JP5096074B2 - IP telephone apparatus with relay function, IP telephone system including the same, and communication relay method - Google Patents

Description

  The present invention relates to an IP (Internet Protocol) telephone apparatus that performs communication in a wide area network including a call control server such as a SIP (Session Initiation Protocol) server, an IP telephone system including the IP telephone system, and a communication relay method.

  2. Description of the Related Art Conventionally, an IP telephone device connected to a company or home LAN (Local Area Network) is generally connected to a WAN (Wide-Area Network) via a relay device such as a router. As such a relay device, there is a device equipped with a NAT function for transparently converting between a private IP address that can be used only in a LAN and a global IP address that can be used to access an external WAN. . According to this NAT function, there is an advantage that the degree of freedom of assigning an IP address to a device connected to the LAN increases and the security of the LAN also increases.

  However, when the relay device has a NAT function, in SIP (Session Initiation Protocol) used as a call control protocol for IP telephones, communication is performed by adding a private IP address or port number to the data portion of the IP packet. The NAT function that converts only the IP address in the header portion of the IP telephone device cannot cope with it, and even if an attempt is made to add a global IP address to the data portion of the IP packet, the IP telephone device uses the global address and port number used by itself. There is a problem that information cannot be grasped by itself. As a result, a so-called NAT traversal problem has occurred, in which communication packets from the WAN side do not reach the LAN side due to the relay device as a wall.

  As a means for solving this NAT traversal problem, a method using STUN (Simple Traversal of User Datagram Protocol) is known. In this method, NAT traversal can be realized by installing a STUN server on the network that provides necessary information (global IP address, port number, NAT type, etc.) in response to a request from the IP telephone apparatus. However, this method may not function properly depending on the NAT type and device characteristics of the relay device.

  Therefore, as a technique for realizing NAT traversal regardless of the NAT type and device characteristics, for example, a relay device that forms a pair across a network firewall (NAT) is provided, and a firewall is provided on the relay device. There is known a communication system that realizes NAT traversal by mediating intercommunicating communications (see Patent Document 1).

JP 2006-254160 A

  However, in the prior art described in Patent Document 1, it is necessary to provide a relay device that mediates communication inside and outside the firewall (NAT) in each network to which a terminal that performs communication belongs, which increases the cost of the system. Also, there was a problem that it was poor in versatility.

  As a technique for realizing NAT traversal regardless of the NAT type and device characteristics, there is a method in which a relay server such as an SBC (Session Border Controller) performs relay processing of SIP messages and voice data. However, when transferring SIP messages and voice data executed between IP telephone devices via a relay server, there is a problem that an increase in traffic and an excessive processing load on the relay server may occur.

  The present invention has been devised in view of such problems of the prior art, and it is possible to connect between IP telephone apparatuses with a simple and inexpensive configuration regardless of the NAT type of the relay apparatus to which the IP telephone apparatus is connected. It is an object to provide an IP telephone device with a relay function, an IP telephone system including the IP telephone device, and a communication relay method.

An IP telephone apparatus with a relay function according to the present invention includes an IP telephone system having a plurality of IP telephone apparatuses connected to a wide area network through a relay apparatus having a NAT function and a call control server for controlling calls of these IP telephone apparatuses. The IP telephone device with a relay function used in the above-mentioned network is connected to the wide area network via a relay device having a complete cone type NAT function, and performs voice communication with the IP telephone device A normal mode for performing the operation and a relay mode for relaying voice communication performed between other IP telephone devices. When the operation mode is in the relay mode, the call control server A transfer instructing unit for instructing transfer of a call control message transmitted / received between other IP telephone devices via the communication device, and the call control server based on the transfer instruction. Receiving a call control message transferred from, recognizes the destination and source IP address and port number included in the call control message, the originating IP address and ports in the SDP of the call control messages The message relay means for rewriting the number to the IP address and port number of the own device and returning the rewritten call control message to the call control server, and the voice between the other IP telephone device by transmitting and receiving the call control message After the connection for executing communication is established, communication data transmitted from one of the IP telephone devices to the IP address and port number of the own device is the original transmission destination based on the recognition. Voice relay means for relaying the voice communication by transferring to That.

  As described above, according to the present invention, when an IP telephone device with a relay function does not perform voice communication by itself, an IP telephone device that performs voice communication is connected by relaying voice communication between other IP telephone devices. Regardless of the NAT type of the relay device, the voice communication between the IP telephone devices can be realized with a simple and inexpensive configuration.

A first invention made to solve the above problems includes a plurality of IP telephone apparatuses connected to a wide area network via a relay apparatus having a NAT function, and a call control server for controlling calls of these IP telephone apparatuses. An IP telephone apparatus with a relay function used in an IP telephone system having the IP telephone apparatus with the relay function, wherein the IP telephone apparatus with the relay function is connected to the wide area network via a relay apparatus having a complete cone type NAT function. And a normal mode for performing voice communication and a relay mode for relaying voice communication performed between other IP telephone devices. When in the relay mode, the call control server Transfer instruction means for instructing transfer of a call control message transmitted and received between other IP telephone devices via the call control server, and based on the transfer instruction, Receiving the forwarded call control message from the control server, recognizes the destination and source IP address and port number included in the call control message, the source IP address included in the SDP of the call control messages And message relay means for rewriting the port number to the IP address and port number of the own device and returning the call control message after the rewriting to the call control server, and the other IP telephone device by transmitting and receiving the call control message Communication data transmitted from one of the IP telephone devices to the IP address and port number of the own device after the connection for performing voice communication is established at the original destination based on the recognition. structure that includes a voice relay means for relaying the voice communication by transferring to an other To.

  According to this, when the IP telephone device with a relay function does not perform voice communication by itself, by relaying voice communication between other IP telephone devices, the NAT of the relay device to which the IP telephone device that performs voice communication is connected Regardless of the type, voice communication between IP telephone devices can be realized with a simple and inexpensive configuration. In addition, an IP telephone apparatus that performs voice communication has an advantage that voice communication can be executed in the same procedure as before without requiring a special function.

  In this case, the IP telephone device with a relay function executes relay processing (relay mode) when voice communication is not performed by itself, thereby affecting the normal voice communication processing (normal mode) of the own device (processing delay or the like). ) Can be suppressed.

A second invention has been made in order to solve the above problems, the transfer instruction means instructs the transfer of the call control messages by continuously repeatedly transmit the transfer instruction request packet to the call control server On the other hand, the transfer instruction is canceled by stopping transmission of the packet.

  According to this, packets are continuously transmitted to enable inbound communication when voice communication between other IP telephone devices is relayed, and the call control message is transferred to the call control server depending on the presence / absence of this packet transmission. Therefore, it is possible to increase security by minimizing the time required to access a predetermined port of the own device from the outside.

