JP6011793B2 - Relay device and communication system - Google Patents

Description

  The present invention relates to a relay apparatus and a communication system for relaying communication between a radio and a telephone via a network.

  Conventionally, a system has been proposed in which a wired telephone line is connected to a wireless device (master device) via a relay device so that communication between the wireless device (slave device) and the telephone can be performed wirelessly. For example, Patent Document 1). In the system described in Patent Document 1, when a voice signal instructing a call is input from the slave unit, the call message is set to be transmitted to one telephone number stored in advance.

JP 2011-135290 A

  However, since the system described in Patent Document 1 can make a call only to one phone number stored in advance, one of a plurality of phone numbers is selected and a call is made to that phone number. Can not do it. In addition, since a recent digital transceiver has a small number of buttons, there is a problem that it is difficult to directly input a telephone number from the transceiver.

  An object of the present invention is to provide a relay device and a communication system capable of making a call by determining a telephone number of a call destination from destination information input from a wireless device.

The relay apparatus of the present invention is connected to a network that is a data communication network, and is connected to a network interface that communicates with a network device including a telephone connected to the network, and a relay radio, and the relay radio A radio interface for receiving a voice signal embedded with destination information having the same configuration as the call information used for selective call in communication between radios, from a radio of a slave unit communicating with the relay radio; When a voice signal in which the destination information is embedded is received from the radio interface, a call destination telephone number is determined based on the destination information, and a call message is generated with the determined telephone number as a destination. And a control unit for transmitting via the network interface.

  In the present invention, when a voice signal is input from the radio of the slave unit via the radio interface, the telephone number of the call destination is determined based on the destination information embedded in the voice signal. Then, a call message having the telephone number as destination information is sent to the network. Thus, since the telephone number is determined based on the destination information received from the wireless device, the telephone number can be switched depending on what kind of destination information is embedded in the wireless device, and the degree of freedom of the call destination is increased. be able to.

  In the above invention, the control unit includes a table in which the destination information and the telephone number are associated with each other, and when the control unit receives an audio signal including the destination information from the relay radio, the table is The telephone number of the call destination corresponding to the destination information may be determined with reference to the destination information.

  In the above invention, the radio interface may transmit and receive the audio signal by packet communication with the relay radio.

A communication system according to the present invention is a communication system including the relay device, a relay radio connected to the radio interface of the relay device, and a slave radio that communicates with the relay radio. The wireless device transmits an audio signal in which destination information having the same configuration as the call information used for selective calling in communication between the wireless devices is embedded, and the relay wireless device is transmitted by the wireless device. An audio signal is received and input to the relay device. The relay device inputs the audio signal, determines a telephone number based on destination information embedded in the audio signal, and uses the telephone number as a destination. A call message is created and sent to the network.

  According to the present invention, since the telephone number is determined based on the destination information received from the wireless device, it becomes possible to switch the telephone number depending on what kind of destination information is embedded in the wireless device, and the degree of freedom of the call destination Can be increased.

It is a block diagram of the communication system which is embodiment of this invention. It is a figure which shows the memory content of the radio | wireless machine which is a subunit | mobile_unit. It is a block diagram of the relay apparatus of a communication system. It is a figure which shows the memory content of the memory | storage part of a relay apparatus. It is a flowchart which shows operation | movement of a relay apparatus.

A communication relay device and a communication system of the present invention will be described with reference to the drawings.
FIG. 1 is a configuration diagram of a communication system according to an embodiment of the present invention. This communication system is a system that enables outgoing calls, incoming calls, and telephone calls between the transceiver 4 and the IP telephone set 6 or the telephone set 8 using the relay device 2.

