JP2007049415A - Voice data conversion apparatus, network system, and control method and program - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To realize call connection among different voice codec terminals in a VoIP network. <P>SOLUTION: In a voice conversion apparatus 102 arranged on networks 10, 20, 100, 200, 201 for performing VoIP communication, among terminals 11, 12, 21, 22 constituting the networks 10, 20, 100, 200, 201, voice codec information to which respective terminals 11, 12, 21,22 correspond is stored in each call connection, and voice codec conversion is performed so that communication can be mutually performed among respective terminals 11, 12, 21, 22. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention relates to an audio data converter, a network system, a control method, and a control program, and more particularly to an audio data converter, a network system, and a control method for converting audio data flowing on an Internet Protocol (IP) network. And a control program.

  In a voice call using the Internet protocol (IP), one of protocols used for call connection is SIP (Session Initiation Protocol). In SIP, the audio codec type supported by the terminal is notified by storing it in the message body of the SIP message as SDP (Session Description Protocol). Each terminal determines the codec type of audio data to be transmitted / received by referring to this message. Therefore, in order to make a call connection, both terminals must support the same audio codec type.

  At present, terminals with different codec types cannot be connected to each other, so it is necessary to install terminals with matching corresponding codecs in advance. A technique described in Patent Document 1 is known as a technique for performing voice data conversion by arranging a gateway device at a network boundary. However, the gateway device is intended to realize call connection between terminals in different networks, and no mechanism for performing voice codec conversion between terminals on the IP network has been provided. Further, as a technique for performing audio codec conversion, the technique described in Patent Document 2 is known, but the technique described in Patent Document 2 is intended for communication in an ISDN network.

JP 2003-37683 A JP 2000-22756 A

  Accordingly, for communication connection between terminals having different voice codecs on the VoIP network, there are no known communication procedures including the SIP protocol, and there is no apparatus that can substitute for voice conversion. Therefore, on the VoIP network, since it is not possible to establish a call connection between IP telephone terminals having different voice codecs, the corresponding voice codec of the terminal is different from the terminal codec for each connection destination when the VoIP system is introduced. There was a problem of having to be conscious.

This problem is caused by the fact that the communication control protocol for making a voice call connection between terminals on an IP network called SIP does not support connection between terminals of different voice codec types. In addition, in order to establish a call connection, it is necessary to match the voice codec used between the connected terminals, but this causes a new problem that the flexible operation cannot be performed.

  Accordingly, an object of the present invention is to provide an audio data conversion device, a network system, a control method, and a control program that can realize a call connection between different audio codec terminals in a VoIP network.

  In order to solve the above-described problems, the present invention provides a voice conversion device arranged in a network that performs VoIP communication, and each voice corresponds to each terminal for each call connection between the terminals constituting the network. The audio codec conversion is performed so as to hold codec information and to enable communication between the terminals. According to this configuration, for each call connection between the terminals constituting the network, information on the voice codec supported by each terminal is held, and the voice codec conversion is performed so that the terminals can communicate with each other. Therefore, it is possible to realize call connection between different voice codec terminals in the VoIP network.

  In this case, according to a call connection procedure of a predetermined protocol, codec information identifying means for identifying the voice codec included in a connection request message and a connection response message, a data table for storing information on the identified voice codec, Information registration means for registering the voice codec information for each call connection to the data table and codec conversion means for referring to the data table and performing voice codec conversion may be provided.

  The voice conversion device is disposed between the SIP network and another SIP network, or between the SIP network and a network under management of other communication carriers including the PSTN network, the PHS network, and the mobile phone network. You may do it.

  Furthermore, the codec conversion means may perform the audio codec conversion by selecting an audio codec with a smaller amount of data when making a call connection addressed to a terminal under network management with a narrower band.

  Furthermore, when the codec conversion means handles a call connection request message by SIP and transfers it to the connection destination network, the codec conversion means notifies the plurality of voice codecs to be used, so that the connection destination network The audio codec conversion may be performed between the audio codec notified as the usable audio codec and the audio codec notified by the connection source network.

  The information registration means acquires voice codec information for each of the plurality of terminals that are broadcast destinations in a call connection for performing broadcast communication, registers the information in the data table, and the codec conversion means You may make it provide the transmission means which copies transmission data with respect to the said several terminal of a communication destination, and transmits.

