JP2006229550A - VoIP-GW APPARATUS - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a VoIP-GW (Voice Over Internet Protocol Gate Way) apparatus which does not need to be conscious of a line for every SIP (Session Initiation Protocol) server by an exchange side. <P>SOLUTION: The VoIP-GW apparatus 1 provided between a non-IP network side and an IP network includes an IP network side interface 11, an SIP controller 14 which controls a communication with the SIP server of the IP network side, a non-IP network side interface 12, an RTP controller 16 which controls the communication with the non-IP network side, a main control unit 13 having an SIP route distribution table which controls the operation of the entire apparatus and describes the correspondence with a landing number and the SIP server, and an internal switch controller 15 which changes a mutual connection at the interface 11, the SIP controller 14, the interface 12, the RTP controller 16, and the main control unit 13. With reference to the SIP route distribution table, the route to the SIP server is determined based on a contractor identification number in an incoming number described in a connection request from the non-IP network side to the SIP server. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

  The present invention is for avoiding a practical problem of a multi-ISP compatible VoIP (Voice on IP) -GW (Gateway).

  IP telephones whose dial numbers start with 050 are becoming common. An IP phone is a service provided by an IP phone carrier, and call control is performed via a server in an ISP (Inter-Net Service Provider) in order to establish End-End communication.

  Most of the intra-ISP server protocols for IP telephones employ SIP (Session Initiation Protocol) and communicate via the SIP server. The SIP communication protocol itself is a protocol standardized by RFC 3261, RFC 3264, etc., but it cannot be said that the protocol is completely unified by the original specification of each IP telephone carrier's SIP server. There is also a problem with the interconnection.

  In order to cope with a plurality of IP telephones corresponding to IP telephone service providers, as shown in the conventional multi-ISP compatible connection configuration diagram shown in FIG. 9, a private branch exchange (PBX) 3A to 3P for an ISP in which an IP telephone is accommodated 3C is connected to the media converter 5 via the VoIP-GWs 9A to 9C, and is connected to the SIP servers 71A to 71C connected to the Internet 7 via the physical line 51. Each of the VoIP-GWs 9A to 9C includes interfaces 91A to 91C for ISP. It is necessary to prepare VoIP-GWs 9A to 9C for each of the SIP servers 71A to 71C, and the exchanges 3A to 3C have to select GW routes based on the numbers.

In addition, a telephone is connected to a public telephone line and an internet circuit via an adapter that performs switching connection according to an identification signal, and the telephone number that identifies the adapter in the internet circuit and corresponding to this. It has already been proposed to provide a VoIP server having a gatekeeper function having data corresponding to an IP address to be used so that a telephone call can be made between telephones connected to the Internet via an adapter (for example, , See Patent Document 1).
JP 2003-264644 A

  In the above prior art, there is a problem in that it is necessary to select a route (GW) based on the number on the exchange side, and it is necessary to prepare an interface for each SIP server.

  It is an object of the present invention to provide a VoIP-GW that does not require the exchange side to be aware of the route for each SIP server. It is another object of the present invention to provide a VoIP-GW that does not require an interface for each SIP server and can reduce the occurrence of busy due to a shortage of interface channels.

  In order to achieve the above object, the present invention performs termination control of the SIP protocol for each global IP address allocated from each ISP, and controls the entire apparatus including conventional voice interface (non-IP interface) control. Solve the problem.

  That is, according to the present invention, in a VoIP-GW device that converts voice communication between a non-IP communication network and an IP communication network, communication between an interface on the IP communication network side and a SIP server installed on the IP communication network side is performed. It controls the SIP control unit to be controlled, the plurality of interfaces on the non-IP communication network side, the plurality of RTP control units for controlling communication with the non-IP communication side, and the operation of the entire apparatus, Between the main control unit having a SIP route distribution table describing the correspondence between the number and the SIP server, the interface on the IP communication network side, the SIP control unit, the interface on the non-IP communication network side, the RTP control unit, and the main control unit An internal switch control unit that controls an internal switch that switches the mutual connection of the non-IP communication network, and there is a connection request to the SIP server from the non-IP communication network side , Based on the vendor identification number of the called party number described in the connection request, and to determine the route to SIP server with reference to the SIP route sorting table.

