KR910005331B1 - Digital telephone system - Google Patents

Abstract

The system provides services of voice for normal telecommunication and non- voice like the facsimile, teletext, or videotex through the digital transmission line. The system includes an automatic equaliser circuit (21) having an automatic line build-out circuit providing constant gain, a controller (1) recognising the selection state of the inner line and communication state of the terminal, a COMBOCODE (17) converting the digital voice signal of a DRRZDG (22) to an analog one or analog voice signal of an analog multiplexer (18) to a digital one, and a DTIM (20) transmitting the non-voice signal between the DRRZCG and an interface (19).

Description

Digital Telephony Provides Voice and Non-Voice Information

1 is a circuit diagram of the present invention.

* Explanation of symbols for main parts of the drawings

1: microprocessor 2: clock generator

3: Roman 4: First display unit

5 key matrix and second display unit 6 second display control unit

7: Display and Kisscan Circuit 8: Ring Signal Generation Circuit

9: confirmation tone generating circuit 10: microphone

11: speakerphone 12: handset

13, 14, 16: 1st, 2nd, 3rd amplifier 15: speaker

17: Combo Codec 18: Analog Multiplexer

19: interface 20: DTIM

21: automatic equalization circuit 22: DRRZDG

23: power supply section 24: DSS / BLF

T1: First Transformer

The present invention relates to an apparatus for providing a communication service for voice and non-voice information to a subscriber by being connected to a public telecommunication network. In particular, a digital telephone capable of simultaneously providing voice and non-voice information by being connected to a private exchange or a kitellipon system. It is about.

In recent years, the communication media and the communication means efficiently store and deliver the knowledge or information in accordance with the congestion of knowledge and information required by humans. In the telephones that provide non-voice services, such as facsimile, telex, and videotex that provide non-voice services in the phone, the variety of communication media such as facsimile, teletex, and videotex are being used. Advances in transmission have improved the quality and speed of information transmission.

With the development of the communication medium and communication means, service of voice and non-voice information can be provided through digital transmission lines even in current private exchanges and key telephone systems.

However, terminals that are currently transmitting means are designed to receive either information service of voice or non-voice information according to each communication purpose. Recently, the terminal is connected to the Kitellipon system and can receive voice information service from the voice information service. In the case of the kitelliphone, the non-voice information service is merely a real-time or arithmetic information controlled by the kitelliphone system and cannot be provided with the non-voice communication service. .

Accordingly, an object of the present invention is to provide a digital telephone capable of receiving voice and non-voice communication services through a private exchange or a digital telephone system digital transmission line.