  In addition, the present invention is a network system, and holds information on a plurality of terminals arranged in a network for performing VoIP communication and voice codec supported by each terminal for each call connection between the terminals. And an audio conversion device that performs the audio codec conversion so that the terminals can communicate with each other.

  In this case, the voice conversion device stores codec information identification means for identifying the voice codec included in the connection request message and the connection response message according to a call connection procedure of a predetermined protocol, and information on the identified voice codec. A data table, information registration means for registering the voice codec information for each call connection to the data table, and codec conversion means for referring to the data table and performing voice codec conversion. Also good.

  The present invention also provides a method for controlling a voice conversion device arranged in a network that performs VoIP communication, wherein a voice codec supported by each terminal is connected for each call connection between the terminals constituting the network. The method includes a step of holding information and a step of performing the voice codec conversion so that the terminals can communicate with each other.

  Further, in a control method of a network system comprising a network, a plurality of terminals arranged in the network and performing VoIP communication, and a voice conversion device that performs voice codec conversion so that the terminals can communicate with each other A process of holding information on a voice codec supported by each terminal for each call connection between the terminals, and a process of performing the voice codec conversion so that the terminals can communicate with each other. It is characterized by providing.

  The present invention is also a control program for controlling a voice conversion device arranged in a network for performing VoIP communication by a computer, wherein each terminal corresponds to each call connection between terminals constituting the network. The voice codec conversion is performed such that the information of the existing voice codec is held and the terminals can communicate with each other.

  A network system comprising a network, a plurality of terminals arranged in the network and performing VoIP communication, and a voice conversion device that performs the voice codec conversion so that the terminals can communicate with each other is provided by a computer. In a control program for controlling, information about the voice codec supported by each terminal is held for each call connection between the terminals constituting the network so that the terminals can communicate with each other. To perform the audio codec conversion.

  As described above, according to the audio data conversion apparatus and the like according to the present invention, it is possible to realize a voice call without being conscious of the fact that the other party's terminal and the audio codec are different.

  FIG. 1 is a schematic configuration block diagram of a network system including a voice conversion device according to the present invention. This network system includes a first VoIP network 10 and a second VoIP network 20 configured as a LAN. IP terminals 11 and 12 are connected to the first VoIP network 10, and IP terminals 21 and 22 are connected to the second VoIP network 20.

  Further, the first VoIP network 10 and the second VoIP network 20 are connected to a telecommunications carrier network 100 that provides VoIP voice call connection to these two networks. Connected to the telecommunications carrier network 100 are the Internet 200, another telecommunications carrier network 201, and another network 202 that operates with a protocol other than IP. Here, the network 202 other than IP corresponds to a network other than IP, such as a PSTN network and a mobile network such as PHS.

  The first VoIP network 10 and the second VoIP network 20 are networks used in different organizations, and the communication carrier network 100 provides a voice call service to these organizations. Each of the IP terminals 11, 12, 21, and 22 is configured as an IP telephone device or a PC (Personal Computer) capable of connecting and calling using SIP and RTP.

  The communication carrier network 100 functions as a management network that realizes a voice call connection with the first VoIP network 10, the second VoIP network 20, and a network 202 other than IP. For this reason, the communication carrier network 100 includes a SIP server 101 that performs call control for voice call connection by SIP, a voice conversion device 102 that performs voice data conversion (voice codec conversion) between networks, and the Internet 200. And a router 103 used for voice communication with a terminal managed by another communication carrier network 201, and a gateway device 104 for connecting to another network.

  The voice conversion device 102 functions as a relay point for SIP control messages and voice data packets (RTP packets) on a communication path in a call between terminals, and by editing media information of SIP messages, different terminals can communicate with each other. The voice call is realized by performing the call connection and the voice codec conversion of the RTP packet.

  As shown in FIG. 2, the voice conversion apparatus 102 includes a packet receiving unit 301 that receives an external packet input, a protocol control unit 302, a call control unit 303, a media information conversion unit 304, and an RTP packet termination unit. 305, an audio codec conversion unit 306 that converts the audio codec of the received RTP packet, and a packet transmission unit 307.