  As the network 1, for example, a LAN configured by Ethernet (registered trademark) or the Internet can be applied. One or a plurality of relay devices 2 (two in FIG. 1) are connected to the network 1. Each relay device 2 (2A, 2B) covers different communication areas A, B, respectively. A stationary transceiver is connected to the relay device 2 as a relay transceiver 3 (3A, 3B). The relay transceiver 3 may be a so-called push-to-talk (PTT) half-duplex communication device or a full-duplex communication device capable of transmitting and receiving simultaneously. Within the communication range of the relay transceiver 3, one or a plurality (two in the figure) of transceivers 4 (4A-1, 2, 4B-1, 2) exist. The relay transceiver 3 and the transceiver 4 can communicate with each other. The relay transceiver (master device) 3 corresponds to the relay radio device of the present invention, and the transceiver (slave device) 4 corresponds to the radio device of the present invention.

  In addition, an IP telephone 6 and a VoIP gateway 7 are connected to the network 1. A telephone 8 is connected to the VoIP gateway 7. The VoIP gateway is a gateway device that relays a VoIP (Voice over Internet Protocol) procedure for transmitting a voice signal via the network 1 and the telephone 8 that is a voice call device.

  The relay device 2 has a built-in SIP server function. The SIP server function is a function of performing a call control that associates a telephone number (URI) with an IP address and calls and connects a telephone of a communication partner using the SIP protocol.

  The transceiver 4 and the relay transceiver 3 are so-called digital transceivers. The digital transceiver can communicate audio signals and data converted into digital signals, and can simultaneously communicate control information such as destination information in parallel with audio signal communication. The relay transceiver 3 packetizes the digital signal received from the transceiver 4 and inputs the packet to the relay device 2. The relay transceiver 3 extracts the voice signal and control information from the packet input from the relay device 2. Convert to digital signal of axis and send.

  As shown in FIG. 2, the transceiver 4 stores an ID (radio device identification information) for identifying the own station and a group number to which the own station belongs, and the received digital signal includes destination information as destination information. If the ID or the group number to which the own station belongs is embedded, this signal is demodulated and an audio signal is output from a speaker or the like. If the received digital signal is not embedded with the ID of the own station or the group number to which the own station belongs, the digital signal is discarded. The ID of the own station and the group number to which the own station belongs are inserted, for example, as a squelch code in the received audio signal.

  FIG. 3 is a block diagram of the relay device 2. The relay device 2 includes a control unit 20 and includes a radio interface 21 at the downstream (relay transceiver 3 side) end and a network interface 22 at the upstream (network 1 side) end. For example, an Ethernet (registered trademark) physical layer connector can be applied to both the radio interface 21 and the network interface 22 and is in charge of the physical layer and data link layer of digital communication.

  The radio interface 21 inputs the packet input from the relay transceiver 3 to the control unit 20 and sends the packet input from the control unit 20 to the relay transceiver 3.

  The network interface 22 inputs a packet from a communication partner received via the network 1 to the control unit 20 and sends the packet input from the control unit 20 to the network 1.

  The control unit 20 includes a microcomputer and functionally includes a SIP processing unit 23, a storage unit 24, an upstream packet processing unit 25, and a downstream packet processing unit 26. In the storage unit 24, the local station IP address storage area 200, the local station telephone number storage area 201, the outgoing telephone number conversion table 203, and the incoming radio number conversion table 204 shown in FIG. 4 are set.

  The SIP processing unit 23 performs processing such as making, receiving, and disconnecting telephone communications, and has a SIP server function, and performs processing for calling a communication partner by associating a telephone number with an IP address.

  The upstream packet processing unit 25 reads the voice signal from the upstream voice packet input from the relay transceiver 3 via the radio interface 21, converts it into an RTP packet, and transfers it to the network interface 22. The control signal is read from the voice packet, and the SIP processing unit 23 is caused to execute a predetermined process according to the content of the control signal. For example, when a packet including call destination information is received, the SIP processing unit 23 is caused to execute call processing for the destination telephone number.

  The downstream packet processing unit 26 has a VOX (Voice Operated Relay) function, and monitors the level and duration of the audio signal included in the RTP packet input from the network interface 22. When the level and duration of the input audio signal exceeds a predetermined threshold value, it is assumed that speech is being input from the telephone of the other party, and this RTP packet is embedded with call information in the audio signal. The voice packet is converted and transferred to the relay transceiver 3. When a voice packet is input from the relay device 2, the relay transceiver 3 turns on the PTT and transmits a voice signal included in the voice packet.