  According to a third aspect of the invention for solving the above problem, the transfer instruction request packet is transmitted from a port used for voice relay in the own apparatus.

  According to this, it becomes possible to perform instructing or canceling the call control message and UDP hole punching for relaying voice communication in conjunction with each other, and it is possible to simplify the operation and increase the efficiency.

  According to a fourth aspect of the present invention for solving the above problems, when the mode is shifted to the normal mode while the relay mode is being executed, the IP telephone device is in communication with one of the IP telephone devices that performs the voice communication. Instructing the call control server to send a call control message for re-establishing the connection and transferring the call control message transmitted / received between the IP telephone devices to the other IP telephone device with a relay function. A relay change instruction means for instructing is further provided.

  According to this, when an IP telephone device with a relay function needs to perform voice communication by itself during execution of voice communication, it can easily take over the relay processing to another IP telephone device with a relay function. Thus, it is possible to avoid the inconvenience that the voice communication being relayed is stopped.

  A fifth invention made to solve the above problems is an IP telephone system including the IP telephone device with a relay function according to any one of the first to fourth inventions.

  A sixth invention made to solve the above-described problem is that the call control server receives a registration request from the IP telephone apparatus and uses the IP telephone apparatus as a relay terminal that relays a call control message or communication data. The call control comprising: a relay terminal registration means for registering in the relay terminal registration table; and a transfer means for transferring a call control message transmitted / received between other IP telephone devices to the relay terminal registered in the relay terminal registration table A server comprising: a relay terminal registration table means for registering a terminal that relays a call control message or communication data; and a relay terminal registration means for receiving a registration request from the terminal and registering the terminal as a relay terminal; To do.

  According to this, the call control server can automatically determine to which terminal the call control message is relayed, and the setting operability of the system is improved.

A seventh invention made to solve the above problems includes a plurality of IP telephone apparatuses connected to a wide area network via a relay apparatus having a NAT function, and a call control server for controlling calls of these IP telephone apparatuses. A communication relay method used in an IP telephone system, wherein at least one of the IP telephone devices is connected to the wide area network via a relay device having a complete cone type NAT function, and the IP telephone device and voice There are two operation modes, a normal mode for performing communication and a relay mode for relaying voice communication performed between other IP telephone devices. When in the relay mode, the call control server is informed of the call mode. Instructs transfer of a call control message transmitted / received between other IP telephone devices via the control server, and is transferred from the call control server based on the transfer instruction. Receiving a call control message, recognizes the destination and source IP address and port number included in the call control message, the IP address and port number of the calling side contained in the SDP of the call control message itself Connection for rewriting the IP address and port number of the device, returning the rewritten call control message to the call control server, and performing voice communication between the other IP telephone devices by transmitting and receiving the call control message Is established, the communication data transmitted from one of the IP telephone devices to the IP address and port number of the own device is transferred to the other, which is the original transmission destination, based on the recognition. Relay communication.

  According to this, when the IP telephone device with a relay function does not perform voice communication by itself, by relaying voice communication between other IP telephone devices, the NAT of the relay device to which the IP telephone device that performs voice communication is connected Regardless of the type, voice communication between IP telephone devices can be realized with a simple and inexpensive configuration.

  Hereinafter, embodiments of the present invention will be described with reference to the drawings.

  1 is a system configuration diagram showing an outline of an IP telephone system including an IP telephone terminal with a relay function according to the present invention, FIG. 2 is a functional block diagram of the IP telephone terminal, and FIG. 3 is a functional block of a SIP server. FIG.

  The IP telephone system 1 has a configuration in which a plurality of IP telephone terminals A to E and a SIP server (call control server) 2 are connected to each other via the Internet (wide area network) 3 so as to communicate with each other. IP telephone terminals A to E belong to LANs (Local Area Networks) 4 to 8, respectively, and are connected to the Internet 3 via NAT routers (relay devices having a NAT function) A to E.

  The IP telephone terminals A to E function as UAC (User Agent Client) of SIP (Session Initiation Protocol). The IP telephone terminals C and D, like the IP telephone terminals A, B, and E, are in a normal mode for performing voice communication with other IP telephone terminals and a relay mode for relaying voice communication performed between the other IP telephone terminals. When the IP telephone terminal with a relay function (hereinafter referred to as “relay terminal”) has two operation modes, and does not perform voice communication by itself, the relay mode is executed and the voice between other IP telephone terminals is Communication can be relayed.

  As shown in FIG. 2, the relay terminals C and D include a call control unit 11, a dial operation unit 12, a voice packet transmission / reception unit 13, a handset unit 14, a relay terminal registration unit (transfer instruction unit) 15, and a relay operation setting unit 16. SIP message relay unit (message relay unit, relay change instruction unit) 17, relay address information storage unit 18, voice packet relay unit (voice relay unit) 19, and network I / F unit 20. Here, the relay terminal registration unit 15, the relay operation setting unit 16, the SIP message relay unit 17, the relay address information storage unit 18, and the voice packet relay unit 19 are related to the relay function of the relay terminals C and D. Other configurations are the same as those of the IP telephone terminals A, B, and E.

  The call control unit 11 establishes a call connection while exchanging SIP messages with the partner device (another IP telephone terminal) via the SIP server 2 based on SIP in accordance with the operation of the dial operation unit 12 by the user. Signaling processing is performed. Further, the call control unit 11 controls the processing of each unit so as to selectively execute the two operation modes of the normal mode and the relay mode. Various methods can be used for selecting the operation mode depending on the usage mode of the IP telephone system 1. For example, the execution time can be set in advance for each operation mode, or the relay mode can always be executed when the own device does not perform voice communication with other IP telephone terminals.

  The voice packet transmitting / receiving unit 13 executes various processes related to voice communication performed with a partner apparatus based on VoIP (voice over internet protocol), starting with transmission / reception of voice packets after establishment of a call connection. The voice packet transmission / reception unit 13 includes an A / D and D / A conversion function for transmission data and reception data, and a voice codec based on a predetermined voice coding method (for example, ITU-T recommendation G.711, G.729, etc.) A voice analog signal input from the handset unit 14 used by the user is A / D converted and encoded to be transmitted data, while the encoded received data is decoded and D / The analog signal generated by the A conversion is output to the handset unit 14.

  The relay terminal registration unit 15 executes processing for registering that the local apparatus is in the relay mode with respect to the SIP server 2 (hereinafter referred to as “relay terminal registration”). More specifically, when the own device is in the relay mode, the relay terminal registration unit 15 continuously and repeatedly transmits a packet including the request for registering the relay terminal to the SIP server 2, while the own device is in the normal mode. If there is, stop sending the packet. Information necessary for relay terminal registration executed by the relay terminal registration unit 15 is set in the relay operation setting unit 16. For example, information on a packet transmission destination, packet transmission timing, and information on a port number for sending a packet, etc. Is set.

