JP2009272668A - GATEWAY APPARATUS, VoIP NETWORK MUTUAL CONNECTION SYSTEM, COMMUNICATION METHOD, AND PROGRAM - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a technique for mutually connecting between a plurality of VoIP networks, at low cost. <P>SOLUTION: A gateway apparatus includes: a conversion rule storage means for storing a conversion rule for performing conversion processing between a call control message of a specification corresponding to a VoIP network and a common expression correspondingly to each VoIP; a means for converting the call control message into the common expression, according to the conversion rule corresponding to a certain VoIP network, when the call control message is received from the VoIP network; a means for performing the call control processing by using the obtained common expression; and a means for converting the common expression into a call control message of a specification that corresponds to a VoIP network of a transmission destination according to a conversion rule, corresponding to the VoIP network which is the transmission destination of the call control message, based on the processed common expression, and transmitting the call control message to the VoIP network of the transmission destination. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present invention relates to a gateway device that interconnects a plurality of VoIP networks that provide telephone services over an IP network such as the Internet.

  In recent years, IP telephone services that can use 050 numbers and 0AB to J numbers similar to fixed telephones have become widespread. In such an IP telephone service, a certain quality guarantee is made, and the above telephone numbers are assigned on the assumption.

  On the other hand, with the spread of VoIP (voice over Internet protocol), VoIP services without quality assurance are provided on the Internet by various service providers.

  In this specification, the IP telephone service with quality assurance as described above is referred to as a 050 telephone service, and the network is referred to as a 050 telephone network or a quality assurance VoIP network. In addition, a service without quality assurance is called an Internet telephone service, and the network is called an Internet telephone network. In addition, these services are collectively called a VoIP service, and the network is called a VoIP network.

  As described above, there are many VoIP services at present, and if these VoIP services can be interconnected, the user's callable range is expanded and the convenience of the users is improved. In particular, if an interconnection between the 050 telephone network connected to the public switched telephone network (PSTN) and the Internet telephone network is realized, a terminal connected to the PSTN from a terminal connected to the Internet telephone network Therefore, convenience for the user is greatly improved.

Now, all VoIP services currently provided are basically based on SIP technology and have a common technical foundation. However, the networks related to these multiple VoIP services exist individually, and at present, the interconnection is hardly realized.
"Second Study Report on the Way of Telecommunication Numbers in the IP Era", June 2006, Ministry of Internal Affairs and Communications

Although there is a need for interconnection between a plurality of VoIP services as described above, there is a problem of cost for realizing the interconnection as described below as a reason why it is hardly realized.
Each company's VoIP network differs in the details of the SIP specifications used therefor, so it is difficult to communicate between networks simply by connecting the networks. For this reason, a gateway device for absorbing SIP specification differences is required for interconnection. However, in order to interconnect a VoIP network with other N VoIP networks, N gateway devices are required, which is expensive.

  The present invention has been made in view of the above points, and an object thereof is to provide a technique for interconnecting a plurality of VoIP networks at a low cost.

  In order to solve the above-described problems, the present invention is a gateway device for interconnecting a plurality of VoIP networks, and includes a call control message having a specification corresponding to the VoIP network and a call having a specification independent of the VoIP network. A conversion rule storage means for storing a conversion rule for performing a conversion process between the common expression as a control message in association with each VoIP network, and when a call control message is received from a certain VoIP network, the VoIP network Call conversion using the common expression obtained by the message reception conversion means and the message reception conversion means for reading the conversion rule corresponding to the message from the conversion rule storage means and converting the call control message into a common expression according to the conversion rule. Call control processing means for processing, and a conversion rule corresponding to the VoIP network to which the call control message is transmitted based on the common expression processed by the call control processing means. And the common expression is converted into a call control message having a specification corresponding to the destination VoIP network according to the conversion rule, and the call control message is transmitted to the destination VoIP network. And a message conversion / transmission means.

  In the gateway device, the message reception conversion means may identify the VoIP network that is the transmission source of the call control message based on the IP address of the transmission source of the received call control message.

  The gateway apparatus may further include an identification information correspondence table storage unit that stores a correspondence table in which terminal identification information using an identifier and terminal identification information using a telephone number are associated with each other. The processing means may comprise means for converting the identification information of the transmission source terminal or the identification information of the transmission destination terminal in the common expression by referring to the correspondence table stored in the identification information correspondence table storage means. .

  Further, the call control processing means performs voice communication between the call connection request source terminal and the call connection request destination terminal via voice relay means provided in the gateway device or outside the gateway device. The call control processing unit may perform the call control processing as performed, and the call control processing unit may perform voice communication directly between the terminal of the call connection request source and the terminal of the call connection request destination. Call control processing may be performed.

  The present invention also provides the gateway device, a call control device (SIP server) that is connected to the gateway device and provides a VoIP service in a predetermined VoIP network, and is connected to the gateway device and connected to the PSTN. It can also be configured as a VoIP network interconnection system characterized by comprising a call control device for providing a VoIP service in a guaranteed VoIP network.

