JP2011135290A - Relay apparatus and communication system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a relay apparatus that responds to an incoming call from a telephone on the condition that a sure response instruction is issued from a wireless communication device of a slave unit, and a communication system. <P>SOLUTION: The relay apparatus 2 intermittently transmits a ringback tone from a relay transceiver 3 when receiving an incoming call and monitors a response from a transceiver 4 in the pause interval thereof. The response is a voice satisfying prescribed first conditions. When receiving the response from the transceiver 4, the relay apparatus replies an OK message to a SIP server 6 so as to request the establishment of a session. The same telephone number (the broadcasting-incoming-call telephone number) is set in a plurality of relay apparatus 2. When the SIP server 6 receives a call message of the telephone number, it allows all of the relay apparatus 2 to receive the incoming call. <P>COPYRIGHT: (C)2011,JPO&INPIT

Description

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

  Conventionally, a system has been proposed in which a wired telephone line is connected to a wireless communication device (parent device) via a relay device, and communication between the telephone and the wireless communication device (child device) used by the user is possible (for example, Patent Documents 1 and 2). In the systems described in Patent Documents 1 and 2, when a call is received from a wired telephone line, the relay device answers the telephone almost unconditionally. That is, in the case of the horn patch device of Patent Document 1, “if the radio device 200 (master device) is in an empty line state, the relay 11 is switched to the break contact B side, and the telephone line 100 enters the hybrid circuit 4”. Is connected. Further, in the case of the receiving device 1 of Patent Document 2, “when a telephone enters the wired communication terminal 2 from an external wired communication terminal or the like, the signal incoming detection circuit 6 connected to the wired communication terminal 2 is provided. And the line switching control circuit 7 is activated, and the signal sound emitted from the wired communication terminal 2 is analyzed via each circuit of the communication relay device 5 and the host computer 15 connected to the circuit. Thus, the incoming call is answered unconditionally.

Japanese Patent Publication No. 3-44698

Japanese Utility Model Publication No. 5-60067

  However, in the configuration that responds unconditionally to an incoming call, it responds even when there is no slave unit that can make a call, so it is possible to answer this slave unit from the partner device or not (absent) There was a problem that I did not know if there was. In addition, if there is a response in the absence, the frequency of telephone communication may be wasted. Furthermore, if the incoming call is unconditionally answered, there is a problem that it is impossible to make a broadcast transmission in which a plurality of counterpart devices (relay devices) are simultaneously called and one of them is answered.

  An object of the present invention is to provide a relay apparatus and a communication system that responds to an incoming call from a telephone on the condition that there is a reliable response instruction from the wireless communication device of the child device.

  According to the first aspect of the present invention, there is provided a network interface connected to a network which is a data communication network, a telephone communication processing unit for controlling a telephone communication session via the network, and a telephone number storage for storing a telephone number of the own station. A radio interface for connecting the radio communication device to and inputting / outputting audio signals to / from the radio communication device and outputting a PTT signal as a signal for switching the radio communication device to a transmission mode; and the telephone communication When receiving notification that the first telephone number has received an incoming call from the processing unit, the ringback tone and the ringback for outputting the ringback tone and the PTT signal as voice indicating the incoming call to the radio interface Performs a calling process that repeats a pause interval in which tones and PTT signals are stopped at a predetermined interval. An audio signal input from the radio interface is monitored, and when the audio signal satisfies a predetermined first condition, the calling process is stopped and the telephone communication processing unit should respond to the incoming call. And a control unit that performs a response process for instructing.

  According to a second aspect of the present invention, in the first aspect of the invention, the telephone number storage unit further stores a second telephone number of a local station different from the first telephone number, and the control unit is configured to perform the telephone communication. Instructing the telephone communication processing unit to respond to the incoming call without performing the call processing when receiving a notification from the processing unit that the second telephone number has been received. It is characterized by that.

  According to a third aspect of the present invention, in the first and second aspects of the invention, when the input audio signal exceeds a first sound volume threshold for a first duration, the control unit performs the predetermined first operation. It is determined that the condition is satisfied.

  According to a fourth aspect of the present invention, in the first to third aspects of the present invention, the control unit monitors an audio signal input from the radio interface during standby when a telephone communication session is not established, and the audio When the signal satisfies a predetermined second condition, the telephone communication processing unit is instructed to instruct the telephone communication processing unit to make a call to a predetermined telephone number, and the telephone communication processing unit When the notification that the incoming call is canceled is received from the mobile phone, the calling process is stopped, and a masking process for prohibiting the calling process for a predetermined mask time is performed.

  According to a fifth aspect of the present invention, in the invention of the fourth aspect, the control unit receives an incoming call from the telephone communication processing unit to the first telephone number or the second telephone number even during the mask time. When the notification is received, the masking process is stopped, and the operation corresponding to the telephone number that has received the call is executed.

