Abstract:
The present invention relates to a relay device enabling efficient management of a voice message and a caller to wait for transmission of the voice message in a short time. The relay device is located at a boundary between an existing telephone network and an IP network, converts and relays a voice signal and an IP packet. The relay device comprises a receiving unit receiving voice source data into which a voice message is digitized, a packet processing unit converting the received voice source data into a voice source data packet as the IP packet, a storage unit storing the converted voice source data packets, and a transmitting unit transmitting, when transmitting a designated voice message to a designated destination, the voice source data packet corresponding to the designated voice message among the stored voice source data packets.

Description:
BACKGROUND OF THE INVENTION 
   1. Field of the Invention 
   The invention relates a relay device for relaying data in a way that mutually converts voice signals and IP packets between an existing telephone network and an IP (Internet Protocol) network. 
   2. Description of the Related Art 
   At the present, utilization of a technology for performing a voice talk is underway, wherein the existing Public Switched Telephone Networks (which will hereinafter be abbreviated to PSTNs) are connected to each other via an IP (Internet Protocol) network. This type of technology involves employing VOIP (Voice Over Internet Protocol) etc. for transferring and receiving the voice signals over the IP network. 
   This type of VOIP-based conventional voice talk system (which will hereinafter be referred to as a conventional system) will be explained with reference to  FIG. 17 .  FIG. 17  is a view showing an example of a network architecture of the conventional telephone system. In the conventional system shown in  FIG. 17 , a VOIP gateway (which will hereinafter be abbreviated to VOIPGW) located at a boundary between the PSTN and the IP network voice-packetizes digital signals (STM (synchronous Transport Module)-1, STM-4, etc.) transferred and received over the PSTN by use of a self-equipped CODEC (Coder/Decoder) etc. and forwards the voice packets to the IP network, thereby actualizing voice communications. 
   The conventional system is that, as illustrated in  FIG. 17 , telephones  215  as subscriber terminals connected to a PSTN  211  are connected to telephones  216 , etc. connected to another PSTN  212  via an IP network  210 . Further, the PSTNs  211  and  212  are connected to the IP network  210  via VOIP gateways  213  and  214 , respectively. Moreover, a call agent (which will hereinafter abbreviated to CA) is connected to the IP network  210 , wherein this CA controls calls from the respective telephones  215  and  216 . Further, an FTP (File Transfer Protocol) server  218  is connected to the IP network  210 . The FTP server  218  retains digital data (which will hereinafter be referred to as voice source data) etc. into which a guidance message of a talkie etc. is voice-coded by utilizing a μ-LAW 64 kbs PCM (Pulse Coded Modulation (ITU-T G.711) system and so on. 
   Next, an operation of the conventional system on the occasion of providing a service for flowing the guidance message of the talkie etc. to the telephone as the subscriber terminal, will be explained with reference to  FIGS. 17 and 18 .  FIG. 17  is a view showing a network architecture of the conventional system and also illustrating how the voice source data are transferred from the FTP server.  FIG. 18  is a view showing how the voice source data are sent to the PSTN from the VOIPGW in response to an instruction of the CA in the conventional system shown in  FIG. 17 . 
   To start with, as preprocessing, the FTP server  218  transfers, as shown in  FIG. 17 , the voice source data to the VOIP gateways  213  and  214  (S 219 ). Then, the VOIP gateways  213  and  214  receiving the voice source data store memories with the voice source data. Namely, the voice source data related to the message of the talkie etc. are stored on the respective VOIP gateways. 
   Next, the operation of the conventional system for actually flowing the message to each telephone, will be described. The conventional system sends the message in response to a call from the telephone as the user terminal. In this case, the CA notifies each VOIPGW of call control information such as call setting, a voice source data add instruction, etc. (S 221 ). The VOIPGW notified of the call control information adds the voice source data to a designated timeslot in the STM, thereby sending the voice source data to a target telephone (S 222 ). 
   An operation of the VOIPGW stored with voice source data and sending the stored voice source data to the target telephone, will be described with reference to  FIG. 19 .  FIG. 19  is a diagram showing a configuration of the VOIPGW in the conventional system and also illustrating how the VOIPGW is stored with the voice source data and sends the voice source data. Note that  FIG. 19  shows a functional configuration by exemplifying the VOIPGW  213 . 
   The VOIPGW  213  is constructed of an IP switch unit  231  serving as an interface with the IP network, an STM switch control unit  232  serving as an interface with the PSTN, a control unit  233 , a CODEC unit  234 , a packet processing unit  235  and a packet buffer  236 . The STM switch control unit  232  is further constructed of a voice source data storage memory  237 , a voice source data add unit  238 , etc. 
   In the case of storing the voice source data given from the FTP server  218 , the VOIPGW  213  receives the voice source data from the IP network  210  and stores the voice source data on the voice source data storage memory  237  within the STM switch control unit  232  via the IP switch unit  231 , the packet processing unit  235  and the control unit  233  (a data flow indicated by a dotted line in  FIG. 19 ). 
   On the other hand, in the case of sending the voice source data to the telephone, the VOIPGW  213  receives a call control signal from the CA. The VOIP gateway  213  receiving the call control signal from the CA instructs the packet processing unit  235 , the CODEC unit  234  and the STM switch control unit  232  to perform call setting in accordance with the call control signal (a data flow indicated by one-dotted broken line in  FIG. 19 ). Next, the VOIPGW  213  receives a voice source data add instruction from the CA. Upon receiving the instruction, the control unit  233  instructs the voice source data add unit  238  to send (add) the voice source data into a channel (call) designated in the voice source data add unit  238  (a data flow indicated by a solid line in  FIG. 19 ). 
   An operation of the voice source data add unit  238  will be explained in greater detail with reference to  FIG. 20 . FIG.  20  is a diagram showing a detailed functional configuration of the voice source data add unit  238  in the conventional VOIPGW. The voice source data add unit  238  adds, based on the voice source data add instruction given from the control unit  233 , the designated voice source data into the designated channel. Further, respective functional units of the voice source data add unit  238 , as one frame is transmitted and received at an interval of 125 micro second (μs) when the PSTN employs the STM-1 communication system, execute the following processes within this interval. 
   A voice source data readout control unit  241 , in accordance with the voice source data add instruction given from the control unit  233 , on a channel-by-channel basis, calculates a readout address, reads the voice source data from the voice source data storage memory  237 , retains the readout data on an add data storage register  234  (channel unit), and updates and retains the readout address on a voice source data readout address storage register  244  (channel unit). Moreover, a voice source data add processing unit  246  reads the add data from the add data storage register  243  and adds the add data in synchronization with a channel-by-channel transmission timing. 
   As described above, in the conventional system, the voice source data transmitted by the FTP server  218  are stored on the voice source data storage memory  237  of the VOIP gateway  213 . Then, in the case of sending the voice source data in accordance with the call given from the subscriber terminal, the VOIP gateway  213  reads the voice source data on the channel-by-channel basis (the channel unit) from the voice source data storage memory  237 , and the voice source data add processing unit  246  adds (allocates) the voice source data to a predetermined timeslot of the STM. 
   Note that the conventional art related to the present invention of the application is disclosed in the following document. The conventional art document is “Japanese Patent Application Laid-Open Publication No.4-239254”. 
   The voice source data storage/transmission method in the conventional system, however, has the following problems. 