  In the above configuration, the packet receiving unit 301 receives an IP packet and performs a process of passing packet data to the RTP packet terminating unit 305 and the protocol control unit 302 according to the type of the packet. The packet receiving unit 301 inspects the protocol type of the IP header of the packet received by the network interface in the apparatus, and performs processing of passing packet data to the RTP packet terminating unit 305 if the protocol type is RTP. Similarly, when the IP header protocol type indicates UDP, and the UDP source port and destination port number = “5060” or “5062”, the packet receiver 301 receives packet data. Is transferred to the protocol control unit 302.

  The protocol control unit 302 has a function of handling a SIP message and analyzing and assembling a header part. The protocol control unit 302 receives SIP message data from the packet receiving unit, disassembles the SIP message, reads out the dialog information, and passes it to the call control unit 303. In addition, after editing the media information of the SIP message, the SIP message is assembled, and the SIP message data is passed to the packet transmission unit 307.

  The call control unit 303 has a function of performing call identification management from dialog information of the SIP message. The call control unit 303 acquires dialog information from the SIP header information received from the protocol control unit 302, holds the dialog information as internal data, and passes the SIP header information to the media information conversion unit 304. In addition, the call control unit 303 generates call information at the start of call connection and deletes call information at the time of call disconnection.

  The media information conversion unit 304 has a function of editing the media information of the SDP unit of the SIP message according to the codec type of the terminal to which the call is connected. The media information conversion unit 304 writes all compatible codec types in the INVITE message from the calling terminal, and passes the data to the packet transmission unit 307.

  The RTP packet termination unit 305 has functions of handling and terminating RTP packets in order to convert the codec of audio data included in the data portion of the RTP packet. The RTP packet termination unit 305 checks the protocol type of the IP header part of the received packet. If the protocol type of the received packet is RTP, the RTP packet termination unit 305 handles the RTP packet and loads the packet data into the queue. Next, the RTP packet termination unit 305 takes out the packet data loaded in the queue from the queue in the order of reception, that is, the order of loading in the queue, and checks the contents of the header part.

  Next, the RTP packet termination unit 305 acquires the transmission source IP address and the transmission source port from the IP and UDP header of the received packet, and passes them to the audio codec conversion unit 306 together with the packet data. The process proceeds to 306.

  The audio codec conversion unit 306 converts the audio codec of the RTP packet according to the codec of the connection destination terminal based on the codec information held in the apparatus. The codec conversion process affects the processing capability of the apparatus, and is preferably processed by dedicated hardware. Next, the voice codec conversion unit 306 passes the RTP packet data after the voice codec conversion processing to the packet transmission unit.

  The packet transmission unit 307 has a function of transmitting an SIP message and an RTP packet. The apparatus of the present invention transmits a packet after the media information conversion processing of the SIP message in the media information conversion unit 304 and a packet after the codec conversion processing of the voice data of the RTP packet of the voice codec conversion unit 306 from an appropriate network interface. It has a function to transmit.

  In order to execute conversion of the voice codec, conversion information is held in a memory in the apparatus for each call between terminals. The voice media information is stored in the device when a SIP message transmitted from the terminal is received by the device, or when the device receives an RTP packet transmitted between the terminals to which a call connection is made and performs conversion of the voice codec. Is stored as internal data, and the registered audio media information is read out, and the audio codec conversion of the RTP packet is executed according to the acquired data.

  As shown in FIG. 3, the voice conversion apparatus 102 holds two types of tables, a media information table 402 (originating side 403 and terminating side 404) associated with a call connected to the call information management table 401. Thus, the media information conversion processing of the signaling message is appropriately performed. The configuration of these tables is shown in FIGS. 4, 5, and 6. FIG.

  FIG. 4 shows a call information management table. This call information management table 500 includes call information identification number information 501 and call identification information (dialog) 502.

  FIG. 5 shows a media information table. The media information table 510 includes a codec list 512 including interface (I / F) identification number information 511 and a list of usable audio codecs.

  FIG. 6 shows a codec conversion table. The codec conversion table 520 includes transmission destination IP address information 521 for specifying a transmission destination terminal, corresponding port number information 522, and a conversion destination codec corresponding to the transmission destination terminal. Information 523 is provided.

  The voice conversion device 102 holds the codec conversion table 405 shown in FIG. 3 in order to perform voice codec conversion of RTP packets.

