KR20080086803A - Multi communication processing device by ip phone - Google Patents

Abstract

A multi-party call processing device is provided to allow a participant of multi-party communication to make a user located in a foreign country or in a remote area participate in multi-party communication when one of the participants is connected with a VoIP(Voice over Internet Protocol) phone of the user. A VoIP phone connection unit(110) is connected with multiple VoIP phones that perform communication via a voice packet, and is connected with a transmitter/receiver connection terminal of the VoIP phones to receive an audio signal from the VoIP phones. An audio signal processing unit(120) provides the audio signal received via the VoIP phone connection unit to an audio receiving terminal of a neighbor VoIP phone, and prevents the audio signal from being fed back from the audio signal provided side. The VoIP phone connection unit is connected with a transmitter/receiver connection terminal provided in each VoIP phone, and is connected with a speaker and a microphone of the transmitter/receiver connection terminal.

Description

Multi communication processing device by IP phone

1 is a diagram illustrating an example in which a VOIP phone is connected to a network switch;

2 conceptually illustrates a conventional VOIP phone connection method;

3 is a conceptual diagram of a connection of a multiparty call processing apparatus according to an embodiment of the present invention;

4 is a conceptual diagram of a connection of a multiparty call processing device according to another embodiment of the present invention;

5 is a block diagram of a multi-party call processing apparatus according to an embodiment of the present invention;

6 conceptually illustrates an internal signal flow of an audio splitter;

7 is a conceptual block diagram of a personal conference processing device according to another embodiment of the present invention;

8 is a block diagram according to an example of the call circuit shown in FIG. 7, and

9 illustrates an example of implementing one-way transmission using an amplifying circuit.

* Description of the symbols for the main parts of the drawings *

110: connection unit 120: audio signal processing unit

121: switch 122: audio splitter

123: switch controller 124: key control panel

130: display device 140: amplifier

150: handset 151: speaker

152: microphone

The present invention relates to a multi-party call processing apparatus, in particular, a multi-party call using a VOIP phone, without the need for expensive servers, and gateways for the multi-party call, a multi-party call using a single network line When performing the present invention, the present invention relates to a multi-party call processing apparatus that can handle this at a single line cost.

With the development of networks, traditional landline telephones are gradually turning to VOIP phones. The VOIP phone transmits voice packets to the other terminal (VOIP phone or computer) instead of voice signal in the almost free network network. The rate is greatly reduced. Since VOIP phones transmit and receive voice packets to and from each other via ADSL, cable, and other optical LANs using telephone lines, the multi-party call through the VOIP phone is usually made through a server. This will be described with reference to FIGS. 1 and 2.

First, FIG. 1 shows an example in which a VOIP phone is connected to a network switch.

As shown, in a typical environment each VOIP phone 10a-10n is connected to the network switch 20 and used. The network switch 20 is a device such as a router, a hub, and a router, and enables one VOIP phone 10a to 10n to use one authorized IP. Although only the VOIP phones 10a to 10n are shown in the network switch 20 in the drawing, it is common for the network switch 20 to further add a computer, a printer, and various other network devices in addition to the VOIP phones 10a to 10n. . This connection environment is most common in an office environment where a single authorized IP is shared by several people. In the figure, the network switch 20 is connected to the Internet using one PUBLIC IP, and the network switch 20 assigns one public IP to several VOIP phones 10a to 10n. If each VOIP phone 10a to 10n wants to directly connect with another VOIP phone outside the network switch 20, the VOIP phones 10a to 10n transmit voice packets to the IP of the other VOIP phone and The voice packet must be received from the VOIP phone. However, since the network switch 20 shows only one public IP when viewed from the outside, the external VOIP phone can identify each VOIP phone 10a to 10n connected to the network switch 20. none. Therefore, each VOIP phone 10a to 10n is difficult to make a direct call with another external VOIP phone in an environment outside the internal network of the network switch 20, and more difficult to make a multi-party call. Therefore, the existing VOIP phone must be connected via a server, as shown in FIG.

2 conceptually illustrates a conventional VOIP phone connection method.

