JP6226799B2 - IP-PBX system and sound source data providing method thereof - Google Patents

Description

  The present invention relates to an IP-PBX (Private Branch Exchange) system for connecting telephone terminals via a communication network via an IP (Internet Protocol) communication network and a method for providing sound source data thereof.

  Conventionally, in an IP-PBX system, an IP telephone terminal is connected to an IP communication network such as the Internet. In addition, the IP-PBX system connects a call control server and a line terminator to the IP communication network so that even an analog telephone terminal that does not support IP can be connected to the IP telephone terminal via the IP communication network. Composed. Furthermore, some IP telephone terminals use SIP messages using SIP (Session Initiation Protocol) as VoIP (Voice over Internet Protocol). The IP-PBX system is configured by connecting a SIP exchange device to an IP communication network in order to establish a call connection with such an IP telephone terminal that supports SIP.

  The call control server has an original protocol other than SIP, accommodates analog telephone terminals and IP extension telephone terminals using the original protocol, and makes, disconnects and holds calls related to these telephone terminals. Take control. This call control server is configured to be able to transmit and receive IP packets based on a unique protocol via an IP communication network.

  The SIP exchange device controls the call connection between an IP telephone terminal that supports SIP and another telephone terminal that does not support SIP. SIP messages used on the IP telephone terminal side and other telephone terminals The protocol with the IP packet used on the side is converted to enable transmission / reception between the two.

  One line terminator is connected to an analog telephone terminal and is controlled by a call control server to realize IP communication of the analog telephone terminal. This one line termination device manages channel resources for path connections with analog telephone terminals. Also, this one line terminating device converts analog call voice data used on the analog telephone terminal side into an RTP (Real-time Transport Protocol) packet so that it can be transmitted / received via the IP communication network.

  Other line terminators store sound source data such as music on hold, guidance, and DTMF (Dual-Tone Multi-Frequency) and are controlled by a call control server to realize the provision of sound source data. is there. Other line terminators manage channel resources for path connections for providing sound source data. In addition, the other line terminator converts the sound source data into an RTP packet so that it can be transmitted / received via the IP communication network.

  For example, in the telephone exchange system described in Patent Document 1, an IP-PBX is connected to a normal public telephone line network, so that a normal analog telephone connected to a public telephone line network is connected to a local IP telephone terminal or an external IP telephone terminal. Allows calls to and from the phone. A packet transfer apparatus connected to this IP-PBX via a local IP network has a music on hold sound source.

  For example, in the telephone system described in Patent Document 2, a call control server, a plurality of media servers, a plurality of IP telephone terminals, and a plurality of distribution servers are connected to an IP network. Then, the IP telephone terminal receives the hold tone packet from the media server.

JP 2005-72673 A JP 2011-199417 A

  In conventional IP-PBX systems, voice / IP conversion equipment for converting sound source data and channel resources for path connection in order to transmit and receive sound source data such as music on hold, guidance, and DTMF via the IP communication network It is necessary to provide a channel switch for managing the above-mentioned other line terminating device.

  Moreover, since a large-scale IP-PBX system accommodates a large number of telephone terminals and stores a large number of sound source data, it is necessary to provide a device such as a large-capacity server or database. For this reason, it is necessary to separately provide a line terminator for IP communication of an analog telephone terminal and a line terminator for providing sound source data, and each device must be provided with a voice / IP conversion device and a channel switch. . Therefore, in the IP-PBX system, the number of components such as voice / IP conversion devices and channel switches increases, so it is not possible to efficiently install the main components for call connection, and the cost is high. It will increase.

  If the line termination device for IP communication of analog telephone terminals and the line termination apparatus for providing sound source data are shared, path connection requests from many telephone terminals are congested and channel resources are insufficient. As a result, channel resources may not be properly secured. As a result, call loss occurs and smooth call connection and sound source data cannot be provided.