A sixth aspect of the present invention is a communication system in which a telephone communication server that controls a telephone communication session via a network and a plurality of relay apparatuses according to the first to fifth aspects are connected to a network. ,
The plurality of relay devices store the same telephone number as the first telephone number in the telephone number storage unit, and the telephone communication server receives the calling message for the first telephone number when the call message arrives. The call message is transferred to the relay apparatus to be called, and then a session is established between the relay apparatus that has responded earliest and the communication device that has transmitted the call message.

  According to the present invention, when a telephone call is received, a ringback tone is transmitted by radio, and when an audio signal satisfying the first condition is input during the pause period, the incoming call is answered. Thus, when there is no user (radio device) who can make a call, there is no need to respond in vain.

  Also, by waiting for a reliable response instruction and responding to an incoming call in this way, a broadcast call that allows a plurality of relay devices to receive incoming calls simultaneously becomes possible.

It is a block diagram of the communication system which is embodiment of this invention. It is a figure explaining the operation mode of the relay apparatus of the said communication system. It is a figure explaining the VOX parameter which a relay apparatus determines an audio | voice signal. 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. It is a flowchart which shows operation | movement of a relay apparatus. It is a flowchart which shows operation | movement of a relay apparatus. It is a flowchart which shows operation | movement of a relay apparatus. It is a flowchart which shows operation | movement of a relay apparatus. It is a flowchart which shows operation | movement of a relay apparatus. It is a flowchart which shows operation | movement of a relay apparatus. It is a flowchart which shows operation | movement of a relay apparatus. It is a flowchart which shows operation | movement 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 telephone communication via the network 1 using a transceiver 4 that is a wireless communication device, that is, IP phone call, call reception, and call by relaying by the relay device 2. As the network 1, for example, a LAN configured by Ethernet (registered trademark) or the Internet is applicable. One or more relay apparatuses 2 are connected to the network 1. A transceiver 3 is connected to the relay device 2 as a relay transceiver. The 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. One or a plurality of (two in the figure) transceivers 4 (4-1, 2) exist in the communication area of the transceiver 3. The transceiver 3 and the transceiver 4 are of a type that can communicate with each other.

  In addition, a SIP server 6 and a VoIP gateway 7 are connected to the network 1, and an IP telephone 8 is connected to the VoIP gateway 7. The SIP server 6 is a server device that performs call control by using a SIP protocol and associating a telephone number (URI) with an IP address to call and connect a call partner. 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. Although not shown, a VoIP router or PSTN device may be connected to the network 1. The SIP server 6 corresponds to the telephone communication server of the present invention.

  When the user transmits an audio signal using the transceiver 4 (4-1, 2), the audio signal is input to the relay device 2 via the relay transceiver 3. The relay device 2 has the function of an IP telephone, and controls call processing such as call origination / incoming call / response based on the input voice signal.

  The relay device 2 is connected only to the relay transceiver 3 and the microphone cable to maintain the versatility that any transceiver can be connected, and does not exchange special signals. That is, the relay device 2 and the relay transceiver 3 input / output only audio signals and PTT signals, thereby controlling the above-described calling, incoming calls, and telephone calls. Thus, any transceiver that can be externally connected with a microphone cable can be used as the relay transceiver 3 by connecting to the relay device 2 regardless of the model.

  FIG. 2 is a diagram for explaining the state transition of the relay device 2. The relay apparatus 2 is used to control telephone communication between the transceiver 4 and other telephone communication devices (for example, an IP telephone 8, another relay apparatus 2, a general telephone connected via a PSTN apparatus). While making a transition to the state (mode) shown in FIG. The relay device 2 takes five modes: a standby mode, a SIP call mode, a radio call mode, a call mode, and a mask mode.

  The standby mode is a state of waiting for a call instruction from the transceiver 4 (relay transceiver 3) or an incoming call from the partner apparatus (other telephone communication device) via the network 1. The incoming call from the partner apparatus via the network 1 is reception of the INVITE message of the SIP procedure. That is, when an INVITE message is sent from the SIP server 6, it shifts to a mode (call mode or wireless call mode) corresponding to a call from the partner device. Here, two telephone numbers (URI) are set in the relay device 2. One is a telephone number used for a single incoming call of only the relay apparatus 2, and the other is a telephone number used for a broadcast incoming call to a plurality of apparatuses including the relay apparatus 2. When an incoming call is received at a single incoming phone number, the mode is switched to a call mode. When an incoming call is received at a broadcast incoming phone number, the mode is shifted to a wireless call mode.

  On the other hand, assume that the relay apparatus 2 is instructed to make a call from the transceiver 4 when a predetermined voice signal is input from the relay transceiver 3 in the standby mode. That is, when a voice signal exceeding the control voice parameter 200 shown in FIG. 3 is input from the relay transceiver 3, the voice signal is determined to be a call instruction signal, and the relay apparatus 2 enters the call mode. Transition.

