital matching board unit, trunk line matching board unit, and digital signal processing unit which receive data from the other computer and transmit the received data to the other computer or receive the digital data transmitted from the other computer. The present invention provides an IP-PBX that supports multiple channels using an SC-BUS, which includes an SC-BUS functioning as a channel through which data and digital data transmitted from a counterpart computer move.

Hereinafter, with reference to the accompanying drawings will be described in detail a preferred embodiment of the present invention.

3 is a block diagram schematically showing an internal configuration of an IP-PBX 200 according to an embodiment of the present invention.

The IP-PBX 200 is a control unit 310 for controlling the overall operation of the IP-PBX 200, a subscriber switch module (SSM) unit 320 connected to an extension telephone, a PSM connected to a trunk line telephone. It includes a (Public Switch Module, trunk line matching board) unit 330 and the VSM (VoIP Switch Module, digital signal processing board) unit 340 connected to the computer via the Internet.

The control unit 310 functions to control call channel allocation through the IP-PBX 200 and transmission and reception of various data through the allocated channel, and the CPU 1 (312) and DPRAM (Dual Port). RAM) 1 (314).

The SSM unit 320 includes a DPRAM 2 322, a CPU 2 324, an SC-BUS control unit A 326, a SLIC (Subscriber Line Interface Circuit, an extension matching unit) 328, and the like.

DPRAM 2 322 temporarily stores data transmitted / received from DPRAM 1 314, or transfers data received from DPRAM 1 314 to CPU 2 324, which will be described later, or data received from CPU 2 324. To the DPRAM 1 314.

The CPU 2 324 generally controls and manages the SC-BUS control unit A 326, the SLIC 328, etc. constituting the SSM unit 320 to communicate with the control unit 310.

The SC-BUS control unit A 326 performs a function of transmitting and receiving voice data through the SC-BUS 350. That is, the SC-BUS control unit A 326 converts the data received from the SLIC 328 which will be described later to conform to the transmission standard of the SC-BUS 350 or SLIC the data received from the SC-BUS 350. The overall control of the SC-BUS 350, such as forwarding to (328).

Here, the SC-BUS 350 is a path through which data divided in a time division manner is moved using a SCSA ^TM (Signal Computing System Architecture, SCSC ^TM ) standard. Modulation and flow of data using the time division method will be described in detail with reference to FIGS. 4 and 5. SCSA ^TM is a standard developed by Dialogic Inc. and other 70 companies in 1993. It is a standard interface for processing voice, fax, and data on personal computers. Detailed description thereof will be omitted since it departs from the spirit of the present invention.

The SC-BUS controller is a chip for transmitting and receiving data through the SC-BUS 350 and can accommodate up to 2,048 voice channels. That is, since a voice channel requires a transmission channel for transmitting data and a reception channel for receiving data, the number of communication lines that one SC-BUS controller can provide is 1,024, which is half of the total channels. These 1,024 call lines are the maximum number of simultaneous call channels that the IP-PBX 200 can support.

Referring to FIG. 1, the SLIC 328 basically supplies power to the premises communication service unit 130 to which the premises phones 132 and 133 are connected, and provides a ring for the premises phones 132 and 133. Generates a signal. In addition, CPT (Call Progress Tone) related to the ring signal, busy signal and response state is generated from the telephones 132 and 133, and the state of pressing the Dual Tone Multi Frequency (DTMF) key is pressed from the telephones 132 and 133. It performs the electrical matching function of the premises communication service unit 130 to detect the.

In addition, the SLIC 328 includes a codec unit (not shown) to encode or decode a voice signal transmitted and received. The process in which the SLIC 328 encodes the analog voice signal input from the premises telephones 132 and 133 will be described in detail with reference to FIG. 5.

The PSM unit 330 includes a DPRAM 3 332, a CPU 3 334, an SC-BUS control unit B 326, a codec unit 337, a DAA (Data Access Arrangement), and the like. It is composed.

Since the DPRAM 3 332, the CPU 3 334, and the SC-BUS control unit B 326 perform the same functions as described in the SSM unit 320, a detailed description thereof will be omitted.

The codec unit 337 functions to encode an analog voice signal received from the DAA 338 to be described later or to decode data received from the SC-BUS control unit B 336. A process of encoding the analog voice signal by the codec unit 337 will be described in detail with reference to FIG. 5.