   First, in the conventional system, the storage of the voice source data involves preparing a dedicated memory such as a ROM (Read Only Memory) etc. in the STM switch control unit  232  within the VOIPGW in order to store the voice source data. The STM switch control unit  232  is normally mounted with only a small-capacity memory. Therefore, it is required that a memory for storing the voice source data be provided for this purpose. A flash ROM suited to accessing on a 1-byte basis is in many cases employed for this dedicated memory. This is because in the case of adding the data into a timeslot corresponding to each channel, the data are required to be added on the 1-byte basis in terms of STM communications standards. Further, in the case of having a necessity of storing plural categories of voice source data, even when using a large-capacity flash ROM, a plurality of memories are needed. For example, the flash ROM having an 8-megabyte (MB) capacity is stored with only the voice source data on the order of 16 min as a total. 
   Second, on the occasion of adding the voice source data into the STM timeslot, there can be no perfect assurance for searching out the head of the voice message, corresponding to a call of every subscriber terminal. This is derived from the following reasons. In the case of adding the voice source data into the STM timeslot, it is required that the voice source data be separately readout for every channel corresponding to the call. If the number of channels which should be added at a time increases, there must be a rise in data size of the data to be read out within a predetermined frame interval (e.g., 125 μs at 64 Kbps), and hence the memory access speed does not catch up with this rise. 
   Concerning this problem, there is proposed a method that the voice source data are previously read out at a certain fixed interval in order to search out the head of the voice message, and the closest readout data is selected and added into the target timeslot (refer to “Japanese Patent Application Laid-Open Publication No.4-239254”). In this method also, however, if there are plural categories of voice messages, it follows that a limit of a voice source data readout interval is determined from the memory access time, and hence there is no perfect head-search-out function. For instance, in the case of the voice source data on the order of  16  min as a total, supposing that the access time to the flash ROM is 90 nanoseconds (ns), the readout interval is equal to or larger than approximately 700 milliseconds (ms). In the case of the voice source data on the order of 32 min as a total, the readout interval is equal to or larger than approximately 1.4 sec. 
   Third, the memory management of the memory (the voice source data storage memory  237 ) for storing the voice source data of the voice message becomes troublesome. The conventional system, in the case of storing the voice source data corresponding to the voice message, requires ensuring a memory area for a maximum length of the voice source data that should be stored previously. Further, the voice source data, if not stored in one area, are divided into equal data segments and thus stored. Under such a condition, when changing the voice message, especially when changing into a voice message having a different message length, it is required that the memory area already stored with the voice source data be released and that the segmented memories be reallocated to the voice source data for the change. 
   SUMMARY OF THE INVENTION 
   It is an object of the present invention to actualize a relay device enabling efficient management of a voice message and a caller to wait for transmission of the voice message in a short time in the relay device that converts a voice signal between an existing public switched telephone network and an IP network. 
   The invention adopts the following configurations in order to solve the problems. Namely, the present invention is a relay device located at a boundary between an existing telephone network and an IP network and relaying a voice signal and an IP packet in a way that mutually converts the voice signal transferred and received over the existing telephone network and the IP packet transferred and received over the IP network, which the relay device comprises a receiving unit receiving voice source data into which a voice message is digitized, a packet processing unit converting the received voice source data into a voice source data packet as the IP packet, a storage unit storing the converted voice source data packets, and a transmitting unit transmitting, when transmitting a designated voice message to a designated destination, the voice source data packet corresponding to the designated voice message among the stored voice source data packets to the designated destination. 
   In the present invention, the voice source data packets, which convert the voice source data corresponding to the voice message into IP packets, are retained, and the voice source data packet corresponding to the designated voice message among the retained voice source data packets are transmitted. 
   Accordingly, the relay device according to the present invention handles the IP packets when transmitting the voice message and is therefore capable of high-speed processing as compared with the conventional system handling the voice source data themselves in conformity with a communication mode over the existing telephone network. Owing to this, it is possible to increase the number of simultaneously-transmittable destinations and to transmit the voice message without the caller&#39;s waiting in a long time. 
   Further, the relay device according to the present invention further comprises a transfer unit transmitting, toward the IP network, the voice source data packet converted by the packet processing unit so that the packet is addressed to other relay device. Moreover, the relay device of the present invention further comprises a packet receiving unit receiving the voice source data packet transmitted from the other relay device, and a packet storage unit storing the received voice source data packets. 
   In the present invention, the voice source data are packetized, and the voice source data packet is transferred to other relay device. On the other hand, in the case of receiving the voice source data packet transferred from the other relay device, the received voice source data packet is stored direly on the self storage unit. 
   Therefore, according to the present invention, it is possible to limit the number of the relay devices each having the function of packetizing the voice source data and to build up a system that restrains the cost in the system employing the relay device according to the present invention. 
   Further, the relay device further comprises a data transmitting unit transmitting, toward the IP network, the voice source data packet stored on the storage unit not as a voice packet but as a data packet so that the packet is addressed to the other relay device. Moreover, the relay device further comprises a data receiving unit receiving the data packet transmitted from the other relay device, wherein the voice source data packet in the data packet received by the data receiving unit is stored. 
   In the present invention, the voice source data packet retained by the relay device according to the present invention is transferred as a normal data packet to the other relay device. Then, the other relay device receiving the transferred data packet retains the self storage unit with the voice source data packet in the data packet. 
   Hence, according to the present invention, the voice source data can be handled and transferred in the same way as the normal data can be without transmitting the data with a predetermined period etc. as in the case of the voice packet. This makes it possible to copy the voice source data held by one relay device to the plurality of relay devices by a simple method. 
   Note that the present invention may be a program for actualizing any one of the functions given above. Further, the present invention may also be a readable-by-computer stored with such a program. 
   According to the present invention, it is possible to actualize the relay device enabling the efficient management of the voice message and the caller to wait for transmission of the voice message in the short time in the relay device that converts the voice signal between the existing public switched telephone network and the IP network. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
       FIG. 1  is a diagram showing a network architecture in a first embodiment; 
       FIG. 2  is a diagram showing a diagram showing a functional configuration of a VOIPGW in the first embodiment; 
       FIG. 3  is a diagram showing a diagram showing a detailed functional configuration of a voice source data transfer unit in the first embodiment; 
       FIG. 4  is a diagram showing a diagram showing a detailed functional configuration of a packet processing unit in the first embodiment; 
       FIG. 5  is a diagram showing a flowchart showing a voice source data storage process of the VOIPGW in the first embodiment; 
       FIG. 6  is a diagram showing a flowchart showing the voice source data storage process in the packet processing unit in the first embodiment; 
       FIG. 7  is a diagram showing a flowchart showing a voice source data delete process of the VOIPGW in the first embodiment; 
       FIG. 8  is a diagram showing a flowchart showing the voice source data delete process in the packet processing unit in the first embodiment; 
       FIG. 9  is a diagram showing a flowchart showing a voice source data transmission process of the VOIPGW in the first embodiment; 
       FIG. 10  is a diagram showing a flowchart showing the voice source data transmission process in the packet processing unit in the first embodiment; 
       FIG. 11  is a diagram showing a diagram showing an outline of voice source data storage/delete/transmission processes; 
       FIG. 12  is a diagram showing a diagram showing an example of a buffer memory access in the voice source data transmission process; 
       FIG. 13  is a diagram showing a diagram showing a configuration of the VOIPGW and a control flow in a second embodiment; 
       FIGS. 14A and 14B  are diagrams showing a diagram showing an other-device transfer sequence in the second embodiment; 
       FIG. 15  is a diagram showing a diagram showing a configuration of the VOIPGW and a control flow in a third embodiment; 
       FIG. 16  is a diagram showing a diagram showing an inter-buffer transfer process sequence of the VOIPGW in the third embodiment; 
       FIG. 17  is a diagram showing a diagram showing a network architecture in a conventional system; 
       FIG. 18  is a diagram showing a diagram showing call setting/voice source data transmission to VOIPGW in the conventional system; 
       FIG. 19  is a diagram showing a configuration of a VOIPGW and a control flow in the conventional system; and 
       FIG. 20  is a diagram showing a detailed functional configuration of a voice source data adding unit in the conventional system; 
   

   DESCRIPTION OF THE PREFERRED EMBODIMENTS 
   First Embodiment 
   A VOIP (Voice Over Internet Protocol) gateway device (which will hereinafter be abbreviated to VOIPGW) according to a first embodiment of the present invention, will be described with reference to the drawings. A configuration of the first embodiment is an exemplification, and the present invention is not limited to the configuration of the first embodiment. 