  The present invention can also be configured as a communication method corresponding to the processing of the gateway device and a program that causes a computer to function as each means of the gateway device.

  According to the present invention, a conversion rule for performing conversion processing between a call control message having a specification corresponding to the VoIP network and a common expression which is a call control message having a specification independent of the VoIP network is supported for each VoIP network. The gateway device is provided with a conversion rule storage means for storing the call control message. When a call control message is received from a certain VoIP network, the call control message is converted into a common expression according to the conversion rule, and the call control processing is performed using the common expression. The common expression is converted into a call control message according to the conversion rule corresponding to the destination VoIP network, and the call control message is transmitted to the destination VoIP network. Specification difference can be absorbed. For example, even when N VoIP networks are connected to each other, it is only necessary to install one gateway device. It is possible to provide a technology for interconnecting at low cost.

  Embodiments of the present invention will be described below with reference to the drawings. In this embodiment, an example is shown in which SIP is used as a call control method, but the present invention can also be applied to call control methods other than SIP.

(Overall configuration, operation overview)
FIG. 1 shows a configuration example of an interconnection network using a gateway device according to an embodiment of the present invention.

  As shown in FIG. 1, the interconnection network in the present embodiment is configured by connecting an Internet telephone network A2, an Internet telephone network B3, an Internet telephone network C4, and a 050 telephone network 5 to one gateway device GW1. Yes. The 050 telephone network 5 is connected to the PSTN 7 via the PSTN gateway device 6. The interconnection between the 050 telephone network 5 and the PSTN 7 can be implemented using existing technology.

  In the present embodiment, SIP server A8 performs SIP call control processing in Internet telephone network A2, SIP server B9 performs call control processing in Internet telephone network B3, and SIP server C10 in Internet telephone network C4. Performs call control processing, and in the 050 telephone network 5, the SIP server 050 (11) performs call control processing.

  In the 050 telephone network 5, a telephone number starting from 050 is used as identification information indicating the transmission source and destination, and in each Internet telephone network, starting from 050 as identification information indicating the transmission source and destination. It is assumed that information other than the telephone number (name, etc., which is called an identifier) is used. In the following description, in order to avoid complication, the description of the apparatus and network shown in FIG. 1 will be described without the reference numbers used in FIG.

  As described above, the details of the specifications of SIP used in each VoIP network are generally different. For example, a location (header, etc.) in a SIP message that contains information used as source information in a VoIP network (SIP server) is used as a location in which the source information is described in a SIP message in another VoIP network In some VoIP networks (SIP servers), optional information that must be included in SIP messages is optional information that is not used in other VoIP networks.

  Therefore, the gateway device GW in the present embodiment absorbs the difference in specifications as described above and uses a common SIP expression (referred to as a common SIP expression), so that call control processing based on SIP between VoIP networks. It is possible to do.

  An outline of the operation of the gateway apparatus GW having such a function will be described with reference to FIG. FIG. 2 shows an example in which a call connection is made from a terminal (referred to as terminal A) connected to the Internet telephone network A having SIP server A to a terminal (referred to as 050 terminal) in the 050 telephone network having SIP server 050. Is shown.

  First, the terminal A in the Internet telephone network A performs a call connection operation (calling operation), whereby a call connection request message (INVITE message) is sent to the gateway device GW via the SIP server A (step 1, step). 2). The SIP server A is registered to transfer the SIP message to the gateway device GW for the call connection to the VoIP network other than the Internet telephone network A, and is sent from the terminal A based on the registered content. The call connection request message is transferred to the gateway device GW. This function itself is based on the normal SIP server relay function.

The call connection request message transferred to the gateway device GW is input to the gateway device GW. FIG. 2 shows that the call connection request message received from the SIP server A is input from the route a. In addition, a route b and a route c corresponding to the SIP server B and the SIP server C are also shown, respectively. This “route” is a logical interface associated with each VoIP network connected to the gateway device GW. The gateway device GW is the IP address of the SIP server that configures the connected VoIP network. Identifying the road.
Therefore, at the stage of receiving the call connection request message in FIG. 2, the gateway device GW is based on the transmission source IP address (IP address of SIP server A) included in the call connection request message (IP packet header, etc.). The route a is identified.

  The gateway device GW has a description that uses the SIP message used in the VoIP network for each route (that is, for each VoIP network) regardless of the type of the VoIP network (this is referred to as a common SIP expression). SIP conversion table for conversion to The gateway device GW performs call control processing using this common SIP expression. This SIP conversion table is also referred to in order to convert the common SIP expression into a SIP message corresponding to the SIP message destination path.

  As shown in FIG. 2, the gateway device GW that has received the call connection request message in the path a refers to the SIP conversion table, and converts the call connection request message configured in accordance with the SIP specification in the VoIP network A into a common SIP expression. Convert (step 3).

  Then, the gateway device GW performs a call control process using the common SIP expression (step 4). This call control process is based on the SIP protocol, and determines the SIP server that is the message transfer destination based on the destination information, and the message control for providing the terminal with information necessary for the terminal to perform voice communication. This is a process for generating and transferring. In the present embodiment, a number conversion process, which will be described later, is performed as necessary in the call control process.