  SUMMARY OF THE INVENTION In view of such problems, the present invention provides an IP-PBX system and a sound source data providing method thereof that facilitate call connection and sound source data path connection without increasing the number and cost of components required for providing sound source data. The purpose is to provide.

  In order to solve the above-described problems, the present invention provides an IP-PBX system in which a non-SIP compatible telephone terminal and a SIP compatible IP telephone terminal are connected via an IP communication network. A call control server that accommodates SIP non-compliant telephone terminals, and a SIP converter that accommodates the IP telephone terminals and converts communication signals between the SIP non-compliant telephone terminals and the IP telephone terminals into SIP messages; The call control server detects call information of the non-SIP-compliant telephone terminal, and if the call information is a sound source reproduction call that causes the IP telephone terminal to reproduce sound source data, the other party telephone terminal of the call information A call control unit for determining whether or not the IP phone terminal is the IP phone terminal, and when the other party phone terminal is the IP phone terminal under the control of the call control unit, Replay request to this SIP converter for playback And a path control unit that transmits the sound, and the SIP conversion device includes a sound source storage unit that stores the sound source data reproduced by the IP telephone terminal in response to the call information of the SIP non-compliant telephone terminal. Features.

  In addition, the sound source data providing method of the IP-PBX system in which a non-SIP compatible telephone terminal that does not support SIP and an IP telephone terminal that supports SIP is connected via an IP communication network accommodates this IP telephone terminal. In the SIP converter that converts the communication signal between the SIP non-compliant telephone terminal and the IP telephone terminal into a SIP message, the IP telephone terminal reproduces the call according to the call information of the non-SIP telephone terminal. The sound information stored in the sound source storage unit, and the call control server that accommodates the SIP non-compliant telephone terminal detects the call information of the SIP non-compliant telephone terminal. In the case of a sound source reproduction call that causes the terminal to reproduce the sound source data, a call control step for determining whether or not the other party telephone terminal of the call information is the IP telephone terminal, and the call control step by the call control server As a result of determination by When the other party's telephone terminal is this IP telephone terminal, a path control process for transmitting a reproduction request to this SIP conversion apparatus to reproduce this sound source data on this IP telephone terminal, and this SIP conversion apparatus, A sound source data acquisition step of acquiring the sound source data to be reproduced by the IP telephone terminal from the sound source storage unit in accordance with the call information of the non-SIP compatible telephone terminal.

  According to the present invention, in the IP-PBX system, it is not necessary to provide a dedicated line terminator for providing sound source data such as music on hold, guidance, and DTMF, and a dedicated voice / IP converter and a dedicated channel switch are not required. Therefore, it is possible to reduce the equipment cost. In addition, in the IP-PBX system, a space for arranging hardware devices for providing sound source data is not required, so that other component devices can be efficiently arranged.

It is a block diagram which shows the schematic structure in one Example of the IP-PBX system which concerns on this invention. 2 is a timing chart showing an operation procedure of sound source data reproduction in the IP-PBX system of the embodiment shown in FIG. 10 is a timing chart showing an operation procedure of sound source data reproduction in another embodiment of the IP-PBX system according to the present invention.

  Next, an embodiment of an IP (Internet Protocol) -PBX (Private Branch Exchange) system according to the present invention will be described in detail with reference to the accompanying drawings. For example, as shown in FIG. 1, the IP-PBX system 10 of the present embodiment includes a call control server 12, a SIP (Session Initiation Protocol) converter 14, a first line terminator 16, and a second line. A terminal device 18 is connected via an IP communication network 20 such as the Internet. Connected to the IP communication network 20 is an IP telephone terminal 22 that uses SIP as VoIP (Voice over Internet Protocol). Hereinafter, a telephone terminal using SIP is referred to as an IP telephone terminal. In addition, an IP extension telephone terminal 24 that uses the unique protocol of the call control server 12 is connected to the IP communication network 20. An analog telephone terminal 26 not compatible with IP is connected to the first line termination device 16. Note that portions not directly related to understanding the present invention are not shown and redundant description is avoided.