As shown, each VOIP phone 10a-10n is networked with the server 50 and transmits a voice packet to the desired destination via the server 50. The server 50 analyzes the voice packet transmitted from each VOIP phone 10a to 10n to determine a destination to which the voice packet should be transmitted, transmits the voice packet to the corresponding destination, and any one of the VOIP phones 10a to 10n. If a multi-party call is desired, the packet transmitted from the VOIP phone (any one of 10a to 10n) is also transmitted to another VOIP phone (not shown) to which the multi-call is desired. Therefore, as the number of VOIP phones participating in the conference increases, the load on the server 50 increases, which results in a decrease in voice call quality.

On the other hand, when a conventional VOIP phone has a conference call with several people, the billing is determined according to the number of VOIP phones participating in the conference. For example, assuming that VOIP phone A has a multiparty call with VOIP phone B, VOIP phone C, and VOIP phone D, VOIP phone A sends and receives voice packets with VOIP phone B, VOIP phone C, and VOIP phone D, respectively. 3 times as much as 1: 1 communication. Accordingly, the VOIP phone A must pay the usage fee of the voice packet in proportion to the number of VOIP phones participating in the conference. Accordingly, the present applicant is to reduce the billing according to the number of people participating in the multi-party call, and propose a multi-party call processing apparatus that allows a large number of users to make a multi-party call without going through a separate server for the multi-party call.

Accordingly, an object of the present invention is to provide a multi-party call processing apparatus that does not require a separate server for mediating a multi-call when performing a multi-party call through a VOIP phone. In addition, another object of the present invention is to provide a multi-party call processing apparatus such that even if the number of people participating in the multi-party call increases, the call charge does not increase by more than a certain amount. In addition, another object of the present invention is to provide a multi-party call processing apparatus that can perform a multi-party call while using as it is, without modifying the existing VOIP phone.

According to the present invention, the above object is connected to a plurality of VOIP phones for making a call through a voice packet, a VOIP phone connection unit connected to a handset connection terminal of the VOIP phone to receive an audio signal from the VOIP phone, and the An audio signal received through a VOIP phone connection unit is provided to an audio receiving terminal of a neighboring VOIP phone, and is provided by a signal processor that prevents the audio signal from being fed back to the side providing the audio signal.

Preferably, the VOIP phone connection part is connected to a handset connection terminal provided in each VOIP phone, and is connected to a speaker and a microphone of the handset connection terminal.

The VOIP phone connection unit is connected to a handset connection terminal provided in the analog phone, and is preferably connected to a speaker and a microphone provided in the handset connection terminal of the analog phone.

The signal processor may provide an audio signal received through the analog phone to an audio receiving terminal of any one of a neighboring VOIP phone and a neighboring analog phone.

Preferably, the apparatus further includes an amplifier provided between the VOIP phone connection unit and the signal processing unit and amplifying the audio signal.

It is preferable that the said VOIP phone connection part is connected with said each VOIP phone using RJ-11 terminal.

The signal processing unit transmits the audio signal received through the VOIP phone connection unit in one direction only to another VOIP phone.

The signal processor, when the audio signal is transmitted from any one of the plurality of VOIP phones, the repeater for one-way transmission of the audio signal to the other VOIP phone except the VOIP phone that transmitted the audio signal, and each VOIP phone It is preferable to include a recording unit for recording the shared audio signal.

Preferably, the apparatus further includes a memory for storing the audio signal.

Preferably, the audio signal processor includes a key controller and a switch controller configured to set whether to allow a communication channel for each of the plurality of VOIP phones in response to an operation of the key controller.

Preferably, the apparatus further includes a display device indicating whether the communication channel is allowed.

According to the present invention, the above object is connected to a plurality of network lines, and each network line is provided with a call circuit for a VOIP phone, each of which performs a telephone call using a voice packet, a display device, a keypad, and the plurality of A call circuit selected by the keypad of the call circuits is displayed on the display device, and a telephone number input through the keypad is provided to the selected call circuit to achieve a call connection.

The control unit may further include an audio signal processor that provides an audio signal output from each of the communication circuits to another communication circuit except for a communication circuit that outputs the audio signal.

The keypad is jointly used by each call circuit to request a call connection, and each call circuit, when selected by the keypad, may perform call connection according to a telephone number input through the keypad.

