KR20020016645A - System of internet-phone gateway with containing pstn and voip ciruit - Google Patents

Abstract

PURPOSE: An Internet phone gateway system containing a PSTN/VoIP interface circuit is provided to automatically conduct the origination and reception of either an Internet phone or a PSTN phone according to a user's intention. CONSTITUTION: An Internet phone gateway system containing a PSTN/VoIP interface circuit consists of a DSP(700), a CODEC(200), a SLIC(300), a control block(400), a detector(500), and an Ethernet switch(600). The DSP(700), connected with the CODEC(200) and the Ethernet switch(600), is composed of a DSP main, a channel drive, and a host communication interface. The CODEC(200) converts a 4-wire analog voice signal outputted from a SLIC(300) into PCM data and transmits the data to the DSP(700). Also the CODEC(200) converts PCM data into a 4-wire analog voice signal and transmits it to the SLIC(300). The SLIC(300) converts a 2-wire analog voice signal outputted from the control block(400) into a 4-wire signal or a 4-wire analog voice signal into a 2-wire signal. The control block(400), connected to the SLIC(300) and the detector(500), executes a switch function to connect a voice path to a PSTN or a VoIP. The detector(500), connected with the control block(400) through the PSTN, conducts hook detection and ring detection for the PSTN. The Ethernet switch(600), necessary for data communication, is connected with a WAN and a LAN respectively.

Description

Internet phone gateway system with PSTN and BIP matching circuit {SYSTEM OF INTERNET-PHONE GATEWAY WITH CONTAINING PSTN AND VOIP CIRUIT}

The present invention relates to an Internet phone gateway system, and more particularly, to an Internet phone gateway system having a PSTN and VoIP matching circuit for automatically and inexpensively using a phone line and a VoIP phone at an internet phone gateway to conveniently transmit and receive each phone call. It is about.

In general, a general telephone connected to a public switched telephone network (PSTN) and a voice over IP (VoIP) connected to the Internet network are used to pass through a remote party. It is divided into.

Currently, the Internet is not limited to a specific field but is widely used, and as a part of real life, the number of Internet users using the Internet is rapidly increasing, and thus the number of service providers providing specific services to the Internet users is increasing rapidly.

In particular, Internet phones that can connect to the remote network and talk to the remote party are very expensive due to the explosive increase in the number of computer users. In the case of international calls, public telephones are very expensive. The use of Internet phones that can make calls is increasing. The Internet phone refers to equipment or software provided to a client that enables a telephone call through the Internet of a computer. When the computer user connects to an Internet phone provider through the Internet, the Internet phone connects the user to the desired telephone number. In the case of long distance or international calls, which are expensive due to the use of a computer sound card under the name of the telephone client program, it is possible to make a call with a remote party by connecting to the Internet without using a general telephone network. By using the Internet, the burden on user fees is reduced.

However, in the conventional VoIP network, the telephone type allows the user to make a telephone call only to the physical form of the VoIP network, and thus cannot simultaneously use the PSTN in use. Therefore, there is a problem that it is possible to have both a call to PSTN and a call to VoIP.

Therefore, in recent years, by installing an adapter that can be used as a general telephone commercially available to the general phone through the public telephone exchange network and the Internet network, it is possible to use the Internet phone or a regular phone freely by arbitrary choice Although there have been attempts to diversify the functions of the device and provide convenience in use, it is expensive to add the PSTN and Internet phone selection function by adding the exchange function in the equipment of the adapter function (VoIP). Considering the cost, products that are not able to call PSTN at all are the mainstream.

Accordingly, an object of the present invention is to provide an Internet telephone gateway system incorporating a PSTN and VoIP matching circuit which automatically enables transmission and reception of Internet telephone and PSTN telephone according to the user's intention to use the telephone.

1 is a system configuration diagram of an Internet phone gateway incorporating a PSTN and VoIP matching circuit according to the present invention.

2 is an operational configuration diagram in the system according to FIG.

3 shows a specific circuit diagram of the system according to FIG. 1;

4 (A), (B) is a phone call flow chart using the Internet phone gateway system including a matching circuit of the PSTN and VoIP according to an embodiment of the present invention.

5 is a system circuit diagram of a gateway of an Internet phone incorporating a PSTN and a VoIP matching circuit according to an embodiment of the present invention.