  In FIG. 1, one IP telephone terminal 22, one IP extension telephone terminal 24, and one analog telephone terminal 26 are shown, but the IP-PBX system 10 includes a plurality of IP telephone terminals 22, a plurality of IP extension telephones. A terminal 24 and a plurality of analog telephone terminals 26 may be provided.

  In this embodiment, the IP-PBX system 10 establishes a call connection of two or more telephone terminals among the IP telephone terminal 22, the IP extension telephone terminal 24, and the analog telephone terminal 26. The IP-PBX system 10 can make a call connection with any combination of the IP telephone terminal 22, the IP extension telephone terminal 24, and the analog telephone terminal 26.

  The call control server 12 accommodates an IP extension telephone terminal 24 and an analog telephone terminal 26, and manages call information such as calling, disconnection and holding of these telephone terminals. For example, as shown in FIG. 1, the call control server 12 includes a call control unit 30, a terminal information storage unit 32, a line control unit 34, a path control unit 36, an IP leased line communication control unit 38, and an IP communication network communication control unit. 40 and an IP communication interface 42. The call control server 12 is connected to the IP communication network 20 via the IP communication interface 42. The call control server 12 has a unique protocol other than SIP, and an IP packet using the unique protocol (hereinafter simply referred to as an IP packet) can be transmitted / received via the IP communication network 20.

  The call control unit 30 is connected to the IP extension telephone terminal 24 or analog telephone terminal 26 and the IP telephone terminal 22 or other IP extension telephone terminal 24 or other analog telephone terminal 26, that is, in the call connection with the other party telephone terminal. Call information of the telephone terminal 24 or the analog telephone terminal 26 is managed. The call control unit 30 controls the line control unit 34, path control unit 36, IP leased line communication control unit 38, and IP communication network communication control unit 40 to execute call control. For example, when the call control unit 30 detects call information transmitted from the analog telephone terminal 26 or call information transmitted to the analog telephone terminal 26, the path control unit 30 processes the path connection according to the call information. The unit 36 is controlled.

  Further, the call control unit 30 is configured such that when the call information indicates a sound source reproduction call such as a hold for causing the IP telephone terminal 22 to reproduce sound source data, the other party telephone terminal to be held is the IP telephone terminal 22 or another IP extension telephone terminal. 24 or other analog telephone terminal 26 is determined. The call control unit 30 controls the path control unit 36 so that the SIP conversion device 14 is requested to reproduce the hold sound when the other party telephone terminal to be held is the IP phone terminal 22. On the other hand, when the other party telephone terminal to be held is another IP extension telephone terminal 24 or another analog telephone terminal 26, the call control unit 30 requests the second line terminating device 18 to reproduce the hold sound. In this way, the path control unit 36 is controlled.

  The terminal information storage unit 32 stores terminal information such as the telephone numbers of the IP extension telephone terminal 24 and the analog telephone terminal 26 accommodated in the call control server 12.

  The line control unit 34 controls communication on the analog telephone terminal 26 side when the analog telephone terminal 26 connected to the public switched telephone network (not shown) is connected to another telephone terminal. For example, the line control unit 34 transmits the analog signal including voice data such as voice, hold tone, guidance, and DTMF (Dual-Tone Multi-Frequency) to the analog telephone terminal 26 via the public switched telephone network, It is also possible to control transmission from the analog telephone terminal 26 via the public switched telephone network.

  The path control unit 36 controls a path connection for transmitting and receiving voice signals such as call voice, hold tone, guidance, and DTMF in a call connection between the IP extension telephone terminal 24 or the analog telephone terminal 26 and the other party telephone terminal. . The path control unit 36 of the present embodiment is controlled by the call control unit 30, and according to the call information related to the IP extension telephone terminal 24 or the analog telephone terminal 26, the IP extension telephone terminal 24 or the analog telephone terminal 26 and the other party telephone Controls establishment, disconnection, or hold of a path connection with a terminal.