  FIG. 3 is a graph showing the VOX parameters. The VOX parameter is a threshold value for the VOX processing units 24 and 27 (see FIG. 4) built in the relay apparatus 2 to output a VOX signal when a predetermined audio signal is input. The control voice parameter 200 is a threshold value of the volume level and duration (attack time) of the voice signal that the relay device 2 recognizes as a call instruction. As shown in FIG. 3, the VOX parameter includes the volume level of the input signal and its duration. The reason why a large volume level and a long duration such as the control voice parameter 200 are set in the call instruction voice signal is to prevent an erroneous call from being generated by the noise radio wave received by the relay transceiver 3. It is to do. The control voice parameter 200 is also used to determine a call cancel instruction voice signal (to be described later) and a connection request voice signal (to the caller counterpart apparatus).

  Further, when a telephone communication session is established and the operation of the relay apparatus 2 shifts to the call mode, the uplink call voice parameter 201 and the downlink call voice parameter 202 are used to detect the presence / absence of uplink / downlink call voice. It is done. These VOX parameters are threshold values that are more gradual than those of the control voice parameter 200 because they respond to a small voice and are less likely to malfunction like the control voice. Further, the downlink voice signal input from the network 1 has less noise than the uplink voice signal input from the relay transceiver 3, and the PTT signal needs to be controlled by the downlink voice signal, so that the reaction speed is high. Since it is required, the downlink voice parameter 202 is more gentle and has a short attack time.

  In this embodiment, the same VOX parameter (threshold value) is used to determine the voice signal of the call instruction, the call cancel instruction, and the connection request. However, even if different VOX parameters are set, respectively. Good.

  Returning to FIG. 2, in the calling mode, the calling process is performed by the SIP procedure using the telephone number (URI) of the connection destination stored in advance. In this state, when there is a response from the partner device, the communication mode is entered. If an instruction to cancel the call is issued from the relay transceiver 3 during the call, the call process is stopped and the standby mode is restored. Here, it is assumed that the relay device 2 is instructed to cancel the call from the transceiver 4 when a voice signal exceeding the control voice parameter 200 is input from the relay transceiver 3 in the call mode. Note that the relay device 2 transmits a ringback tone (RBT) to the transceiver 4 via the relay transceiver 3 during a call, and the detection of the voice of the call cancel instruction is performed during a period during which the ringback tone is suspended. To do.

  In the standby mode, when there is an incoming call to the broadcast incoming telephone number, the radio call mode is entered. In the radio call mode, the fact that an incoming call has been received is transmitted to the transceiver 4 via the relay transceiver 3, and when there is a connection request from the transceiver 4, the incoming call is answered (an OK message is returned) and the telephone is called. A process of establishing communication (session) is performed. That is, since the broadcast incoming call is a procedure in which only one device that has responded first is connected to the partner device (the telephone communication device that made the call), the device (transceiver 4) that allows the user to talk to the partner device. In order to connect, an OK message is returned after waiting for a connection request from the transceiver 4. Here, it is assumed that the relay apparatus 2 receives a connection request from the transceiver 4 when a voice signal exceeding the control voice parameter 200 is input from the relay transceiver 3 in the radio call mode. After responding to the partner device (sending an OK message), the communication mode is entered.

  In the wireless call mode, if a CANCEL message is sent from the SIP server 6 before the relay apparatus 2 responds, the mode shifts to the mask mode. The case where the CANCEL message is sent before the relay device 2 responds is the case where another device that received the broadcast has responded first, or the counterpart device that is the transmission source cancels the call. When the broadcast mode is immediately switched to the standby mode when the incoming broadcast is canceled, when the user of the transceiver 4 transmits the voice exceeding the control voice parameter 200 for the purpose of the connection request immediately after the cancellation, this is a call instruction in the standby mode. Therefore, there is a risk of making an unnecessary call. Therefore, a mask time is set for not accepting voice input from the transceiver 4 for a certain period of time (about 10 seconds) when the incoming broadcast is canceled. In the mask mode, a busy tone (BT: busy tone) is generated and transmitted to the transceiver 4 at this mask time, and processing for not accepting the uplink control sound 200 is executed. After the mask time has elapsed, the mode shifts to the standby mode.

  In the call mode, the RTP procedure is operated, and voice packets for uplink (transceiver 4 (relay transceiver 3) to the partner device) and downlink (addresses from the partner device to transceiver 4 (relay transceiver 3)) are sent. Send and receive. Then, when a BYE message for instructing the end of the call is transmitted from the partner apparatus or when the call voice is interrupted for a predetermined time or more, the call is ended and the standby mode is entered. If the voice packet from the partner apparatus does not arrive continuously for a certain time (about 5 seconds), it is determined that the call with the partner apparatus becomes impossible due to a network failure or the like, and the call (session) is terminated. In this case, the user of the transceiver 4 is notified that the call has been terminated due to an unexpected failure, and the mode is shifted to the mask mode in order to prevent erroneous calls.