Preferably, the apparatus further includes a recording unit for recording and storing the audio signal shared by each VOIP phone.

The controller may display an identifier for the call circuit selected by the keypad on the display device, and the identifier may be any one of numbers, letters, figures, and images.

Preferably, the apparatus further comprises an analog call circuit connected to an analog phone to perform a telephone call using a voice signal, wherein the control unit connects a call using any one of the analog call circuit and the VOIP phone call circuit. Can be derived.

The above object is connected to a plurality of VOIP phones having a handset connection terminal, and a handset connection terminal of each VOIP phone according to the present invention, when any one of the plurality of VOIP phones transmits an audio signal, It is achieved by a system having a multiparty call processing device for providing an audio input of a VOIP phone to share the audio signal with the plurality of VOIP phones.

According to the present invention, the above object is connected to a plurality of VOIP phones belonging to the first call area, and the audio signal transmitted by any one of the VOIP phones is provided to another VOIP phone to belong to the first call area. The audio signal transmitted by any one of the VOIP phones belonging to the first multi-call processing device and the plurality of VOIP phones belonging to the second call area and shared by each VOIP phone to be transmitted are the audio signals transmitted by one of the VOIP phones. And a second multiparty call processing device which is provided to another VOIP phone belonging to the second call area and shared to each VOIP phone belonging to the second call area, and any one of the VOIP phones belonging to the first call area. Performs a multi-party call with the VOIP phone belonging to the second call area, the audio signal shared in the first area is provided to the second call area, and the audio signal shared in the second call area is Provided by the first call area is achieved by a multi-party call processing system in which the audio signal of the first call area and the audio signal of the second call area are shared with each other.

The first conference call processing device and the second conference call processing device are connected to at least one analog phone, and the analog phone is connected to any one of the VOIP phone and the analog phone that are adjacent to each other in the call area. Audio signals can be shared.

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

3 is a conceptual diagram illustrating a connection of a multiparty call processing apparatus according to an embodiment of the present invention.

The multi-party call processing apparatus 100 according to the present embodiment allows a multi-party call to be simultaneously made with a plurality of other VOIP phones 51 to 56 using a handset (not shown) connected to the multi-party call processing apparatus 100. . VOIP phones 61 to 66 for call connection are disposed between the other VOIP phones 51 to 56 and the conference call processing device 100. The user calls the other party's VOIP phone (e.g., 51 to 56) that wishes to have a multi-party call using the VOIP phones 61 to 66, and the VOIP phones 51 to 56, which have successfully connected to each VOIP phone (61 to 66), And voices of the user and the user of other VOIP phones 61 to 66 are shared with each other in the handset.

On the other hand, the multi-party call processing apparatus 100 in Fig. 3 is connected to a plurality of VOIP phones (61 to 66), but can also be connected to an analog phone (usually a phone that provides a communication service by connecting to a PSTN network). have.

The multi-party call processing apparatus 100 according to the present invention is connected to an audio signal or a VOIP phone 61 to 66 that is transmitted and received through a handset of an existing analog phone, which exchanges an analog signal using a telephone line of a public network (PSTN). Audio signals transmitted and received through the handset can be shared with each other. This is because the multi-party call processing apparatus 100 according to the present invention shares an audio signal with a VOIP phone (or an analog phone) participating in the multi-party call using an analog audio signal that is finally interfaced with a user. Although not shown in the drawings, the multi-party call processing apparatus according to the present invention may be connected to a mobile phone capable of connecting a microphone and a speaker, and when using a mobile phone, the multi-party call processing apparatus 100 may transmit and receive an analog signal with the mobile phone. Can be connected with the cable. For example, in Korea, the conference call processing device 100 and a mobile phone may be connected to each other through a standard 24-pin connector or another pin connector for connecting a mobile phone and a heterogeneous device, and receives an audio signal (analog signal) provided by the mobile phone. It can be shared with other VOIP phones, or can provide audio signals shared by other VOIP phones to the microphone input of the mobile phone through a 24-pin standard connector or other pin connector.