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

1 is a circuit diagram of the present invention, which controls a system of a digital telephone and communicates through a D channel, and generates a clock pulse train of a predetermined period and supplies the same to the microcontroller 1. A clock generator 2 for storing the program, a ROM for storing a program for controlling the digital telephone system in the microcontroller 1, and character information such as caller ID and real-time information. A first matrix 4 and a second matrix 5 for displaying an operating state of the digital telephone and for inputting a subscriber number and various function selection information at the time of origination, the key matrix and the second display unit A second display control unit 6 for controlling the display state of the display unit 5, and a first display unit controlled by the address applied from the microcontroller 1, and the key matrix and the second A display and kisscan circuit (7) for inputting key data from the display unit (5) to output the microcontroller (1) or inputting control data to the microcontroller (1) to control the second display controller (6). And a ring signal generation circuit 8 for generating a ring signal, a confirmation tone generation circuit 9 for generating a confirmation tone when a key is input, and a speaker phone for amplifying a voice signal input from the microphone 10. Phone 11, a first amplifier 13 for amplifying the voice signal input from the handset 12, a second amplifier 14 for amplifying the voice signal to be output to the handset 12, and a speaker A third amplifier 16 for amplifying the audio signal to be output to 15 and a combo codec for converting the analog voice signal into digital voice information and converting the digital voice information into an analog voice signal (COMOCODEC) 17, the speakerphone 11 or the first The audio signal input from the aeration 13 is transmitted to the combo codec 17 or the audio signal input from the combo codec 17 and the confirmation tone input from the confirmation tone generating circuit 9 are second or third amplifiers 14. 16, an analog multiplexer 18 for transmitting the ring signal input from the ring signal generation circuit 8 to the third amplifier 16, and an interface 19 for relaying the data terminal and the digital telephone system. A data terminal interface module 26 for performing communication of non-voice information with the data terminal through the interface 19, a transmission line L1, L2, and a digital telephone. An automatic equalization circuit for transformer T1 for matching impedance between systems and a received signal connected to the secondary side of the transformer T1 and having a loss according to the frequency characteristics of the line automatically. 2B +, which is a basic access channel structure (Alternative Mark Inversion: AMI) signal received through the automatic (Equalier Circuit) 21 and the automatic equalization circuit (21) After demodulating the D-type data, the voice information is separated into non-speech information and signaling information, and voice information is transmitted to the combo codec 17, and non-voice information is transmitted to the DTIM 20, and signaling information is transmitted to the microcontroller 1. The voice information is input to the combo codec 17, non-voice information is input from the DTIM 20, and signaling information is input from the microprocessor 1, and then combined into data of 2B + D form, and then modulated into an AMI signal. Data Recovery Return to Zero Grnerater 22, which is transmitted to the automatic equalization circuit 21, is referred to as DRRZDG 22 and the transmission lines L1 and L2. Power supply unit 23 for supplying the operating voltage to each part of the digital system by inputting the power supply to be transmitted, and direct station selection / busy tamp field for direct call selection and monitoring of the call state of the data terminal. Station Selection / Busy Tamp Field (hereinafter referred to as DSS / BLF) (24).

The present invention will be described in detail based on the above configuration.

First, the power supply unit 23 inputs the supply power (-48V, GND) supplied from the private exchange or the key phone system through the transmission lines (L1, L2) to supply voltages of ± 5, ± 12V. Is generated and supplied to each part of the circuit.

The automatic equalization circuit 21 is composed of an automatic gain control circuit incorporating an automatic line build out circuit so that a constant gain can be obtained even in the case of the frequency characteristics, the length of the line, and the gain variation due to the characterization. .

In addition, the DRRZDG 22 that receives or transmits an AMI signal through the automatic equalization circuit 21 demodulates the received AMI signal into data having a channel structure of 2B + D, and then voice information B1 and non-voice. Separate the information (B2) and signaling information (D) channels, respectively, to the combo codec 17, the DTIM 20, and the microcontroller 1, and transmit the voice information from the combo codec 17 at the time of transmission. ) Inputs non-voice information and microcontroller 1 inputs signaling information, combines it into a channel structure of 2B + D, codes it in the form of an AMI signal, and transmits it to the transmission line L1 through the automatic equalization circuit 21. To L2).

The microcontroller 1 operates by inputting the clock pulse train supplied from the generator 2, and exchanges data in a serial manner through a dedicated transmission / reception line with the DSS / BLF 24 to check the station selection state and the call state of the data terminal. On the other hand, the program stored in the ROM 3 is read through the data bus, and the character information data is output to the second display section 4, and the DTIM 20 and the DRRZDG 22 and the display and kisscan circuit ( 7) send and receive data. In addition, the microcontroller 1 has a built-in RAM for temporarily storing information generated during system operation.

In the above description, the second display unit 4 is configured with an LCD to display various service messages such as character information, that is, channel message display or subscriber station subscriber number during real-time call-in / out of speaker control.

On the other hand, the key matrix and a plurality of keys of the second display unit 5 and the LED is configured to indicate the operating state of the function keys and the operation state of the system. The DSS / BLF 24 is composed of a plurality of keys for extension selection and LEDs for indicating an extension call state and a data terminal call state.

Meanwhile, the combo codec 17 converts the digital voice information input from the DRRZDG 22 to an analog multiplexer 18 and outputs the analog voice signal to the analog multiplexer 18 or converts the analog voice signal input from the analog multiplexer 18 into digital voice information. The clock signal required for outputting to the DRRZDG 22 is supplied from the DRRZDG 22.