  After that, the gateway device GW refers to the SIP conversion table again based on the destination route (in this case, the route 050) determined by the call control process, and displays the common SIP expression as the transfer destination VoIP network 050. The message is converted into a call connection request message according to the SIP specification of the telephone network (step 5), and the call connection request message is transmitted to the SIP server 050 (step 6).

  The call connection request message is transmitted to the 050 terminal via the SIP server 050. Thereafter, a response from the 050 terminal is returned to the terminal A via the SIP server 050, the gateway device GW, and the SIP server A. Also in this case, the gateway device GW converts the SIP message into a common SIP expression based on the reception route, and after the call control process, converts the common SIP expression into a SIP message based on the transmission route and sends it out. By exchanging such messages, each terminal acquires information necessary for voice communication with the partner terminal, and voice communication is performed between the terminals.

  When the B2BUA method (RFC3261) is adopted as a form of call control processing in the gateway apparatus GW, the gateway apparatus GW responds to the call connection request message received from the SIP server A as a UAS (User Agent Server). Is transmitted to the terminal A via the SIP server A, and a response to the call connection request message transmitted to the SIP server 050 is received as a UAC (User Agent Client). Then, by exchanging messages between the gateway device GW and the terminal A and between the gateway device GW and the 050 terminal, information necessary for voice communication is exchanged, and voice communication between the terminal and the terminal is performed. Is possible.

  In the example of FIG. 2, the call connection process between the Internet telephone network A and the 050 telephone network is shown, but the same process as the process shown in FIG. 2 is performed in any combination of VoIP telephone networks. Hereinafter, the configuration and operation of the gateway device GW will be described in more detail.

(Device configuration)
FIG. 3 is a functional configuration diagram of the gateway device GW in the present embodiment. As shown in FIG. 3, the gateway device GW in the present embodiment includes a SIP message processing unit 21 and a voice relay unit 22. The SIP message processing unit 21 is a functional unit that generates, transmits and receives a SIP message so that a voice call can be performed between a terminal connected to a certain VoIP network and a terminal connected to another VoIP network. The voice relay unit 22 is a functional unit for relaying voice communication between terminals. That is, the voice relay unit 22 has a function of performing voice communication with a terminal connected to a certain VoIP network, performing voice communication with a terminal connected to another VoIP network, and connecting these voice communications. Yes. The voice relay unit 22 is given a global address.

  Note that the gateway device GW does not necessarily include the voice relay unit 22. Further, the voice relay unit 22 may be provided as a device different from the gateway device GW.

  The SIP message processing unit 21 includes a SIP message transmission / reception unit 23, a SIP message conversion unit 24, a call control unit 25, a number conversion unit 26, a SIP conversion table 27, and a number conversion table 28.

  The SIP message transmission / reception unit 23 is a functional unit for transmitting / receiving a SIP message via a network. When the SIP message conversion unit 24 receives a SIP message from a SIP server of the VoIP network connected to the gateway device GW, the SIP message conversion unit 24 selects a route based on the source IP address of the SIP message (IP address of the source SIP server). A receiving route), a SIP conversion rule corresponding to the route is acquired from the SIP conversion table 27, and the SIP message is converted into a common SIP expression according to the rule, and connected to the gateway device GW. When a SIP message is sent to a SIP server in the VoIP network, the route (transmission route) is determined based on the IP address (IP address of the destination SIP server) that is the destination of the SIP message. A SIP conversion rule corresponding to the route is acquired from the SIP conversion table 27, and a function for converting the common SIP expression into a SIP message conforming to the specification of the destination VoIP network is provided according to the rule. That.

  FIG. 4 shows an example of the SIP conversion table 27. In the example shown in FIG. 4, for example, for a SIP message received from route a, information obtained by extracting PAI header (P-Asserted-Identity header) information as a transmission source and deleting optional parameters from To header information Is extracted as a transmission destination, and using these pieces of information, a common SIP expression including at least the transmission source and the transmission destination is created. When a SIP message is transmitted to the route a, a PAI header is provided, information on the transmission source in the common SIP expression is described therein, information on the transmission destination in the common SIP expression is attached to the To header, and an optional parameter is set. Create a SIP message by appending. Of course, in the process of converting the received SIP message into the common SIP expression or creating the SIP message to be transmitted from the common SIP expression, other rules than the above rule are also used. Similar processing is performed using each rule. In addition, the SIP message information used in each VoIP network may be used as it is for a portion of the SIP message that has no difference between VoIP networks.

  The same applies to the route b and the route 050. For example, when a SIP message is transmitted to the route 050, the source information in the common SIP expression is described in the From header, and the destination information in the common SIP expression is described. A SIP message is created by adding a character string “user = phone” to the To header. In FIG. 4, X-Header is deleted when a SIP message received from route b is converted into a common SIP expression, and when a SIP message is transmitted to route b, X-Header is added to the SIP message. The addition of is also shown.

