HK1245564B - Relay device, communication packet relay method, and sound communication system - Google Patents

Description

Relay device, communication packet relay method, and voice communication system

Technical Field

The present invention relates to a voice communication system using a network, and more particularly to a technology for relaying voice communications connecting a plurality of networks.

Background Art

A local area network (LAN) transceiver system that communicates using a LAN with an operational feel similar to that of conventional handheld transceivers has been put into practical use (e.g., Patent Document 1). This system uses a single-segment LAN to transmit audio signals, and each device is assigned a dedicated address valid within its LAN.

Prior art literature

Patent Literature

Patent Document 1: International Publication No. WO/2015/068663

Summary of the Invention

-Problems to be solved by the invention-

In order to expand the communication range even for the above-mentioned voice communication system using a LAN, there is a case where it is desired to connect different networks to build a system. The LAN transceiver system is constructed as a system on the LAN using a dedicated address. In the Internet Protocol (IP), since the address range of the dedicated address is determined, the address space of the two connected networks may overlap, and the dedicated addresses of the devices existing in each network may overlap. In such a case, although the address of the device of any network can be changed to avoid duplication, the management of the address is extremely troublesome. In addition, although it is possible to connect the two networks to each other by installing a NAT router in each network to change the address, the NAT router is required for each network, so the structure becomes complicated and the setting is troublesome.

Therefore, an object of the present invention is to realize a voice communication system that communicates between two networks using overlapping address spaces.

-Methods for solving the problem-

The present invention is a gateway device that connects a first network and a second network, each of which has at least a portion of an overlapping address space. The gateway device comprises: a first protocol stack that manages the sending and receiving addresses of communication packets in the first network; a second protocol stack that manages the sending and receiving addresses of communication packets in the second network; and a call control unit that, when receiving a communication packet from any network, determines the resending object network and resending object address of the payload contained in the packet based on the call object information contained in the packet.

The present invention is a gateway device that connects a first network and a second network, each of which has at least a portion of an overlapping address space. The gateway device comprises: a first interface connected to the first network; a second interface connected to the second network; and a call control unit, which establishes a corresponding call object table for the identification information of terminals existing in the first and second networks and the addresses used to send packets to the terminals. When a communication packet is received from any interface, the re-sending object network and re-sending object address of the payload contained in the packet are determined based on the call object information including the identification information contained in the packet.

In the above invention, the first and second networks may be a local area network and a mobile phone network.

In the above invention, the first network and the second network may be connected to one or more servers respectively, and each server may send and receive voice signals to one or more subordinate communication terminals respectively. The call control unit makes its own device independent of the first network and the second network, and sets it to operate in master mode or slave mode. In the master mode, all the voice signals received by the server are summarized and distributed to each server using communication packets. In the slave mode, the voice signals received from the subordinate communication terminals are sent to the master mode device (master server) using communication packets, and the voice signals received from the master server are sent to the subordinate communication terminals. When the call control unit is in master mode, it transmits the voice signals received from the server to other servers except the server.

In the slave mode, a sound signal received from the master server is transmitted to a server on the network opposite to the master server, and a sound signal received from a server on the network opposite to the master server is transmitted to the master server.

The present invention is a method for relaying communication packets, which relays communication packets between a first network and a second network, each of which has at least a portion of an overlapping address space. The method for relaying communication packets comprises: a step of receiving a communication packet from the first network using a first protocol stack that manages the sending and receiving addresses of the communication packet in the first network; a step of determining a resending object address of a payload contained in the received communication packet based on call object information contained in the packet; a step of generating a packet having the determined resending object address and containing a payload; and a step of sending the generated packet to the second network using a second protocol stack that manages the sending and receiving addresses of the communication packet in the second network.

The present invention is a method for relaying communication packets, which relays communication packets between a first network and a second network, each of which has at least a portion of an overlapping address space. The method for relaying communication packets comprises: a step of receiving a communication packet from a first interface connected to the first network; a step of establishing a corresponding call object table based on call object information contained in the received communication packet, with reference to identification information of terminals existing in the first and second networks and an address for sending packets to the terminal, and determining a retransmission object address for a payload contained in the packet; a step of generating a packet having the determined retransmission object address and containing a payload; and a step of sending the generated packet from a second interface connected to the second network.

The voice communication system of the present invention comprises: a first server device, connected to a first network, for transmitting and receiving voice signals to and from a first communication terminal; a second server device, connected to a second network, at least a portion of the address space of which overlaps with that of the first network, for transmitting and receiving voice signals to and from a second communication terminal; and the gateway device.

The voice communication system of the present invention comprises: one or more servers connected to a first network; one or more servers connected to a second network whose address space at least partially overlaps with that of the first network; and the gateway device operating in master mode with respect to both the first and second networks. Each server transmits and receives voice signals to and from one or more subordinate communication terminals. The server transmits voice signals received from subordinate communication terminals to the gateway device, and transmits voice signals received from the gateway device to subordinate communication terminals. The gateway device forwards voice signals received from a server to other servers other than that server.