   Network Architecture 
   To begin with, a network architecture of a VOIP call system configured by the VOIPGW according to the first embodiment, will be explained referring to  FIG. 1 .  FIG. 1  is a view showing the network architecture of this VOIP call system. 
   The network in this VOIP call system is configured by public switched telephone networks (which will hereinafter be abbreviated to PSTNs)  10 ,  20 , and  30 , and an IP network  3 , wherein the PSTNs  10 ,  20  and  30  are respectively connected to the IP network  3  via VOIPGWs  11 ,  21  and  31  according to the embodiment. Telephones  15 ,  16 ,  25 ,  26  and  35  serving as subscriber terminals are connected to the PSTNs, wherein the VOIP call system provides a call service to each of these telephones. The respective PSTNs are built up by STM (Synchronous Transport Module), in which the VOIPGWs  11 ,  21  and  31  voice-packetize digital signals (STM-1, STM-4, etc.) transferred and received over the PSTNs and relay the voice packets toward the IP network  3 , thereby actualizing interconnections. Note that the subscriber terminal may be an IP telephone connectable directly to the IP network  3  as in the case of an H323 terminal  6  in  FIG. 1 . Further, an FTP (File Transfer Protocol) server  1  and a call agent (which will hereinafter be abbreviated to CA)  2  are connected to the IP network  3 . 
   The FTP server  1  retains digital data (which will hereinafter be referred to as voice source data) etc. into which a guidance message of a talkie etc. is voice-coded by utilizing an ITU-T G.711 system (μ-Law 64 kbs PCM (Pulse Coded Modulation) and so on, and provides the voice source data to each VOIPGW. The CA  2  controls a call from the subscriber terminal and makes the IP network  3  function as a relay switched network between the respective PSTNs. Accordingly, the VOIPGWs  11 ,  21  and  31 , the FTP server  1 , etc. execute the respective functions based on control instructions given from the CA  2 . 
   Voice Message Service 
   The VOIP call system has a function of sending the guidance message of the talkie etc. to each telephone as the subscriber terminal. The following is an explanation of an outline of an operation of each device within the VIOP call system on the occasion of providing the voice message service. The following discussion will exemplify a case that the VOIP call system sends the voice message to the telephone  15  in the network architecture shown in  FIG. 1 . 
   In the VOIP call system, when providing the voice message service, at first, the voice source data retained on the FTP server  1  are transferred to the VOIPGW  11  in response to an instruction signal from the CA  2 . Namely, the CA  2  instructs the FTP server  1  to transfer the voice source data to the VOIPGW  11 , and the FTP server  1  transfers the voice source data to the VOIPGW  11  in response to this instruction. The VOIPGW  11  receiving the voice source data retains the transferred voice source data. 
   When actually providing the voice message service, the VOIP call system controls each device to send a target voice message among pieces of voice source data retained on the VOIPGW  11  in response to a call from the subscriber terminal. To be specific, the CA  2  performs call control in response to the call of the telephone  15  and, as a result of this, notifies the VOIPGW  11  of call control information such as call setting, a voice source data add instruction, etc. The VOIPGW  11  notified of the call control information adds the voice source data into a designated timeslot (a channel corresponding to the call) in the STM from the PSTN  10 , thus sending the voice source data to the target telephone  15 . 
   Configuration of Device 
   Next, a functional configuration of the VOIP gateway device according to the first embodiment will be explained with reference to  FIGS. 2 through 4 . The following discussion will exemplify the VOIPGW  11  shown in  FIG. 1 .  FIG. 2  is a block diagram showing the functional configuration of the VOIPGW  11 .  FIG. 3  is a block diagram showing a detailed functional configuration of a voice source data transfer unit in the VOIPGW  11 .  FIG. 4  is a block diagram showing a detailed functional configuration of a packet processing unit in the VOIPGW  11 . Note that each of the VOIPGWs  11 ,  21 , and  31  in the first embodiment is the same device and has the same functional configuration. 
   The VOIPGW in the first embodiment is constructed of, as shown in  FIG. 2 , a control unit  110 , an STM switch control unit  111 , a voice source data transfer unit  112 , a CODEC unit  113  (corresponding to a packet processing unit, a transmitting unit and a packet transmitting unit according to the present invention), a packet processing unit  114 , a packet buffer  115 , and an IP switch unit  117  (corresponding to a receiving unit, a packet receiving unit and a data receiving unit according to the present invention). The packet buffer  115  further includes a voice source data storage buffer  116  (corresponding to a storage unit and a packet storage unit according to the present invention). The following are individual descriptions of the respective function units. 
   Control Unit  110   
   The control unit  110  receives a call control instruction, a voice source data transfer instruction, a voice source data add instruction, etc. from the CA  2 , and transmits instruction signals corresponding these instructions to other respective function units. The control unit  110  extracts a variety of instruction information of the CA  2  from a control packet transmitted from the CA  2 . The call control instruction represents a control instruction about the call given from the subscriber terminal within the PSTN  10 . This call control instruction contains instructions related to various types of control for establishing a call channel between a self-telephone and a partner telephone in response to the call from the telephone during a period till the call is disconnected since the call was connected. 
   In the case of receiving the voice source data transferred from the FTP server  1 , the control unit  110  transfers the voice source data to the voice source data transfer unit  112  (a dotted line with an arrowhead shown in  FIG. 2 ). Then, the control unit  110  transmits control signals to the voice source data transfer unit  112 , the CODEC unit  113  and the packet processing unit  114  so as to transfers the voice source data via the CODEC unit  113  to the packet processing unit  114  and further transfer the voice source data to the voice source data storage buffer  116  (one-dotted chain lines with arrowheads shown in  FIG. 2 ). 
   STM Switch Control Unit  111   
   The STM switch control unit  111  takes in an STM line from the PSTN  10  and serves as a PSTN interface. The STM switch control unit  111  executes, based on the call control instruction given from the CA  2 , control such as associating a call from the subscriber terminal with the STM channel, and so forth. The STM switch control unit  111  outputs a switch-controlled STM channel to the CODEC unit  113 . 
   Voice Source Data Transfer Unit  112   
   The voice source data transfer unit  112  temporarily stores the voice source data transferred from the FTP server  1 , and transfers the voice source data to the voice source data storage buffer  116  by the instruction of the control unit  110 . The voice source data transfer unit  112  transfers the voice source data toward the CODEC unit  113  by using a specified voice source data transfer channel designated by the CA  2  in the STM between the STM switch control unit  111  and the CODEC unit  113 . 
     FIG. 3  is a diagram showing a detailed functional configuration of the voice source data transfer unit  112  of the VOIPGW in the first embodiment. The detailed functional configuration of the voice source data transfer unit  112  will be explained with reference to  FIG. 3 . The voice source data transfer unit  112  is constructed of a local bus control unit  131 , a voice source data transfer control unit  132  and a voice source data temporary memory  133 . These function units will be described as below. 
   Voice Source Data Temporary Memory  133   
   The voice source data temporary memory  133  is a memory for temporarily storing the voice source data transferred from the FTP server  1 . 