  In addition, by providing identification information for identifying the route and associating the identification information with the IP address of the SIP server as a table, the route identification information ("route in FIG. 4") is stored. a "etc.) may be discriminated and the SIP conversion table 27 may be referred to based on the discriminating information, or the route identification information (such as" Route a "in FIG. 4) itself may be used as the IP address.

  The call control unit 25 in FIG. 3 is a functional unit that performs call control processing using the common SIP expression converted by the SIP message conversion unit 24. The call control process here is basically the same as the process of the SIP relay server (SIP proxy server, SIP server having the B2BUA function, etc.) when the common SIP expression is viewed as a normal SIP message. However, in this embodiment, the call control unit 25 uses the number conversion unit 26 to convert the transmission source information or the transmission destination information from information using the identifier to information using the telephone number, and vice versa. It has a function to perform conversion. A specific example of this number conversion will be described later.

  The number conversion unit 26 receives the number conversion request including the transmission source information and the transmission destination information from the call control unit 25, and refers to the number conversion table 28 based on the transmission source information or the transmission destination information. If there is registration information corresponding to these pieces of information in the table 28, it is a functional unit that converts the information and returns the converted information to the call control unit 25.

  FIG. 5 shows an example of the number conversion table 28. In the example of FIG. 5, the SIP-URI in the Internet telephone network using the identifier for the user part and the SIP-URI in the 050 telephone network using the 050 number for the user part are stored in association with each other. Yes.

  In the present embodiment, a user of the Internet telephone network is assigned a 050 number when receiving a VoIP network interconnection service using the gateway device GW according to the present embodiment. In any VoIP network, in the case of connection between VoIP networks, a call connection can be established to a partner terminal using a 050 number. In addition, the PSTN connected to the 050 telephone network can be connected from any VoIP network using the PSTN telephone number, and from the PSTN to any VoIP network connected to the gateway device GW. Can be connected using 050 number.

  For example, when a call connection is made from terminal A connected to the Internet telephone network A to a 050 terminal connected to the 050 telephone network, the terminal A includes the 050 number (SIP-URI) of the 050 terminal as the transmission destination and transmits A call connection request message including the SIP-URI having the identifier of the terminal A as a source is transmitted. In the gateway apparatus GW, the transmission destination is compared with the right column of the number conversion table 28 in FIG. 5, but the number conversion is not performed because there is no entry for the terminal connected to the 050 telephone network. On the other hand, the sender is compared with the left column, and there is an entry, and the SIP-URI having the identifier is converted into a 050 number format SIP-URI. Then, a call connection request message having the converted number information is transmitted to the 050 telephone network, and the 050 terminal is notified of the 050 number as the transmission source.

  Also, for example, when a call connection is made from a terminal A connected to the Internet telephone network A to a terminal B connected to the Internet telephone network B, the user of the terminal A knows the 050 number of the terminal B in the interconnection network The terminal A transmits a call connection request message including the 050 number (SIP-URI) of the terminal B as the transmission destination and the SIP-URI having the identifier of the terminal A as the transmission source. Then, in the gateway device GW, the destination is compared with the right column of the number conversion table 28 in FIG. 5, and there is an entry. Therefore, the 050 number of the terminal B is SIP-URI having an identifier (used in the Internet telephone network B). To be converted). Also, the sender is compared with the left column, and there is an entry, and the SIP-URI having the identifier is converted into a 050 number format SIP-URI. Then, a call connection request message having the converted number information is transmitted to the Internet telephone network B, and the terminal B is notified of the 050 number of the terminal A as a transmission source.

  The gateway device GW can be realized by installing a program corresponding to the processing of each functional unit in a computer having a CPU, a storage device, and the like. The data of each table is stored in the storage device of the computer. Even when the gateway device GW has a configuration that does not include the voice relay unit 22, the gateway device GW can be realized by installing a program corresponding to the SIP message processing unit 21 in the computer.

  The program can be read into a computer from a recording medium such as a portable memory, or can be downloaded to the computer via a network.

(Operation of gateway device GW)
Next, the operation of the gateway device GW will be described in detail. In the following processing, the call control unit 25 of the gateway device performs an operation based on B2BUA (RFC3261) as an operation related to call control.

<Call connection processing from Internet telephone network A>
First, a call connection process from a terminal A connected to the Internet telephone network A to a 050 terminal connected to the 050 telephone network will be described with reference to FIGS. A PSTN terminal connected to the 050 telephone network via a PSTN gateway device is also referred to as a 050 terminal for convenience. In this example, it is assumed that the voice relay unit 22 relays voice communication between the terminal A and the 050 terminal. That is, in the call control process, voice communication is performed between the terminal A and the voice relay unit 22, and processing such as information exchange necessary for voice communication between the 050 terminal and the voice relay unit 22 is performed.