  In the call mode, the presence / absence of uplink call voice is determined based on whether or not an audio signal exceeding the uplink call voice parameter 201 in FIG. 3 is input from the relay transceiver 3, and the presence / absence of downlink call voice is determined in FIG. The determination is made based on whether or not an audio signal exceeding the downlink call parameter 202 is input from the network 1.

  FIG. 4 is a block diagram of the relay device 2. The relay device 2 includes a radio interface 21 at the downstream end (relay transceiver 3 side) and a network interface 22 at the upstream end (network 1 side). The radio interface 21 has a cable connector, a buffer circuit, and the like that connect the relay transceiver 3. The network interface 22 is a circuit unit in charge of the physical layer and data link layer of the network.

  An A / D converter 23, a VOX (Voice Operated Relay) processing unit 24, and a voice packet transmitting unit 25 are provided between the radio interface 21 and the network interface 22 in the upstream direction, and voice packets are received in the downstream direction. Section 26, VOX processing section 27, and D / A converter 28. In addition, the relay device 2 includes a control unit 20 that is a microcomputer that controls the operation of the entire device, and a SIP processing unit 29 that performs processing such as calling, incoming call, and disconnection of telephone communication. The control unit 20 includes a storage unit 20A that stores a telephone number (URI) of a connection destination.

  FIG. 5 is a diagram showing the contents stored in the storage unit 20A. In the storage unit 20A, the own station IP address, SIP address, connection destination telephone number (URI), own station single incoming telephone number (URI), own station broadcast incoming telephone number (URI), and those shown in FIG. Information such as VOX parameters is stored.

  In FIG. 4, the radio interface 21 and the relay transceiver 3 are connected by a so-called microphone cable. The microphone cable is an audio input / output signal line, that is, a speaker signal line for inputting the audio signal received by the relay transceiver 3 to the relay device 2, and outputs an audio signal from the relay device 2 to the relay transceiver 3. A cable having a microphone signal line and a PTT signal line.

  The radio interface 21 inputs the audio signal input from the relay transceiver 3 to the A / D converter 23. The A / D converter 23 converts the input audio signal into a digital signal and inputs the digital signal to the VOX processing unit 24 and the voice packet transmission unit 25.

  The VOX processing unit 24 monitors the level and duration of the audio signal input from the relay transceiver 3. When the level and duration of the input audio signal exceed the VOX parameter set at that time (see FIG. 3), the VOX signal is output to the control unit 20 as having uplink voice. When the VOX signal is input from the VOX processing unit 24, the control unit 20 performs processing according to the operation mode at that time (calling, calling cancellation, response to calling, call voice transfer, etc.).

  The voice packet transmission unit 25 converts the voice signal input from the AD converter 23 into a voice packet and transmits the voice signal to the partner apparatus in the call mode. The IP address of the partner apparatus is given from the SIP server 6 when a telephone communication session is established.

  The voice packet receiving unit 26 monitors reception of voice packets from the network 1 in the call mode, and selects and receives packets addressed to the local station IP address among the packets received from the network 1. When receiving the voice packet, the voice packet receiving unit 26 combines the voice packets and demodulates the audio stream signal. This audio signal is input to the VOX processing unit 27 and the D / A converter 28. The D / A converter 28 converts the input digital audio signal into an analog signal and inputs the analog signal to the radio interface 21. This audio signal is input to the relay transceiver 3 via a microphone cable.

  The VOX processing unit 27 monitors the level and duration of the audio signal input from the voice packet receiving unit 26. When the level and duration of the input audio signal exceed the parameters for downlink voice (see FIG. 3), a VOX signal is output to the control unit 20 as there being downlink voice.

  When the VOX signal is input from the VOX processing unit 27, the control unit 20 outputs a PTT signal to the radio interface 21. The PTT signal is input to the relay transceiver 3 via the microphone cable. The relay transceiver 3 is switched to the transmission mode by the PTT signal, and transmits the audio signal input via the microphone cable to the transceiver 4 by superimposing it on the transmission radio wave.

The operation of the relay device 2 will be described in detail with reference to the flowcharts of FIGS.
FIG. 6 is a flowchart showing the main routine. The main routine is a routine for determining the current mode based on the status flag and executing the processing of that mode. This main routine is executed at regular intervals (for example, every 100 milliseconds). When the power is turned on (at initialization), the status flag is set to the standby mode.