   Local Bus Control Unit  131   
   The local bus control unit  131 , based on the instruction given from the control unit  110 , transmits and receives the control signals within the local bus, thereby controlling the voice source data add control unit  131  and the voice source data temporary memory  133 . For instance, the voice source data transferred from the FTP server  1  are stored on the voice source data temporary memory  133  in accordance with the control signal given from the local bus control unit  131 . 
   Voice Source Data Transfer Control Unit  132   
   The voice source data transfer control unit  132  adds the voice source data stored on the voice source data temporary memory  133  onto a specified voice source data transfer channel by the voice source data transfer instruction given from the CA  2  (the control unit  110 ). The voice source data transfer control unit  132  adds the voice source data in a way that reads the data on a byte-by-byte basis from the voice source data temporary memory  133  for every STM frame (125 μs) sent from the PSTN  10 . Further, the voice source data transfer control unit  132 , when finishing adding a last piece of voice source data, notifies the local bus control unit  131  of the end of the voice source data transfer. The notification showing the end of the voice source data transfer is delivered eventually to the CA  2 . 
   CODEC Unit  113   
   The CODEC unit  113  has the voice data subjected to data compression etc. by a predetermined method, thus voice-packetizing the voice data. As a CODEC method, there are standardized methods as defined by ITU-T G.711, ITU-T G.729, etc. The CODEC unit  113  performs encoding/compressing etc. of the voice data in accordance with the CODEC method (CODEC type) contained in the control information given from the CA  2 . The CODEC unit  113 , when the voice source data are transferred to the voice source data storage buffer  116  from the voice source data transfer unit  112 , encodes the voice source data corresponding to the CODEC type, and packetizes the encoded voice source data with a predetermined packet translation period (e.g., 20 ms). The thus-packetized voice source data are transmitted as IP packets to the packet processing unit  114 . The IP packet transmitted at that time involves using, e.g., an RTP (Real-time Transport Protocol)/RTCP (RTP Control Protocol) packet, wherein a predetermined port number for transferring the voice source data may be set in a UDP (User Datagram Protocol) header field. The packet processing unit  114  can know that the received packet is the voice source data transfer packet, by referring to this port number. 
   Conversely, with respect to the voice packet transferred from the packet processing unit  114 , the CODEC unit  113  decodes the voice data contained in the voice packet in accordance with the CODEC type which the packet processing unit  114  notifies of, thus effecting conversion into STM digital signals. The CODEC unit  113  adds the voice source data thus-converted to STM digital signals onto a target STM channel. 
   Packet Processing Unit  114   
   The packet processing unit  114  receives the voice source data packet packetized by the CODEC unit  113 , or the VOIP packet transmitted from the IP switch unit  117 , and executes a variety of processes corresponding to the received packets. When receiving the voice source data packet from the CODEC unit  113 , the packet processing unit  114  stores the received voice source data packet on the voice source data storage buffer  116 . Further, the packet processing unit  114  controls the transmission of the voice source data packet stored on the voice source data storage buffer  116  by the instruction of the control unit  110 . 
     FIG. 4  is a diagram showing a detailed functional configuration of the packet processing unit of the VOIPGW in the first embodiment. The detailed functional configuration of the packet processing unit  114  will be explained with reference to  FIG. 4 . The packet processing unit  114  actualizes the packet processing by use of, in addition, a receiving unit  141 , a transmitting unit  142 , a local bus control unit  143 , a voice source data transmission control unit  144 , a voice source data storage control unit  145 , a voice source data storage information table  147  and a voice source data transmission management table  146 . These function units will be described as follows. 
   Receiving Unit  141 , Transmitting Unit  142   
   The receiving unit  141  receives the IP packet, and the transmitting unit  142  transmits the IP packet. The receiving unit  141  receives the IP packet transmitted from the IP switch unit  117  and the IP packet transmitted from the CODEC unit  113 . The received IP packets are transferred to the packet processing unit  114 . The transmitting unit  142  transmits the predetermined IP packet to the IP switch unit  117  or the CODEC unit  113 . 
   Local Bus Control Unit  143   
   The local bus control unit  143  performs the control for notifying the respective function units in order to execute the packet processing based on the instruction given from the control unit  110 . 
   Voice Source Data Transmission Control Unit  144   
   The voice source data transmission control unit  144 , based on a voice source data add instruction given from the CA  2  (the local bus control unit  143 ), refers to the voice source data storage information table  147 , and notifies the packet processing unit  114  of various items of information (a storage address etc.) about the designated voice source data. Upon the notification from the voice source data transmission control unit  144 , the packet processing unit  114  reads the designated voice source data packet from the voice source data storage buffer  116 . The packet processing unit  114 , which has read the voice source data packet, updates a destination of this voice source data packet into an address indicating the CODEC unit  113 , further updates the UDP port number into a port number indicating a transmission destination subscriber, and transmits the packet toward the CODEC unit  113 . The control unit  110  previously notifies of the port number indicating the transmission destination subscriber terminal. 
   Voice Source Data Storage Control Unit  145   
   The packet processing unit  114 , based on a voice source data storage starting instruction given from the CA  2  (the local bus control unit), when judging from the port number of the received IP packet that this IP packet is a packet for transferring the voice source data, stores the voice source data packet on the voice source data storage buffer  116 . The voice source data storage control unit  145  receives information about storing the voice source data from the packet processing unit  114 , and stores the information in the voice source data storage information table  147 . Note that the packet processing unit  114  may continue to store the received packet on the voice source data storage buffer  116  till a voice source data storage finishing instruction comes from the local bus control unit  143 . 
   Voice Source Data Transmission Management Table  146   
   The voice source data transmission management table  146  manages management information of the voice source data packet stored on the voice source data storage buffer  116  on a destination-by-destination basis of the voice source data transmission. The voice source data transmission management table  146  has, on the destination-by-destination basis, has pieces of information about a port number, a message number, header information, directional information, a next transmission voice source data packet buffer address and a timer. The port number is an ID assigned to every transmission destination subscriber terminal and is set as a UDP port number. The message number is an ID determined for every voice message and is the same as the information stored in the voice source data storage information table  147 . The header information is information used for updating the header when transmitting the voice source data packet. The directional information is information indicating a transmitting direction (toward PSTN/IP network) of the voice source data packet. The next transmission voice source data packet buffer address is information representing a storage address of the data that should be transmitted next in the case of sequentially sending the voice source data packets. The timer has setting of a packet transmission interval period determined based on the voice-packetization, and is employed for taking a timing when sending the next packet. 
   Voice Source Data Storage Information Table  147   
   The voice source data storage information table  147  is a table stored with, on a voice-source-data-by-voice-source-data basis (e.g., a guidance message), pieces of information about the voice source data packet stored on the voice source data storage buffer  116 . The voice source data storage information table  147  is stored with, on the voice-source-data-by-voice-source-data basis, pieces of information such as a message number, a CODEC type, a start buffer address, a last buffer address, a chain count and an in-use count. The message number is an ID determined for every voice message, and the CA  2  gives an instruction to send the predetermined voice source data by use of this message number. The CODEC type is a type of the CODEC for the voice source data. The start buffer address, the last buffer address and the chain count are information representing addresses where the voice source data are stored on the voice source data storage buffer  116 , and, if stored in division, the number of divisions (the chain count). The in-use count is information showing whether or not the voice source data are being transmitted at the present, wherein the in-use count may be, for example, counted up each time the voice source data are transmitted and may also be cleared (becomes “0”) if there is no partner destination to which the data are being transmitted. 
   Packet Buffer  115   
   The packet buffer  115  is a memory area used when transmitting and receiving the VOIP packets. 