  For example, the path control unit 36, when the call information indicates a call from the analog telephone terminal 26 to the IP telephone terminal 22, the terminal information of the source analog telephone terminal 26 and the terminal information of the destination IP telephone terminal 22 And a path connection signal including a path connection start request with the IP telephone terminal 22 is transmitted to the SIP converter 14.

  Further, the path control unit 36, when the call information indicates the hold from the analog telephone terminal 26 to the IP telephone terminal 22, the terminal information of the source analog telephone terminal 26, the terminal information of the destination IP telephone terminal 22, A path connection signal including a request for stopping the path connection with the IP telephone terminal 22 is transmitted to the SIP conversion device 14, and a path connection signal for requesting a music-on-hold connection is transmitted to the first line termination device 16. At this time, the path control unit 36 may designate one sound source data among a plurality of types of sound source data and transmit a path connection signal to the first line termination device 16.

  Further, the path control unit 36, when the call information indicates disconnection from the analog telephone terminal 26 to the IP telephone terminal 22, the terminal information of the originating analog telephone terminal 26, the terminal information of the destination IP telephone terminal 22, Then, a path connection signal including a request for disconnecting the path connection with the IP telephone terminal 22 is transmitted to the SIP conversion device 14.

  The IP leased line communication control unit 38 communicates with the IP extension phone terminal 24 when the IP extension phone terminal 24 connected to the IP extension or the IP leased line (not shown) is connected to another phone terminal. To control. For example, the IP leased line communication control unit 38 uses an IP extension or an IP leased line for IP packets including voice data such as call voice, music on hold, guidance, and DTMF, that is, RTP (Real-time Transport Protocol) packets. It is possible to control transmission to the IP extension telephone terminal 24 and transmission from the IP extension telephone terminal 24 via the IP extension line or the IP dedicated line.

  The IP communication network communication control unit 40 controls communication on the IP telephone terminal 22 side when the IP extension telephone terminal 24 or the analog telephone terminal 26 is connected to the IP telephone terminal 22 connected to the IP communication network 20. To do. For example, the IP communication network communication control unit 40 transmits the RTP packet as described above from the IP extension telephone terminal 24 or the analog telephone terminal 26 to the IP telephone terminal 22 via the IP communication network 20, and the IP telephone terminal 22 To the IP extension telephone terminal 24 or the analog telephone terminal 26 via the IP communication network 20 can be controlled.

  The SIP conversion device 14 accommodates an IP telephone terminal 22 that uses SIP. The IP telephone terminal 22 is connected to another IP telephone terminal 22 or an IP extension telephone terminal 24 or an analog telephone terminal 26, that is, a partner telephone terminal. The call connection is controlled, and when the other party's telephone terminal uses an IP packet, the protocol of the IP packet and the SIP message used in the IP telephone terminal 22 is converted. For example, the SIP conversion device 14 includes a call control unit 50, a sound source storage unit 52, a call control server side IF unit 54, a voice control unit 56, a SIP communication control unit 58, and an IP communication interface 60, as shown in FIG. Consists of. The SIP conversion device 14 is connected to the IP communication network 20 via the IP communication interface 60.

  The call control unit 50 manages call information of the IP telephone terminal 22, and controls the call control server IF unit 54, the voice control unit 56, and the SIP communication control unit 58. For example, when the call control unit 50 receives a path connection signal from the path control unit 36 of the call control server 12, whether or not the IP telephone terminal 22 indicated by the terminal information of this path connection signal is accommodated in the SIP converter 14 or not. Search for. Further, the call control unit 50 controls the voice control unit 56 in response to the start request, stop request, or disconnection request indicated by the path connection signal regarding the path connection with the searched IP telephone terminal 22.