  First, in S1 to S4, the current mode is determined based on the status flag. In the case of the standby mode (YES in S1), standby mode processing (S5: see FIG. 8) for monitoring the occurrence of outgoing / incoming calls is executed. In the calling mode (YES in S2), RBT generation processing (S6: see FIG. 9) and calling mode processing (S7: see FIG. 10) for outputting a ringback tone (RBT) to the radio interface 21 are performed. Execute. In the radio call mode (YES in S3), RBT generation processing (S8: see FIG. 9) and radio call mode processing (S9: see FIG. 11) are executed. In the case of the call mode (YES in S4), the call mode process for controlling the call between the transceiver 4 and the partner device (S10: see FIG. 13). A silence detection process (S11: see FIG. 14) for performing the process and a packet loss monitoring process (S12: see FIG. 15) for monitoring abnormal disconnection of communication due to a network failure or the like are executed. Further, in the case of the mask mode (NO in S1 to S4), a mask mode process (S13: see FIG. 12) that does not accept the uplink control voice for a certain time is executed.

FIG. 7 is a flowchart showing a transition process for performing setting according to the next mode when shifting from the current mode to the next mode.
FIG. 7A is a flowchart showing standby mode transition processing executed when transitioning from another mode to standby mode. This process is also executed when the apparatus is activated in the standby mode when the power is turned on. First, the status flag is set to the standby mode (S20). As a result, standby mode processing is executed in the next main routine. The control voice parameter is set as the VOX parameter in the upstream VOX processing unit 24 (S21), and the downstream VOX processing unit 27 stops (S22).

  FIG. 7B is a flowchart showing a calling mode transition process executed when shifting from the standby mode to the calling mode. This process is executed when an audio signal exceeding the control audio parameter is input from the radio interface 21 in the standby mode (see FIG. 8). First, the status flag is set to the calling mode (S25). As a result, the calling mode process is executed in the next main routine. A timer t for measuring the elapsed time of the calling operation is started from 0 (S26). The control voice parameter is set as the VOX parameter in the upstream VOX processing unit 24 (S27), and the downstream VOX processing unit 27 stops (S28).

  FIG. 7C is a flowchart showing a radio call mode transition process executed when shifting from the standby mode to the radio call mode. This process is executed when an incoming call is received at the telephone number of the broadcast incoming call in the standby mode (see FIG. 8). First, the status flag is set to the radio call mode (S30). As a result, in the next main routine, the radio call mode process is executed. The control voice parameter is set as the VOX parameter in the upstream VOX processing unit 24 (S31), and the downstream VOX processing unit 27 is stopped (S32).

  FIG. 7D is a flowchart showing call mode transition processing executed when transitioning from another mode to the call mode. First, the call mode is set in the status flag (S35). Next, the up silence flag and the up / down silence flag are reset (S36), and the up silence timer and the up / down silence timer are reset (S37). These flags and timers are used in silence detection processing for measuring the time when the call voice is interrupted and terminating the call (session). Next, the uplink voice parameter is set as the VOX parameter in the uplink VOX processing unit 24 (S38), and the downlink voice parameter is set as the VOX parameter in the downlink VOX processing unit 27 (S39). Then, the voice packet transmitting unit 25 and the voice packet receiving unit 26 are activated to execute the RTP procedure (S40).

  FIG. 7E is a flowchart showing a mask mode transition process executed when shifting from the wireless calling mode to the mask mode. First, the status flag is set to the mask mode (S45). Next, the upstream VOX processing unit 24 is stopped and the input of the audio signal from the radio interface 21 is masked (S46). Then, the mask timer is started (S47), and a PTT signal and a busy tone (BT) are output to the radio interface 21 (S48, S49).

Hereinafter, processing in each mode will be described in detail.
FIG. 8 is a flowchart showing the standby mode process. First, whether an ON signal is input from the upstream VOX processing unit 24, that is, whether an input of control voice that is a call request is detected (S50), and whether an INVITE message is received, that is, an incoming call is received from another telephone communication device. (S51) is determined.

  When the ON signal is input from the upstream VOX processing unit 24 (YES in S50), the calling process is performed according to the SIP procedure. That is, an INVITE message is transmitted to the connection destination telephone number stored in the storage unit 20A (S56), and the calling mode transition process shown in FIG. 7B is executed (S57).

  On the other hand, when the INVITE message is received via the network interface 22 (YES in S51), the destination telephone number of the INVITE message is the single incoming telephone number of this relay apparatus 2 or the broadcast incoming telephone number. It is determined whether it exists (S52). In the case of a single incoming phone number, an OK message is immediately returned to the partner device (S53), and a call mode transition process for shifting to the call mode is executed (S54). If the telephone number is a broadcast incoming call, a wireless call mode transition process is executed (S55).

  FIG. 9 is a flowchart showing RBT (ringback tone) generation processing executed in the calling mode and the radio calling mode. This RBT generation process is a process for generating a ringback tone in a repeating pattern of ringing for 1 second and pausing for 2 seconds. However, the ringback tone generation / pause pattern is not limited to this.