Here, it is not limited to FIG. 3 that the multi-party call processing apparatus according to the present invention is connected to an analog phone and a mobile phone in addition to the VOIP phone. The same applies to the whole of the present invention. The present invention is not limited to the communication method of the equipment (VOIP phone, analog phone, mobile phone, etc.) for the telephone call. This is due to the multi-party call processing apparatus 100 according to the present invention to share the signal of the final interface of the equipment, that is, the speaker and the microphone to interface with the user. Therefore, in the following description of the present invention including FIG. 3, even if the Applicant does not separately mention an analog phone (or a mobile phone), the multi-party call processing device is connected to the analog phone and the mobile phone to connect peripheral equipment (VOIP phone, analog phone). , And mobile phones, etc.) can be shared in advance.

4 is a conceptual diagram illustrating a connection of a multiparty processing apparatus according to another embodiment of the present invention.

In the multi-party call processing apparatus according to the present embodiment, any one of the VOIP phones (for example, reference numeral 56) performing the multi-party call in the area A (the first call area) is the multi-party call in the other area B (the second call area). When connected to a VOIP phone (eg, reference numeral 71) that performs the above, an embodiment in which both the VOIP phones 51 to 56 in the A area and the VOIP phones 71 to 76 in the B area can share voices with each other It is about. The multi-party call processing apparatus 100 for implementing a multi-party call in the area A is implemented such that the VOIP phones 51 to 56 connected to each other share audio signals, and the multi-party call processing device for implementing the multi-party call in the area B ( In addition, the VOIP phones 71 to 76 connected to each other share an audio signal with each other.

Therefore, if a VOIP phone (e.g., 56) participating in the conference in the area A makes a two-way call with a VOIP phone (e.g. 71) participating in the conference in the area B, the VOIP phone participating in the two areas (A, B) 51 to 56 and 71 to 76 can share audio signals shared by each other. A multi-party call processing apparatus for enabling this will be described with reference to FIG. 5.

The illustrated multiparty processing apparatus includes a connection unit 110, an audio signal processing unit 120, an amplifier 140, a handset 150, and a display device 130, and includes a handset 150 for a user. The handset 150 includes a microphone 152 and a speaker 151 for transmitting the user's voice to the outside.

The connection unit 110 is connected to the handset connection terminal of the VOIP phones 61 to 66 shown in FIG. VOIP phones that transmit and receive audio signals via voice packets contain network interface modules and digital circuits such as DSPs (not shown), but the final interface with the user is through analog devices (speakers, microphones). do.

When a user transmits his / her voice through a microphone to a VOIP phone, the VOIP phone converts it into a digital signal and then converts it into a network packet.

When the voice packet is transmitted from another VOIP phone, the VOIP phone parses the voice packet to extract the voice packet, and decodes the extracted voice packet to play the speaker. Applicant takes advantage of this feature of the VOIP phone, that is, the final interface of each VOIP phone is treated as analog. Each VOIP phone (51 to 56, 61 to 66, 71 to 76, and 81 to 86) is provided with at least two terminals, one of which is an RJ-45 connection terminal connected to a network switch (not shown) such as a router. And the other is an RJ-11 terminal connected to a handset (built-in microphone and speaker). The connection unit 110 is connected to the RJ-11 terminal of each VOIP phone. Accordingly, the connection unit 110 may receive an analog audio signal transmitted from each VOIP phone 51 to 56, 71 to 76, and may transmit an analog audio signal to each VOIP phone 51 to 56, 71 to 76. . Hereinafter, the VOIP phone will be described with reference to FIG. 3, and reference numerals 71 to 76 and 81 to 86 of FIG. 4 will be cited only when necessary.

The audio signal processor 120 provides an audio signal of each VOIP phone 61 to 66 received through the connection unit 110 to the handset 150 and the microphone input terminal of each VOIP phone 61 to 66.

For example, when the audio signal is received from the VOIP phone 61, the audio signal processing unit 120 provides an audio signal of the VOIP phone 61 to the microphone input terminal of the VOIP phones 62 to 66, thereby providing the respective VOIP phones 62 to 62. 66) share the audio signals of the VOIP phone 61 together. At this time, the audio signal processing unit 120 provides the audio signal of the VOIP phone 61 to the handset 150, so that other VOIP phones 62 to 66 and the user of the handset 150 participate in the conference call. The audio signal transmitted at 61 can be shared. This applies to all other VOIP phones 62 to 66. In addition, the audio signal processor 120 may selectively turn on / off one of the VOIP phones participating in the conference call (any one of 61 to 66). This may be used when a specific person occupies most of the call time, and the user may process it by using a key manipulation unit (not shown) provided in the conference call processing device.