  As shown in FIG. 6, the INVITE message transmitted from the terminal A reaches the gateway device GW via the SIP server A, and the gateway device GW receives it (step 11). The SIP message conversion unit 24 receives the INVITE message via the SIP message transmission / reception unit 23, determines the route (route a) based on the IP address of the transmission source of the received INVITE message, and stores the SIP conversion table 27. By referencing, the INVITE message is converted into a common SIP expression (step 12).

  FIG. 7A shows an example of the INVITE message in this case, and FIG. 7B shows a part of the common SIP expression converted from the INVITE message. The SIP message conversion unit 24 follows the rule corresponding to the route a in the SIP conversion table (FIG. 4), and the information of the P-Asserted-Identity header of the INVITE message shown in FIG. (Sip: user9999@a.example.net) is used, and information (sip: 012345678@050.example.net) obtained by deleting the option parameter from the To header information is used as the destination information. Thereby, a common SIP expression as shown in FIG. 7B is obtained.

Subsequently, the call control unit 25 performs number conversion in the common SIP expression using the number conversion unit 26 (step 13). Since the number conversion table 28 (FIG. 5) records 05099999999@050.example.net corresponding to user9999@a.example.net, the number conversion unit 26 stores user9999@a.example.net. The data is converted to 05099999999@050.example.net, and the converted source information is passed to the call control unit 25. As a result, the common SIP expression is as shown in FIG. Note that the destination information 012345678@050.example.net is not registered because it is not registered in the number conversion table 28. The number conversion of the sender information is mainly intended to notify the destination of the 050 number assigned to the sender terminal, so that the sender information is in the 050 number format (domain is 050.example.net). In the case of (1), transmission source information is not converted.
Thereafter, the call control unit 25 performs call control processing based on the common SIP expression (step 14). The call control processing here is specifically to generate a message (common SIP expression) to be transmitted and received. This message generation process is performed based on the SIP protocol, and includes a message type, a destination determination, a description of information (SDP information) necessary for performing a voice call between terminals, and the like.
In the example of the sequence shown in FIG. 6, since it is assumed that the B2BUA operation is performed, the call control unit 25 creates a response (response) to the INVITE message received from the SIP server A using the common SIP expression. Further, the call control unit 25 generates an INVITE message based on the common SIP expression to be transmitted to the call connection destination terminal (050 terminal) based on the common SIP expression of the INVITE message. Here, the call control unit 25 recognizes from the domain name 050.example.net of the transmission destination information 012345678@050.example.net in the message that the network to which the call connection request destination terminal belongs is the 050 telephone network. Therefore, the INVITE message based on the common SIP expression is a message addressed to the SIP server 050 (route 050).

  Then, the SIP message conversion unit 24 converts the response message based on the common SIP expression sent to the route a into a response message corresponding to the route a (step 15), and transmits it to the SIP server A ( Step 16). Further, the SIP message conversion unit 24 converts the INVITE message based on the common SIP expression into the INVITE message corresponding to the route 050 using the conversion rule corresponding to the route 050 (step 17), and converts it to the SIP server 050. Transmit (step 18). Note that message communication between the terminal A and the gateway device GW and message communication between the 050 terminal and the gateway device GW may be performed asynchronously. The same applies to other examples.

  In this conversion, the transmission source information in the common SIP expression shown in FIG. 7C is set as the From value of the INVITE message according to the rule corresponding to the route 050 in FIG. 4, and the transmission destination in the common SIP expression Information is set as the value of To and "; user = phone" is added. By performing such conversion, an INVITE message shown in FIG. 7D is generated and transmitted to the SIP server 050.

  Note that the above shows a part of the SIP message transmitted and received between the terminal and the gateway device GW. In actuality, the SIP protocol is used to send and receive only messages necessary for call connection (exchange of information necessary for voice communication). Based on However, for any message, the gateway device GW converts the SIP message into a common SIP expression based on a rule corresponding to the route, and converts the common SIP expression into a SIP message.

  Next, a voice communication path will be described with reference to FIG. As described above, in this example, the voice relay unit 22 of the gateway device GW relays voice communication between the terminal A and the 050 terminal. Therefore, by performing the above-described call control processing based on SIP between the terminal A and the gateway device GW, the terminal A obtains information (address, port, media information, etc.) necessary for performing voice communication with the voice relay unit 22. The SIP message processing unit 21 of the gateway device GW acquires information necessary for the voice relay unit 22 to perform voice communication with the terminal A. Similarly, the 050 terminal performs voice communication with the voice relay unit 22. The SIP message processing unit 21 of the gateway device GW acquires information necessary for the voice relay unit 22 to perform voice communication with the 050 terminal (step 100).

  Then, the SIP message processing unit 21 gives a voice relay command to the voice relay unit 22 by passing the information obtained as described above to the voice relay unit 22 (step 110). In accordance with this voice relay command, the voice relay unit 22 terminates voice packet (UDP packet) communication from the terminal A, generates a voice packet to the 050 terminal, and transmits the packet to the 050 terminal. The voice packet communication is terminated, a voice packet to terminal A is generated, and a process of transmitting it to terminal A is performed (step 120).