  As will be described later, the packet transmission by the relay operation setting unit 16 functions as a SIP message transfer instruction to the SIP server 2, and the SIP server 2 that receives this packet transmits / receives a SIP message between other IP telephone terminals. Is appropriately transferred to the relay terminal that is registering (that is, the relay terminal in which the SIP server 2 is receiving the packet). This transfer instruction is canceled when the registration of the relay terminal is canceled (that is, reception of a packet from the relay terminal is stopped). Further, the packet is sent out from a port (hereinafter referred to as “voice relay port”) used for relaying subsequent voice communication as UDP hole punching. Inbound communication from other IP telephone terminals is possible.

  The SIP message relay unit 17 receives the SIP message transferred from the SIP server 2 based on the relay terminal registration, and stores it in an SDP (Session Description Protocol) description document (hereinafter simply referred to as “SDP”) on the SIP message. The IP address and port number indicating the communication destination of the included voice communication are rewritten to the predetermined IP address and port number of the own device, respectively, and the rewritten SIP message is returned to the SIP server 2 again. Here, the IP address and port number of the own device to be rewritten are stored in advance in the relay address information storage unit 18 as relay address information. The port number of the own device used here is the number of the voice relay port described above. The SIP message relay unit 17 executes only the rewriting process of the IP address and the port number, and the call control unit 11 executes the transmission / reception process of the SIP message related to the rewriting process in the same way as a normal SIP message. It can also be.

  The voice packet relay unit 19 relays voice packets transmitted and received between the IP telephone terminals in the media session after the call connection between the other IP telephone terminals is established by executing the relay mode.

  The network I / F unit 20 is connected to the LAN and executes transmission / reception of SIP messages and voice packets with a partner apparatus according to a predetermined communication protocol.

  The NAT routers A to E transparently transmit a private IP address used for communication by the IP telephone terminals A to E in the respective LANs 4 to 8 and a global IP address used for communication via the external Internet 3. This is a broadband router equipped with a NAT function for mutual conversion. The NAT routers A to E allow outbound communication started from the inside to the outside of the NAT by the NAT function, while rejecting inbound communication started from the outside to the inside of the NAT in principle. By deleting the entry (registration contents) of the NAT conversion table as described above, it also serves as a firewall. Note that the term “NAT” used here is not limited to a technique for converting a private address and a global address, but includes NAPT (Network Address Port Translation) that converts a port number in addition to IP address conversion. Similar technology shall be included.

  Here, the NAT functions of the NAT routers A, B, and E are classified into NAT types (Full Cone), Restricted Cone (Restricted Cone), and Port Cone (Port). It may belong to any of Restricted Cone) and Symmetric). On the other hand, the NAT function of the NAT routers C and D to which the relay terminals C and D are connected is such that the mapping between the same private IP address and port number and the same global IP address and port number can be made to any communication partner. It is necessary to be a complete cone type to be held. In some cases, a configuration in which the relay terminals C and D are connected to the Internet 3 without going through a NAT router is also possible.

  The SIP server 2 functions as a SIP UAS (User Agent Server). As shown in FIG. 3, the SIP server 2 includes a terminal registration unit 31, a terminal table 32, a SIP message relay unit 33, a relay terminal registration unit 34, and a relay terminal table 35.

  The terminal registration unit 31 receives the REGISTER message from each of the IP telephone terminals A to E, thereby acquiring the location information (IP address, port number, etc.) of the IP telephone terminals A to E, and registering and managing them. It has a function. The location information of the IP telephone terminals A to E is stored in the terminal table 32 and updated as necessary.

  The SIP message relay unit 33 relays a SIP message for call connection between the devices in response to requests from the IP telephone terminals A to E.

  The relay terminal registration unit 34 executes relay terminal registration in accordance with a request from the relay terminal registration unit 15 of the relay terminals C and D. More specifically, the relay terminal registration unit 34 recognizes that the relay terminal is in the relay mode while continuously receiving packets including the relay terminal registration request from the relay terminal, and detects the position information ( (IP address, port number, etc.) are stored in the relay terminal table 35. On the other hand, when the reception of the packet is stopped, the position information is deleted from the relay terminal table 35 in order to cancel the registration of the relay terminal.

  When a plurality of relay terminals in the relay mode are registered, the relay terminal registration unit 34 sets a predetermined priority (for example, registration order) for the relay terminals, and each relay terminal is based on the priority. Can execute relay processing. As will be described later, when the relay terminal shifts to the normal mode while the relay terminal is executing the relay mode, the relay terminal registration unit 34 executes a process for taking over the relay process to another relay terminal.

  The network I / F unit 36 is connected to the Internet 3 and executes transmission / reception of SIP messages and the like with the IP telephone terminals A to E on the network according to a predetermined communication protocol.

  FIG. 4 is a sequence diagram showing the basic operation of the IP telephone system according to the present invention. Here, a case is shown in which the IP telephone terminal A and the IP telephone terminal B are the transmission side and the reception side, respectively, and the voice communication is relayed by the relay terminal C.

  As shown in FIG. 4, the IP telephone terminals A and B and the relay terminal C each appropriately transmit a REGISTER message to the SIP server 2 after activation and receive a 200 OK message from the SIP server 2 to receive location information and the like. Registration processing is performed (1001). In this registration process, the SIP server 2 determines the IP address IP-A and the port number PA of the IP telephone terminal A, the IP address IP-B and the port number P- of the IP telephone terminal B from the header information of each REGISTER message. B, IP address IP-C and port number PC of relay terminal C are acquired.

  When the relay terminal C determines that it does not perform voice communication by itself and shifts to the relay mode, the relay terminal C starts transmission of a packet including a relay terminal registration request to the SIP server 2 (1002). This packet transmission is transmitted from the voice relay port S3 used when relaying the subsequent voice communication, and is continuously repeated until the voice communication is completed.

  Next, upon receiving a dial operation by the user (1003), IP telephone terminal A transmits an INVITE message addressed to IP telephone terminal B to SIP server 2 (1004). Here, the notation of “SDP: IP-A, S1” in the INVITE 1004 (SDP: IP-A, S1) is the IP address IP-A that is the communication destination of the subsequent voice communication in the SDP on the INVITE message. It indicates that the port number S1 is included (hereinafter, including the case of the 200 OK message).

  When receiving the INVITE message, the SIP server 2 transfers the message to the relay terminal C (1005). When the relay terminal C receives the transferred INVITE message, the IP address IP-A and the port number S1 of the voice communication destination included in the SDP on the INVITE message are changed to the IP address IP-C and the port number of the own device. Each is rewritten to S3, and the rewritten INVITE message is returned to the SIP server 2 (1006). The SIP server 2 transfers the INVITE message as it is to the IP telephone terminal B (1007).