The voice communication system of the present invention comprises: one or more servers connected to a first network; one or more servers connected to a second network whose address space at least partially overlaps with that of the first network; and a gateway device that operates in slave mode with respect to the first network and in master mode with respect to the second network. One of the servers in the first network is designated as the first master server, the other servers are designated as first slave servers, and the servers in the second network are designated as second slave servers. The first master server aggregates all voice signals received by the first slave servers and distributes them to the first slave servers and the gateway device using communication packets. The first slave server transmits voice signals received from a subordinate communication terminal to the first master server using communication packets, and transmits voice signals received from the first master server to subordinate communication terminals other than the communication terminal. The second slave server transmits voice signals received from a subordinate communication terminal to the gateway device using communication packets, transmits them to subordinate terminal devices other than the communication terminal, and transmits voice signals received from the gateway device to subordinate communication terminals. The gateway device transmits the audio signal received from the first master server to the second slave server, and transmits the audio signal received from the second slave server to the second slave servers other than the slave server and the first master server.

The voice communication system of the present invention comprises: one or more slave mode servers connected to a first network; one or more slave mode servers connected to a second network whose address space overlaps at least partially with that of the first network; and the gateway device operating in a master mode, wherein each server transmits and receives voice signals to and from one or more subordinate communication terminals. During group communication in which the communication terminals belonging to a group of two or more communication terminals communicate with each other, the server and the gateway device operate as follows. The server transmits the voice signal received from the subordinate communication terminal to the terminal device and gateway device belonging to the subordinate group other than the communication terminal, and transmits the voice signal received from the gateway device to the communication terminal belonging to the subordinate group. The gateway device transmits the voice signal received from the server to a server other than the server that manages the communication terminals belonging to the group.

The voice communication system of the present invention comprises: one or more servers connected to a first network;

One or more servers are connected to a second network whose address space overlaps at least partially with that of the first network; and the gateway device operates in slave mode, with each server transmitting and receiving audio signals with one or more subordinate communication terminals. One of the servers of the first network is set as the first master server, and the other servers are set as the first slave servers. The servers of the second network are set as the second slave servers. The gateway device is set in slave mode with respect to the first network and in master mode with respect to the second network. When group communication is performed in which the communication terminals in a group to which two or more communication terminals belong communicate with each other, each server and the gateway device operates as follows.

The first slave server transmits a voice signal received from a subordinate communication terminal to terminal devices belonging to the subordinate group other than the communication terminal and the first master server. The first slave server transmits a voice signal received from the first master server to a communication terminal belonging to the subordinate group.

The first master server transmits a voice signal received from a subordinate communication terminal to communication terminals belonging to the subordinate group other than the communication terminal and to a first slave server that manages the communication terminals belonging to the group. The first master server transmits a voice signal received from a first slave server to first slave servers that manage the communication terminals belonging to the group other than the slave server and to communication terminals belonging to the subordinate group. When the communication terminal belonging to the group is in the second network, the first master server transmits the voice signal to the gateway device.

The second slave server transmits the audio signal received from the subordinate communication terminal to the terminal devices and the gateway device belonging to the subordinate group other than the communication terminal. The second slave server transmits the audio signal received from the gateway device to the communication terminal belonging to the subordinate group.

The gateway device transmits the audio signal received from the first master server to the second slave server that manages the communication terminals belonging to the group. The gateway device transmits the audio signal received from the second slave server to the second slave server that manages the communication terminals belonging to the group other than the slave server and the first master server.

The voice communication system of the present invention comprises: one or more servers connected to a first network; one or more servers connected to a second network whose address space at least partially overlaps with that of the first network; and the gateway device operating in slave mode, wherein each server transmits and receives voice signals to and from one or more subordinate communication terminals. One of the servers of the first network is set as the first master server, and the other servers are set as the first slave servers. One of the servers of the second network is set as the second master server, and the other servers are set as the second slave servers. When group communication is performed in which the communication terminals in a group to which two or more communication terminals belong communicate with each other, each server and the gateway device operates as follows.

The first slave server transmits a voice signal received from a subordinate communication terminal to terminal devices belonging to the subordinate group other than the communication terminal and the first master server. The first slave server transmits a voice signal received from the first master server to a communication terminal belonging to the subordinate group.

The first master server transmits a voice signal received from a subordinate communication terminal to communication terminals belonging to the subordinate group other than the communication terminal and to a first slave server that manages the communication terminals belonging to the group. The first master server transmits a voice signal received from a first slave server to first slave servers that manage the communication terminals belonging to the group other than the slave server and to communication terminals belonging to the subordinate group. When the communication terminal belonging to the group is in the second network, the first master server transmits the voice signal to the gateway device.