Preferably, the audio signal processing unit 120 includes a key operation unit 124, a switch 121, a switch controller 123, a recording unit 125, a memory 126, and an audio distributor 122.

The key operation unit 124 has a button corresponding to each of the VOIP phones 61 to 66, and the switch controller 123 has each of the VOIP phones 61 to 66 and the audio splitter 122 depending on whether each button is selected. Turns the signal path between) on and off. The VOIP phone (for example, 61) whose signal path is blocked cannot transmit audio signals to other VOIP phones 62 to 66, and the other VOIP phones 62 to 66 do not hear the audio signal transmitted from the VOIP phone 61. . This is useful when the multi-party call processing apparatus according to the present invention uses the telephone for English education. For example, assuming that an English teacher is located in a multi-party call processing apparatus according to the present invention, and that each VOIP phone 61 to 66 is connected to a learner's VOIP phone (for example, reference numerals 51 to 56), the teacher is assigned to each VOIP phone 51. 56) through a multi-party call with a learner connected to the caller, and when the learner's speech is too long or is too far from the learning subject, the key control unit 124 is used to receive an audio signal transmitted from the VOIP phone of the learner. You can block. At this time, the specific learner can hear the contents of the call of the other learner and the teacher, but cannot transmit his / her remarks to others.

The switch 121 is operated on-off by the switch controller 123 in response to the key manipulation unit 124, and may be implemented as a relay and an electronic switch.

The recording unit 125 records audio signals shared by the persons participating in the conference. The recording unit 125 records the audio signal transmitted to the handset 150 connected to the conference processing apparatus 100 and the audio signal transmitted from the handset 150 to each VOIP phone 51 to 56. The recording unit 125 may record a conversation of all persons participating in the conference by recording an audio signal received by the handset 150 or transmitted from the handset 150 to the VOIP phones 51 to 56. This is due to the fact that the audio signal is shared through the handset 150 in the multiparty processing apparatus 100 according to the present invention. The recording unit 125 stores the audio signal as it is or compresses the memory 126. The memory 126 may be any one of a flash memory and a hard disk.

The audio distributor 122 transmits the audio output signal of each VOIP phone 61 to 66 connected to the connection unit 110 to the neighboring VOIP phone so that the handset 150 and each VOIP phone 61 to 66 are connected to each other. Allow audio signals to be shared. An internal wiring method of the audio splitter will be described with reference to FIG. 6 together.

6 conceptually illustrates the internal signal flow of the audio splitter 122.

First, "T" is an abbreviation of Transmitter and becomes the microphone part of the handset. The same applies to T1, T2, and T3.

Reference symbol "R" stands for Receiver, which is the speaker part of the handset, which applies equally to R1, R2, and R3.

In addition, R1 and T1 means a handset composed of a microphone and a speaker.

In the figure, when the audio signal transmitted from the VOIP phone 51 is transferred to the personal conference processing apparatus according to the present invention, the reception port R1 of the personal conference processing apparatus receives it and transmits it to the relay A1. The repeater A1 is a device for transmitting the audio signal only in one direction, and transmits the audio signal transmitted from R1 to T2 and T3. The simplest method of configuring the repeater A1 is a diode which forms a signal path in the direction of R1 to T2. In addition, an amplifier circuit is formed by transistors, and the signal is transmitted in one direction to T2 or T3, or an integrated circuit is formed. Of course, the same effect can be implemented using. Here, a method of transmitting the audio signal in one direction using the amplifying circuit will be described with reference to FIG. 9.

9 illustrates an example of implementing one-way transmission using an amplifying circuit.