  The IP telephone terminal B that has received the INVITE message transmits a 180 Ringing message addressed to the IP telephone terminal A that notifies the calling state to the SIP server 2, and the 180 Ringing message is similar to the previous INVITE message. It is sent to IP telephone terminal A via relay terminal C (1008). Subsequently, when the off-hook is executed in the IP telephone terminal B (1009), the IP telephone terminal B transmits a 200 OK message addressed to the IP telephone terminal A to the SIP server 2. The SDP on the 200 OK message includes an IP address IP-B and a port number S2, which are communication destinations for later voice communication (1010). When the SIP server 2 receives the 200 OK message, it forwards it to the relay terminal C (1011). When the relay terminal C receives the forwarded 200 OK message, it rewrites the IP address IP-B and port number S2 included in the SDP on the 200 OK message with its own IP address IP-C and port number S3, respectively. The rewritten 200 OK message is returned to the SIP server 2 (1012). The SIP server 2 transfers the 200 OK message as it is to the IP telephone terminal A (1013).

  The IP telephone terminal A that has received the 200 OK message transmits an ACK message addressed to the IP telephone terminal B to the SIP server 2 in order to notify the reception, and the ACK message includes the previous 180 Ringing message and Similarly, it is sent to the IP telephone terminal B via the relay terminal C (1014).

  By exchanging a series of such SIP messages (1015), a call connection between the IP telephone terminals A and B is established. The IP telephone terminals A and B recognize that the destination of the voice packet to be executed in the subsequent media session is the IP address IP-C and the port number S3 of the relay terminal C.

  When voice communication is started, IP telephone terminal B starts transmission of a voice packet from port S2 of its own device to port S3 of relay terminal C (1016). Similarly, IP telephone terminal A starts transmitting a voice packet from port S1 of its own device to port S3 of relay terminal C (1017). Here, the symbol ● attached on the arrow indicating the transmission of each message and the like in FIG. 4 and the symbol ○ connected with the dashed arrow are the inbounds indicated by the symbol ○ by outbound communication marked with the symbol ●. It shows that communication is possible (the same applies to FIG. 9). That is, a voice packet from 1016 IP telephone terminal B can reach port S3 of relay terminal C by continuously executing 1002 packet transmission. Similarly, a voice packet from 1017 IP telephone terminal A can reach port S3 of relay terminal C by continuously executing 1002 packet transmission. With such a configuration, it is possible to increase the security by minimizing the time required to access the voice relay port S3 of the own apparatus from the outside.

  The relay terminal C that has received the voice packet from the IP telephone terminal B 1016 acquires the port number S2 of the IP telephone terminal B as the information on the voice packet transfer destination. Similarly, the relay terminal C that has received the voice packet from the IP telephone terminal A 1017 acquires the port number S1 of the IP telephone terminal A as the information on the transfer destination of the voice packet. As a result, the relay terminal C transfers the voice packet from the IP telephone terminal B 1016 to the port S1 of the IP telephone terminal A (1018), and the voice packet from the IP telephone terminal B 1017 is transferred to the IP telephone terminal. It is possible to transfer to A port S1 (1019). At this time, 1018 voice packets can reach port S1 of IP telephone terminal A by transmission of 1017 voice packets, and 1019 voice packets can be transmitted to port S2 of IP telephone terminal B by transmission of 1016 voice packets. Can be reached. Thereafter, the same voice packet transmission / reception is repeatedly executed until the voice communication in the media session ends (1020).

  When the voice communication is completed, the IP telephone terminal A transmits a BYE message addressed to the IP telephone terminal B to the SIP server 2 in order to notify the disconnection of the session. The BYE message is the previous 180 Ringing message or the like. In the same manner as described above, it is sent to the IP telephone terminal B via the relay terminal C (1021). Upon receiving this BYE message, IP telephone terminal B transmits a 200 OK message addressed to IP telephone terminal A to SIP server 2 in order to notify the reception, and the 200 OK message is the previous 180 Ringing message. In the same manner as in the above, it is sent to the IP telephone terminal A via the relay terminal C (1022).

  In this way, when the relay terminal C does not perform voice communication by itself, the relay terminal C relays voice communication between the IP telephone terminals A and B, so that the NAT routers A and B to which the IP telephone terminals A and B are connected are relayed. Voice communication can be realized with a simple and inexpensive configuration regardless of the type of NAT and without requiring a relay server such as SBC (Session Border Controller). Further, IP telephone terminals A and B that perform voice communication can perform voice communication in the same procedure as before without requiring a special function.

  Next, with reference to FIG. 5 to FIG. 8, details of processing of the IP telephone terminals A and B, the relay terminal C, and the SIP server in the basic operation of the IP telephone system will be described. Here, for convenience of explanation, processing will be described in accordance with the operation in the IP telephone system shown in FIG.

  FIG. 5 is a flowchart showing the processing procedure of the IP telephone terminal A on the calling side. First, when the IP telephone terminal A is activated, the call control unit 11 transmits a REGISTER message to the SIP server 2 (ST101), and receives a 200 OK message from the SIP server 2 to perform registration processing of location information and the like. (ST102). After that, when the dial operation unit 12 detects the user's dial operation (ST103: YES), the call control unit 11 generates an INVITE message addressed to the destination IP telephone terminal B and transmits it to the SIP server 2 (ST104). ). At this time, the IP address IP-A and the port number S1 of the IP telephone terminal A that is the communication destination of the subsequent voice communication are set in the SDP on the INVITE message.

  Next, the call control unit 11 sequentially receives the 180 Ringing message and the 200 OK message from the IP telephone terminal B via the relay terminal C (ST105, ST106). At this time, in the SDP on the 200 OK message, the IP address IP-C and the port number S3 of the relay terminal C that is the communication destination of the subsequent voice communication are set. Receiving the 200 OK message, call control unit 11 transmits an ACK message addressed to IP telephone terminal B to SIP server 2 (ST107).

  Thereafter, voice packet transmitting / receiving section 13 starts transmitting voice packets addressed to IP telephone terminal B (ST108). This voice packet is sent from port S1 of IP telephone terminal A to port S3 of relay terminal C. Subsequently, voice packet transmitting / receiving section 13 receives a voice packet from IP telephone terminal B (ST109). The received voice packet is transmitted from the port S3 of the relay terminal C to the port S1 of the IP telephone terminal A.