The second slave server transmits the audio signal received from the subordinate communication terminal to the terminal devices belonging to the subordinate group other than the communication terminal and the second master server. The second slave server transmits the audio signal received from the second master server to the communication terminal belonging to the subordinate group.

The second master server transmits a voice signal received from a subordinate communication terminal to communication terminals belonging to the subordinate group other than the communication terminal in question and to a second slave server that manages the communication terminals belonging to the group. The second master server transmits a voice signal received from a second slave server to a second slave server that manages the communication terminals belonging to the group other than the slave server in question and to a communication terminal belonging to the subordinate group. When the communication terminal belonging to the group is in the first network, the second master server transmits the voice signal to the gateway device.

The gateway device transmits the audio signal received from the first host server to the second host server, and transmits the audio signal received from the second host server to the first host server.

-Effects of the Invention-

According to the present invention, it is possible to connect two networks having overlapping address spaces and perform voice communication across the networks.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG1 is a block diagram of a voice communication system according to an embodiment of the present invention.

FIG2 is a functional block diagram of a gateway, a LAN server, and an LTE server.

FIG3 is a diagram showing an example of the structure of an audio group.

FIG4 is a block diagram of a LAN communication terminal.

FIG. 5 is a diagram showing a call destination table provided in a gateway.

FIG. 6 is a diagram showing a call destination table provided in the LTE server.

FIG. 7 is a diagram showing a call destination table provided in the LAN server (S1).

FIG. 8 is a diagram showing a call destination table provided in the LAN server (S2).

FIG. 9 is a diagram illustrating a process of transmitting an audio signal.

FIG. 10 is a diagram showing the flow of audio signals when the gateway (G) functions as a master server.

FIG. 11 is a diagram showing the flow of audio signals when the LAN server ( S1 ) functions as a master server.

FIG. 12 is a diagram showing an example of a call destination table for each server.

FIG. 13 is a diagram showing the flow of audio signals when group communication is performed in an audio communication system in which a LAN server (S1) is configured as a master server.

DETAILED DESCRIPTION

The voice communication system of the present invention will be described with reference to the accompanying drawings. FIG1 is a block diagram of a voice communication system according to an embodiment of the present invention. This voice communication system connects a LAN system 10 using a wireless LAN 16, such as Wi-Fi, and an LTE system 20 using a mobile phone communication network, such as LTE 21, via a gateway 12. Communication terminals 14 and 24 belonging to each system perform voice communication with each other.

The LAN system 10 includes a LAN server 11 and a LAN communication terminal 14. The LAN communication terminal 14 and the LTE communication terminal 24 have nearly identical configurations, each shaped like a handheld transceiver for wireless communication equipped with a PTT (Push to Talk, PoC) switch. Functionally, they are wireless network devices that transmit and receive voice signals via a wireless access point (AP) 13 or a base station 23. The LAN server 11 and the LTE server 22, acting as relay devices, relay communications between the LAN communication terminal 14 and the LTE communication terminal 24. Hereinafter, the wired network 15 and the wireless LAN 16 of the LAN system 10 will be collectively referred to as the network 17, and the network of the LTE system 20 will be referred to as the LTE network 21. For the wireless LAN 16, a communication method based on IEEE 802.11, such as Wi-Fi, can be used.

In Figure 1, a wired network 15 is connected to two server devices 11, a gateway 12, and a SIP server 19. In both the LAN system 10 and the LTE system 20, the gateway 12 functions as a master server, controlling the destination of voice signal transmission. Transmission of voice signals here refers to the process of transmitting the voice signal, contained as the payload in a received voice packet, within a voice packet corresponding to the destination network. When multiple voice packets are received and these payloads, i.e., multiple voice signals, are transmitted to the same destination, a signal resulting from the mixing of these voice signals is included as the payload in the transmitted voice packet. Furthermore, the two server devices 11 function as slave servers S1 and S2, which transmit voice signals transmitted from the master gateway 12 to a LAN communication terminal 14. Slave server S1 communicates with LAN communication terminal 14, whose terminal ID is 001, via wireless access point 13, managing this LAN communication terminal 14 as a subordinate terminal. The server device S2 communicates with the LAN communication terminal 14 whose terminal ID is 011 via the wireless access point 13, and manages the LAN communication terminal 14 as a subordinate terminal.

In addition, although in this embodiment, the gateway 12 is set as the master server for both the LAN-side system 10 and the LTE-side system 20, the LAN server 11 and/or the LTE server 22 may also be set as the master server. When the server devices 11 and 22 are set as the master server, the gateway 12 operates in a slave mode (as a slave server) with respect to the systems 10 and 20 on that side. In addition, in the LAN-side system 10, the number of LAN servers 11 is not limited to two, and can be one or three or more. In addition, when configuring a system with three or more network segments connected, the number of gateways 12 may also be two or more.