  The sound source storage unit 52 stores sound source data such as music on hold, guidance, and DTMF, or sound source element data that can generate these sound source data. The sound source storage unit 52 may store multiple types of music on hold, multiple types of guidance, and multiple types of DTMF, and can store multiple types of sound source element data to generate multiple types of sound source data. It may be. For example, in the sound source storage unit 52, the holding sound and the guidance are G.I. 711 format or G. 729 format, DTMF is an uncompressed G.264 format. 711 format.

  The call control server side IF unit 54 controls communication performed from the IP telephone terminal 22 to the call control server 12 based on the unique protocol of the call control server 12. For example, the call control server-side IF unit 54 transmits / receives an IP packet based on the unique protocol of the call control server 12 to / from the call control server 12.

  The voice control unit 56 once terminates an RTP packet including voice data transmitted / received between the IP telephone terminal 22 and the IP extension telephone terminal 24 or the analog telephone terminal 26. Then, the voice control unit 56 rewrites and rewrites the header information such as the IP address header and RTP header of the RTP packet between the SIP and the original protocol of the call control server 12 according to the destination telephone terminal. Transfer RTP packets to the destination telephone terminal. In addition, the voice control unit 56 opens and closes the RTP port of the IP telephone terminal 22 in response to a start request, a stop request, or a disconnection request indicated by the path connection signal from the path control unit 36 of the call control server 12. Thus, the path connection on the IP telephone terminal 22 side is controlled.

  The SIP communication control unit 58 performs protocol conversion between the SIP message used in the IP telephone terminal 22 and the IP packet or RTP packet used in the other party telephone terminal under Web maintenance. For example, the SIP communication control unit 58 converts an IP packet or RTP packet transmitted from the analog telephone terminal 26 to the IP telephone terminal 22 into a SIP message, and is transmitted from the IP telephone terminal 22 to the analog telephone terminal 26. Convert SIP messages to IP packets or RTP packets.

  The first line terminating device 16 implements IP communication via the IP communication network 20 of the connected analog telephone terminal 26. For example, the first line termination device 16 includes a line package 70, a voice / IP conversion device 72, a channel switch 74, and an IP communication interface 76, as shown in FIG. The first line termination device 16 is connected to the IP communication network 20 via the IP communication interface 76.

  The line package 70 accommodates a line of the public switched telephone network, and the analog telephone terminal 26 is connected to the line package 70. For example, the line package 70 receives a call voice analog signal transmitted from the analog telephone terminal 26 (hereinafter referred to as an analog call signal) and transmits it to the voice / IP conversion device 72. Further, the line package 70 transmits the analog call signal to the analog telephone terminal 26 converted by the voice / IP conversion device 72 to the analog telephone terminal 26.

  The voice / IP conversion device 72 converts an analog call signal used in the analog telephone terminal 26 and an RTP packet based on the unique protocol of the call control server 12. The voice / IP conversion device 72 is configured to be mounted on, for example, a digital signal processor (DSP). For example, the voice / IP conversion device 72 receives an analog call signal from the analog telephone terminal 26 via the line package 70 and converts the analog signal into an RTP packet. The voice / IP conversion device 72 receives the RTP packet for the analog telephone terminal 26 via the IP communication interface 76, and converts the RTP packet into an analog call signal.

  The channel switch 74 controls the line occupation status of the line package 70, that is, the channel resource. For example, when the channel switch 74 receives a path connection instruction from the path control unit 103 of the call control server 12 via the IP communication interface 76, the channel switch 74 switches the channel resource of the circuit package 70 in accordance with the path connection instruction.

  When the IP extension telephone terminal 24 or the analog telephone terminal 26 makes a call connection with another IP extension telephone terminal 24 or another analog telephone terminal 26, the second line termination device 18 holds the IP extension telephone terminal 24 or The provision of sound source data to the analog telephone terminal 26 is realized. For example, as shown in FIG. 1, the second line termination device 18 includes a sound source storage unit 80, a voice / IP conversion device 82, a channel switch 84, and an IP communication interface 86. The second line terminating device 18 is connected to the IP communication network 20 via the IP communication interface 86.