<Call connection processing from Internet telephone network B>
Next, a call connection process from terminal B connected to the Internet telephone network B to 050 terminal connected to the 050 telephone network will be described with reference to FIGS. Also in this example, it is assumed that the voice relay unit 22 relays voice communication between the terminal B and the 050 terminal.

  As shown in FIG. 9, the INVITE message transmitted from the terminal B reaches the gateway device GW via the SIP server B, and the gateway device GW receives it (step 21). The SIP message conversion unit 24 receives the INVITE message via the SIP message transmission / reception unit 23, determines the route (route b) based on the IP address of the transmission source of the received INVITE message, and stores the SIP conversion table 27. By referencing, the INVITE message is converted into a common SIP expression (step 22).

  FIG. 10A shows an example of the INVITE message in this case, and FIG. 10B shows a part of the common SIP expression converted from the INVITE message. The SIP message conversion unit 24 follows the rule corresponding to the route b in the SIP conversion table (FIG. 4), and includes the From information (sip: nick8888 @) of the INVITE message shown in FIG. b.example.net), the To header information (sip: 012345678@050.example.net) is adopted as the destination information, and the X-Header is deleted (ignored). Thereby, a common SIP expression including the expression shown in FIG.

  Subsequently, the call control unit 25 converts nick8888@b.example.net to 05088888888@050.example.net using the number conversion unit 26 (step 23). As a result, the common SIP expression is as shown in FIG.

  Thereafter, the call control unit 25 generates a response to the INVITE message received from the SIP server B using the common SIP expression, and generates an INVITE message based on the common SIP expression addressed to the SIP server 050 (route 050). The call control process including the process to be performed is performed (step 24).

  Then, the SIP message conversion unit 24 converts the response message based on the common SIP expression sent to the route b into a response message corresponding to the route b (step 25), and transmits it to the SIP server B (step 25). Step 26). Further, the SIP message conversion unit 24 converts the INVITE message based on the common SIP expression into an INVITE message corresponding to the route 050 using the conversion rule corresponding to the route 050 (step 27), and converts it to the SIP server 050. Transmit (step 28).

  In this conversion, the source information in the common SIP expression shown in FIG. 10C is set as the From value of the INVITE message corresponding to the route 050 according to the rule corresponding to the route 050 in FIG. The destination information in the SIP expression is set as the value of To, and “; user = phone” is added. By performing such conversion, the INVITE message shown in FIG. 10 (d) is generated and transmitted to the SIP server 050.

  Next, a voice communication path will be described with reference to FIG. In this example, similarly to the example described above, by performing the above-described call control processing based on SIP between the terminal B and the gateway device GW, the terminal B obtains information necessary for voice communication with the voice relay unit 22. The SIP message processing unit 21 of the gateway device GW acquires information necessary for the voice relay unit 22 to perform voice communication with the terminal B. Similarly, the terminal B performs voice communication with the voice relay unit 22. The SIP message processing unit 21 of the gateway device GW acquires information necessary for the voice relay unit 22 to perform voice communication with the terminal B (step 200).

  Then, the SIP message processing unit 21 gives the voice relay unit 22 a voice relay command by passing the information obtained as described above to the voice relay unit 22 (step 210). In accordance with this voice relay instruction, the voice relay unit 22 terminates voice packet communication from the terminal B, generates a voice packet to the 050 terminal, transmits it to the 050 terminal, and voice packet communication from the 050 terminal. , And generates a voice packet to terminal B and transmits it to terminal B (step 220).

<Connection process from Internet telephone network A to Internet telephone network B>
Next, an example in which a call connection is made from a terminal A connected to the Internet telephone network A to a terminal B connected to the Internet telephone network B will be described with reference to FIGS. In this example, it is assumed that the user of terminal A knows the 050 number in the VoIP interconnection network of terminal B and uses that 050 number to make a call connection.

  As shown in FIG. 12, the INVITE message transmitted from the terminal A reaches the gateway device GW via the SIP server A, and the gateway device GW receives it (step 31). The SIP message conversion unit 24 receives the INVITE message via the SIP message transmission / reception unit 23, determines the route (route a) based on the IP address of the transmission source of the received INVITE message, and stores the SIP conversion table 27. By referencing, the INVITE message is converted into a common SIP expression (step 32).

  FIG. 13A shows an example of the INVITE message in this case, and FIG. 13B shows a part of the common SIP expression converted from this INVITE message. The SIP message conversion unit 24 follows the rule corresponding to the route a in the SIP conversion table (FIG. 4), and information on the P-Asserted-Identity header of the INVITE message shown in FIG. (Sip: user9999@a.example.net) is used, and information (sip: 012345678@050.example.net) obtained by deleting the option parameter from the To header information is used as the destination information. As a result, a common SIP expression including the expression shown in FIG. 13B is obtained.

  Subsequently, the call control unit 25 uses the number conversion unit 26 to convert user9999@a.example.net to 05099999999@050.example.net. In addition, since nick8888@b.example.net is recorded in the number conversion table 28 corresponding to the transmission destination information 05088888888@050.example.net, the call control unit 25 calls 05088888888@050.example.net. Is converted to nick8888@b.example.net. Thereby, the common SIP expression after number conversion including the expression as shown in FIG. 13C is obtained (step 33).