  The calling telephone number conversion table 203 set in the storage unit 24 calls the destination information included in the voice packet transmitted from the transceiver 4, that is, the voice packet input to the relay device 2 via the relay transceiver 3. It is a table for converting into numbers.

  A general digital transceiver has a small number of buttons and it is difficult to directly input a telephone number. For this reason, the transceiver 4 uses a simple numeric string that can be input with a small number of buttons as the destination information. The relay device 2 converts this simple destination information into the telephone number of the IP telephone 6 or telephone 8 connected via the network 1. The outgoing telephone number conversion table 203 stores the simplified destination information and telephone numbers in association with each other.

  The outgoing telephone number table 203 has columns of Call_type (group / individual), destination ID and destination telephone number. In the transceiver 4, when the user selects a group / individual and an ID by operating a button and turns on the PTT button, a voice packet including the destination information (Call_type + destination ID) is transmitted. This destination information is the same as the call information used for the selective call (individual call / group call) in the communication between the transceivers 4.

  The incoming radio number conversion table 204 set in the storage unit 24 is a table for converting telephone numbers input from the telephones 6 and 8 into destination information (Call_type + destination ID). When a telephone call is made from the telephones 6 and 8 to the transceiver 4, a selective call (individual call or group call) can be made to an individual telephone by inputting a radio number after the telephone number of the relay device 2. The radio number is a simple numerical value (about two digits) that can be dialed. Each radio number is associated with a Call_type (group / individual) and a destination ID. Here, the destination information including Call_type (group / individual) and the destination ID corresponds to the embedded radio identification information of the present invention, and the radio number corresponds to the designation radio identification information.

  When the relay transceiver 3 can communicate with a plurality of transceivers 4, the radio number input from the communication partner (IP telephone 6 or the like) connected via the network 1 is used as destination information (Call_type (group / individual) and destination ID). ), Any one of the transceivers 4 can be designated, and only the designated transceiver 4 can make a call.

  A digital signal transmitted and received between digital transceivers has a control information frame for writing control information such as destination information between audio signals, and the destination information is written in this control information frame. Further, the destination information may be written as a squelch code. The digital signal in which the destination information is written is demodulated and reproduced only by the transceiver 4 specified by the destination information. Using this function, the individual call by the selective call is realized.

With reference to the flowchart of FIG. 5, the operation | movement at the time of the outgoing call of the relay apparatus 2 and an incoming call is demonstrated.
When an uplink voice packet is input from the radio interface 21 (S10), it is determined whether or not the currently talking flag is set (S11). If the in-call flag is set (YES in S11), the process is terminated without doing anything in this flowchart because the current call / incoming call process or call is in progress. If the busy flag is reset (NO in S11), the busy flag is set (S12). The call destination information is read from the input voice packet (S13), and the call phone number conversion table 203 is searched using this call destination information to read the corresponding phone number (S14). Then, an INVITE message is created with this telephone number as the SIP address and sent to the network 1 via the network interface 22 (S15).

  When a provisional response indicating that the request is received and processing is received (S16), a ringback tone (RBT) voice packet is generated and output to the radio interface 21 (S17), and the SIP procedure is advanced. I will. When a formal response is received from the partner station which is the communication partner telephone (S18), the ringback tone is stopped (S19), and the transceiver 4 and the call destination telephone (IP telephone 6 or telephone 8) are in a talking state. To.

  The call state is as follows. When the relay transceiver 3 is half duplex, the PTT process is performed with priority on the upstream packet input from the relay transceiver 3. While the upstream packet is being input, the downstream packet input from the network 1 (the partner station) is discarded and only the upstream packet is transferred. While the upstream packet is not input (while the PTT of the transceiver 4 is turned off), when the level of the voice signal transmitted in the downstream packet is equal to or higher than a predetermined threshold, the downstream packet is transmitted to the relay transceiver 3. Then, the PTT of the relay transceiver 3 is turned on, and the voice signal of the partner station is transmitted to the transceiver 4 (VOX processing).