The present invention for achieving the above object in the Internet phone gateway system with a PSTN and VoIP matching circuit,

Ethernet switch divided into WAN and LAN required for data communication;

A detector for detecting a ring signal in an idle state detected in a trunk line and a hook on / off state in a busy state;

A control block for monitoring a ring signal detected by the detector and a hook-on state and driving a switch for connecting a voice path to VoIP and PSTN;

A slick converting two-wire analog audio signals output from the DSP into four wires, or converting four-wire analog audio signals into two wires and performing functions such as battery feeding and ringing, ring trip, and hook on / off detection;

DTMF tone transfer unit for transmitting DTMF tones in the absence of feeding;

A codec for converting the analog voice signal converted from the slick into four wires to PCM data and transmitting the same to a DSP, or converting the PCM data into four wire analog voice signals to decompress the voice signal for an Internet telephone; And

And a DSP connected to the codec and the Ethernet switch and controlling a signal for an overall processor or a data transmission and an interface.

DETAILED DESCRIPTION Hereinafter, the present invention will be described with reference to embodiments in order to specifically describe the present invention. Hereinafter, the present invention will be described in detail with reference to the accompanying drawings.

First, FIG. 1 illustrates a system configuration diagram of an Internet phone gateway incorporating a PSTN and a VoIP matching circuit according to an embodiment of the present invention.

In FIG. 1, the gateway of the present system includes a DSP, a CODEC, a slick, a control block, a detector, and an Ethernet switch.

The DSP 700 is largely composed of a DSP main, a CNL (Channel) drive, and a Host Communication Interface (HST). The DSP 700 is connected to a Coder Decoder (CODEC) (200) and an Ethernet switch 600, respectively. It handles line input / output, tone generation and encode / decode interfaces, data transmission, and signal control of transmission status and event interfaces.

CODEC 200 converts the analog audio signal converted into four-wire output from slick 300 to PCM data and transmits it to DPS 700, or converts the PCM data into four-wire analog voice signal to slick 300. send.

The slick 300 converts the 2-wire analog audio signal output from the control block 400 into 4 wires, or converts the 4-wire analog audio signal into 2 wires. In addition, the CODEC 200 and the control block 400 are connected to perform functions such as battery feeding and ringing, ring trip, and hook on / off.

The DTMF Tone Path Block 100 does not form a loop between the SLIC 300 and the telephone because the hardware path cannot be connected to VoIP in the situation where the PSTN is set to PSTN. However, if the user wants to switch to VoIP, the user should detect the DTMF (Dual Tone Multi Frequency) signal, but the digital signal is blocked and only the analog signal is passed through so that the SLIC 300 recognizes the analog signal and converts it into the signal. It is used for DTMF detection. In this system, the capacitor and the DTMF Tone Path Block are identical.

The control block 400 is connected to the slick 300 and the telephone and the detector 500 to perform a switch function for connecting a voice path to the VoIP and the PSTN. The switch function is performed in the DSP 700. In case of power failure, the switch position is switched to the PSTN line position.

The detector 500 is connected to the control block 400 by a general trunk line (PSTN) and is a block for track detection and ring detection for a general trunk line. Since the switch in the control block 400 is located in VoIP (SLIC), ring detection is required when receiving a general trunk line (PSTN), and the control block 400 is monitored by monitoring the state of the line after completion of the call in the trunk line (PSTN) call. The switch position in the system is controlled by the VoIP position.

Ethernet switch 600 is required for data communication, and is separated from the WAN and LAN parts.

2 is an operational configuration diagram in the system according to FIG. 1.

Referring to the drawings, the operation is described as follows.

The general telephone is connected to the terminals 20 and 20 ', and makes a call by making an Internet phone (VoIP) through the path of the slick 300, or by selecting a PSTN side connected to the terminals 21 and 21'. Selection may be made using the relay 60. At this time, the loop detector 30 connected to the trunk line (PSTN) is connected in series with the subscriber card of the telephone station, and the relay 60 contacts are connected with the telephone, so that when the user lifts the handset (hook off state), the system outputs an output signal. The control unit transmits the data to the DSP. In addition, the ring detector 40 connected in parallel to the trunk line PSTN detects the ring call signal by detecting the ring call signal when the ring detector 40 receives a ring signal from the telephone station regardless of which position the relay 60 is located. Transfer to the system control unit (DSP). The relay drive 60 is driven according to a control signal for selecting a path of the slick 300 side to use the trunk line PSTN side or the Internet phone VoIP in the system controller DSP.

FIG. 3 is a detailed circuit diagram of the gateway of FIG. 1.

As shown, the circuit line drawn from the terminal side (co1, 2) is connected to the control block 400 through the ground terminal. A condenser c87 is connected in parallel to the control block 400, and the control block 400 is also connected to the slick 200. In addition, the detector 500 is connected at the terminal sides co1 and 2. The circuit diagram for the codec 200 is shown to the right of the detector 500.