  In the VoIP-GW apparatus, the main control unit is provided with a channel management table that describes the minimum number of guaranteed channels and the maximum number of communication channels to be used for each SIP server. In such a case, whether or not an incoming call to the SIP server is possible is determined from the current usage status of the route of the SIP server, the minimum number of guaranteed channels, and the maximum number of communication channels.

  Furthermore, the present invention provides communication between an interface on the IP communication network side and a SIP server installed on the IP communication network side in a VoIP-GW device that converts voice communication between the non-IP communication network and the IP communication network. It controls the SIP control unit to be controlled, the plurality of interfaces on the non-IP communication network side, the plurality of RTP control units for controlling communication with the non-IP communication side, and the operation of the entire apparatus, A main control unit having a called number conversion table describing a number and a conversion destination of the called number, an interface on the IP communication network side, an SIP control unit, an interface on the non-IP communication network side, an RTP control unit, and a main control unit An internal switch control unit for controlling an internal switch for switching the mutual connection between the IP communication network side and the connection request when the connection request is made from the IP communication network side to the IP communication network side. Based on the vendor identification number of numbers, and to determine the root after conversion by referring to the called number conversion table.

  According to the present invention, the interface on the PBX side can be kept to a minimum, the system cost can be kept down, and the physical number of VoIP-GWs can be reduced, thereby reducing the management cost and maintenance cost. Connected. Furthermore, according to the present invention, it is possible to reduce the occurrence of busy due to an interface shortage based on physical and cost constraints.

  Examples of the present invention will be described below. A multi-ISP compatible connection configuration using VoIP-GW according to the present invention will be described with reference to FIG. The VoIP-GW 1 according to the present invention is connected to PBXs 3A to 3C to which IP telephones are connected, and is connected to SIP servers 71A to 71C connected to the Internet 7 via the media converter 5 and the physical line 51.

  A detailed configuration of the VoIP-GW 1 according to the present invention will be described with reference to FIG. The VoIP-GW 1 includes a plurality of Internet-side interfaces 11A to 11C, a plurality of PBX-side interfaces 12A to 12C, a main control unit 13, a plurality of SIP control units 14A to 14C, an internal switch control unit 15, and a plurality of RTP control units 16A to 16C.

  The Internet side interface 11 is a physical interface on the Internet side. The Internet side interface 11 is displayed with three interfaces for easy understanding. However, since most of the Internet side interfaces use IEEE802.3, only one physical external interface may be used. .

  The PBX side interface 12 is an interface for analyzing and creating a call control message and signal on the PBX side, and the final judgment of connection or disconnection is determined by the main control unit. The interface 12 on the PBX side is a PBX interface as an example, but is not limited to the PBX, and indicates general non-IP communication including PSTN (Public Switched Telephony Networks).

  The main control unit 13 is a control unit that controls each control unit via the internal switch control unit 15. That is, the main control unit 13 is a control unit that controls the entire apparatus based on information from each control unit, and performs PBX side outgoing / incoming control, internal switch switching operation instruction, Internet side outgoing / incoming control, etc. Do.

  The SIP control unit 14 is a means for analyzing and creating an Internet side child control message, and the main control unit 13 makes a final determination of connection or disconnection. The SIP control unit 14 is a control unit capable of terminating the SIP protocol for each SIP server (ISP), and uses a global IP address assigned by the ISP. In this example, three SIP control units are described, but this SIP control unit requires the number of multiple connections. In addition, although the SIP control unit is used for explanation, the protocol is not limited to SIP, and represents a call control unit in a broad sense.

  The internal switch control unit 15 is a control unit that performs control communication between the main control unit 13 and each of the control units 14 and 16, and performs layer 2 and layer 3 header conversion and switching functions of TCP / IP data for communication including RTP data. Have Further, the internal switch control unit 15 controls the operation of the internal switch, transmits a control message within the VoIP-GW 1, and determines a switching method according to an instruction from the main control unit 13. Packet) switching transmission, Internet side child control message (IP packet) switching transmission in which a switching method is determined by an instruction from the main control unit 13 and the like.