On the other hand, in the LTE side system 20 , the LTE server 22 communicates with the LTE communication terminals 24 having terminal IDs 021 and 022 via the LTE network 21 , and manages these LTE communication terminals 24 as subordinate terminals.

1 , each device in the LAN system 10 and the LTE system 20 is assigned a dedicated IP address for network 17 and a dedicated IP address for LTE network 21. The respective address spaces at least partially overlap, and some devices in the two systems 10 and 20 have the same IP address.

Figure 2 is a functional block diagram of the LAN server 11, gateway 12, and LTE server 22. The LAN server 11 includes a call control unit 100, a LAN terminal interface 101, a mixer 102, and an interface for another location 103. The call control unit 100 controls the destination of the voice signal contained in the received voice packet based on the communication control information of the packet. When multiple voice signals are sent to the same destination, the mixer 102 is instructed to mix these voice signals. The LAN terminal interface 101 is an interface for communicating with the LAN communication terminal 14. The interface for another location 103 is an interface for communicating with the gateway 12 and other LAN servers 11. The physical layer including the network 17 and the protocol stack 104 are commonly used in the LAN terminal interface 101 and the interface for another location 103. The protocol stack 104 includes an application for directly transmitting voice signals without using the SIP procedure.

In addition, the LTE server 22 also has: a call control unit 120, an LTE terminal interface 121, a mixer 122, and an other base interface 123. The LTE terminal interface 121 is an interface for communicating with the LTE communication terminal 24. In addition, the other base interface 123 is an interface for communicating with the gateway 12. The call control unit 120 controls the transmission destination of the sound signal contained in the sound packet based on the communication control information of the received packet. When multiple sound signals are sent to the same destination, the mixer 122 is instructed to mix these sound signals. The physical layer and protocol stack 124 including the LTE network 21 are commonly used in the LTE terminal interface 121 and the other base interface 123. The protocol stack 124 has an application for directly transmitting sound signals without using the TCP/IP and SIP procedures on the LTE network 21.

The gateway 12 connects the LAN server 11 (LAN-side system 10) and the LTE server 22 (LTE-side system 20). The gateway 12 includes a (LAN-side) other-site interface 112 for communicating with the LAN server 11, a (LTE-side) other-site interface 114 for communicating with the LTE server 22, a call control unit 110, and a mixer 111. The LAN-side other-site interface 112 includes a protocol stack 113 for communicating with the LAN server 11 via the network 17. This protocol stack 113 is substantially identical to the protocol stack 104 of the LAN server 11. Furthermore, the LTE-side other-site interface 114 includes a protocol stack 115 for communicating with the LTE server 22 via the LTE network 21. This protocol stack 115 is substantially identical to the protocol stack 124 of the LTE server 22. Voice signals contained in voice packets input from the other-site interfaces 112 and 114 are mixed by the mixer 111 and re-edited into voice packets transmitted from the other-site interface on the opposite side or the other-site interface on the same side. The call control unit 110 determines which other-site interface, or server, to which to transmit the voice signal based on the communication control information contained in the received voice packet. The LTE-side other-site interface 114 is a wireless (LTE) interface that uses the same communication channel as mobile phones.

The call control unit 110 of the gateway 12 is implemented by the cooperation of a computer and a communication application. The communication application includes a program for determining the interface of the server to be called and a server program for communicating with the LTE server 22 and the LAN server 11.

Referring to Figure 3, the structure of the voice packet used in the voice communication system is explained. The voice packet includes: a destination IP address, a destination port number, header information including the source IP address and source port number, and a data body (payload) including voice data and communication control information. The voice data is a digitized and compressed voice signal of one thread (for example, 20 milliseconds). The destination IP address and source IP address are the IP addresses indicating the direct destination and source of the voice packet. The communication control information is information that identifies the communication session in which which communication terminal is communicating with which communication terminal, and includes the call type, call recipient ID, and call source ID. The call type is information indicating the form of communication and is any information indicating individual communication, group communication, or overall communication. The call recipient ID is the terminal number of the communication recipient in the case of individual communication and the group number in the case of group communication.

For example, when a call is made from the LTE communication terminal 24 (021) to the LAN communication terminal 14 (001), the call destination ID and the call source ID in the communication control information are recorded as 001 and 021, respectively. Voice signals are transmitted in the order of LTE communication terminal 24 (021) → LTE server 22 → gateway 12 → LAN server 11 (S1) → LAN communication terminal 14 (001). However, each time a voice packet containing the voice signal is transmitted, a new voice packet is generated, and the source IP address and destination IP address at that time are recorded. Furthermore, this transmission order represents the order when the gateway 12 is the primary server. When another LAN server 11 is set as the primary server, the gateway 12 first transmits the voice signal to the primary server, and the primary server transmits the voice signal to the LAN server 11 (S1). However, when the LAN server S1 is the primary server, the above order may be sufficient.