  Thereafter, the call control unit 25 generates a response to the INVITE message received from the SIP server A using the common SIP expression, and generates an INVITE message based on the common SIP expression addressed to the SIP server B (route b). A call control process including the process to be performed is performed (step 34). Here, the call control unit 25 determines that the network to which the call connection request destination terminal belongs is the Internet telephone network B from the domain name b.example.net in the destination information nick8888@b.example.net in the common SIP expression of the INVITE message. And an INVITE message based on the common SIP expression addressed to SIP server B (route b) is generated.

  Then, the SIP message conversion unit 24 converts the response message based on the common SIP expression transmitted to the route a into a response message corresponding to the route a (step 35), and transmits it to the SIP server A ( Step 36). Further, the SIP message conversion unit 24 converts the INVITE message based on the common SIP expression into the INVITE message corresponding to the route b using the conversion rule corresponding to the route b (step 37), and converts it to the SIP server B. Transmit (step 38).

  In this conversion, the source information in the common SIP expression shown in FIG. 13 (c) is set as the From value of the INVITE message corresponding to the route b according to the rule corresponding to the route b in FIG. The destination information in the SIP expression is set as a value of To, and an X-header is added. By performing such conversion, an INVITE message shown in FIG. 13 (d) is generated and transmitted to the SIP server B.

  Next, a voice communication path will be described with reference to FIG. The basic processing is the same as in FIGS. 8 and 11, and the terminal A performs voice communication with the voice relay unit 22 by performing the above-described call control processing based on SIP between the terminal A and the gateway device GW. The SIP message processing unit 21 of the gateway device GW acquires information necessary for the voice relay unit 22 to perform voice communication with the terminal A. Similarly, the terminal B obtains information necessary for voice communication with the voice relay unit 22, and the SIP message processing unit 21 of the gateway device GW allows the voice relay unit 22 to perform voice communication with the terminal B. Necessary information is acquired (step 300).

  Then, the SIP message processing unit 21 gives the voice relay unit 22 a voice relay command by passing the information obtained as described above to the voice relay unit 22 (step 310). In accordance with this voice relay command, the voice relay unit 22 terminates voice packet communication from the terminal A, generates a voice packet to the terminal B, transmits it to the terminal B, and voice packet communication from the terminal B. , And generates a voice packet for terminal A and transmits it to terminal A (step 320).

  Further, voice communication may be performed between terminals without using the voice relay unit 22. FIG. 15 shows a voice communication path in such a case. In this case, in the call control process based on SIP described above, terminal A acquires information necessary for voice communication with terminal B, and terminal B acquires information necessary for voice communication with terminal A. (Step 400), perform voice communication with each other. FIG. 15 shows an example of voice communication between the terminal A and the terminal B, but the same direct communication is possible between the terminal A and the 050 terminal and between the terminal B and the 050 terminal.

  8, 11, and 15, by adopting a method of relaying voice communication between terminals by the voice relay unit 22 to which a global address is assigned, either or both terminals are connected to NAT ( Even when the network address translation device is under control, there is an effect that the possibility of voice communication between terminals is increased. This is because, in general, even when the terminal is inside the NAT device, packet communication between the terminal and the device assigned with the global address can often be performed without any problem.

(Effect of embodiment)
As described above, according to the embodiment of the present invention, the gateway device GW that absorbs SIP specification differences in a plurality of VoIP networks and performs call control processing operation is introduced. Even when two VoIP networks are connected to each other, it is only necessary to install one gateway device, and interconnection between a plurality of VoIP networks is possible at a low cost.

  In this embodiment, the 050 telephone network, which is a quality assurance network connected to the PSTN, and the Internet telephone network are interconnected. In other words, one relay VoIP network (050 telephone network) in which the gateway device is arranged is interconnected in a form of a hub with another VoIP network. As a result, the service provider of the Internet telephone network can connect to the PSTN without installing a connection device to the PSTN, and can expand the target range of communication at low cost. Furthermore, by connecting between different Internet telephone networks via the gateway device, the range of call destinations for the users of the respective telephone networks is expanded, and convenience is improved.

  The present invention is not limited to the above-described embodiments, and various modifications and applications are possible within the scope of the claims.

It is an example of composition of an interconnection network using a gateway device concerning an embodiment of the invention. It is a figure for demonstrating the outline | summary of operation | movement of the gateway apparatus GW. It is a functional block diagram of gateway apparatus GW. It is a figure which shows an example of the SIP conversion table. It is a figure which shows an example of the number conversion table. 7 is a sequence diagram showing call connection processing from a terminal A connected to the Internet telephone network A to a 050 terminal connected to the 050 telephone network. FIG. It is a figure which shows the example of the message in the example shown in FIG. It is a figure which shows the path | route of the audio | voice communication in the example shown in FIG. FIG. 6 is a sequence diagram showing call connection processing from a terminal B connected to the Internet telephone network B to a 050 terminal connected to the 050 telephone network. It is a figure which shows the example of the message in the example shown in FIG. It is a figure which shows the path | route of the audio | voice communication in the example shown in FIG. FIG. 4 is a sequence diagram showing a call connection process from a terminal A connected to the Internet telephone network A to a terminal B connected to the Internet telephone network B. It is a figure which shows the example of the message in the example shown in FIG. It is a figure which shows the path | route of the audio | voice communication in the example shown in FIG. It is a figure which shows the other example of the path | route of the audio | voice communication in the example shown in FIG.