  When the SIP control message is received, if the received control message is an INVITE message, dialog information is acquired from the message. If the call is a newly connected call, a call information management table 500 is newly generated. Then, a unique number is registered in the call information identification number field 501, and the call identification information (dialog) field 502 of the call information management table 500 is registered with the Call-ID parameter, From-Tag parameter, Via-branch as SIP dialog information. Register parameters.

  When SDP media information is stored in the SIP message, each field of the media information table 510 includes an I / F number 511 corresponding to each interface (I / F) on the receiving side and transmitting side, and The codec list 512 corresponding to the terminal is registered.

  When “200 OK” is received as the status code, the transmission destination IP address 521, the port number 522, and the conversion destination codec 523 are registered in each field of the RTP codec conversion table 520.

  Based on the codec conversion table 520, the audio conversion device 102 converts the audio codec information of the received RTP packet into a registered codec type and transmits it to an appropriate port.

The following three types of operation patterns of the audio media information conversion process are conceivable.
(1) When two terminals that perform communication have the same corresponding voice codec (2) When two terminals that perform communication have a part of the corresponding voice codec (3) In two terminals that perform communication, When all supported audio codecs are different

  First, the case where the two corresponding audio codecs are the same in the two terminals that perform communication will be described with reference to FIG.

  When the voice conversion apparatus 102 receives an initial INVITE message including information of a voice codec (X, Y in FIG. 7) corresponding to the terminal A from one terminal A, the voice conversion apparatus 102 receives the voice. A process of writing an audio codec (corresponding to X, Y, and Z in FIG. 7) that can be processed by the conversion apparatus 102 into the SDP header of a message to be transmitted to the other terminal B is performed (step S11).

  Then, “200 OK” is received as the status code included in the message transmitted from the terminal B, and the information of the voice codec (X, Y in FIG. 7) corresponding to the terminal B is included in this message. If all of the corresponding audio codecs are the same, the audio conversion device transfers the message without performing media information conversion processing (step S12). Then, codec conversion is not executed for RTP packets.

  Next, a case where the corresponding audio codec is partially the same in two terminals performing communication will be described with reference to FIG.

  If some of the corresponding audio codecs are the same between the terminals, media information addition processing (step S21) at the time of initial INVITE reception, and media information that does not match at the time of reception of a response message from the destination terminal side from the message Delete and transfer (step S22). Therefore, for the signaling message, media information conversion processing is performed when initial INVITE is received and when a 200 response is received from the called side. Also, codec conversion is not performed for RTP packets.

  Next, a case where the corresponding audio codecs are different in the two terminals that perform communication will be described with reference to FIG.

  If the corresponding audio codec is different between terminals, the media information addition process (step S31) at the time of initial INVITE reception and the media information on the calling terminal side from the media information table 510 upon reception of a response message from the destination terminal side. The message is acquired, and the media information of the message is rewritten to the information of the calling terminal and transferred (step S32). For RTP packets, codec conversion is executed according to the codec conversion information table for the corresponding call.

  Also, the above voice codec conversion process is a process of rewriting the media information of the SDP header stored in the message body part of the SIP message.

Here, the editing process of the SDP media information of the control message will be described.
(1) INVITE message reception Information on all voice codecs that can be supported by the gateway device is added to the (m =) field of the SDP. The codec uses the RTP payload type defined by RFC3551 and IANA.
(2) Reception of status code 200 When receiving a response message of status code 200 from receiving side terminal B, check whether the media information of the received message matches the media information of terminal 701 that is the calling side terminal To do. If the media information of both terminals match, the message is transferred without rewriting the codec information.
If the media information of the two terminals does not match, the voice conversion device 102 selects one from the list of codec information of the calling terminal A, rewrites the codec, and transfers it.
(3) Reception of status code 180 When the SDP is included in the received message, it is the same as when the status command 200 response is received.
(4) Re-INVITE and UPDATE message reception Media information registration update is performed. This is the same as when the status code 200 response is received.

  Next, the operation of the speech conversion apparatus having the above configuration will be described with reference to the operation sequence diagram of FIG.

  First, the voice conversion device 703 arranged on the relay path receives a connection request message from the terminal 701 to the terminal 705 (step S50).

  Next, the type of the received message is checked, and if it is an initial INVITE message, a call memory is newly generated and call identification dialog information is registered (step S51). On the other hand, if the type of the received message is not the initial INVITE message, a call information search is performed.