In addition, the analog multiplexer 18 transmits the voice signal input from the second amplifier 14 or the speakerphone 119 to the combo codec 17 or the voice signal and confirmation tone generation circuit input from the combo codec 17 ( 9) to switch the confirmation tone input from the second or third amplifiers 14 and 16 and to output the ring signal input from the ring signal generation circuit 8 to the third amplifier. It is controlled through the input / output mode of (1).

In addition, the confirmation tone generating circuit 9 which generates confirmation tone for recognizing that a key has been input at the time of key input operates under the control of the microcontroller 1.

Meanwhile, the DTIM 20 transmits non-voice information input from the DRRZDG 22 to the interface 19 by signaling with the microcontroller 1 through the data bus, that is, under the control of the microprocessor 1, and the interface 19. By transmitting the non-voice information input from the DRRZDG 22, the non-voice information between the DRRZDG 22 and the interface 19 is relayed.

The interface 19 then transmits the non-voice information input from the DTIM 20 to the data terminal and the non-voice information input from the data terminal side to the DTIM 20, which is RS-232C. Can be used.

As described above, the present invention has the advantage of simultaneously serving voice and non-voice information, and the medium providing non-voice information service has the advantage of using a host computer or a general data terminal.

Claims (1)

An apparatus for providing voice and non-voice information service through a public telecommunication network, the microcontroller 1 controlling a system of a digital telephone and communicating through a D channel, and generating a clock pulse train of a predetermined period to generate the microcontroller ( 1) a clock generator (2) for supplying to, a ROM (3) storing a program for controlling the digital telephone system in the microcontroller (1), and character information such as subscriber number and real time information at the time of origination. A key matrix and a second display portion 5 for displaying a first display portion 4 for displaying, an operation state of the digital telephone, and inputting a subscriber number and various function selection information at the time of origination, and the key matrix and second display portion The second display control unit 6 for controlling the display state of (5) and the first display unit 4 are controlled by the address applied from the microcontroller 1. A display for controlling the second display controller 6 by inputting key data from the key matrix and the second display unit 5 and outputting the key data to the micro controller 1 or inputting control data to the micro controller 1; Amplify the kisscan circuit 7, the ring signal generating circuit 8 for generating a ring signal, the confirmation tone generating circuit 9 for generating a confirmation tone upon key input, and the audio signal input from the microphone 10 A speakerphone 11 for processing, a first amplifier 13 for amplifying the voice signal input from the handset 12, a second amplifier 14 for amplifying the voice signal to be output to the handset 12, and And a third amplifier 16 for amplifying the voice signal to be output to the speaker 15, and a combo codec for converting the analog voice signal into digital voice information and converting the digital voice information into an analog voice signal ( 17) and the speakerphone 11 The voice signal input from the first amplifier 13 is transmitted to the combo codec 17 or the voice signal input from the combo codec 17 and the confirmation tone input from the confirmation tone generating circuit 9 are second or third amplifiers. An analog multiplexer 18 which transmits the ring signal inputted from the ring signal generation circuit 8 or the ring signal generation circuit 8 to the third amplifier 16, and an interface 19 for relaying the data terminal and the digital telephone system. And a DTIM 26 for communicating non-voice information with the data terminal through the interface 19, a transformer T1 for matching impedance between transmission lines L1 and L2 and the digital telephone system. And an automatic equalization circuit 21 connected to the secondary side of the transformer T1a to automatically compensate a received signal having a loss due to the frequency characteristic of the line, and an AMI signal received through the automatic equalization circuit 21. basic Demodulate data in 2B + D format, which is an access channel structure, and separate the speech and non-voice information and signaling information.The voice information is the combo codec 17, and the non-voice information is the DTIM 20. Transmits to the controller 1, combines voice information from the combo codec 17, non-voice information from the DTIM 20, and signaling information from the microprocessor 1, and combines the data into 2B + D type data. Supplying an operating voltage to each part of the digital system by inputting a DRRZDG 22 which is modulated by an AMI signal and transmitted to the automatic equalization circuit 21 and the supply power transmitted through the transmission lines L1 and L2. And a DSS / BLF (24) for directly selecting an extension and monitoring the call state of the data terminal.

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