Explanation of symbols

1 Gateway device GW
2 Internet telephone network A
3 Internet telephone network B
4 Internet telephone network C
5 050 telephone network 6 PSTN gateway device 7 PSTN
8 SIP server A
9 SIP server B
10 SIP server C
11 SIP server 050
21 SIP message processing unit 22 Voice relay unit 23 SIP message transmission / reception unit 24 SIP message conversion unit 25 Call control unit 26 Number conversion unit 27 SIP conversion table 28 Number conversion table

Claims (8)

A gateway device for interconnecting multiple VoIP networks,
A conversion rule for storing a conversion rule for conversion processing between a call control message having a specification corresponding to the VoIP network and a common expression which is a call control message having a specification independent of the VoIP network in association with each VoIP network. Storage means;
When a call control message is received from a certain VoIP network, a conversion rule corresponding to the VoIP network is read from the conversion rule storage unit, and a message reception conversion unit that converts the call control message into a common expression according to the conversion rule;
Call control processing means for performing call control processing using a common expression obtained by the message reception conversion means;
The conversion rule corresponding to the VoIP network that is the transmission destination of the call control message based on the common expression processed by the call control processing means is read from the conversion rule storage means, and the common expression is converted to the destination VoIP according to the conversion rule. A gateway device comprising: message conversion and transmission means for converting the call control message into specifications corresponding to a network and transmitting the call control message to the destination VoIP network.   2. The gateway apparatus according to claim 1, wherein the message reception conversion unit specifies a VoIP network that is a transmission source of the call control message based on an IP address of a transmission source of the received call control message. The gateway device further includes identification information correspondence table storage means for storing a correspondence table in which terminal identification information using an identifier and terminal identification information using a telephone number are associated with each other,
The call control processing means includes means for converting the identification information of the transmission source terminal or the identification information of the transmission destination terminal in the common expression by referring to the correspondence table stored in the identification information correspondence table storage means. The gateway device according to claim 1, wherein:   The call control processing means performs voice communication between a call connection request source terminal and a call connection request destination terminal via a voice relay means provided in the gateway device or outside the gateway device. 4. The gateway device according to claim 1, wherein the call control process is performed as described above.   4. The call control processing means performs call control processing so that voice communication between a call connection request source terminal and a call connection request destination terminal is directly performed between the terminals. The gateway apparatus of any one of these. The gateway device according to any one of claims 1 to 5,
A call control device connected to the gateway device and providing a VoIP service in a predetermined VoIP network;
A call control device connected to the gateway device and providing a VoIP service in a quality assurance VoIP network connected to the PSTN;
A VoIP network interconnection system characterized by comprising: A gateway device for interconnecting multiple VoIP networks, which performs conversion processing between call control messages with specifications compatible with VoIP networks and common expressions that are call control messages with specifications independent of VoIP networks. A communication method executed by a gateway device having conversion rule storage means for storing conversion rules to be associated with each VoIP network,
When a call control message is received from a certain VoIP network, a conversion rule corresponding to the VoIP network is read from the conversion rule storage means, and a message reception conversion step of converting the call control message into a common expression according to the conversion rule;
A call control processing step for performing a call control process using the common expression obtained by the message reception conversion step;
The conversion rule corresponding to the VoIP network that is the transmission destination of the call control message based on the common expression processed in the call control processing step is read from the conversion rule storage means, and the common expression is converted to the destination VoIP according to the conversion rule. A message conversion and transmission step of converting the call control message to a specification corresponding to the network and transmitting the call control message to the destination VoIP network. A program for causing a computer to function as a gateway device for interconnecting a plurality of VoIP networks,
A conversion rule for storing a conversion rule for conversion processing between a call control message having a specification corresponding to the VoIP network and a common expression which is a call control message having a specification independent of the VoIP network in association with each VoIP network. Storage means,
When a call control message is received from a certain VoIP network, the conversion rule corresponding to the VoIP network is read from the conversion rule storage unit, and the message reception conversion unit converts the call control message into a common expression according to the conversion rule,
Call control processing means for performing call control processing using the common expression obtained by the message reception conversion means;
The conversion rule corresponding to the VoIP network that is the transmission destination of the call control message based on the common expression processed by the call control processing means is read from the conversion rule storage means, and the common expression is converted to the destination VoIP according to the conversion rule. A message conversion transmission means for converting the call control message into a specification corresponding to the network and transmitting the call control message to the destination VoIP network;
Program to function as.

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