4 (A) and (B) are flowcharts of a telephone call using an internet telephone phone gateway incorporating a PSTN and VoIP matching circuit according to an exemplary embodiment of the present invention.

Explained according to the flow chart shown, the call procedure of the telephone user C first determines within the receiver internet phone gateway whether the slick is hooked off (700). If it is determined that the hook is off (700) (YES), the relay switch has a procedure for determining the first digit generated by pressing the button of the telephone by the user's operation while switching to the trunk line. If the first digit is a certain digit (#) set arbitrarily, the relay switch switches to the slick side and drives the dial tone at the same time, determining that the call is desired by VoIP. However, if the first digit is not a specific digit (703) (NO), it is determined that the call is desired by the trunk line PSTN, and the call can be executed from the trunk line. When the COS hooks on (YES) to terminate the call, the relay switch returns to the slick side (711).

On the other hand, when the slick is not hooked off, that is, when the slick is on (700) (NO), when a ring is detected on the trunk line (712) (YES), when the VoIP is busy, it notifies the PSTN call or sends a special call. It will drive the tone, and VoIP will run the ring when it is not busy.

5 is a system configuration diagram of a gateway of an Internet phone incorporating a PSTN and a VoIP matching circuit according to an embodiment of the present invention.

First, the initial state will be described. In the gateway 900 incorporating a matching circuit according to the present invention, a relay is connected to a state in which a PSTN call is possible in a power off state. When the user turns on to use the system 900 incorporating the matching circuit according to the present invention, the user enters an idle state. When the power is turned on, it is connected to the SLIC / COMBO 905 by controlling the relay. Therefore, in the idle state, the relay is connected to the VoIP call enabled state.

The following describes the operation status for each of the origination and reception of PSTN and VoIP.

In the case of the PSTN call, the user enters the hook-off state when the user picks up the handset for the purpose of using the telephone. When the hook on / off detector 902 detects the hook off and sends it to the DSP 901, the DSP 901 controls a relay corresponding to the hooked-off telephone so that the switch 907 connects from the VoIP to the PSTN. That is, the switch 907 connected to the SLIC / COMBO 905 is turned off and turned on to the switch 907 connected to the PSTN.

When the user sends the first digit on the telephone button to call the called party, the DSP 907 detects the first digit and does not control the relay if the first digit is not #.

This is the case when a user typically uses a telephone with a PSTN. Therefore, the user can call the called party in the same manner as the conventional telephone call method when using the system with the matching circuit according to the present invention.

If the user calls the called party to end the call and hang up the handset, the phone in the hooked off state is hooked on. When the hook on / off detector 902 detects the hook on and notifies the DSP 901, the DSP 901 recognizes the hook on, controls the corresponding relay, connects to the VoIP from the PSTN, and enters the idle state again. That is, the switch connected to the PSTN is turned off and turned on to the switch 907 connected to the SLIC / COMBO 905.

Accordingly, the user can use the telephone 903 connected to the system 900 incorporating the matching circuit according to the present invention in the same manner as the conventional PSTN telephone using method.

A system having a matching circuit according to the present invention may be used by connecting a plurality of telephones. In the exemplary embodiment of the present invention, a case in which four telephones 903 are connected to each other is illustrated and described, but is not limited thereto.

If one of the four telephones is talking to the PSTN and the telephone is trying to talk to the PSTN through the other telephone, it is preferable to send a busy tone to the telephone to be called. Furthermore, it is more desirable to detect the hook-off of the phone attempting to talk to the PSTN and to make a call by connecting to VoIP.

In the embodiment according to the present invention, the user uses a specific digit # on the telephone button to determine whether to use the PSTN phone or the VoIP phone, but is not limited thereto. That is, in addition to the # digits, the digits that are not confused with a prefix or a station number among Arabic numerals of 0 to 9 may be utilized or * digits.

In the case of a PSTN incoming call, when the ring detector 902 detects a PSTN ring signal and notifies the DSP 901 in an idle state, the DSP 901 rings all phones not currently in use. ) Generates a signal.

When the user picks up the telephone receiver to answer the ringing telephone, the user is hooked off. The DSP 901 detects the hook-off state through the SLIC / COMBO 905, and the DSP 901 controls the relay corresponding to the telephone of which the hook-off state is detected and connects the VoIP to the PSTN by the switch 908. . That is, the switch 907 connected to the SLIC / COMBO 905 is turned off and turned on to the switch 907 connected to the PSTN.