  When the transmission / reception of the voice packet with the IP telephone terminal B as described above is repeatedly executed and finally the call ends (ST110: YES), the call control unit 11 sends the address to the IP telephone terminal B to the SIP server 2. A BYE message is transmitted (ST111), and a 200 OK message from IP telephone terminal B is received via relay terminal C (ST112). After that, returning to ST103, IP telephone terminal A repeatedly executes the same processing as described above.

  FIG. 6 is a flowchart showing the processing procedure of the IP telephone terminal B on the receiving side. First, when IP telephone terminal B is activated, call control unit 11 executes registration processing similar to ST101 and ST102 of FIG. 5 (ST201, ST202). Thereafter, when call reception unit 11 detects an incoming call (receives an INVITE message from IP telephone terminal A) (ST203: YES), it transmits a 180 Ringing message addressed to IP telephone terminal A to SIP server 2 (ST204). ).

  Next, when dial operation unit 12 detects off-hook, call control unit 11 transmits a 200 OK message addressed to IP telephone terminal A to SIP server 2 (ST205). At this time, in the SDP on the 200 OK message, the IP address IP-C and the port number S3 of the relay terminal C that is the communication destination of the subsequent voice communication are set. Subsequently, call control unit 11 receives an ACK message from IP telephone terminal A via relay terminal C (ST206).

  Thereafter, similar to ST108 and ST109 in FIG. 5, voice packet transmission / reception is performed by the voice packet transmission / reception unit 13 (ST207, ST208). When the call is finally finished (ST209: YES), the call control unit 11 receives the BYE message from the IP telephone terminal A via the relay terminal C (ST210), and the IP telephone terminal is connected to the SIP server 2. A 200 OK message addressed to A is transmitted (ST211). After that, returning to ST203, IP telephone terminal B repeatedly executes the same processing as described above.

  FIG. 7 is a flowchart showing the processing procedure of the relay terminal C. First, when the relay terminal C is activated, the call control unit 11 executes registration processing similar to ST101 and ST102 in FIG. 5 (ST301 and ST302). After that, the relay terminal registration unit 15 transmits a packet including a relay terminal registration request from the voice relay port S3 to the SIP server 2 in order to shift to the relay mode when confirming that the own device is not in a call state. Start (ST303).

  Subsequently, when the SIP message relay unit 17 receives the INVITE message addressed to the IP telephone terminal B transferred from the SIP server 2 (ST304), the IP address indicating the communication destination of the voice communication included in the SDP on the INVITE message IP-A and port number S1 are rewritten to IP address IP-C and port number S3 of the own device, respectively, and the rewritten INVITE message is returned to SIP server 2 (ST305).

  Next, SIP message relay unit 17 receives the 180 Ringing message addressed to IP telephone terminal A transferred from SIP server 2, and returns the 180 Ringing message to SIP server 2 (ST306). Subsequently, when the SIP message relay unit 17 receives the 200 OK message addressed to the IP telephone terminal A transferred from the SIP server 2, the IP address IP− of the communication destination of the voice communication included in the SDP on the 200 OK message. B and port number S2 are rewritten to the IP address IP-C and port number S3 of the own device, respectively, and the 200 OK message after the rewriting is returned to SIP server 2 (ST307). Furthermore, SIP message relay unit 17 receives the ACK message addressed to IP telephone terminal B transferred from SIP server 2, and returns the ACK message to SIP server 2 (ST308).

  Next, voice packet relay unit 19 receives a voice packet addressed to IP telephone terminal A from IP telephone terminal B, and obtains its source IP address IP-B and port number S2 (ST309). Subsequently, the voice packet relay unit 19 receives the voice packet addressed to the IP telephone terminal B from the IP telephone terminal A, and acquires the source IP address IP-A and the port number S1 (ST310). Further, voice packet relay section 19 transfers the voice packet received from IP telephone terminal A in ST310 to IP address IP-B and port number S2 of IP telephone terminal B acquired in ST309 (ST311). Voice packet relay section 19 forwards the voice packet received from IP telephone terminal B in ST309 to IP address IP-A and port number S1 of IP telephone terminal A acquired in ST310 (ST312). The voice packet relay unit 19 repeatedly executes such voice packet transfer processing until the end of the call.

  When the talk is finally over (ST313: YES), the voice packet relay unit 19 receives the BYE message addressed to the IP telephone terminal B from the IP telephone terminal A (ST314), and returns the BYE message to the SIP server 2. (ST315). Subsequently, voice packet relay unit 19 receives a 200 OK message addressed to IP phone terminal A from IP phone terminal B (ST315), and returns the 200 OK message to SIP server 2 (ST316). After that, returning to ST304, relay terminal C repeatedly executes the same processing as described above.

  FIG. 8 is a flowchart showing the processing procedure of the SIP server. The terminal registration unit 31 of the SIP server 2 appropriately receives REGISTER messages from the IP telephone terminals A and B and the relay terminal C, acquires their position information, and performs registration processing (ST401). If the registration is successful, terminal registration unit 31 appropriately transmits a 200 OK message to IP telephone terminals A and B and relay terminal C (ST402).

  Next, relay terminal registration unit 34 receives a packet including a relay terminal registration request from relay terminal C (ST403). Accordingly, the relay terminal registration unit 34 records the position information of the relay terminal C and the like in the relay terminal table 35 so as to register the relay terminal C as a relay terminal. The relay terminal registration unit 34 can appropriately accept registrations other than the relay terminal C by receiving similar packets.

  Next, when receiving an INVITE message addressed to IP telephone terminal B from IP telephone terminal A (ST404: YES), SIP message relay section 33 refers to relay terminal table 35 to refer to IP address and port number of relay terminal C. And the INVITE message is transferred to relay terminal C (ST405). Subsequently, the SIP message relay unit 33 receives the INVITE message returned from the relay terminal C and transfers it directly to the IP telephone terminal B (ST406).

  Thereafter, the SIP message relay unit 33 transfers the 180 Ringing message addressed to the IP telephone terminal A from the IP telephone terminal B to the relay terminal C, and transfers the 180 Ringing message returned from the relay terminal C to the IP telephone terminal A. (ST407, ST408). Similarly, SIP message relay section 33 transfers the 200 OK message from IP telephone terminal B to relay terminal C, and forwards the 200 OK message returned from relay terminal C to IP telephone terminal A (ST409, ST410). . Further, SIP message relay unit 33 transfers the ACK message from IP telephone terminal A to relay terminal C, and transfers the ACK message returned from relay terminal C to IP telephone terminal B (ST411, ST412).

  Thereafter, IP telephone terminal A and IP telephone terminal B perform voice communication by relay of relay terminal C, but SIP server 2 does not intervene during this time.