   Voice Source Data Storage Buffer  116   
   The voice source data storage buffer  116  is a memory area provided in the packet buffer  115  and serving to store the voice source data packets. The voice source data are stored in a state of being packetized by the CODEC unit  113 . Note that a storage mode may be direct storage of the voice source data packet given the header etc. or may also be storage of only the voice source data in the packet. 
   IP Switch Unit  117   
   The IP switch unit  117  becomes an interface with the IP network  3 . 
   Example of Operation 
   Next, an example of the operation of the VOIP gateway device in the first embodiment will be described with reference to  FIGS. 5 through 7 . Herein, the operation of the VOIP gateway device is explained in separation into a case of storing the voice source data transferred from the FTP server  1  on the voice source data storage buffer  116 , a case of deleting the stored voice source data, and a case of sending the stored voice source data to the PSTN. The following discussion will exemplify the VOIPGW  11  shown in  FIG. 1 , wherein an assumption is a case that the VOIPGW  11  sends the voice message to the telephone  15 . 
   Voice Source Data Storage Process 
   To begin with, an operation of the VOIPGW  11  in the case of storing the voice source data storage buffer  116  with the voice source data transferred from the FTP server  1 , will be described with reference to  FIG. 5 .  FIG. 5  is a flowchart showing voice source data storing procedure of the VOIPGW in the first embodiment. 
   The VOIPGW  11 , when storing the voice source data, receives the voice source data and the voice source data information from the FTP server  1  connected to the IP network  3  (S 501 ). These pieces of information are transmitted as, e.g., FTP packets from the IP network  3  and are therefore received by the control unit  110  via the IP switch unit  117  and the packet processing unit  114 . The voice source data information in the packet contains a message number showing a voice message serving as a source of the voice source data, a data size of the voice source data, a CODEC type, an STM channel information for transferring the voice source data, a UDP port number, etc. 
   The control unit  110 , which has received the voice source data and the voice source data information, transmits the voice source data and the data size to the voice source data transfer unit  112  (S 502 ). With this operation, the voice source data transfer unit  112  stores the voice source data temporary memory  133  with the received voice source data by the notified data size. Then, the control unit  110  notifies the packet processing unit  114  of the message number of the should-be-transferred voice source data, the CODEC type and the port number, and also notifies the packet processing unit  114  of a voice source data storage starting instruction (S 503 ). 
   Subsequently, the control unit  110  notifies the CODEC unit  113  of the CODEC type and the port number, and instructs the CODEC unit  113  to open the voice source data transfer channel (S 504 ). Following this instruction, the CODEC unit  113  opens the voice source data transfer channel, and prepares for acquiring the voice source data that will be transferred. Then, when acquiring the voice source data, the CODEC unit  113  encodes the voice source data, corresponding to the notified CODEC type, thus voice-packetizing the voice source data. The CODEC unit  113 , when effecting this voice-packetization, translates the notified port number into a UDP port number and transmits the UDP port number to the packet processing unit  114 . 
   The control unit  110  instructs the voice source data transfer unit  112  to start transferring the voice source data (S 505 ). Based on this voice source data transfer starting instruction, the voice source data are relayed sequentially to the STM switch control unit  111  and the CODEC unit  113  and thus transferred to the packet processing unit  114  (S 506 ). Thereafter, the control unit  110  waits till receipt of a transfer end notification informing of an end of transferring the voice source data from the voice source data transfer unit  112  (S 507 , S 507 ; No). Upon receiving the transfer end notification (S 507 ; YES), the control unit  110  instructs the packet processing unit  114  to finish storing the voice source data (S 508 ). The control unit  110  further instructs the CODEC unit  113  to close the voice source data transfer channel (S 509 ). With this operation, the VOIPGW  11  terminates the voice source data transfer process (S 510 ; NO). Note that if there exist other voice source data (S 510 ; YES), the transfer process is continuously executed (S 502 ). 
   An operation of the packet processing unit  114  with respect to such a voice source data storage process will hereinafter be explained with reference to  FIG. 6 .  FIG. 6  is a flowchart showing the voice source data storage process in the packet processing unit  114 . 
   The packet processing unit  114 , when receiving a voice source data storage starting instruction from the control unit  110  (S 601 ), executes the following voice source data storage process. The packet processing unit  114 , following the instruction, receives a message number of the should-be-transferred voice source data, a CODEC type and a port number from the control unit  110  (S 602 ). With this receipt, the packet processing unit  114 , for a start of storing the voice source data, waits for an IP packet in which the notified port number is set, i.e., the voice source data packet (S 603 , S 603 ; NO). 
   Upon receiving the IP packet containing the port number set therein, the packet processing unit  114  judges that this packet is the voice source data packet, and stores this packet on the voice source data storage buffer  116  (S 604 ). At this time, a single voice message is received in a segmented state into a plurality of voice source data packets, and hence the packet processing unit  114  stores the voice source data packets in a way that generates a chain for every packet. 
   As the storage of the voice source data is ended, the packet processing unit  114  judges whether or not the voice source data storage finishing instruction is received from the control unit  110  (S 605 ), and, if not received (S 605 ; NO), the packet processing unit  114  comes again to the waiting state for receiving the voice source data packet (S 603 ). Namely, the packet processing unit  114  continues the voice source data storage process till the receipt of the voice source data storage finishing instruction from the control unit  110 . When receiving the storage finishing instruction from the control unit  110 , the packet processing unit  114  recognizes an end of the should-be-stored voice source data, and stores various items of information about the stored voice source data in the voice source data storage information table  147  (S 606 ). 
   Voice Source Data Delete Process 
   Next, an operation of the VOIPGW  11  in the case of deleting the stored voice source data will be described with reference to  FIG. 7 .  FIG. 7  is a flowchart showing a voice source data delete process of the VOIPGW  11  in the first embodiment. 
   The control unit  110  of the VOIPGW  11 , when deleting the voice source data, receives a voice source data delete instruction from the CA  2  (S 701 ). Notification of this instruction is given through a control packet, wherein a should-be-deleted message number is contained in this control packet. The control unit  110  receiving this instruction designates the should-be-deleted message number and instructs the packet processing unit  114  to delete the voice source data (S 702 ). The packet processing unit  114 , upon receiving this instruction, confirms that the voice source data corresponding to the designated message number is not in the process of its transmission, and, if not so, deletes the same voice source data (S 703 ). 
   An operation of the packet processing unit  114  with respect to such a voice source data delete process will be described as below with reference to  FIG. 8 .  FIG. 8  is a flowchart showing the voice source data delete process in the packet processing unit  114 . 
   The packet processing unit  114 , when receiving the should-be-deleted message number and the voice source data delete instruction from the control unit  110  (S 801 ), executes the following voice source data delete process. The packet processing unit  114 , based on the instruction, judges whether or not the voice source data corresponding to the designated message number are under the transmission, by referring to an in-use count in the voice source data storage information table  147  (S 802 ). When judging from the in-use count that the target voice source data are not in use (S 802 ), the packet processing unit  114  refers to the start buffer address, the last buffer address, the chain count, etc., of the voice source data storage information table  147  and thus deletes the chain of the voice source data stored in the voice source data storage buffer  116  (S 803 ). With this operation, it follows that the designated voice source data are deleted from the voice source data storage buffer  116 . Finally, the packet processing unit  114  deletes all the information about the deleted voice source data from the voice source data storage information table (S 804 ). 
   Voice Source Data Transmission Process 
   Next, an operation of the VOIPGW  11  in the case of sending the stored voice source data to the PSTN will be explained with reference to  FIG. 9 .  FIG. 9  is a flowchart showing the voice source data transmission process of the VOIPGW in the first embodiment. 