When the gateway 12 and LAN server 11 receive audio signals from the LAN communication terminal 14 or other servers 11 and 22, they extract the data body from the audio signal, determine the destination of the audio data based on the communication control information, generate a new audio signal, and transmit it to the destination. At this time, they perform processing such as mixing multiple audio data as needed.

FIG4 is a block diagram of the LAN communication terminal 14. As described above, the LAN communication terminal 14 is functionally a wireless network device that transmits and receives voice signals via the wireless access point (AP) 13 of the wireless LAN. The control unit 40 that controls the operation of the device is composed of a microprocessor. The control unit 40 has a storage unit 41 that stores various data. The storage unit 41 stores setting data such as the call object ID that can be communicated through the terminal device. The control unit 40 is connected to the operation unit 42, the display unit 43, the audio circuit 44, and the wireless LAN communication unit 45. The operation unit 42 includes a push switch such as the PTT switch 220, which receives the user's operation and inputs the operation signal to the control unit 40. The display unit 43 includes a liquid crystal display. The identification number of the communication object selected by the user's operation, the identification number of the received communication object, etc. are displayed on the liquid crystal display.

The audio circuit 44 also includes a microphone 240 and a speaker 241. The control unit 40 decodes the received audio signal and inputs it to the audio circuit 44. The audio circuit converts the decoded audio signal into an analog signal and outputs it from the speaker 241. Furthermore, the audio circuit 44 converts the audio signal input from the microphone 240 into a digital signal and inputs it to the control unit 40. The control unit 40 converts the digital audio signal into audio packets and inputs them to the wireless LAN communication unit 45. The wireless LAN communication unit 45 includes a circuit for wireless communication using the communication method compliant with IEEE 802.11. The wireless LAN communication unit 45 transmits packets input from the control unit 40 to the wireless access point 13 and inputs packets received from the wireless access point 13 to the control unit 40.

In the LAN communication terminal 14 configured as described above, when the user presses the PTT switch 220 and inputs a voice into the microphone 240 , the LAN communication terminal 14 edits the voice signal into a voice packet and transmits the packet to the LAN server 11 .

4 , the LAN communication terminal 14 is described, but the LTE communication terminal 24 has substantially the same configuration. The LTE communication terminal 24 includes an LTE communication unit instead of the wireless LAN communication unit 45 .

Figures 5, 6, 7, and 8 are diagrams showing call destination tables provided in the gateway 12, LTE server 22, and LAN server 11 (S1, S2), respectively. When the LTE server 22 and LAN server 11 receive voice packets from their subordinate communication terminals 14, 24, they transmit the voice signal contained in the voice packet to the gateway 12, which is the master server. When the gateway 12 receives a voice packet from the server 11, 22, it searches the table of Figure 5 based on the communication type and call destination ID contained in the communication control information of the voice packet, infers the server to which the voice signal (payload) contained in the voice packet is to be transmitted, and transmits the voice signal to the transmission destination. At this time, the voice packet is edited using the protocol corresponding to the network of the transmission destination, and the voice signal is transmitted using the voice packet. Furthermore, when the LAN server 11 other than the gateway 12 is set as the master server, the voice signal is transmitted to the gateway 12 via the master server.

The call object table includes: call type, call object ID, belonging terminal information, and other base information (server information). The call type is information that distinguishes individual calls/group calls/all calls. The call object ID is information that identifies the terminal 14, 24 or group of the communication object. In the case of an individual call, the identification information of the terminal device 14 of the communication object, that is, the terminal ID, becomes the call object ID. In other words, the call object ID is the same as the terminal information. In the case of a group call, the group number assigned to the group (a number starting from 1 in the case of this embodiment) becomes the call object ID. The belonging terminal information and other base information are information indicating the transmission object of the sound signal. In addition, the belonging terminal information of the call object table of the main server (the gateway 12 in this embodiment) can also include communication terminals that are not its own subordinates to register the numbers of all communication terminals belonging to the group.

Figure 5 shows the call object table of the gateway 12. In the call object table of the gateway 12, as other base information, LAN side other base information and LTE side other base information are stored separately. Since the gateway 12 does not have subordinate communication terminals, the terminal information corresponding to all call object IDs is blank. The sound signal is transmitted to the other servers 11 and 22 based on the other base information. In addition, here, the other base information (LS, S1, S2) includes the IP address and port number of each server. Although there is a case where the local IP address is repeated in the LAN side system 10 and the LTE side system 20, these will not be confused because the storage columns in the table and the other base interfaces 112 and 114 including the physical layer are different.