  The RTP (Real-time Transport Protocol) control unit 16 is a control unit that performs conversion from non-IP communication to IP communication, and performs decoding or encoding conversion between PCM data and RTP (IP packetized voice) data. It is a means for performing processing.

  The main control unit 13 includes an SIP route distribution table based on called numbers from the PBX side interface 12 shown in FIG. 3, a SIP server unit channel management table 132 shown in FIG. 4 (an example in which the PBX side channel total is 60 Ch), A number conversion table 133 for incoming calls from the Internet side is provided.

  The SIP route distribution table 131 based on the called number from the PBX side interface is a table in which the called number is associated with the route to the SIP server 71 and corresponds to the called number extracted from the connection request received by the PBX side interface 12. Reference is made when the main control unit 13 selects a route to the SIP server.

  The SIP server unit channel management table 132 is a table describing the minimum guaranteed channel number and the maximum communication channel number for each SIP server.

  The called number conversion table 133 for incoming calls from the Internet side is a table describing a converted called number for converting a called number from the Internet side to a called number of another route.

  As shown in the SIP route distribution table 131, the number starting with 050 is the SIP server (ISP) of the communication destination (destination) by the carrier identification code (for example, ABCD, EFGH, IJKL, etc.) indicated by the next four digits. Therefore, it is possible to identify the SIP server route by referring to the SIP route distribution table 131, and it is not necessary to prepare the VoIP-GW and the switch side interface for each SIP server (ISP).

  With this configuration, in the case of outgoing from the PBX interface side, the outgoing event and called number are passed from the PBX side interface 12 to the main control unit 13, and the main control unit 13 sends the route of the SIP server to be sent from the SIP route distribution table 131. Is identified. The main control unit 13 issues instructions to the internal switch control unit 15, the SIP control unit 16, and the RTP control unit 16, and controls the SIP call control and switching of RTP communication of the VoIP-GW1.

  With reference to FIG. 6, processing for distributing SIP routes based on incoming calls from the PBX side interface 12 will be described. When the PBX side interface 12A receives a transmission signal (number: 050-ABCD-XXXX) from the PBX 3A, it transmits it to the main control unit 13 via the internal switch control unit 15 (S2). The main control unit 13 refers to the SIP route distribution table 131 using the called party identification code (ABCD), selects the SIP server A route (S3), and transmits a call request to the SIP control unit 14A. (S4). The SIP control unit 14A generates a SIP message (INVITE) toward the SIP server A 71A, and transmits it to the SIP server 71A on the Internet 7 from the Internet side interface 11A via the internal switch control unit 15 (S5).

  The VoIP-GW 1 receives the SIP message connection completion (200K) received from the SIP server 71A via the Internet side interface 14A via the internal switch control unit 15 to the SIP control unit 14A (S6). The SIP control unit 14A transmits connection completion to the main control unit 13 (S7). The main control unit 13 sends a switching instruction to the internal switch control unit 15 (S8), and the internal switch control unit 15 sets the switching data of the RTP control 16A and the Internet side interface 11A (S9). At the same time, the main control unit 13 notifies the RTP control unit 16A of connection completion (S10), and transmits a connection completion signal via the PBX side interface 12A (S11).

  The PCM voice data is received from the PBX 3A side via the PBX side interface 12A (S12). The RTP control unit 16A converts the PCM voice data into RTP voice data (S13), and the RTP voice data is sent to the internal switch control unit 15. Via the Internet side interface 11A, it transmits to the SIP server 71A (S14).

  RTP audio data received from the SIP server 71A via the Internet-side interface 11A and the internal switch control unit 15 (S15) The RTP control unit 16A converts the RTP audio data into PCM audio data (S16), and converts the PCM audio data to PBX. The data is transmitted to the PBX 3A side via the side interface 12A (S17).

  In this way, when there is a connection request to the SIP server 71A from the PBX 3 side, both can be connected via the SIP control unit 14A of the VoIP-GW 1 and there is a connection request to the SIP server 71B. Similarly, both can be connected via the SIP control unit 14B.