   The control unit  110  of the VOIPGW  11 , when sending the voice source data, receives a voice source data transmission instruction from the CA  2  (S 901 ). This voice source data transmission instruction contains pieces of information such as a destination IP address, a channel number/UDP port number, a message number, a CODEC number, etc. An address of the CODEC unit  113  that should transmit the voice source data packet in the case of transmitting the voice source data to the PSTN is set to the destination IP address field. Then an IP address of the partner destination user terminal to which the voice source data should be transmitted in the case of transmitting the voice source data toward the IP network is designated to the destination IP address field. If the partner destination user terminal is the H323 terminal  6  shown in  FIG. 1 , an IP address of the H323 terminal  6  is designated. A channel number/UDP port number associated with the call of the transmission destination user terminal is designated in the channel number/UDP port number field. 
   The control unit  110 , upon receiving the instruction, notifies the CODEC unit  113  of a channel number associated with the transmission destination user terminal, a CODEC type for decoding the voice source data packet transmitted from the packet processing unit  114  and a UDP port number associated with the call of the transmission destination user terminal, and instructs the CODEC unit  113  to open the designated channel (S 902 ). 
   Subsequently, the control unit  110  notifies the packet processing unit  114  of the designated message number, the destination IP address and the UDP port number, and instructs the packet processing unit  114  to start transmitting the voice source data (S 903 ). The packet processing unit  114  receiving the transmission starting instruction reads the voice source data packet associated with the designated message number from the voice source data storage buffer  116 , updates the UDP port number in this voice source data packet into the designated UDP port number, and transmits the voice source data packet to the CODEC unit  113  specified by the destination IP address (S 904 ). 
   The control unit  110  waits till receipt of transmission end notification informing of an end of the voice source data transmission from the CA  2  (S 905 , S 905 ; NO). When receiving the transmission end notification (S 905 ; YES), the control unit  110  instructs the CODEC unit  113  to close the channel corresponding to the transmission destination user terminal (S 906 ). 
   An operation of the packet processing unit  114  with respect to such a voice source data transmission process will be described as below with reference to  FIG. 10 .  FIG. 10  is a flowchart showing the voice source data transmission process in the packet processing unit  114 . 
   The packet processing unit  114 , when receiving a voice source data transmission starting instruction from the control unit  110  (S 1001 ), executes the following voice source data transmission process. The packet processing unit  114 , based on the instruction, registers information about the target voice source data in the voice source data transmission management table  146  (S 1002 , S 1003 , S 1004 ). The port number specifying the transmission destination user terminal that is set in the UDP header of the to-be-transmitted packet (S 1002 ) and the address in the voice source data storage buffer  116  stored with the voice source data that should be transmitted next time (S 1003 ), are registered as the information about the voice source data, and the timer value is cleared (S 1004 ). 
   The packet processing unit  114  judges whether the timer in the voice source data transmission management table is cleared or not (S 1005 ). If the timer is judged to be cleared (S 1005 ; YES), the voice source data packet is read from the voice source data storage buffer  116  on the basis of the address set in the next transmission voice source data buffer address field in the voice source data transmission management table (S 1006 ). The packet processing unit  114  updates the destination IP address, the UDP port number, etc. of the readout voice source data packet, and sends the updated packet to the CODEC unit  113  (S 1006 ). The packet processing unit  114 , for registering the address where the voice source data packet, which should be transmitted when reading next, are stored, updates the next transmission voice source data buffer address in the voice source data transmission management table (S 1007 ). Further, the packet processing unit  114  resets the timer in the voice source data transmission management table  146  to a packet transmission period as the initial value (S 1008 ). 
   The packet processing unit  114 , upon finishing the process, judges whether or not the voice source data transmission finishing instruction comes from the control unit  110  (S 1009 ). If the voice source data transmission finishing instruction comes in (S 1009 ; YES), the packet processing unit  114  deletes a record containing the port number specifying the user terminal becoming the transmission destination this time is entered in the port number field in the voice source data transmission management table (S 1010 ). Whereas if the voice source data transmission finishing instruction does not come from the control unit  114  (S 1009 ; NO), the packet processing unit  114  continues the voice source data transmission process (S 1005 ). The packet processing unit  114  continues the voice source data transmission process till the voice source data transmission finishing instruction comes from the control unit  110 , in other words, till the call of the transmission destination user terminal is disconnected or otherwise and till the CA  2  judges that the transmission of the voice message to its transmission destination is ended. 
   Operations/Effects in First Embodiment 
   Herein, operations and effects of the VOIP gateway device in the first embodiment discussed above, will be described.  FIG. 11  is a diagram showing an outline of control flows of the respective function units with respect to the voice source data storage process, the voice source data delete process and the voice source data transmission process of the VOIPGW  11 . The description of the operation might involve referring to  FIG. 11  as the necessity arises. 
   In the VOIPGW  11  in the first embodiment, on the occasion of providing the guidance message of a talkie etc. to each telephone as the subscriber terminal, at first, the guidance message is digitized, and the packetized voice source data packets are stored on the voice source data storage buffer  116 . 
   In this voice source data storage process, the VOIPGW  11  receives the voice source data retained on the FTP server  1 , and temporarily stores the received voice source data on the voice source data transfer unit  112 . Thereafter, in the VOIPGW  11 , the CODEC unit  113  packetizes the voice source data during a period till a voice source data storage finishing instruction is received since a voice source data storage starting instruction was received from the CA  2 , and the voice source data storage buffer  116  is sequentially stored with the packetized voice source data packets (a process ( 1 ) shown in  FIG. 11 ). 
   The VOIPGW  11 , upon receiving the voice source data storage finishing instruction given from the CA  2 , registers the stored information about the voice source data packet stored this time in the voice source data storage information table  147 , wherein the message number as the voice message ID is used as a key (a process ( 2 ) shown in  FIG. 11 ). 
   Thus, the VOIP gateway device in the first embodiment digitizes and packetizes the voice message provided to the subscriber terminal such as the telephone etc., and stores the voice source data packet on the voice source data storage buffer in the packet buffer. With this operation, the VOIP gateway device in the first embodiment has no necessity of adding any memory dedicated to the voice source data to within the device, and part of the buffer within the memory that is normally employed for the packet processing is efficiently used, thus enabling the voice source data to be retained. Further, the CODEC unit  113 , when assembling the voice source data packet, effects the voice-compression, whereby a data capacity itself of the should-be-stored voice source data can be reduced, and, by the same token, the memory capacity for storing the voice source data can be saved. 
   The VOIPGW  11 , when actually providing the voice message service, sequentially reads the voice source data packets stored beforehand on the voice source data storage buffer  116  and transmits the voice source data packets in accordance with the voice source data transmission starting instruction given from the CA  2 . When reading the voice source data, the VOIPGW  11  assigns the predetermined port number to every transmission destination and registers the voice source data transmission information in the voice source data transmission management table  146  (a process ( 3 ) shown in  FIG. 11 ). Further, as the voice source data are packetized and thus stored, the packet transmission timing is managed by the timer in the voice source data transmission management table  146 , and the stored voice source data packet is transmitted with the period set in the timer (processes ( 4 ) and ( 5 ) shown in  FIG. 11 ). The VOIPGW  11  deletes the record concerning the target partner destination terminal registered previously in the voice source data transmission management table  146  in accordance with the voice source data transmission finishing instruction (a process ( 6 ) shown in  FIG. 11 ). 