Figures 6, 7, and 8 respectively represent the call destination tables of the LTE server 22 and the LAN server 11 (S1, S2). Among the communication terminals called according to the call destination ID, the terminal numbers of the communication terminals belonging to the server are recorded in the "Subordinate Terminal Number" column. When a communication terminal that is not a subordinate of the server is called according to the call destination ID, the voice signal corresponding to the call destination ID is transmitted to the gateway 12, which is the master device. Therefore, the IP address and port number of the gateway 12 are recorded in the "Other Location Information." In addition, the "Other Location Information" of the gateway 12 on the LAN side and the "Other Location Information" on the LTE side are assigned separate IP addresses (having separate MAC addresses) different from the physical layer. Therefore, the "Other Location Information" of the gateway 12 written to the call destination table (Figure 6) of the LTE server 22 and the "Other Location Information" of the gateway 12 written to the call destination table (Figures 7 and 8) of the LAN server 11 are different information.

Next, a process of transmitting a voice signal from the LTE communication terminal 24 of the LTE side system 20 to the LAN communication terminal 14 of the LAN side system 10 will be described with reference to FIG. 9 .

(Step S101) The LTE communication terminal 24 (021) transmits a communication packet destined for the LAN communication terminal 14 to the LTE server 22. The transmission source IP address of the communication packet is 10.10.0.1, and the destination IP address is 192.168.0.1.

(Step S102) The LTE server 22 refers to the call destination table (FIG. 6) to infer the destination and transmits the communication packet to the gateway 12. The transmission source IP address of the communication packet is 192.168.0.1, and the destination IP address is 10.10.0.2.

(Step S103 ) The gateway 12 receives the communication packet arriving at the other site interface (LTE interface) 114 (IP address 10.10.0.2) on the LTE side via the LTE protocol stack 115 .

(Step S104) The call control unit 110 (internal communication application) of the gateway 12 refers to the call destination table (Figure 5) based on the communication control information in the received communication packet and determines that the server to be called is the server connected to the LAN-side other site interface (LAN interface) 112. At this time, the calling IP address and the connection destination port number are also obtained.

(Step S105 ) The call control unit 110 of the gateway 12 performs communication protocol conversion toward the network 17 and transmits an instruction to the LAN protocol stack 113 .

(Step S106) The gateway 12 transmits a communication packet from the LAN interface 112 (IP address 192.168.0.2) to the LAN server 11 (S1) via the LAN protocol stack 113. The transmission source IP address of the communication packet is 192.168.0.2, and the destination IP address is 192.168.0.1.

(Step S107) The LAN server 11 refers to the internal call destination table (FIG. 7) to estimate the destination and sends the communication packet to the LAN communication terminal 14 (001). The source IP address of the communication packet is 192.168.0.1, and the destination IP address is 192.168.0.10.

(Step S108) The LAN communication terminal 14 receives the communication packet and ends the transmission of the communication packet.

When the LAN communication terminal 14 transmits a communication packet to the LTE communication terminal 24, communication is performed in the reverse process to that shown in FIG9 . Furthermore, although this embodiment describes the transmission of audio signals to the opposite side of the gateway 12, it is also possible to transmit communication packets received from the LTE server 22 or the LAN server 11 to both the LTE server 22 and the LAN server 11, or to transmit communication packets to another server on the receiving side network, by setting the call destination table of the gateway 12.

In the above embodiment, an example is given in which the gateway 12 acts as the master server (M) and the other servers (LAN server 11, LTE server 22) act as slave servers. Figure 10 shows the flow of sound signals in this case. The sound signals of all communications are temporarily collected by the gateway 12, which acts as the master server. The master server determines the arrival point of each sound signal and transmits the sound signal to the slave server. When the master server and each slave server receive multiple sound signals, they mix them and set them as one sound signal for transmission. In addition, although the number of terminals under each server is 2, 0, 1, and 1 respectively, the number of terminals is not limited to this. In addition, the gateway 12 can also communicate directly with the communication terminal.

Figure 11 shows the flow of audio signals when a server other than gateway 12 (in this example, LAN server 11 (S1)) functions as a master server (M), and other servers, including gateway 12, function as slave servers in the LAN system 10. All audio signals from communications are collected by LAN server 11 (S1), the master server. The master server determines the arrival point of each audio signal and transmits it to a slave server or its subordinate LAN communication terminal 14. Upon receiving multiple audio signals, the master server and each slave server mix them and transmit them as a single audio signal. In this case, in the LTE system 20, gateway 12 can simply function as the master server.

The flow of audio signals during group communication in the system illustrated in Figure 11, where LAN server 11 (S1) functions as the master server, will be described. Group communication refers to a configuration in which multiple (generally three or more) communication terminals communicate with each other simultaneously. In this case, audio signal transmission and reception are preferably performed using a full-duplexer.

As shown in Figures 5, 6, 7, and 8, a call destination table is set up in each of the gateway 12, LTE server 22, and LAN server 11 (S1, S2). When group communication is performed, the group is registered in the call destination table of each server. The example described below shows an example of group communication performed by group 102 consisting of three communication terminals (001, 011, 021). As shown in Figure 12, group 102 is registered in the call destination table of each server. In addition, the subordinate terminals of LAN server 11 (S1), which is the main server, are only communication terminal 14 (001), but the numbers of all communication terminals belonging to the group can also be registered in the call destination table of the main server.