When the called party ends the call with the sender and hangs up, the phone is hooked on. When the hook on / off detector 902 detects the hook on state and notifies the DSP 901, the DSP 901 controls a relay corresponding to the telephone on which the hook on state is detected and connects to the VoIP from the PSTN to idle state. Becomes That is, the switch connected to the PSTN 907 is turned off and the switch connected to the SLIC / COMBO 905 is turned on.

In the case of VoIP call origination, when the user picks up the handset for the purpose of using the telephone, it is hooked off. The DSP 901 detects the hook-off state through the SLIC / COMBO 905, and the DSP 901 recognizes the hook-off and controls the corresponding relay to connect from the VoIP to the PSTN. That is, the switch connected to the PSTN is turned off and the switch connected to the SLIC / COMBO 905 is turned on.

When the user sends the first digit to call the called party, the DSP 901 detects the first digit and, if the first digit is #, controls the relay to connect from the PSTN to the VoIP. That is, the switch connected to the SLIC / COMBO 905 is turned on and turned off to the switch connected to the PSTN.

This is the case when a user uses a VoIP telephone using a system incorporating a matching circuit according to the present invention. The user can call the called party using a VoIP phone by sending # to the first digit.

The hook-on signal occurs when the user calls the called party to end the call and hang up the handset. The DSP 901 recognizes the hook on and enters the idle state again. In this case, no separate relay control is necessary. Thus, the user can use a VoIP phone by sending # to the first digit.

In the case of the VoIP incoming call, when the DSP 901 recognizes the VoIP incoming call in the idle state, a ring signal is generated to the corresponding telephone.

To answer the ringing telephone, the user picks up the phone (hooks off) and talks. In this case, the DSP 901 does not need to perform separate relay control.

When the called party ends the call with the sender and hangs up the phone (hooks on), the call is ended and put back into the idle state. In this case too, the DSP 901 does not need to perform relay control.

The terms used to describe the embodiments of the present invention above are used for the purpose of describing the present invention, and are not used to limit the scope of the present invention as defined in the claims or the claims.

As described above, the present invention connects one or more ordinary telephones to a PSTN telephone or

By selectively using as a VoIP phone, it is possible to secure a cheap and stable call quality.

In addition, it is possible to automatically switch to the VoIP phone and PSTN phone system, even if the power is turned off to the personal computer has the effect that can be called by phone to phone.

Claims (7)

In the Internet phone gateway system with PSTN and VoIP matching circuit, Ethernet switch divided into WAN and LAN required for data communication; A detector connected to the trunk line to detect a ring signal in an idle state and a hook on / off state in a busy state; A control block for monitoring a ring signal detected by the detector and a hook on / off state to drive a switch for connecting a voice path to VoIP and a PSTN; A slick converting two-wire analog audio signals output from the DSP into four wires, or converting four-wire analog audio signals into two wires and performing functions such as battery feeding and ringing, ring trip, and hook on / off detection; DTMF tone transfer unit for transmitting DTMF tones in the absence of feeding; A codec for converting the analog voice signal converted from the slick into four wires to PCM data and transmitting the signal to a DSP or converting the PCM data into four wire analog voice signals to perform compression and restoration of the voice signal for an Internet phone; And And a digital signal processor (DSP) connected to the codec and the Ethernet switch and controlling a signal for an overall processor or data transmission and an interface; and an internet phone gateway system having a PSTN and a VoIP matching circuit. The control block of claim 1, wherein the control block switches a path by relay driving under DSP control according to a loop detector for recognizing a trunk line (PSTN) state and a ring detector for detecting a ring signal which is a call signal. Internet phone gateway system with integrated PSTN and VoIP matching circuits. The method of claim 1, wherein the relay position in the initial state of the control block is a PSTN callable position when the relay is powered off, and when the gateway is powered on, the VoIP call is available in an idle state. Internet phone gateway system with built-in PSTN and VoIP matching circuit. The Internet phone gateway system according to claim 1, wherein the phone connected to the gateway can use one or a plurality of phones capable of making a VoIP phone call or a PSTN phone call. The Internet phone gateway system according to claim 1, wherein in the outgoing call using the gateway circuit, the first dial digit is used to determine whether to make a PSTN and a VoIP call. 6. The method of claim 5, wherein the first dial digit is detected and controlled by a DSP and relayed to a VoIP or PSTN call when the first digit is not detected, and an Arabic numeral from 0 to 9 other than # digits is confused with a prefix or a station number. Internet phone gateway system with built-in PSTN and VoIP matching circuits, utilizing unused digits or digits. The Internet phone gateway system as set forth in claim 1, wherein the DTMF tone transfer unit includes a condenser element connected in parallel.