  When the call ends due to on-hook or the like, the SIP message relay unit 33 receives the BYE message from the IP telephone terminal A, transfers it to the relay terminal C, receives the BYE message returned from the relay terminal C, and receives the IP message. Transfer to telephone terminal B (ST413, ST414). Similarly, SIP message relay unit 33 transfers the 200 OK message from IP telephone terminal B to relay terminal C (ST415), and transfers the 200 OK message returned from relay terminal C to IP telephone terminal A (ST416). ). After that, returning to ST404, the SIP server 2 repeatedly executes the same processing as described above.

  Next, in the operation of the IP telephone system shown in FIG. 4, the relay change operation when the transition to the normal mode becomes necessary while the relay terminal C is executing the relay mode will be described.

  FIG. 9 is a sequence diagram showing the relay change operation of the IP telephone system according to the present invention. 4 shows a case where there is an incoming call from the IP telephone terminal E to the relay terminal C in the operation shown in FIG. 4 and the relay processing being executed by the relay terminal C is taken over by the relay terminal D. The operation of the IP telephone system shown in FIG. 9 is started during the voice communication process 1020 in FIG.

  When the relay terminal D shifts to the relay mode after executing the registration process similar to 1001 in FIG. 4, the relay terminal D starts transmission of a packet including a relay terminal registration request to the SIP server 2 (2001). This packet transmission is transmitted from the voice relay port S4 used when relaying the subsequent voice communication, and is continuously repeated until the voice communication ends.

  When the SIP server 2 receives a packet including a relay terminal registration request from the relay terminal D, the SIP server 2 adds the position information (IP address, port number, etc.) of the relay terminal D to the relay terminal C in the relay terminal table. Remember.

  A case where the IP telephone terminal E makes a call to the relay terminal C while the relay terminal C is relaying will be described below. Upon receiving a dial operation by the user (2002), the IP telephone terminal E transmits an INVITE message addressed to the relay terminal C to the SIP server 2 (2003). At this time, the IP address IP-E and the port number S5 of the IP telephone terminal E that is the communication destination of the subsequent voice communication are set in the SDP on the INVITE message.

  When receiving the INVITE message, the SIP server 2 transfers the message to the relay terminal C (2004). As a result, the relay terminal C confirming the incoming call from the IP telephone terminal E transmits a REFER message to the IP telephone terminal A via the SIP server 2 (2005). The reason why the relay terminal C transmits the REFER message is to instruct the SIP server 2 to transfer the SIP message transmitted / received between the IP telephone apparatuses to another relay terminal. The REFER message includes a command for sending a RE-INVITE message (call control message for reestablishing a connection between the IP telephone apparatuses A and B) to the IP telephone terminal A.

  When receiving the REFER message, IP telephone terminal A transmits a RE-INVITE message addressed to IP telephone terminal B to SIP server 2 (2006). When the SIP server 2 receives the RE-INVITE message, the SIP server 2 transfers the message to the relay terminal D (2007). The relay terminal D rewrites the IP address IP-A and the port number S1 indicating the communication destination of the voice communication included in the SDP of the transferred INVITE message with the IP address IP-D and the port number S4 of the own device, respectively. The subsequent INVITE message is returned to the SIP server 2 (2008). The SIP server 2 transfers the INVITE message as it is to the IP telephone terminal B (2009).

  Next, the IP telephone terminal B transmits a 200 OK message addressed to the IP telephone terminal A to the SIP server 2 (2010). The SIP server 2 transfers the 200 OK message to the relay terminal D (2011). The relay terminal D rewrites the forwarded 200 OK message with the IP address IP-B and the port number S2 included in the above SDP to the IP address IP-D and port number S4 of its own device, and the rewritten 200 OK message. The message is returned to the SIP server 2 (2012). The SIP server 2 transfers the 200 OK message as it is to the IP telephone terminal A (2013).

  The IP telephone terminal A that has received the 200 OK message transmits an ACK message addressed to the IP telephone terminal B to the SIP server 2, and the ACK message is sent to the IP telephone terminal B via the relay terminal D (2014). ). Subsequently, the IP telephone terminal A receives the 200 OK message from the IP telephone terminal B through the SIP server 2 after receiving the 200 OK message in which the SDP is rewritten with the IP address and port number of the relay terminal D in 2013. A NOTIFY message is transmitted to the relay terminal C (2015). This NOTIFY message includes information on permission of relay termination to the relay terminal C (that is, the fact that a RE-INVITE message sending command by the previous REFER message has been accepted). The relay terminal C that has received this NOTIFY message determines that the relay process has succeeded in the other relay terminal, starts connection with the IP telephone terminal E (2016), and receives the IP telephone via the SIP server 2. A 200 OK message is transmitted to terminal A (2017). When the SIP server 2 relays the NOTIFY message in 2015, it deletes the relay terminal C from the relay terminal table.

  By exchanging the RE-INVITE message and the 200 OK message, the call connection between the IP telephone terminals A and B is established again. The IP telephone terminals A and B recognize that the destination of the voice packet to be executed in the subsequent media session is the IP address IP-D and the port number S4 of the relay terminal D.

  The processes 2018 to 2021 after the start of voice communication are substantially the same as the processes 1016 to 1019 in FIG. 4 except that the relay terminal D relays voice packets instead of the relay terminal C. Is omitted. Although not shown in FIG. 9, the same processing as 1021 and 1022 in FIG. 4 is executed at the end of the talk. Note that, when the call with the IP telephone terminal E ends, the relay terminal C can shift to the relay mode again and start transmitting a packet including a relay terminal registration request to the SIP server 2.

  With this configuration, the relay terminal C can easily take over the relay process to the relay terminal D when it becomes necessary to perform voice communication by itself while performing the voice communication relay. The inconvenience that the voice communication between the telephone terminals A and B is interrupted can be avoided. Here, the case where the relay process is taken over upon receipt of an incoming call from IP telephone terminal E is shown, but the same process can also be executed when a call is made from relay terminal C to another IP telephone terminal. is there.

  Next, with reference to FIG. 10 to FIG. 14, details of processing of the IP telephone terminals A, B, E, the relay terminals C, D, and the SIP server in the relay change operation of the IP telephone system will be described. Here, for convenience of explanation, processing will be described according to the operation in the IP telephone system shown in FIG.

  FIG. 10 is a flowchart showing the processing procedure of the IP telephone terminal A on the calling side. IP telephone terminal A executes ST501 to ST503 similar to ST101 to ST103 in FIG. 5 and then executes the same processing as ST104 to ST110 in FIG. 5 to indicate that a call is in progress via IP telephone terminal B and relay terminal C. (ST504).

  During a call with IP telephone terminal B, call control unit 11 receives a REFER message from relay terminal C (ST505) and, based on the RE-INVITE message sending command included in the REFER message, Then, a RE-INVITE message addressed to IP telephone terminal B is transmitted (ST506). At this time, the IP address IP-A and the port number S1 of the IP telephone terminal A that is the communication destination of the subsequent voice communication are set in the SDP on the RE-INVITE message.