  Subsequently, when the SDP information is added to the header part of the SIP message received from the terminal 701 on the transmission side, the media information is acquired from the SDP part of the message. The read media information of the calling terminal is stored in the media information table 510 (step S52). Since the terminal 701 is the calling terminal, the codec information to be converted is written in the field corresponding to the calling terminal in the codec conversion information table (step S53). If there is no media information table 510, a new information is created and associated with the call information table.

  Next, all the media information is written in the SDP media information of the SIP request message received from the originating terminal 701 (step S54), and the IP address and port are set and transferred to the terminating terminal 705. (Step S55).

  Subsequently, after the INVITE message reaches the terminal 705 on the receiving side, response messages of the status code 100, the status code 180, and the status code 200 are returned in order from the terminal 705 on the receiving side (steps S60, S70, S80). .

  The voice conversion device 703 transmits the status code 100 response after receiving the INVITE message (step S61), and also retrieves the call memory when the status code 180 response is received (step S70) and when the status code 200 response is received (step S81). Dialog information registration (steps S72 and S82), storage of media information included in the message (steps S73 and S83), and setting of the codec type in the receiving terminal 705 field of the codec conversion table (steps S74 and S84) Do. At this point, information that allows bidirectional codec conversion from the terminal 701 to the terminal 705 and from the terminal 705 to the terminal 701 can be acquired.

  If the codec is different between the terminals 701 and 705, the media information of the received response message is rewritten to the information of the terminal 701 (step S75). The response from the destination terminal is transferred to the terminal 701 side after setting the IP address and port (steps S76 and S86).

  When the RTP audio data is transferred thereafter (step S90), the audio codec conversion is executed based on the specified codec information with reference to this information table.

  As described above, according to the present embodiment, when a call is connected between terminals, the media information of the call connection control message by SIP is edited and the codec of the voice RTP packet is converted between the terminals. Voice communication is possible even between terminals having different audio data formats. In other words, in the current VoIP, call connection and call can be realized even between terminals having different types of supported voice codecs that cannot be realized by the SIP protocol.

  In addition, since the voice codec can be converted into a codec that is compatible with the connection destination terminal, it is not necessary to be aware of the type of the corresponding voice codec when introducing the IP terminal. Thereby, for example, even when a low-function terminal with a small number of corresponding voice codecs is introduced, call connection and telephone call can be reliably realized, and the VoIP system introduction cost can be suppressed.

  Furthermore, for call connections destined for IP terminals belonging to a narrowband network, network bandwidth is used more efficiently by converting to a voice codec with high compression efficiency, and conversely, it is converted to a high-quality voice codec for broadband networks. By doing so, it is possible to realize a voice call service that is easier for the user to hear.

  Furthermore, when applied to a broadcasting system, a plurality of codecs can be used in parallel. As a result, when performing broadcast communication to multiple terminals, for example, content distributors can provide a variety of services using multiple data formats by converting the codec for each destination terminal. It becomes.

  In addition, when distributing content data, there is no need to create a plurality of the same data for each codec type in advance, the data size for content creation can be reduced, and content such as introduction of an auxiliary storage device such as a hard disk device It is also possible to reduce production costs.

1 is a block diagram showing a schematic configuration of a network system including a voice conversion device according to the present invention. It is a functional block diagram which comprises the speech converter concerning this invention. It is the figure which showed the kind of data which the audio | voice conversion apparatus concerning this invention hold | maintains, and its correlation. It is the figure which showed the call information management table used in order to hold | maintain the call information of the speech converter concerning this invention. It is the figure which showed the media information table used in order to hold | maintain the media information of SDP of the speech converter concerning this invention. It is the figure which showed the codec conversion table used in order to hold | maintain the conversion information of the RTP audio | voice codec of the audio | voice conversion apparatus concerning this invention. It is explanatory drawing of the media conversion of the audio | voice conversion apparatus concerning this invention in case two corresponding | compatible audio codecs are the same in two terminals which communicate. It is explanatory drawing of the media conversion of the audio | voice conversion apparatus concerning this invention in case two corresponding audio codecs are the same in two terminals which communicate. It is explanatory drawing of the media conversion of the audio | voice conversion apparatus concerning this invention when the two corresponding | compatible audio codecs differ in two terminals which communicate. It is an operation | movement sequence diagram of the speech converter concerning this invention.