Figure 13 shows the flow of audio signals during group communication by group 102. In the description of this figure, the audio signal transmitted from communication terminal n is referred to as audio signal n. The audio signal resulting from the mixing of audio signals m and n is referred to as audio signals m,n. Furthermore, LAN server 11 (S1) serving as the master server is referred to as master server S1, and LAN server 11 (S2) serving as the slave server is referred to as slave server S2. This figure shows a state in which terminals 001 and 021 are engaged in a conversation, that is, transmitting audio signals, during full-duplex communication.

LTE server 22 (LS), acting as a slave server, receives voice signal 021 from subordinate communication terminal 021. Furthermore, it receives voice signal 001 from communication terminal 001, which is sent from master server S1 via gateway 12 (G). LTE server 22 transmits voice signal 001 received from gateway 12 to communication terminal 021, and sends voice signal 021 received from communication terminal 021 to gateway 12 in order to send it to master server S1. Furthermore, if the communication is global communication (group communication targeting all communication terminals), LTE server 22 mixes voice signal 001 and voice signal 021 and sends the resultant mix to communication terminal 022. This allows the voice of communication terminal 001 to be heard at communication terminal 022. Furthermore, voice signal 021 is not mixed into the voice signal sent to communication terminal 021 to prevent echo.

The gateway 12 receives the sound signal 021 from the LTE server 22 and the sound signal 001 from the main server S1. The gateway 12 converts the sound signal 021 received from the LTE server 22 into a packet format of the LAN (network 17) and transmits it to the main server S1. The gateway 12 also converts the sound signal 001 received from the main server S1 into a packet format of the LTE network 21 and transmits it to the LTE server 22.

Master server S1 receives sound signal 001 from subordinate communication terminal 001. It also receives sound signal 021 from communication terminal 021, which is sent from gateway 12. Master server S1 mixes the received sound signal 001 and sound signal 021, and transmits the result to slave server S2. Furthermore, sound signal 001 received from communication terminal 001 is transmitted to gateway 12. The sound signal transmitted to gateway 12 does not include sound signal 021 received from gateway 12. Sound signal 021 received from gateway 12 is transmitted to communication terminal 001. As a result, the sound of communication terminal 021 can be heard at communication terminal 001. Furthermore, sound signal 001 is not mixed into the sound signal transmitted to communication terminal 001 to prevent echo.

Slave server S2 receives the mixed audio signals 001 and 021 from master server S1, but does not receive audio signals from subordinate communication terminal 011 (although it is a member of the group). It transmits audio signals 001 and 021 received from master server S1 to subordinate communication terminal 011. As a result, communication terminal 011 can hear the audio from both communication terminals 001 and 021. Furthermore, slave server S2 does not receive audio signals from subordinate communication terminals and therefore does not transmit anything to master server S1.

In the embodiments of Figures 11, 12, and 13 described above, two servers (LAN server 11) are set up on the LAN (network) 17 side, and one server (LTE server 22) is set up on the LTE network 21 side, with one of the LAN servers 11 being the master server. The number of servers on the LAN 17 side and the number of servers on the LTE network 21 side are not limited to this. Any server can be used as long as there is one or more. Furthermore, although the master server is set up on the LAN 17 side, this is not limited to this. The master server may also be set up on the LTE network 21 side, or the master server may be set up on the LAN 17 side and the LTE network 21 side.

In a voice communication system where a master server is installed on both the LAN 17 side and the LTE network 21 side, the configuration of the LAN 17 side shown in FIG13 is implemented on both the LAN 17 side and the LTE network 21 side. Communication on the LAN 17 side is performed by aggregating voice signals to the master server on the LAN 17 side, while communication on the LTE network 21 side is performed by aggregating voice signals to the LTE network 21. Communication (transmission and reception of voice signals) from the communication terminal 14 on the LAN 17 side to the communication terminal 24 on the LTE network 21 side proceeds in the following order: LAN communication terminal 24 - LAN slave server - LAN master server - gateway 12 - LTE master server - LTE slave server - LTE communication terminal 24. Communication from the communication terminal 24 on the LTE network 21 side to the communication terminal 14 on the LAN 17 side proceeds in the reverse order. Furthermore, if the communication terminals 14 and 24 are subordinate to the master server, communication does not need to go through the slave server.

Through the above configuration and process, according to this embodiment, when setting up a system connecting two networks, the same IP address can be used in each network, thereby enabling flexible network design. As a result, instead of previously having to centrally manage the IP addresses of the two networks, it is now possible to manage them for each network, thus improving convenience.

Furthermore, when a new network system is connected to an existing network system, the network system can be connected without affecting the address form of the existing network.

Furthermore, in group communication, voice signals can be aggregated to one main server via the gateway 12, so that in group communication between multiple communication terminals, mixing for removing echoes can be performed without complicating communication.