The illustrated amplification circuit shows an example in which the repeaters A1, A2, and A3 shown in FIG. 6 are most simply and inexpensively implemented. In the figure, an audio signal is applied to the base terminal of the transistor Q1, and the audio signal is amplified at the collector terminal and output to the resistor R3. That is, the audio signal may flow only in the collector direction at the base of the transistor Q1. If this is applied to the relay A2 of FIG. 6, the audio signal received at R2 of FIG. 6 is connected to the base terminal of the transistor Q1, and T1 and T3 are connected to the collector terminal of the transistor Q1. The illustrated amplification circuit is one of the most inexpensive and simple ones that can be implemented by the present applicant. In addition, the amplifying circuit shown may be implemented differently using various circuits and integrated circuits.

Thus, the audio signal transmitted from the VOIP phone 51 is provided to R1, A1, T2, T3, and R, and T2 transmits the audio signal received through A1 to R of the VOIP phone 52, where T3 is Since the audio signal is transmitted to R of the VOIP phone 53, all the VOIP phones 51 to 53 and handsets T and R connected to the personal conference processing device according to the present invention are transmitted from the VOIP phone 51. Shared audio signals.

Similarly, the audio signal transmitted from the VOIP phone 52 is provided to T1 and T3 through the repeater A2, and directly to R, so that all VOIP phones 51 to 53, and handsets connected to the personal conference processing device. (T, R) may share the audio signal transmitted from the VOIP phone 52. Here, the repeaters A1, A2, and A3 do not transmit the signal in the direction in which the signal is transmitted, but as shown in the figure, transmit the audio signal only in the direction of the arrow. This is to prevent the audio signal from echoing due to the howling phenomenon when the audio signal reverses. In the present invention, a repeater using a diode and an amplifying circuit is illustrated, but this is merely an example, and any type of circuit for transmitting an audio signal only in one direction may be used.

Figure 7 shows a block diagram of a personal multi-party call processing apparatus according to another embodiment of the present invention.

The illustrated personal conference processing apparatus includes a call circuit unit 210, a control unit 220, a display device 230, a key operation unit 240, and a handset 250.

The call circuit unit 210 includes a plurality of call circuits 210a to 210h having the same shape integrated on one substrate, and each call circuit 210a to 210d includes a circuit board embedded in a conventional VOIP phone and its function. , And configuration are the same.

In addition to the circuit board embedded in the VOIP phone, the circuit board of the analog phone may be applied to the call circuit (for example, any one of 210a to 210d). As mentioned through FIG. 3, it has been found that the multi-party call processing apparatus of the present invention can be connected to an analog phone and a mobile phone in addition to the VOIP phone. If the call circuits 210a to 210d are a board of an analog phone, the board is provided with an RJ-11 connection terminal for connecting to the PSTN network, and a circuit board (for example, 210a to 210d) is provided through the RJ-11 connection terminal. One) this PSTN can be connected.

Each call circuit 210a to 210d includes a processor, a DSP, and a codec for network connection. Detailed descriptions of the call circuits 210a to 210d will be described with reference to FIG. 8.

8 is a block diagram illustrating an example of the call circuit illustrated in FIG. 7.

The illustrated call circuit includes a codec 205, a DSP 203, a processor 202, a memory 204, and a network interface module 201.

The codec 205 converts an audio signal received through the microphone 207 into a digital signal, and digitally-analog converts the digital audio signal received through the DSP 203 to the DSP 203. The network interface module 201 is connected to a network switch (not shown) through an RJ-45 jack, and is connected to an external VOIP phone (for example, reference numerals 71 to 76 of FIG. 4) through the network switch. In this case, the network interface module 201 may be connected to a server (not shown) that arranges a call, instead of being directly connected to an external VOIP phone (for example, reference numerals 71 to 76 of FIG. 4). This means that the conference processing apparatus according to the present invention does not need a server, but does not need to be connected to a server having a function of arranging a conference.

The processor 202 instructs the recording and dial-up of an audio signal in response to the selection of the key manipulation unit 240, and issues a command required for the DSP 203, or the codec 205, and the network interface module 201. The voice packet received through the N / W is provided to the DSP 203. The processor 202 also converts the digital audio signal to be transmitted by the DSP 203 into a network packet signal and provides it to the network interface module 201.

The DSP 203 extracts digital voice packets from packets received through the processor 202. The DSP 203 decompresses the extracted voice packet and provides it to the codec 205. In contrast, when the digital audio signal is transmitted from the codec 205, the digital audio signal is compressed and provided to the processor 202. G.711, PCM, G.726, and various other compression algorithms may be applied to the compression of the digital audio signal.