Explanation of symbols

10, 11 VoIP network 11, 12, 21, 22 IP terminal 100 carrier network 101 SIP server 102 voice converter 103 router device 104 VoIP gateway device 200 internet 201 other carrier network 202 network other than IP (PSTN, PHS, mobile phone network, etc.)
301 packet receiver 302 protocol controller 303 call controller 304 media information converter 305 RTP packet termination unit 306 voice codec converter 307 packet transmitter 401 call information management table 402 media information table 403 media information table (for reception I / F) )
404 Media information table (for sending I / F)
405 Codec conversion table 500 Call information management table 501 Call information identification number 502 Call identification information (dialog)
510 Media information table 511 I / F identification number 512 Codec list 520 Codec conversion table 521 Transmission destination IP address 522 Port number 523 Conversion destination codec 701, 705 Terminal (IP telephone terminal)
702, 704 SIP server 703 voice conversion device

Claims (12)

In a voice conversion device arranged in a network that performs VoIP communication,
For each call connection between the terminals constituting the network, information on the voice codec supported by each terminal is held,
An audio conversion apparatus that performs the audio codec conversion so that the terminals can communicate with each other. Codec information identifying means for identifying the voice codec included in a connection request message and a connection response message according to a call connection procedure of a predetermined protocol;
A data table for storing information relating to the identified audio codec;
Information registration means for registering the voice codec information for each call connection with respect to the data table;
The audio conversion apparatus according to claim 1, further comprising: codec conversion means for performing audio codec conversion with reference to the data table.   The voice conversion device is disposed between the SIP network and another SIP network, or between the SIP network and a network under management of other communication carriers including the PSTN network, the PHS network, and the mobile phone network. The voice conversion device according to claim 1, wherein:   The codec conversion means selects the voice codec with a smaller amount of data and performs the voice codec conversion when making a call connection to a terminal under network management with a narrower bandwidth. The voice conversion device described.   When the codec conversion means handles a call connection request message by SIP and transfers the message to the connection destination network, the codec conversion means notifies the plurality of voice codecs to be used, so that the connection destination network can be used. 5. The audio conversion apparatus according to claim 1, wherein the audio codec conversion is performed between an audio codec notified as an audio codec and an audio codec notified by a connection source network. The information registration means acquires the voice codec information for each of the plurality of terminals of the broadcast communication destination in the call connection for performing the broadcast communication, and registers it in the data table,
6. The voice conversion according to claim 1, wherein the codec conversion unit includes a transmission unit that copies and transmits transmission data to the plurality of terminals that are broadcast destinations. apparatus. A plurality of terminals arranged in a network performing VoIP communication;
A voice conversion device that holds information of a voice codec supported by each terminal for each call connection between the terminals and performs the voice codec conversion so that the terminals can communicate with each other; A network system comprising: The voice conversion device, according to a call connection procedure of a predetermined protocol, codec information identification means for identifying the voice codec included in a connection request message and a connection response message;
A data table for storing information relating to the identified audio codec;
Information registration means for registering the voice codec information for each call connection with respect to the data table;
The network system according to claim 7, further comprising codec conversion means for performing audio codec conversion with reference to the data table. In a method for controlling a voice conversion device arranged in a network that performs VoIP communication,
For each call connection between the terminals constituting the network, a process of holding information on the voice codec supported by each terminal;
And a process for performing the voice codec conversion so that the terminals can communicate with each other. In a control method of a network system, comprising: a network; a plurality of terminals that are arranged in the network and perform VoIP communication; and a voice conversion device that performs the voice codec conversion so that the terminals can communicate with each other.
For each call connection between the terminals, a process of holding information on a voice codec supported by each terminal;
And a step of performing the voice codec conversion so that the terminals can communicate with each other. In a control program for controlling a voice conversion device arranged in a network for performing VoIP communication by a computer,
For each call connection between the terminals constituting the network, the information of the voice codec supported by each terminal is held,
A control program for causing the voice codec conversion to be performed so that the terminals can communicate with each other. A computer controls a network system including a network, a plurality of terminals arranged in the network and performing VoIP communication, and a voice conversion device that performs voice codec conversion so that the terminals can communicate with each other. In the control program of
For each call connection between the terminals constituting the network, the information of the voice codec supported by each terminal is held,
A control program for causing the voice codec conversion to be performed so that the terminals can communicate with each other.

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