Referring to FIG. 1, the DAA 338 is a unit responsible for electrical matching with the PSTN 170. The DAA 338 receives voice data transmitted from the CO lines 180 and 181 from the PSTN 170. The voice data transmitted from the premises telephones 132 and 133 are transferred to the trunk line telephones 180 and 181 through the PSTN 170.

The VSM unit 340 includes a DPRAM 4 342, a CPU 4 344, an SC-BUS control unit C 346, a DSP (Digital Signal Processor) 348, and the like. The VSM unit 340 functions to modulate data received from the SSM unit 320 and the PSM unit 330 by using a VoIP technology to transmit the data through the Internet. That is, among the analog voice signals inputted to the SSM unit 320 and the PSM unit 330, the analog voice signals to be transmitted to the computer, not the telephone, are transmitted to the VSM unit 340. Therefore, when a call channel is formed between extension telephones or trunk lines, or between an extension telephone and a trunk line telephone, the analog voice signals inputted to the SSM unit 320 and the PSM unit 330 do not pass through the VSM unit 340. Is sent directly to the phone.

Since the DPRAM 4 342, the CPU 4 344, and the SC-BUS control unit C 346 perform the same functions as described in the SSM unit 320, detailed description thereof will be omitted.

The digital signal processor (DSP) 348 performs digital data transfer from the SSM unit 320 and the PSM unit 330 to the SC-BUS control unit C 346 via the SC-BUS 350. After receiving from the BUS control unit C (346), it functions to modulate the received data by applying VoIP technology for transmission standards on the Internet. In addition, the received data is compressed to increase the transmission speed in a general network. The compressed and transmitted data among the data transmitted from the client computers 120 of FIG. 1 is received through the SC-BUS control unit C 346 to restore the compressed data. Since the process of compressing the data or restoring the compressed data by the DSP 348 is outside the technical spirit of the present invention, a detailed description thereof will be omitted.

In FIG. 3, since the channels of the SSM unit 320, the PSM unit 330, and the VSM unit 340 are formed and the call is the same, the call process of the SSM unit 320 will be described.

Analog voice signals input from a plurality of extension terminals (not shown) connected to the SSM unit 320 are input to the SLIC 328. The SLIC 328 requests the CPU 2 324 to allocate a channel for transmission of the input voice information via the SC-BUS control unit A 326. The CPU 2 324 requests the controller 310 to allocate a channel through the DPRAM 2 322. The CPU 1 312 of the controller 310 requests channel allocation from the CPU 4 344 of the VSM unit 340 through the DPRAM 1 314 and the DPRAM 4 342. The CPU 4 344 of the VSM unit 340 receives the address of the available channel from the SC-BUS control unit C 346 and transfers the received address to the CPU 1 312 of the control unit 310. The CPU 1 312 of the controller 310 transmits the address of the available channel to the CPU 2 324 of the SSM 320 and allocates the channel. In the SLIC 328, analog voice information is multi-time-divided into digital data and transmitted to the SC-BUS controller A 326, and the SC-BUS controller A 326 uses the SC-BUS 350 to time-divided the digital data. Send to section 340. The VSM unit 340 receives the data transmitted through the SC-BUS control unit C 346, modulates and compresses the received data by applying VoIP technology in the DSP 348, and then returns to the SC-BUS 350. send. Data flowing through the SC-BUS 350 is sent to a receiving device of the other party, for example, a telephone or a computer, through a network to form a call channel.

Similarly, after the call is terminated, the call channel is disconnected by the reverse process as described above. In brief, the VSM unit 340 recognizes the disconnected state and sends a signal indicating that the call is completed. When the controller 310 transmits the received call completion signal to the SSM unit 320 or the PSM unit 330, the SSM unit 320 or the PSM unit 330 transmits a call channel of an extension terminal or a trunk line terminal having a call channel. Quit.

4 is a block diagram illustrating a structure of a data frame in a time division multiple access scheme.

Time Division Multiple Access (TDMA) refers to a carrier signal shared by multiple users using time slots in a time interval called a frame. Although a common frequency band is used in a scheme, it is a multiple access communication technology that transmits and receives data by not allowing time intervals allocated to the user to overlap with other users.