  Next, call control section 11 receives a 200 OK message from IP telephone terminal B via relay terminal D (ST507). At this time, in the SDP on the 200 OK message, the IP address IP-D and the port number S4 of the relay terminal D that is the communication destination of the subsequent voice communication are set. Receiving the 200 OK message, call control section 11 transmits an ACK message addressed to IP telephone terminal B to SIP server 2 (ST508).

  Next, the call control unit 11 recognizes that the relay terminal has been changed from the terminal C to the terminal D in response to the reception of the 200 OK message from the terminal B in 2013, and permits the relay terminal C to end the relay. (ST509) and 200 OK message from relay terminal C is received (ST510).

  The processing of ST511 to ST515 after the start of voice communication is substantially the same as the processing of ST108 to ST112 of FIG. 5 except that the relay terminal D relays the voice packet instead of the relay terminal C, and here, Description is omitted.

  FIG. 11 is a flowchart showing the processing procedure of the IP telephone terminal B on the receiving side. IP telephone terminal B performs the same processing as ST204 to ST209 in FIG. 5 after executing ST601 to ST603 similar to ST201 to ST203 in FIG. 6, and performs a call through IP telephone terminal A and relay terminal C. (ST604).

  During a call, call control unit 11 receives a RE-INVITE message from IP telephone terminal A via relay terminal D (ST605). At this time, in the SDP on the RE-INVITE message, the IP address IP-D and the port number S4 of the relay terminal D that is the communication destination of the subsequent voice communication are set. The call control unit 11 transmits a 200 OK message in which the IP address IP-D of the relay terminal D and the port number S4 are set to SDP to the SIP server 2 to the IP telephone terminal A (ST606). Furthermore, call control section 11 receives an ACK message from IP telephone terminal A via relay terminal D (ST607).

  Thereafter, the voice packet transmitting / receiving unit 13 performs voice packet transmission / reception in the same manner as ST207 and ST208 in FIG. 6 (ST608, ST609). When the call is finally finished (ST610: YES), the call control unit 11 receives a BYE message from the IP telephone terminal A via the relay terminal D (ST611), and receives a 200 OK message addressed to the IP telephone terminal A. Is transmitted to the SIP server 2 (ST612). After that, returning to ST603, IP telephone terminal B repeatedly executes the same processing as described above.

  FIG. 12 is a flowchart showing the processing procedure of the relay terminal C. Relay terminal C executes ST701 to ST704 similar to ST301 to ST304 in FIG. 7 and then executes the same processing as ST405 to ST413 in FIG. 7 to perform relay processing between IP telephone terminal A and IP telephone terminal B (ST705).

  Next, the call control unit 11 receives the INVITE message from the IP telephone terminal E (ST706), whereby the SIP message relay unit 17 transfers the SIP message transmitted / received between the IP telephone apparatuses to another relay terminal. In order to instruct the SIP server 2 to do this, a REFER message addressed to the IP telephone terminal A is transmitted to the SIP server 2 (ST707). Subsequently, SIP message relay unit 17 receives the NOTIFY message from IP telephone terminal A (ST708), and transmits a 200 OK message to IP telephone terminal A (ST709). Upon reception of this NOTIFY message, the SIP message relay unit 17 determines that the relay process has been successfully taken over by another relay terminal, and ends the relay process (ST710). At the same time, the relay terminal registration unit 15 stops sending packets to the SIP server 2.

  Thereafter, call control unit 11 executes the same processing as IP telephone terminal B in ST204 to ST208 of FIG. 6, and performs voice communication with IP telephone terminal E (ST711). When the talk is finally ended (ST712: YES), the call control unit 11 determines that the relay mode can be executed again, and the relay terminal registration unit 15 thereby requests the SIP server 2 to register the relay terminal. The transmission of the packet including is started again (ST713). After that, returning to ST704, relay terminal C repeatedly executes the same processing as described above.

  FIG. 13 is a flowchart showing the processing procedure of the relay terminal D that takes over the relay processing between the IP telephone terminal A and the IP telephone terminal B from the relay terminal C. After the relay terminal D executes ST801 to ST803 similar to ST301 to ST303 by the relay terminal C in FIG. 7, the SIP message relay unit 17 transmits the RE-INVITE message addressed to the IP telephone terminal B transferred from the SIP server 2. Receive (ST804). Thereby, the relay terminal D recognizes that its own device is selected as the relay terminal. Subsequently, the SIP message relay unit 17 uses the IP address IP-A and the port number S1 as the communication destination of the voice communication included in the SDP on the RE-INVITE message as the IP address IP-D and the port number S4 of its own device. And the RE-INVITE message after the rewriting is returned to the SIP server 2 (ST805).

  Next, when the SIP message relay unit 17 receives the 200 OK message addressed to the IP telephone terminal A transferred from the SIP server 2, the IP address IP serving as the voice communication destination included in the SDP on the 200 OK message. -B and port number S2 are rewritten to IP address IP-D and port number S4 of the own device, respectively, and the 200 OK message after the rewriting is returned to SIP server 2 (ST806).

  The subsequent processing of ST807 to ST814 is the same processing as ST308 to ST315 by the relay terminal C in FIG.

  Note that the IP telephone terminal E and the relay terminal C can perform voice communication by establishing a call connection without using a relay terminal, as in the case of a conventional IP telephone. Alternatively, the IP telephone terminal E and the relay terminal C can execute the same processing as in the case of the IP telephone terminal A and the IP telephone terminal B of FIG. 5 via another relay terminal (not shown).

  FIG. 14 is a flowchart showing the processing procedure of the SIP server. After executing ST901 to ST904 similar to ST401 to ST404 in FIG. 8, the SIP server 2 executes call connection processing between IP telephone terminal A and IP telephone terminal B corresponding to ST405 to ST416 in FIG. ST905).

  Subsequently, as in ST903, the relay terminal registration unit 34 receives a packet including a relay terminal registration request from the relay terminal D, and registers the relay terminal D in the relay terminal table in addition to the relay terminal C ( ST906).

  Next, SIP message relay section 33 receives the INVITE message addressed to IP telephone terminal C from IP telephone terminal E, and forwards the message to relay terminal C (ST907). Subsequently, the SIP message relay unit 33 receives the REFER message from the relay terminal C, and transfers this message to the IP telephone terminal A (ST908). Here, the SIP message relay unit 33 transfers the REFER message received from the relay terminal C after receiving the INVITE message addressed to the IP telephone terminal C to the SIP message transmitted / received between the IP telephone apparatuses to another relay terminal. Is recognized as a relay change instruction.