In the embodiment, an example is shown in which a general LAN (such as Ethernet) network 17 is connected to an LTE network 21. However, the present invention is not limited to this example and can be widely applied to connecting multiple networks where local IP addresses may overlap. Furthermore, the mobile phone network is not limited to the LTE network 21. For example, a 3G network may also be used.

-Explanation of Symbols-

10 LAN side system

11 LAN Server

12 Gateway

14 LAN communication terminal

17 Network (LAN)

20 LTE Side System

21 LTE network

22 LTE Server

24 LTE communication terminals

Claims (4)

1. A gateway device connecting a first network and a second network, each having a local IP address and each having one or more servers, wherein the local IP address of the first network is the same as the local IP address of the second network, the gateway device comprising: The first protocol stack is configured to manage the transmit and receive addresses of voice packets, including payloads containing voice signals, in the first network; The second protocol stack is configured to manage the transmit and receive addresses of voice packets containing payloads of voice signals in the second network; and The call control unit is configured to, upon receiving the voice packet from a server in either the first network or the second network, determine the sending target network and the sending target address of the server in the sending target network based on the call type and call target ID included in the communication control information of the packet's payload, and the call target table of the gateway device. The first protocol stack is installed on the first other location interface connected to the first network, and the second protocol stack is installed on the second other location interface connected to the second network. The physical layers of the first other base interface and the second other base interface are different. The call control unit includes the call recipient table. The call object table of the gateway device includes other location information related to the first network and other location information related to the second network. The other location information related to the first network and the other location information related to the second network are stored in different columns, and represent the sending object address of the server in the first network or the second network that has established a corresponding connection with the call object ID used to specify the communication terminal and/or group communication. The other location information related to the first network includes the IP address of the server in the first network, and the other location information related to the second network includes the IP address of the server in the second network. The call control unit is configured to send the voice packets from either the first other location interface or the second other location interface connected to the determined sending target network to the sending target address of a server in the determined sending target network. The call control unit operates in either master mode or slave mode relative to the first network and the second network, respectively. In the main mode, the call control unit is configured to receive one or more audio packets containing audio signals from one or more servers in either the first network or the second network, mix the audio signals into mixed audio packets, and distribute the mixed audio packets to servers that are different from the specific server that is the receiving source of the audio packets and are stored as sending target addresses in the call target table. In the slave mode, the call control unit is configured to receive a first audio packet containing an audio signal from a given server in the first network, convert the received audio packet into a packet format in the second network, and send the converted packet to a master server in the second network; and the call control unit is configured to receive a second audio packet from the master server in the second network and send the second audio packet to the given server in the first network. 2. The gateway device according to claim 1, wherein, The first network is a local area network (LAN), and the second network is a mobile phone network. 3. A method for relaying voice packets, wherein voice packets are relayed between a first network and a second network, each having a local IP address and each having one or more servers, wherein the local IP address of the first network is the same as the local IP address of the second network, the method for relaying voice packets comprising: The step of receiving voice packets from the first network using a first protocol stack that manages the transmit and receive addresses of voice packets containing payloads of voice signals in the first network; Referring to the call target table, based on the call type and call target ID contained in the communication control information of the received voice packet, the steps are to determine the sending target network and the sending target address of the server in the sending target network. The steps of generating a packet having the determined destination address and containing the payload; and The step of sending the generated packet to the second network using a second protocol stack that manages the send and receive addresses of voice packets in the second network. The call target table includes other location information related to the first network and other location information related to the second network. This information is stored in different columns and represents the sending target address of a server in the first or second network that has established a corresponding connection with the call target ID used for specifying communication terminals and/or groups. The first protocol stack is installed on the first other location interface connected to the first network, and the second protocol stack is installed on the second other location interface connected to the second network. The physical layers of the first other base interface and the second other base interface are different. The other location information related to the first network includes the IP address of the server in the first network, and the other location information related to the second network includes the IP address of the server in the second network. The voice packet is sent from either the first or second other location interface connected to the determined destination network to the destination address of a server in the determined destination network. Relative to the first network and the second network operating in either master mode or slave mode, In the main mode, one or more audio packets containing audio signals are received from servers in one or more of the first and second networks. The audio signals are mixed to form mixed audio packets. The mixed audio packets are then distributed to servers that are different from the specific server that is the receiving source of the audio packets and are stored as sending target addresses in the call target table. In the slave mode, a first audio packet containing an audio signal is received from a given server in the first network, the received audio packet is converted into a packet format in the second network, the converted packet is sent to a master server in the second network, and a second audio packet is received from the master server in the second network and sent to the given server in the first network. 4. A voice communication system, comprising: The first server is connected to the first network and sends and receives voice signals with the first communication terminal; The second server, connected to the second network, sends and receives audio signals with the second communication terminal; and The gateway device according to claim 1 or claim 2.