   Thus, the VOIP gateway device in the first embodiment handles, on the packet basis, the voice source data that are packetized and thus retained on the occasion of transmitting the voice source data. With this scheme, in a case such as building up the voice source data storage buffer by use of DRAM (Dynamic Random Access Memory), the high-speed memory accessing can be attained by employing a burst access function of the DRAM, and, by the same token, it is possible to increase the number of voice source data simultaneous transmission channels. 
   In this respect,  FIG. 12  shows a result of examination made by exemplifying such a case that the voice source data storage buffer involves adopting SDRAM (Synchronous DRAM) (32-bit width, 100 mega-Hertz (MHz) memory clock) of DDR (Double Data Rate)  200 . The reason why the effect with respect to the simultaneous transmission channel count in the first embodiment is obtained, will be elucidated with reference to  FIG. 12 . In the example shown in  FIG. 12 , there is given a case in which a voice source data packet transmission interval is set to 10 millisecond (ms), and a voice source data packet length is set to 128 bytes. In the SDRAM in this example, the memory access time is approximately 0.2 microsecond (μs) in terms of specifications thereof, and hence a memory accessible count with one-packet period (10 ms) is considered to be approximately 50,000. Accordingly, simply the simultaneous accessing can be done for 50,000 channels at the maximum. Hence, as compared with the voice source data adding in the general type of STM switch control unit, the simultaneous accessible channel count can be remarkably increased. 
   The VOIPGW  11 , when receiving the voice source data delete instruction from the CA  2 , extracts the storage information about the delete target voice source data from the voice source data storage information table  147 , and deletes the target voice source data from the voice source data storage buffer  116  on the basis of the extracted storage information (processes ( 7 ) and ( 8 ) shown in  FIG. 11 ). 
   Thus, in the VOIP gateway device in the first embodiment, the voice source data management can be conducted as the buffer management. This management mode facilitates, on such an occasion as to add, delete and change the voice message, managing the target voice source data, and enables obviation of troublesomeness of the memory management in the conventional system. 
   Second Embodiment 
   A VOIP gateway device in a second embodiment of the present invention will hereinafter be described. The VOIP gateway device according to the first embodiment discussed earlier receives the should-be-transmitted voice source data from the FTP server and stores the voice source data. The VOIP gateway device in the second embodiment has a function of transferring the voice source data to other VOIP gateway device (which will hereinafter be referred to as an other-device transfer function). The network architecture shall be the same as that in the first embodiment shown in  FIG. 1 . A configuration of the second embodiment that will hereinafter be described is an exemplification, and the present invention is not limited to the following configuration. 
   Configuration of Device 
   The VOIP gateway device according to the second embodiment is constructed of the same function units as those in the first embodiment, however, the operations of the respective function units are somewhat different. The function units operating differently from the first embodiment will be explained with reference to  FIG. 13 .  FIG. 13  is a diagram showing functional configurations of VOIPGWs  11 ,  21  and a control flow in the second embodiment, and showing the control flow in the case of transferring the voice source data packet from the VOIPGW  11  to the VOIPGW  21 . Further, in the following discussion, the explanations of the same function units as those in the first embodiment are omitted. 
   Control Unit  110   
   The control unit  110  of the transmission-side VOIPGW  11 , when executing the other-device transfer, as in the first embodiment, receives the call setting for transferring the voice source data and the voice source data transfer instruction from the CA  2 . This voice source data transfer instruction contains pieces of information such as the message number, the CODEC type, the channel number, the port number and a destination IP address, wherein a different point from the first embodiment is to contain the destination IP address. This destination IP address is employed by the packet processing unit  114 , and an address of the other transfer destination VOIP gateway device (VOIPGW  21 ) is designated as the destination IP address. Other pieces of information are the same as those in the first embodiment. 
   The control unit  110 , upon receiving the voice source data transfer instruction, notifies the packet processing unit  114  of the port number specifying the voice source data packet and the destination IP address associated with this port number (2-dotted chain lines shown in  FIG. 13 ). The notification given to other function units is the same as in the case of transferring voice source data in the first embodiment. 
   Moreover, a control unit  110 - 2  of the receiving-side VOIPGW  21  receives the call setting for transferring the voice source data and the voice source data storage instruction from the CA  2 . This voice source data storage instruction contains pieces of information such as the message number of the voice source data packet that is transferred to this side, the CODEC type and the port number. Based on this voice source data storage instruction, the control unit  110 - 2  controls the respective function units. 
   Packet Processing Unit  114   
   The packet processing unit  114  (corresponding to a transfer unit according to the present invention) has the following function in addition to that in the first embodiment. The transmitting-side packet processing unit  114 , when receiving the voice source data packet transmitted from the CODEC unit  113 , updates the destination IP address of voice source data packet with the destination IP address which the control unit  110  has notified of, and sends the address-updated packet to the IP switch unit  117 . With this operation, it follows that the IP switch unit  117  transmits the packet toward the IP network  3 , and the packet is forwarded to the VOIPGW  21 . Note that the packet processing unit  114 , other than sending the voice source data packet toward the IP network  3 , may store the voice source data storage buffer  116  with the voice source data packet as in the first embodiment. 
   A receiving-side packet processing unit  114 - 2 , upon receiving the packet having setting of the port number which the control unit  110 - 2  has notified of, judges this packet as the voice source data packet, and stores the packet on a self voice source data storage buffer  116 - 2 . 
   Example of Operation 
   Next, an example of the operation of the VOIP gateway device in the second embodiment will be described with reference to  FIGS. 14A and 14B .  FIGS. 14A and 14B  are diagrams showing a sequence of the other-device transfer process in the second embodiment. 
   Other-Device Transfer of Voice Source Data 
   For actualizing the other-device transfer of the voice source data, the VOIPGW needs being stored with the voice source data. Such being the case, to begin with, the FTP server  1  transmits the voice source data to the transmitting-side VOIPGW  11  (S 1401 ). When finishing the transfer, the CA  2  sends a control packet containing the voice source data storage instruction to the receiving-side VOIPGW  21  (S 1402 ), and subsequently transmits the voice source data transfer instruction to the transmitting-side VOIPGW  11  (S 1403 ). The voice source data storage instruction and the voice source data transfer instruction contain a call setting instruction for transferring the voice source data and voice source data transfer information. The voice source data transfer information to the VOIPGW  21  contains a message number about the voice source data that are transferred to this side, a CODEC type and a port number. The voice source data transfer information to the VOIPGW  11  contains a message number about the voice source data that should be transferred, a CODEC type, a channel number, a port number and a destination IP address. 
   The VOIPGW  11  receiving the voice source data and the voice source data transfer instruction executes the same process as the voice source data storage process in the first embodiment. The VOIPGW  11  temporarily stores the voice source data on the voice source data transfer unit  112  (S 1404 ). Thereafter, the VOIPGW  11  instructs the CODEC unit  113  and the packet processing unit  114  to transfer the voice source data as preparation for transferring the voice source data (S 1405 , S 1406 ). The packet processing unit  114  is notified of the port number specifying the voice source data packet, the destination IP address of the transfer destination, etc. (S 1405 ). The CODEC unit  113  is notified of the CODEC type and the port number together with an instruction to open the voice source data transfer channel (S 1406 ). 
   When sending the voice source data transfer starting instruction to the voice source data transfer unit  112  from the VOIPGW  11  (S 1407 ), the transfer of the voice source data is started (S 1409 ). The voice source data are transferred to the CODEC unit  113  via the designated channel of the STM, and, after being voice-packetized (into the voice source data packet) by the CODEC unit  113 , transferred to the packet processing unit  114 . The packet processing unit  114  changes the transmission destination IP address of the voice source data packet that has been transferred to this side into the pre-notified destination IP address, whereby the packet is forwarded toward the IP network  3 . 