  First, based on the duration t of the calling operation, it is determined whether the ringing period has started or the suspension period has started (S60). In this example, since the ringing is repeated for 1 second and paused for 2 seconds, the duration t of the calling operation is divided by 3, and if the remainder m is 0 or more and less than 1, the ringing period is 1 or more and less than 3. It is a suspension period. Therefore, the time when m = 0 is the ringing period start timing. At this time, the PTT signal is turned on and output to the radio interface 21 (S61), and a ringback tone (RBT) is generated and output to the radio interface 21 (S62). On the other hand, the time when m = 1 is the suspension period start timing. At this time, the PTT signal is turned off (S63), and the ringback tone is stopped (S64).

  FIG. 10 is a flowchart showing the calling mode process. In the calling mode, the RBT generation processing operation of FIG. 9 and the calling mode processing are executed sequentially and in parallel. In this process, whether an OK message indicating hook-off has been received from the partner apparatus (S70), whether the call operation duration t has exceeded a predetermined timeout period (for example, 1 minute) (S73), and the upstream VOX process It is monitored whether a VOX ON signal for instructing cancellation of a call is input from the unit 24 (S76).

  If an OK message has been received from the partner device (YES in S70), an ACK message indicating acceptance is returned (S71), and a call mode transition process is executed to shift to the call mode (S72). If the duration t of the calling operation has exceeded the timeout period (YES in S73), a CANCEL message is transmitted to the partner device (S74), and a standby mode transition process (S75) is performed to return to the standby mode. Execute. When the VOX ON signal is input from the upstream VOX processing unit 25 (YES in S76), the process proceeds to S74 and the call operation is terminated.

  When the OK message is not received from the partner device (NO in S70), the duration t is within the timeout time (NO in S73), and the VOX ON signal is not input from the upstream VOX processing unit 24 (NO in S76) This processing is terminated without doing anything, and the calling operation is continued.

  FIG. 11 is a flowchart showing the radio call mode process. In the radio call mode, the RBT generation process operation (S8) and the radio call mode process (S9) of FIG. 9 are executed sequentially and in parallel. In this processing, it is monitored whether a CANCEL message indicating call suspension is received from the caller counterpart device (S80), or whether a VOX ON signal as a connection request is input from the upstream VOX processing unit 24 (S83). . The VOX ON signal is output by the VOX processing unit 24 when an audio signal exceeding the control audio parameter 200 is input from the relay transceiver 3 during the ringback tone pause period in the RBT generation process of S8. .

  When a CANCEL message is input from the partner apparatus or the SIP server 6 (YES in S80), an OK message is returned (S81), and a transition is made to a mask mode in which voice input from the radio interface 21 is blocked for a predetermined time. A mask mode transition process is executed (S82). When the VOX ON signal is input from the upstream VOX processing unit 24 (YES in S83), it is a connection request to the partner device, so that an OK message is returned to the partner device (S84) and the call mode is changed. Processing is executed (S85).

  If no CANCEL message is received (NO in S80) and no VOX ON signal is input (NO in S83), this process is terminated without performing anything, and the radio call operation is continued.

  FIG. 12 is a flowchart showing the mask mode process. First, it is determined whether or not the mask timer started in the mask mode transition process has expired (S90). Until the mask timer expires (NO in S90), the mask mode is continued, so the mask mode is continued. If the mask timer has expired (YES in S90), the PTT signal is turned off to end the mask mode (S92), the mask timer is stopped (S93), and the standby mode transition process (S94) is executed. . However, even during the mask period, when an incoming call is received from another telephone communication device (partner device), that is, when an INVITE message is received (YES in S91), the PTT signal is turned off (S95), The mask timer is stopped (S96), the mask mode is stopped, the process jumps to S52 of the standby mode processing operation (see FIG. 8), and this incoming call is dealt with.

  FIG. 13A is a flowchart showing call mode processing. FIG. 13B is a flowchart showing the voice transfer process executed in the call mode process. In the call mode process, until the BYE message to end the call is received from the partner apparatus, the uplink voice signal input from the radio interface 21 is a voice packet in the voice transfer process of S107 (FIG. Uplink voice transfer processing to be transmitted to the network, and downstream voice transfer processing to convert the voice packet received from the network 1 by the RTP procedure into an analog audio signal and input to the radio interface 21 are executed.

  In FIG. 13A, it is first monitored in S100 whether a BYE message is received. While the BYE message is not received, the voice transfer process is executed (S107), and the DTMF * tone or # tone instructing forcible on / off of the PTT switch is monitored (S101, S102).

  * When a tone is received (YES in S101), the PTT flag is set (S108), and a PTT signal is output to the radio interface 21 (S109). Then, a long beep sound indicating that the PTT switch is turned on is transmitted to the partner apparatus via the voice packet transmitting unit 25 (S110). On the other hand, if the #tone is received (YES in S102), the PTT signal is turned off (S111) and the PTT flag is reset (S112). Then, a short beep (beep) indicating that the PTT switch is turned off is transmitted to the partner apparatus via the voice packet transmitter 25 (S113).