  When a BYE message is input from the network 1 or when the level of the voice signal transmitted in the downlink packet does not exceed the threshold value for a predetermined time or more and the uplink packet is not input for the predetermined time or more, the call is disconnected. (Reset the busy flag) to return to the standby state.

  Thus, the telephone number can be determined based on the destination information included in the voice packet input from the transceiver 4 and an INVITE message can be transmitted to the telephone number. You will be able to make a call by selecting a phone number.

  On the other hand, when the INVITE message is received via the network interface 22 (YES in S20), it is determined whether or not the currently talking flag is set (S21). If the in-call flag is set (YES in S21), the current call / incoming call process or call is in progress and a new incoming call is not accepted, so an error response is returned to the partner device ( S29) End. If the busy flag is reset (NO in S21), the busy flag is set (S22), and the SIP procedure is responded to this call (S23). Then, a second dial instruction tone, which is a voice prompting the input of the wireless device number, is generated and transmitted to the partner station that has sent the INVITE message (S24).

  When a wireless device number is sent corresponding to this second dial instruction tone (S25), the incoming wireless device number conversion table 204 is searched using this wireless device number, and the corresponding destination information is read (S26). Then, a voice packet including the destination information is transmitted to the relay transceiver 3 to make a call (S27). When this wireless device responds (S28), the transceiver 4 and the caller telephone set (IP telephone set 6 or telephone set 8) are put into a call state. The call state is the same as when calling.

  As a result, even when a plurality of transceivers 4 are connected to the relay transceiver 3, only one or a plurality of transceivers 4 can talk to the telephone by embedding the destination radio number in the voice packet. It becomes possible to do.

  In the above embodiment, call information used for individual call and group call between the transceivers 4 is used as the destination information, but information different from this may be used as the destination information.

  In the above embodiment, transmission / reception of the audio signal between the relay transceiver 3 and the relay device 2 is performed by a packet, but may be performed by a digitized time axis signal or an analog signal. In any case, the call information may be superimposed (embedded) in some manner.

  Although the relay device 2 of this embodiment has a built-in SIP server function, the SIP server may be externally arranged on the network. In this case, the SIP message output from the relay apparatus 2 toward the IP telephone 6 or the VoIP gateway 7 is delivered to the IP telephone 6 or the VoIP gateway 7 via an external SIP server. A configuration in which the SIP message is not transmitted to and from the IP telephone 6 or the VoIP gateway 7 via an external SIP server without the SIP server is also included in the technical scope of the present invention.

DESCRIPTION OF SYMBOLS 1 Network 2 Relay apparatus 3 Transceiver 4 Transceiver 6 IP telephone 7 VoIP gateway 8 Telephone

Claims (4)

A network interface connected to a network which is a data communication network and communicating with a network device including a telephone connected to the network;
A destination having the same configuration as the call information used for selective calling in communication between radio units from a radio unit of a slave unit that communicates with the relay radio unit via the relay radio unit. A radio interface for receiving audio signals with embedded information;
When a voice signal in which the destination information is embedded is received from the radio interface, a call destination telephone number is determined based on the destination information, and a call message is generated with the determined telephone number as a destination. A control unit for transmitting via the network interface;
A relay device comprising:   The control unit includes a table in which the destination information and the telephone number are associated with each other. When an audio signal including the destination information is received from the relay radio, the control unit corresponds to the destination information with reference to the table. The relay apparatus according to claim 1, wherein a telephone number of a call destination to be determined is determined.   The relay apparatus according to claim 2, wherein the radio interface transmits and receives the audio signal by packet communication with the relay radio. A relay device according to any one of claims 1 to 3,
A relay radio connected to the radio interface of the relay device;
A slave radio that communicates with the relay radio;
A communication system comprising:
The wireless device transmits an audio signal in which destination information having the same configuration as the call information used for selective calling in communication between wireless devices is embedded,
The relay radio receives the audio signal transmitted by the radio and inputs it to the relay device;
The relay device inputs the voice signal, determines a telephone number based on destination information embedded in the voice signal, creates a call message with the telephone number as a destination, and sends it to the network system.

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