   On the other hand, the receiving-side VOIPGW  21 , based on the voice source data storage instruction (S 1402 ), gives the voice source data storage instruction to a packet processing unit  114 - 2  (S 1408 ). This instruction contains pieces of information (the message number, the CODEC type and the port number) about the voice source data that are transferred to this side. With this operation, the VOIPGW  21  comes to a voice source data packet transfer waiting state, wherein the VOIPGW  21  stores a self voice source data storage buffer  116 - 2  with the voice source data packet each time the packet having setting of the notified port number reaches. 
   The transmitting-side VOIPGW  11  continues to transfer the voice source data till the transfer of the voice source data is completed (S 1410 , S 1410 ; NO), and, when finishing transferring all the voice source data (S 1410 ; YES), notifies of an end of the voice source data transfer (S 1411 ). Upon notifying of the end of the voice source data transfer, the VOIPGW  11 , as a voice source data transfer finishing process, instructs the CODEC unit  113  to close the voice source data transfer channel (S 1413 ) and instructs the packet processing unit  114  to cancel a voice source data transfer route (S 1414 ). 
   The receiving-side VOIPGW  21  waits for the voice source data storage finishing instruction from CA  2  (S 1412 ), then instructs the packet processing unit  114 - 2  to finish storing the voice source data at a point of time when receiving this instruction (S 1415 ), and finishes the voice source data storage process. 
   Operation/Effects in Second Embodiment 
   In the second embodiment, the VOIPGW  11  packetizes the voice source data temporarily stored on the voice source data transfer unit  112  provided in the self STM switch control unit, and transfers the packet to the other VOIPGW  21 . On the other hand, the VOIPGW  21  receiving the transferred voice source data packet stores this voice source data packet on the self voice source data storage buffer  116 - 2 . 
   With this operation, in the second embodiment, the number of the gateway devices provided in the STM switch control unit and each having the voice source data transfer function can be limited. As a matter of course, the multi-function device becomes expensive, and therefore, in the system employing the VOIP gateway device, the system can be configured in a way that restrains the costs. 
   Third Embodiment 
   The following is a description of a VOIP gateway device according to a third embodiment of the present invention. The VOIP gateway device according to the second embodiment discussed earlier has the function of transferring the voice source data transferred from the FTP server to the other gateway device from the voice source data transfer unit provided in the STM switch control unit. The VOIP gateway device according to the third embodiment has a function of transferring the voice source data packet stored on the self voice source data storage buffer to the other gateway device (which will hereinafter referred to as an inter-buffer transfer function). The network architecture shall be the same as that in the first embodiment shown in  FIG. 1 . A configuration of the third embodiment that will hereinafter be described is an exemplification, and the invention is not limited to the following configuration. 
   Configuration of Device 
   The VOIP gateway device according to the third embodiment is constructed of the same function units as those in the first embodiment, however, the operations of the respective function units are somewhat different. The function units operating differently from the first embodiment will be explained with reference to  FIG. 15 .  FIG. 15  is a diagram showing functional configurations of the VOIPGWs  11 ,  21  and a control flow in the third embodiment, and showing the control flow when the voice source data packet stored on the VOIPGW  11  is forwarded to the VOIPGW  21 . Further, in the following discussion, the explanations of the same function units as those in the first embodiment are omitted. 
   Control Units  110 / 110 - 2   
   The control unit  110  on the transmitting-side VOIPGW  11 , for actualizing the inter-buffer transfer function, receives an already-stored voice source data transfer instruction from CA  2 . This transfer instruction contains pieces of information such as a message number, a port number, a destination IP address, etc. The destination IP address is employed by the packet processing unit  114 , and an address of the other VOIP gateway device (VOIPGW  21 ) as the transfer destination is entered in this destination IP address field. The control unit  110 , based on this transfer instruction, controls the packet processing unit  114 . 
   On the other hand, the control unit  110 - 2  on the receiving-side VOIPGW  21  receives the voice source data storage instruction from the CA  2 . This storage instruction contains pieces of information such as a message number, a port number, etc. The control unit  110 - 2  controls the packet processing unit  114 - 2  on the basis of this storage instruction. 
   Packet Processing Unit  114   
   The packet processing unit (corresponding to a data transmitting unit according to the present invention) has the following function in addition to the function in the first embodiment. The transmitting-side packet processing unit  114 , when receiving the already-stored voice source data transfer instruction from the control unit  110 , reads the voice source data packet associated with the designated message number from the voice source data storage buffer  116 . The packet processing unit  114  updates the port number and the transmission destination IP address in the readout voice source data packet into the designated port number and designated destination IP address. Note that on the occasion of transferring this voice source data packet, the packet processing unit  114  may add a predetermined packet header for the inter-buffer transfer function to the readout voice source data packet. The packet processing unit  114  forwards the updated voice source data packet not as the voice packet (taking account of the packet transmission period etc.) but as a normal data packet to the IP switch unit  117  (toward the IP network  3 ). Note that the packet processing unit  114  may also forward the packet by the FTP transfer in which this voice source data packet is handled as the data (datagram). 
   Example of Operation 
   Next, an example of the operation of the VOIP gateway device in the third embodiment will be explained with reference to  FIG. 16 .  FIG. 16  is a diagram showing a sequence of the inter-buffer transfer process of the VOIPGW in the third embodiment. 
   Inter-Buffer Transfer 
   Actualization of the inter-buffer transfer of the voice source data requires storing the voice source data on the VOIPGW, however, the voice source data storage process thereof is the same as that in the first embodiment. The following operation is an operation conducted in such a state that the voice source data packet is already stored on the voice source data storage buffer  116  of the VOIPGW  11 . 
   The CA  2  sends the already-stored voice source data transfer instruction to the transmitting-side VOIPGW  11  (S 1601 ). This already-stored voice source data transfer instruction contains pieces of information such as a port number, a message number, a destination IP address, a call setting instruction for the transfer, etc., which are used for transferring the voice source data packet. Subsequently, the CA  2  sends the voice source data storage instruction to the receiving-side VOIPGW  21  (S 1602 ). This voice source data storage instruction contains a message number, a port number, a CODEC type and a packet count. Herein, the packet count represents the number of the voice source data packets that are transferred to this side. 
   The VOIPGW  11  receiving the already-stored voice source data transfer instruction notifies the packet processing unit  114  of the already-stored voice source data transfer instruction (S 1603 ). The packet processing unit  114 , when receiving this instruction, transfers all the voice source data packets corresponding to the designated voice message as the data packets to the designated destination (S 1604 ). 
   On the other hand, the VOIPGW  21  receiving the storage instruction instructs the packet processing unit  114 - 2  to start storing the voice source data (S 1605 ), and receives the transferred data packets by the notified packet count. The transferred data packet is identified with the packet associated with the voice source data packet from the port number and is stored on the message storage buffer  116 - 2  (S 1606 ). 
   Operation/Effects in Third Embodiment 
   In the third embodiment, the VOIPGW  11  transfers the voice source data packet stored on the self voice source data storage buffer  116  as the normal data packet to the other VOIPGW  21 . On the other hand, the VOIPGW  21  receiving the transferred data packet stores the voice source data packet contained in the data packet on the self voice source data storage buffer  116 - 2 . 
   With this operation, in the third embodiment, the voice source data can be handled and transferred in the same way as the normal data can be without transmitting the data with the predetermined period etc. as in the case of the voice packet. This makes it possible to copy the voice source data held by one device to the plurality of devices by the simple method. 
   Others 
   The disclosures of Japanese patent application No.JP2005-079630, filed on Mar. 18, 2005 including the specification, drawings and abstract are incorporated herein by reference.