  Next, the SIP message relay unit 33 receives the RE-INVITE message addressed to the IP telephone terminal B from the IP telephone terminal A (ST909), refers to the relay terminal table 35, and the IP address and port number of the relay terminal D And transfers the RE-INVITE message to relay terminal D (ST910).

  Thereafter, the SIP message relay unit 33 executes ST911 to ST914 in the same manner as ST409 to ST412 of FIG. Subsequently, the SIP message relay unit 33 receives the NOTIFY message addressed to the relay terminal C from the IP telephone terminal A, and transfers this message to the relay terminal C (ST915). Further, SIP message relay section 33 transfers the 200 OK message from relay terminal C to IP telephone terminal A (ST916).

  Next, relay terminal registration unit 34 receives a packet including a relay terminal registration request from relay terminal C (ST917). As a result, the relay terminal registration unit 34 registers the relay terminal C again as a relay terminal. After that, returning to ST904, the SIP server 2 repeatedly executes the same processing as described above.

  Although the present invention has been described in detail based on specific embodiments, these embodiments are merely examples, and the present invention is not limited to these embodiments. For example, the IP telephone system can be configured to execute the relay processing by the relay terminal only when voice communication between IP telephone terminals cannot be realized by the conventional NAT traversal method. In this case, for example, based on the NAT type of the relay device to which the IP telephone terminal that performs voice communication is connected, it is possible to determine whether or not the relay processing by the relay terminal is necessary.

  The IP telephone apparatus with a relay function according to the present invention, the IP telephone system including the IP telephone system, and the communication relay method are not limited to the NAT type of the relay apparatus to which the IP telephone apparatus is connected. It is possible to realize voice communication between apparatuses, and is useful as an IP telephone apparatus that performs communication in a wide area network including a call control server such as an SIP server, an IP telephone system including the IP telephone apparatus, and a communication relay method.

System configuration diagram of IP telephone system according to the present invention Functional block diagram of IP phone terminal Functional block diagram of SIP server Sequence diagram showing basic operation of IP telephone system FIG. 4 is a flowchart showing the processing procedure of IP telephone terminal A FIG. 4 is a flowchart showing the processing procedure of IP telephone terminal B FIG. 4 is a flowchart showing the processing procedure of the relay terminal C in FIG. The flowchart which shows the process of the SIP server in FIG. Sequence diagram showing relay change operation of IP telephone system FIG. 9 is a flowchart showing the processing procedure of IP telephone terminal A FIG. 9 is a flowchart showing the processing procedure of IP telephone terminal B FIG. 9 is a flowchart showing a processing procedure of the relay terminal C in FIG. FIG. 9 is a flowchart showing the processing procedure of the relay terminal D in FIG. FIG. 9 is a flowchart showing the processing procedure of the SIP server.

Explanation of symbols

1 IP telephone system 2 SIP server (call control server)
3 Internet (wide area network)
4-8 LAN
11 Call control unit 13 Voice packet transmission / reception unit 15 Relay terminal registration unit (transfer instruction means)
17 SIP message relay unit (message relay means, relay change instruction means)
19 Voice packet relay unit (voice relay means)
31 terminal registration unit 33 SIP message relay unit 34 relay terminal registration unit

Claims (7)

An IP telephone apparatus with a relay function used in an IP telephone system having a plurality of IP telephone apparatuses connected to a wide area network via a relay apparatus having a NAT function and a call control server for controlling calls of these IP telephone apparatuses. And
The IP telephone device with a relay function is:
Connected to the wide area network via a relay device having a complete cone type NAT function,
There are two operation modes: a normal mode for performing voice communication with the IP telephone device, and a relay mode for relaying voice communication performed between other IP telephone devices,
When in the relay mode, transfer instruction means for instructing the call control server to transfer a call control message transmitted / received between other IP telephone devices via the call control server;
The call control message transferred from the call control server based on the transfer instruction is received, the destination and the source IP address and port number included in the call control message are recognized, and the SDP of the call control message is Message relay means for rewriting the IP address and port number of the source included in the IP address and port number of the own device, and returning the rewritten call control message to the call control server;
After connection to execute voice communication between the other IP telephone devices is established by transmission / reception of the call control message, it is transmitted from one of the IP telephone devices to the IP address and port number of the own device An IP telephone apparatus with a relay function, comprising: voice relay means for relaying the voice communication by transferring communication data to the other, which is an original transmission destination, based on the recognition . The transfer instruction means instructs the transfer of the call control message by continuously and repeatedly transmitting a transfer instruction request packet to the call control server, while stopping the transmission of the packet. The IP telephone apparatus with a relay function according to claim 1, wherein   The IP telephone apparatus with a relay function according to claim 2, wherein the transfer instruction request packet is transmitted from a port used for voice relay in the own apparatus.   When shifting to the normal mode while the relay mode is being executed, one of the IP telephone apparatuses that perform the voice communication is instructed to send a call control message for re-establishing a connection between the IP telephone apparatuses. And a relay change instruction means for instructing the call control server to transfer a call control message transmitted / received between the IP telephone apparatuses to another IP telephone apparatus with a relay function. The IP telephone apparatus with a relay function according to claim 1 or 2.   An IP telephone system comprising the IP telephone device with a relay function according to any one of claims 1 to 4.   The call control server receives a registration request from the IP telephone apparatus and registers the IP telephone apparatus in a relay terminal registration table as a relay terminal that relays the call control message or communication data; 6. The IP telephone system according to claim 5, further comprising transfer means for transferring a call control message transmitted / received between the IP telephone apparatuses to the relay terminal registered in the relay terminal registration table. A communication relay method used in an IP telephone system having a plurality of IP telephone apparatuses connected to a wide area network via a relay apparatus having a NAT function and a call control server for controlling calls of these IP telephone apparatuses,
At least one of the IP telephone devices is connected to the wide area network via a relay device having a complete cone-type NAT function, and between a normal mode for performing voice communication with the IP telephone device, and between other IP telephone devices. It has two operation modes, the relay mode that relays voice communication to be executed,
When in the relay mode, the call control server is instructed to transfer a call control message transmitted / received between other IP telephone devices via the call control server,
The call control message transferred from the call control server based on the transfer instruction is received, the destination and the source IP address and port number included in the call control message are recognized, and the SDP of the call control message is Rewriting the IP address and port number of the caller included in the IP address and port number of the own device, respectively, and returning the call control message after the rewriting to the call control server,
After connection to execute voice communication between the other IP telephone devices is established by transmission / reception of the call control message, it is transmitted from one of the IP telephone devices to the IP address and port number of the own device A communication relay method for relaying the voice communication by transferring communication data to the other, which is an original transmission destination, based on the recognition .

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