In other words, it is a technology that designates a time-use area limited to a line in advance for each channel so that several channels can be used in one line. In TDMA, the bandwidth allocated for each channel is specified in advance, which means that a channel is always allocated regardless of data transmission and reception, resulting in a waste of line bandwidth. However, when the TDMA technology is used, the delay time in the network equipment can be reduced because the network equipment transmits the input data of each channel to a predetermined time slot without performing a complicated process in the network equipment.

Referring to FIG. 4, a frame consists of a plurality of sub-frames called time slots in the time axis. The time allotted to each time slot may be assigned to callers an extremely short time, such as a few microseconds, so that the other party's voice can be heard naturally. That is, after time passes from time slot 0 to time slot N in the process of transmitting voice data, the switching operation is performed at such a high speed that the voice information contained in time slot 0 is not interrupted again. do. This switching is not accomplished by physical switches, but by electrical signals on the circuit.

On the other hand, in the frequency axis, one slot of data divided in the time axis forms one channel. Therefore, a plurality of call channels are formed for each time within a fixed frequency allocated bandwidth.

5 is a block diagram schematically illustrating a time division multiplexing process of an analog voice signal according to an embodiment of the present invention.

Referring to FIG. 3, analog voice signals inputted from a plurality of trunk lines and extension telephones connected to the SLIC 328 of the SSM unit 320 and the DAA 338 of the PSM unit 330 are SLIC 328. And the codec unit 337. The SLIC 328 and the codec unit 337 time-division convert the received analog voice signal into a digital signal in order to meet the transmission standard of the SC-BUS 350.

In more detail, the SLIC 328 and the codec unit 337 include a pulse code modulation (PCM) unit 500, a multiplexer unit 540, and a switching unit 550. do.

The sampler 510 samples the received analog voice signal at a speed two or more times the maximum frequency of the analog voice signal to create a discrete signal. Typically, analog voice signals are sampled at 8000 cycles per second. The sampled data is approximated to a value closest to the quantization level 2 ⁿ , which is performed by the quantization unit 520. That is, if the discrete signal values extracted by the sampling unit 510 are, for example, 0.7, 1.3, and 3.6, the discrete signal values are passed through the quantizer 520 and the respective discrete signal values are closest to 2 ⁿ , that is, 1 (2 ⁰ ) and 1 ( 2 ¹ ) and 4 (2 ² ). The coding unit 530 converts the quantized data into a binary code. For example, 1 is converted into a binary code consisting of 8 bits (Bit) such as 00000001, 2 is 00000010, and 4 is 00000100. The technique of modulating an analog signal into a digital signal through such a process of sampling, quantization, and encoding is called pulse code modulation. Pulse code modulation is a process that must pass through time division multiplexing to convert an analog signal into a time division digital signal.

The digitally encoded signal from the encoder 530 is input to the multiplexer 540. A multiplexer, also called a multiplexer, is a device that carries multiple signals on one channel in a communication system. In this case, the multiplexing of multi means multiplexes data from multiple input lines and sends them to a high capacity data link. On the receiving side, multiplexed data streams are received to separate data according to channels and output appropriately. Send it on the line. The receiving equipment is called a demultiplexer.

The time division signal is transmitted to the switching unit 550, and the switching unit 550 reads the address number of the received signal and allocates the time division signals to the appropriate time slots. Then, the destination information recorded in the header area of the data is read and time-divided signals are transmitted to the corresponding receiving apparatuses 560 and 562.

As described above, the conventional IP-PBX 200 used in the premises communication network has a disadvantage in that the VoIP channel connected to the private exchange has a limited communication channel depending on the number of ports of the private exchange, but according to the present invention, the IP-PBX 200 In the case of SC-BUS, analog voice signals are transmitted by time division multiplexing, thereby forming a plurality of channels, thereby greatly increasing the number of communication terminals such as telephones that the IP-PBX 200 can accommodate.

In addition, since the IP-PBX according to the present invention uses time division multiplexing, high-speed data transmission is possible, and since the private exchange and the VoIP gateway are connected by a single SC-BUS, the private exchange and the VoIP gateway can be easily connected. There is this.