  Similarly, when an incoming call is received from the Internet side, the incoming event and the called number are transferred from the SIP control unit 14 to the main control unit 13, and the main control unit 13 searches for an available PBX-side interface 12 and if the incoming call is possible. The main control unit 13 issues an incoming event to the PBX-side interface 12 and, similar to the outgoing call, the main control unit 13 issues an instruction to the internal switch control unit 15, the SIP control unit 14, and the RTP control unit 16, and the SIP call in the VoIP-GW 1 Control and switching of RTP communication.

  Next, channel management processing for each SIP server will be described with reference to FIG. In this example, the total channel on the PBX side is 60 Ch, and the channel usage mode of the SIP server unit is the condition described in the channel management table 132 of the SIP unit shown in FIG. 4, that is, the minimum guaranteed channel of the SIP server A The number “30”, the maximum number of communication channels “40”, the minimum guaranteed channel number “10” of the SIP server B and the maximum number of communication channels “20”, the minimum guaranteed channel number “10” of the SIP server C and the maximum number of communication channels “ It is assumed that the number of used channels of the SIP server A is “30 Ch”, the number of used channels of the SIP server B is “20 Ch”, and the number of used channels of the SIP server C is “5 Ch”.

  Now, the PBX side interface 12A that has received the outgoing signal of the called number (050-ABCD-XXXX) from the PBX 3A side sends the outgoing call from the PBX (called number: 050-ABCD-XXXX) via the internal switch control unit 15. To the main control unit 13 (S22). The main control unit 13 extracts the carrier number (ABCD) from the called party number from the PBX, selects the route of the SIP server A by referring to the SIP distribution table 131 using the carrier number, and sets the channel management table 132. It is determined whether or not the route to the SIP server A is usable by referring to it. In this case, if the connection of the route to the SIP server A is permitted, the minimum guaranteed channel number “10” to the SIP server C cannot be maintained. Therefore, it is determined to reject the received connection request (S23), and a disconnection instruction is sent to the PBX. The side interface 12A is notified (S24). The PBX-side interface 12A that has received the disconnection instruction transmits a disconnection signal to the PBX 3A (S25).

  Further, for example, even when there is a surplus in the number of channels used for the SIP server B and the SIP server C, when the connection request for the SIP server A exceeds the maximum number of communication channels “40”, the same applies. Then, the connection is rejected for the connection request directed to the SIP server A. In this way, the minimum guaranteed number of channels for each SIP server can be maintained.

  As described above, the main control unit 13 manages the minimum guaranteed channel and the maximum number of communication channels for each SIP server route shown in the channel management table 132 for each SIP server, and the QOS for each route toward each SIP server ( (Quality of Service) management is possible, and the PBX side interface channel can be used effectively.

  Next, the flow of processing when the incoming number from the Internet side is converted and transferred to another SIP server will be described with reference to FIG. When the SIP message (INVITE) directed to the called number (050-ABCD-1234) is received from the SIP server A 71A on the Internet 7 (S31), the Internet side interface 11A passes through the internal switch control unit 15 and the SIP control unit 14A. (S32). The SIP control unit 14A notifies the control unit 13 of an incoming call (S33). The main control unit 13 converts the called number 050-ABCD-1234 into 050-EFGH-1234 by referring to the called number conversion table 133 using the trader number (ABCD) in the called number (S34), and the SIP server. A transmission request for B is transmitted to the SIP control unit 14B (S35).

  The SIP control unit 14B transmits the SIP message (INVITE) of the number 050-EFGH-1234 via the internal switch control unit 15 from the Internet side interface 11B to the SIP server B 71B (S36). The SIP message of connection completion (200 OK) received from the SIP server B 71B received by the Internet side interface 11B is notified to the SIP control unit 14B via the internal switch control unit 15 (S37). The SIP control unit 14B notifies the main control unit 13 of the connection completion (S38). The main control unit 13 notifies the completion of connection to the SIP control unit 14A (S39).

  The SIP control unit 14A notifies the SIP server 71A of the connection completion (200 OK) from the Internet side interface 11A via the internal switch control unit 15 (S40). The main control unit 13 issues a switching instruction to the internal switch control unit 144 (S41), and the internal switch control unit 15 sets switching data between the Internet side interface 11A and the Internet side interface 11B (S42). When the internal switch is set in this way, RTP voice data from the SIP server 71A is transmitted to the SIP server 71B via the Internet side interface 11A, the internal switch control unit 15, and the Internet side interface 11B (S43). . Similarly, RTP voice data from the SIP server 71B is transmitted to the SIP server 71A via the Internet side interface 11B, the internal switch control unit 15, and the Internet side interface 11A (S44).