The memory 204 provides the temporary workspace required by the DSP 203 and the processor 202. For example, the memory 204 may provide a workspace that the DSP 203 needs for conversion of the digital audio signal when compressing or decompressing the digital audio signal.

The network interface module 201 is connected to a network switch (not shown) or the Internet through an RJ-45 terminal.

The control unit 220 connects the call control unit 240 with any one of the call circuits 210a to 210d selected by the key control unit 240 among the call circuits 210a to 210d. The key operation unit 240 includes a numeric key, special character keys (#, *), and a call circuit selection key (not shown), and the selection key is one or a number of call circuits (for example, 210a to 210d). Can be prepared accordingly. If there is only one selection key, the call circuits 210a to 210d may be selected according to the number of times the selection key is pressed. Preferably, the communication circuits 210a to 210d selected by the selection key are displayed on the display apparatus 230. In this case, it is preferable that an identifier corresponding to each call circuit 210a to 210d is displayed on the display apparatus 230. The identifier is provided to the user to easily distinguish and use each communication circuit by assigning numbers, letters, figures, or images to the communication circuits formed on the electromagnetic plate. Accordingly, a user who wants to make a multi-party call can easily recognize which call circuit (one of 210a to 210d) he or she has selected, and perform a call connection through the call circuit (210a to 210d) selected by the user. Can be. After the user selects one of the call circuits 210a to 210d, the user inputs a desired telephone number by using a numeric key provided in the key control unit 240 to perform a call connection. The circuit (one of 210a to 210d) is selected again, and call connection is performed in the same manner. That is, unlike the embodiment illustrated in FIG. 7, the embodiment illustrated in FIG. 7 does not need to connect several external VOIP phones, and uses one key operation unit 240 to make a call connection for a conference call. There is an advantage to handle.

Preferably, the controller 220 includes an audio signal processor 221. As illustrated in FIG. 6, the audio signal processor 221 sets an audio signal path between another VOIP phone (for example, reference numerals 51, 52, and 53) and a personal conference processing device connected for a conference call. . In addition, the audio signal processor 221 receives an audio signal received through the ports 210e to 210h when the number of external VOIP phones (for example, reference numerals 51, 52, and 53) connected to the ports 210e to 210h increases. In order to prevent the signal output through the and ports 210e to 210h from being attenuated, it may serve to amplify the received audio signal and the output audio signal.

The handset 250 is connected to the control unit 220, so that a user who wants a multi-party call can use it to talk to a plurality of people. Here, the user who wants the conference call does not need a separate VOIP phone, and can perform the conference call using only the handset 250.

As described above, the multi-party call processing apparatus according to the present invention does not require a separate server for implementing the multi-party call, and can easily implement the multi-party call only by connecting with the handset terminal of each VOIP phone. No other conferencing equipment so far has ever implemented such a simple conferencing.

In addition, the multi-party call processing apparatus according to the present invention, if any one of the participants participating in the multi-party call is connected to the user-side VOIP phone belonging to a foreign or distant area, the participant participating in the multi-party call belongs to a foreign or distant area. You can join the user together in the conference. At this time, when a user belonging to a foreign country or a distant region makes a conference call with a plurality of other persons through the conference call processing apparatus according to the present invention, all users of both sides can make a conference call at one line connection cost. This can dramatically reduce the cost incurred by the multilateral currency.

In addition, the present invention satisfies all the conditions for the conference by simply connecting the VOIP phone to the conference. Even if you are not a skilled technician in communication or network, anyone can easily set up an environment for conferencing.