Claims (8)

In the Internet Protocol Private Switching System (IP-PBX), which is connected to an extension telephone, a trunk line telephone, and a computer, which enables voice calls and data communication, A controller for controlling the overall operation of the internet protocol private exchange and establishing a call channel; An extension matching board (SSM) unit connected to the extension telephone to receive an analog voice signal from the extension telephone to transmit the analog voice signal to a counterpart telephone or to convert the analog voice signal into multiple time-division digital data; A trunk line matching board (PSM) unit connected to the trunk line telephone to receive the analog voice signal from the trunk line telephone to transmit the analog voice signal to the counterpart telephone, or to convert the analog voice signal into multiple time-division digital data; A digital signal processing board (VSM) unit which receives data from the extension matching board unit and the trunk line matching board unit and transmits the data received from the counterpart computer or receives digital data transmitted from the counterpart computer; And SC-BUS functioning as a channel through which the digital data transmitted from the extension matching board unit, the trunk line matching board unit, and the digital signal processing unit and the digital data transmitted from the counterpart computer move. Internet protocol private exchange supporting multiple channels using the SC-BUS, characterized in that it comprises a. The method of claim 1, The SC-BUS is an Internet protocol private exchange supporting multiple channels using the SC-BUS, comprising one bus and simultaneously supporting 2,048 time division multiplexed signal channels. According to claim 1, wherein the extension matching board portion A central processing unit (CPU) for controlling the overall operation of the extension matching board unit; Dual Port RAM (DPRAM) for temporarily storing data for executing a command transmitted from the central processing unit or transmitting and receiving data with the control unit; An extension matching unit (SLIC) for time division multiplexing the analog voice signal input to the extension matching board unit and supporting electric matching with an extension subscriber network; And SC-BUS control unit for controlling the SC-BUS, receiving the time-division multiplexed signal from the station matching unit to transmit to the SC-BUS or the digital data transmitted from the counterpart computer Internet protocol private exchange supporting multiple channels using the SC-BUS, characterized in that it comprises a. According to claim 1, wherein the trunk line matching board unit A central processing unit controlling overall operation of the trunk line matching board unit; Dual Port RAM (DPRAM) for temporarily storing data for executing a command transmitted from the central processing unit or transmitting and receiving data with the control unit; A trunk line matching unit (DAA) supporting electrical matching between the analog voice signal inputted to the trunk line matching board unit and a trunk line network; A codec unit for time division multiplexing the analog voice signal or demodulating the digital signal received from the SC-BUS into an analog voice signal; And SC-BUS control chip which controls the SC-BUS and receives the time division multiplexed signal from the codec unit and transmits the SC-BUS to the SC-BUS or receives the digital data transmitted from the counterpart computer. Internet protocol private exchange supporting multiple channels using the SC-BUS, characterized in that it comprises a. The method of claim 1, wherein the digital signal processing board unit A central processing unit for controlling the overall operation of the digital signal processing board; Dual Port RAM (DPRAM) for temporarily storing data for executing a command transmitted from the central processing unit or transmitting and receiving data with the control unit; A digital signal processor (DSP) for receiving and compressing the time division multiplexed signal received from the extension matching board unit and the trunk line matching board unit or receiving the digital data transmitted from the counterpart computer and restoring the analog voice signal; And The SC-BUS control unit controls the SC-BUS and receives the digital signal compressed from the digital signal processor and transmits it to the SC-BUS or receives the digital data transmitted from the counterpart computer. Internet protocol private exchange supporting multiple channels using the SC-BUS, characterized in that it comprises a. The method of claim 1, wherein the Internet Protocol Private Exchange is When the counterpart of the requested call request from the extension telephone and the trunk line telephone is the computer, the analog voice signal inputted from the extension telephone and the trunk line telephone is time-division multiplexed into a digital signal. An Internet Protocol private exchange that supports multiple channels. The method of claim 6, The time division multiplexing of the analog voice signal is performed at the extension matching unit of the extension matching board when the analog voice signal is input from the extension telephone, and when the analog voice signal is input from the trunk line telephone. Internet protocol private exchange supporting multiple channels using SC-BUS, characterized in that performed in the codec section of the matching board. The apparatus of claim 7, wherein the extension matching unit and the codec unit are used. Sampling unit for sampling the analog voice signal at a constant cycle rate to convert to a discrete signal; A quantization unit approximating the discrete signal to a value close to a quantization level (2 ⁿ ); An encoder for converting the quantization level value into an 8-bit binary code value; A multiplexer for receiving the binary code value and time division multiplexing the binary code value; And A switching unit which allocates the time division multiplexed digital signal to an appropriate time slot and reads information of a header portion of the time division multiplexed signal and transmits the digital signal to a destination. Internet protocol private exchange supporting multiple channels using the SC-BUS, characterized in that it comprises a.