  As described above, in this embodiment, when receiving an incoming call from the Internet side, by referring to the called number conversion table 133, conversion (transfer) transmission to a different SIP server route is possible, and only a function as a VoIP-GW is possible. Instead, it is also possible to function as an ALG (application layer gateway) in communication between different SIP servers.

  In the above description, SIP is used as the communication protocol, but the same conversion is possible with other communication protocols including MGCP, H323, and the like.

The figure explaining the system configuration | structure at the time of the multi-ISP correspondence connection concerning this invention The figure explaining the structure of the VoIP-GE apparatus concerning this invention. The figure explaining the structure of the SIP route distribution table provided in the main control part of the VoIP-GW apparatus of this invention. The figure explaining the structure of the channel management table of the SIP server unit provided in the main control part of the VoIP-GW apparatus of this invention. The figure explaining the structure of the called number conversion table provided in the main control part of the VoIP-GW apparatus of this invention. The sequence diagram explaining the process of SIP route distribution in this invention. The sequence diagram explaining the process of the channel management per SIP server in this invention. The sequence diagram explaining the called number conversion process in this invention. The figure explaining the system configuration at the time of the conventional multi-ISP correspondence connection

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 ... VoIP-GW apparatus 11 ... Internet side interface 12 ... PBX side interface 13 ... Main control part 131 ... SIP allocation table 132 ... Channel management table 133 ... Called number conversion table 14 ... SIP control part 15 ... Internal switch control part 16 ... RTP control unit 3 ... PBX / PSTN
5 ... Media converter 51 ... Physical line 7 ... Internet 71 ... ISP SIP server 9 ... VoIP-GW device

Claims (3)

In a VoIP-GW (Voice Over Internet Protocol Gate Way) device that converts voice communication between a non-IP communication network and an IP communication network,
An interface on the IP communication network side;
A SIP control unit that controls communication with a SIP server installed on the IP communication network side;
A plurality of interfaces on the non-IP communication network side;
A plurality of RTP control units for controlling communication with the non-IP communication side;
A main control unit that controls the operation of the entire apparatus and has a SIP route distribution table that describes the correspondence between the called number on the IP communication network side and the SIP server;
An internal switch control unit for controlling an internal switch that switches the mutual connection between the interface on the IP communication network side, the SIP control unit, the interface on the non-IP communication network side, the RTP control unit, and the main control unit;
With
When there is a connection request to the SIP server from the non-IP communication network side, the route to the SIP server is determined by referring to the SIP route distribution table based on the called party identification number described in the connection request. VoIP-GW apparatus characterized by this. The VoIP-GW device according to claim 1,
The main control unit has a channel management table that describes the minimum number of guaranteed channels and the maximum number of communication channels to be used for each SIP server.
VoIP characterized in that, when there is a connection request to the SIP server, it is determined whether or not the SIP server can be received from the current usage status of the route of the SIP server, the minimum number of guaranteed channels, and the maximum number of communication channels. GW device. In a VoIP-GW (Voice Over Internet Protocol Gate Way) device that converts voice communication between a non-IP communication network and an IP communication network,
An interface on the IP communication network side;
A SIP control unit that controls communication with a SIP server installed on the IP communication network side;
A plurality of interfaces on the non-IP communication network side;
A plurality of RTP control units for controlling communication with the non-IP communication side;
A main control unit that controls the operation of the entire apparatus, and has a called number conversion table that describes the called number on the IP communication network side and the destination of the called number,
An internal switch control unit for controlling an internal switch that switches the mutual connection between the interface on the IP communication network side, the SIP control unit, the interface on the non-IP communication network side, the RTP control unit, and the main control unit;
With
When there is a connection request from the IP communication network side to the IP communication network side, a converted route is determined by referring to the called number conversion table based on the called party identification number described in the connection request. VoIP-GW apparatus characterized by this.

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