  On the other hand, when the BYE message is received (YES in S100), an OK message is returned to the partner apparatus (S104), and the operations of the voice packet transmitting unit 25 and the voice packet receiving unit 26 are stopped (S105). ), Standby mode transition processing is executed (S106).

  In the audio transfer process of FIG. 13B, the input of the VOX ON signal from the VOX processing unit 24 is monitored (S114). When the VOX ON signal is input from the upstream VOX processing unit 24 (YES in S114), the voice signal input from the A / D converter 23 to the voice packet transmission unit 25 is converted into a voice packet and transmitted to the network 1. (S115). During the call mode, the voice packet transmitter 25 generates a silent packet and sends it to the network 1 while not receiving the instruction of S115 from the controller 20.

  Next, it is determined whether the PTT flag is set, that is, whether the PTT signal is forcibly output (S116). If the PTT flag is set (YES in S116), the operation is terminated as it is. If the PTT flag is reset (NO in S116), it is determined whether or not a VOX ON signal is input from the downstream VOX processing unit 27 (S117). When a VOX ON signal is input from the downstream VOX processing unit 27 (YES in S117), a PTT signal is output to the radio interface 21 (S118). As a result, the audio signal received via the network 1 and converted into an analog signal is transmitted from the relay transceiver 3. If the VOX on signal is not input from the VOX processing unit 27 (NO in S117), the PTT signal is turned off (S119) and the operation is terminated.

  FIG. 14 is a flowchart showing the silence detection process. Since there is no means for forcibly performing an on-hook operation from the transceiver 4 (radio interface 21) side, the silence detection process is performed on the condition that the call voice has been interrupted for a certain period of time, that is, from the relay device 2 to the partner device ( This is a processing operation for performing a process of transmitting a BYE message toward the SIP server 6).

  First, it is detected whether or not a VOX ON signal is output from the upstream VOX processing unit 24 (S120). When the VOX ON signal is output, that is, when it is considered that the user of the transceiver 4 is talking (YES in S120), the up silence flag, the up / down silence flag, the up silence timer, and the up / down silence timer are all reset. (S121).

  The up silence flag is a flag for storing that there is no call voice input from the transceiver 4 (relay transceiver 3), and the up silence timer is a timer for measuring the duration. The up / down silence flag is a flag for storing that neither the call voice from the transceiver 4 (relay transceiver 3) nor the call voice from the partner apparatus (network 1) is input, and the up / down silence timer continues. It is a timer that measures time. The up / down silence timer is set to about 10 seconds, for example. If the call has been interrupted for 10 seconds or more in both the uplink and the downlink, the call (session) is terminated as the call is terminated.

  The up silence timer is set to about 20 seconds, for example. The uplink silence flag and the uplink silence timer are effective when the counterpart device is an automatic response such as an answering machine, and the transceiver 4 does not respond for about 20 seconds even if the counterpart device outputs a voice by the automatic response. Thus, the call can be terminated. In this flowchart, in the call mode, uplink silence is always detected, and when the uplink silence timer expires, the session is terminated. However, when an incoming call is received from the other device, the uplink silence flag and the uplink silence timer are used. Instead, the session may not be terminated only when the uplink call voice is interrupted.

  When the VOX ON signal is not output from the upstream VOX processing unit 24, that is, when the VOX processing unit 24 is OFF (NO in S120), it is determined whether the upstream silence flag is already set (S122). . If the up silence flag is not set (NO in S122), the up silence flag is set (S123) and the up silence timer is started (S124), assuming that the up silence state is started.

  If the up silence flag has already been set (YES in S122), it is determined whether the up silence timer has expired (S125). If the upstream silence timer has timed out (YES in S125), a BYE message is transmitted to the partner device to perform hook-on processing due to the upstream speech silence (S132), and the voice packet transmitter 25 and voice packet receiver 26 After stopping the operation (S133), standby mode shift processing is executed (S134).

  Further, when the upstream VOX processing unit 24 is off and the upstream silence timer has not timed out (NO in S124 and S125), the downstream VOX processing unit 27 further outputs a VOX ON signal. It is also determined whether or not it is present (S126). When the VOX ON signal is output from the down VOX processing unit 27 (YES in S126), it is assumed that the call voice from the partner device is input, and the up / down silence flag indicating that both the up and down are silent, The up and down silence timer is reset (S127).

  On the other hand, if the downstream VOX processing unit 27 does not output the VOX on signal, that is, if the VOX processing unit 27 is also off (NO in S126), it is determined whether the upper and lower silence flags are already set ( S128). If the up / down silence flag is not set (NO in S128), the up / down silence flag is set (S129) and the up / down silence timer is started (S130), assuming that the up / down silence state is started.

  If the up / down silence flag is already set (YES in S128), it is determined whether the up / down silence timer has expired (S131). If the up / down silence timer has expired (YES in S131), a BYE message is transmitted to the partner device to perform hook-on processing with up / down silence (S132), and the voice packet transmitter 25 and voice packet receiver 26 After stopping the operation (S133), standby mode shift processing is executed (S134).