Claims (20)

It is connected to a plurality of VOIP phones that make calls through voice packets. A VOIP phone connection unit connected to a handset connection terminal of the VOIP phone to receive an audio signal from the VOIP phone; And A VOIP phone comprising an audio signal received through the VOIP phone connection unit to an audio receiving terminal of a neighboring VOIP phone, and a signal processing unit for providing the audio signal so that the audio signal is not fed back. Multi-currency processing device using. The method of claim 1, The VOIP phone connection portion, It is connected to the handset connection terminal provided in each VOIP phone, Multi-party call processing apparatus using a VOIP phone, characterized in that connected to the speaker and the microphone of the handset connection terminal. The method of claim 2, The VOIP phone connection portion, It is connected to the handset connection terminal provided in the analog phone, Multi-party call processing apparatus using a VOIP phone, characterized in that connected to the speaker, and a microphone provided in the handset connection terminal of the analog phone. The method of claim 3, The signal processing unit, And an audio signal received through the analog phone to one of the neighboring VOIP phones and the audio receiver terminal of the neighboring analog phone. The method of claim 2, It is provided between the VOIP phone connection unit and the signal processing unit, And a multiplier for amplifying the audio signal. The method of claim 1, The VOIP phone connection portion, Multi-call processing apparatus using a VOIP phone, characterized in that connected to each of the VOIP phone using an RJ-11 cable. The method of claim 1, The signal processing unit, And a one-way transmission of an audio signal received through the VOIP phone connection unit only to another VOIP phone. The method of claim 1, The signal processing unit, A repeater for unidirectionally transmitting the audio signal to another VOIP phone except for the VOIP phone which has transmitted the audio signal when any one of the plurality of VOIP phones is transmitted; And And a recording unit for recording an audio signal shared by each of the VOIP phones. The method of claim 8, Memory for storing the audio signal; Multi-party call processing apparatus using a VOIP phone further comprises. The method of claim 1, The audio signal processor, Key operation unit; And And a switch controller configured to set whether to allow a communication channel for each of the plurality of VOIP phones in response to an operation of the key manipulation unit. 2. According to claim 1, And a display device for displaying whether or not the communication channel is allowed. A plurality of call circuits for VOIP phones connected to a plurality of network lines, each of which performs a telephone call using voice packets, A display device and a keypad; And a controller configured to display a call circuit selected by the keypad among the plurality of call circuits on the display device and to provide a telephone number input through the keypad to the selected call circuit to induce a call connection. Multiparty call processing device using a VOIP phone. The method of claim 12, The control unit, For an audio signal output from each of the call circuits, And an audio signal processing unit provided to other communication circuits except for a communication circuit for outputting the audio signal. The method of claim 12, The keypad, Each call circuit is jointly used to request a call connection, Each call circuit, When selected by the keypad, a multi-party call processing apparatus using a VOIP phone characterized in that to perform a call connection according to the telephone number input through the keypad. The method of claim 12, And a recording unit for recording and storing the audio signal shared by each of the VOIP phones. The method of claim 12, The control unit, An identifier for the call circuit selected by the keypad is displayed on the display device, The identifier is Personal multi-currency processing apparatus, characterized in that any one of numbers, letters, figures, and images. The method of claim 12, An analog call circuit connected to an analog phone to perform a telephone call using a voice signal; The control unit, And a VOPI phone, wherein the call connection is induced using any one of the analog call circuit and the VOIP phone call circuit. A plurality of VOIP phones having handset connection terminals; And It is connected to a handset connection terminal of each VOIP phone, and when any one of the plurality of VOIP phones transmits an audio signal, it is provided to an audio input terminal of another VOIP phone so that the plurality of VOIP phones share the audio signal. Conference call processing system using a VOIP phone, comprising: a conference call processing device. A first connection connected to a plurality of VOIP phones belonging to a first call area, and providing an audio signal transmitted from one of the VOIP phones to another VOIP phone and sharing the same with each VOIP phone belonging to the first call area; Conference call processing apparatus; And The audio signal transmitted by any one of the VOIP phones belonging to the second call area is connected to a plurality of VOIP phones belonging to the second call area, and is provided to another VOIP phone belonging to the second call area. And a second multiparty call processing device to be shared to each VOIP phone belonging to the two call areas. When one of the VOIP phones belonging to the first call area has a multi-party call with the VOIP phone belonging to the second call area, the audio signal shared in the first area is provided to the second call area. And the audio signal shared in the second call area is provided to the first call area so that the audio signal of the first call area and the audio signal of the second call area are shared with each other. . The method of claim 19, The first conference call processing device and the second conference call processing device, Connected with at least one analog phone, The analog phone, And the audio signal is shared with any one of a neighboring VOIP phone and an analog phone in a call area belonging to each other.

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