  FIG. 15 is a flowchart showing packet loss monitoring processing. The packet loss monitoring process is a process for detecting that a voice packet has not arrived from the other device due to a network failure or the like and performing an on-hook process.

  First, it is detected whether there is a packet loss notification from the downstream VOX processing unit 24 (S140). If there is no packet loss notification, that is, if the voice packet has normally arrived from the partner device (NO in S140), the packet loss flag and the packet loss timer are reset (S141), and the operation is terminated. Note that the packet loss flag is a flag for storing that the voice packet from the partner apparatus is interrupted, and the packet loss timer is a timer for measuring the duration. The packet loss timer is set to about 5 seconds, for example.

    If there is a packet loss notification from the downstream VOX processing unit 24, that is, if a voice packet from the partner device does not arrive (YES in S140), it is determined whether the packet loss flag has already been set (S142). ). If the packet loss flag is not set (NO in S142), the packet loss flag is set (S143) and the packet loss timer is started (S144).

  If the packet loss flag has already been set (YES in S142), it is determined whether the packet loss timer has expired (S145). If the packet loss timer has expired (YES in S145), it is determined that a network failure or the like has occurred, and a BYE message is transmitted to notify the SIP server 6 of the end of the call (S146). The operations of the packet transmitter 25 and the voice packet receiver 26 are stopped (S147). Then, a mask mode transition process is executed in order to prevent an erroneous call due to the termination of the call due to an unexpected failure (S148).

  In this embodiment, as shown in FIG. 13 (B), only audio signals exceeding the parameters for uplink voice are converted into voice packets and transmitted to the counterpart device in the call mode, but the telephone communication is full duplex. Since it is communication, all uplink voices may be converted into voice packets without being subjected to VOX control and transmitted to the partner apparatus. Further, when the relay transceiver 3 connected to the radio interface 21 is a full-duplex transceiver, it is not necessary to VOX control the downstream voice (in this case, the PTT signal is unnecessary). ). Even in these cases, since it is necessary to detect silence for disconnecting a call (session), the VOX processing units 24 and 27 are operated.

DESCRIPTION OF SYMBOLS 1 Network 2,12 Relay apparatus 3 Relay transceiver 4 Transceiver 6 SIP server 7 VoIP gateway 8 IP telephone

Claims (6)

A network interface connected to a network which is a data communication network;
A telephone communication processing unit for controlling a telephone communication session via the network;
A telephone number storage unit for storing the telephone number of the own station;
A radio interface that is connected to the radio communication device, inputs and outputs audio signals with the radio communication device, and outputs a PTT signal that is a signal for switching the radio communication device to a transmission mode;
A ringing section for outputting a ringback tone, which is a voice indicating an incoming call, and the PTT signal to the radio interface when receiving a notification from the telephone communication processing unit that the incoming call has been made to the first telephone number. The ring back tone and the pause period for stopping the PTT signal are repeated at predetermined intervals, and an audio signal input from the radio interface during the pause period is monitored, and the audio signal is set to a predetermined first time. A control unit that performs a response process for instructing the telephone communication processing unit to respond to the incoming call when the condition is satisfied;
A relay device comprising: The telephone number storage unit further stores a second telephone number of the own station different from the first telephone number,
When the control unit receives a notification from the telephone communication processing unit that the second telephone number has been received, the control unit performs the incoming call to the telephone communication processing unit without performing the calling process. The relay device according to claim 1, which instructs to respond to   3. The control unit according to claim 1, wherein the control unit determines that the predetermined first condition is satisfied when the input audio signal exceeds a first volume threshold for a first duration. 4. Relay device. The controller is
While waiting for a telephone communication session to be established, the audio signal input from the radio interface is monitored, and when the audio signal satisfies a predetermined second condition, a call is made to a predetermined telephone number. Performing a calling process to instruct the telephone communication processing unit to
When receiving notification that the incoming call has been canceled from the telephone communication processing unit during the calling process, the calling process is stopped and a masking process for prohibiting the calling process for a predetermined masking time is performed. The relay device according to any one of claims 1 to 3.   When the control unit receives a notification from the telephone communication processing unit that the first telephone number or the second telephone number has been received even during the masking period, the control unit stops the masking process. The relay apparatus according to claim 4, wherein an operation corresponding to a telephone number on which an incoming call is made is executed. A communication system in which a telephone communication server for controlling a telephone communication session via a network and a plurality of relay devices according to claims 1 to 5 are connected to a network,
The plurality of relay devices store the same telephone number as the first telephone number in the telephone number storage unit,
When a call message for the first telephone number arrives, the telephone communication server transfers the call message to the plurality of relay apparatuses to receive a call, and thereafter, the relay apparatus that responded earliest and the relay apparatus A communication system that establishes a session with a communication device that has transmitted a paging message.

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