  The sound source storage unit 80 stores sound source data such as music on hold, guidance, and DTMF, or sound source element data that can generate these sound source data. The sound source storage unit 80 may store multiple types of music on hold, multiple types of guidance, and multiple types of DTMF, and can store multiple types of sound source element data to generate multiple types of sound source data. It may be.

  The voice / IP conversion device 82 converts the analog sound source data stored in the sound source storage unit 80 and the RTP packet based on the unique protocol of the call control server 12. The voice / IP converter 82 is mounted on, for example, a digital signal processor (DSP). For example, when the voice / IP conversion device 82 receives a hold tone reproduction request from the path control unit 36 of the call control server 12 by designating a predetermined destination telephone terminal, the voice / IP conversion device 82 receives from the sound source storage unit 80 in response to the reproduction request. Acquires analog sound source data and converts it into RTP packets. Also, the voice / IP conversion device 82 transmits the converted RTP packet of the sound source data to the destination telephone terminal designated by the path control unit 36 via the IP communication interface 76.

  The channel switch 84 controls channel resources of a connection path that transmits sound source data. For example, when the channel switch 84 receives a hold tone playback request specifying a destination telephone terminal from the path control unit 36 of the call control server 12, the channel switch 84 responds to the playback request with a path to the specified destination telephone terminal. Reserve channel resources for the connection.

  Next, in the IP-PBX system 10 of the present embodiment, an operation example when a hold operation is performed at the analog telephone terminal 26 connected to the IP telephone terminal 22 will be described with reference to the sequence chart of FIG. To do.

  In the initial state of this operation example, a path connection between the IP telephone terminal 22 and the analog telephone terminal 26 is established (step S102). In the channel switch 74 of the first line termination device 16, the channel resource of the line package 70 is occupied for path connection with the IP telephone terminal 22. Therefore, it is possible to transmit and receive call voice between the IP telephone terminal 22 and the analog telephone terminal 26.

  For example, in the call voice transmission process from the analog telephone terminal 26 to the IP telephone terminal 22, first, an analog call signal transmitted from the analog telephone terminal 26 is transmitted to the first line terminating device 16 (step S104). ).

  The analog call signal is converted into an RTP packet based on the original protocol of the call control server 12 by the voice / IP conversion device 72 of the first line termination device 16, and this RTP packet is converted to the SIP via the IP communication network 20. The data is transmitted to the conversion device 14 (step S106).

  The RTP packet is converted into a SIP message by the SIP communication control unit 58 of the SIP conversion device 14, and the SIP message is transmitted to the IP telephone terminal 22 (step S108). The IP telephone terminal 22 can acquire the call voice from the analog telephone terminal 26 by this SIP message.

  The call voice transmission process from the IP telephone terminal 22 to the analog telephone terminal 26 is performed by a processing procedure opposite to the call voice transmission process from the analog telephone terminal 26 to the IP telephone terminal 22 as described above.

  When the user performs a hold operation on the analog telephone terminal 26 in the above-described call connection state between the IP telephone terminal 22 and the analog telephone terminal 26 (step S110), the analog telephone terminal 26 generates a hold notification signal and generates a first signal. Is transmitted to the call control server 12 via the line terminating device 16 (step S112).

  In the call control server 12, the hold notification signal from the analog telephone terminal 26 is detected by the call control unit 30. The call control unit 30 determines whether the other party telephone terminal to be held in the hold notification signal is the IP telephone terminal 22, the other IP extension telephone terminal 24, or the other analog telephone terminal 26. Here, since the other party telephone terminal is the IP telephone terminal 22, the call control unit 30 issues a hold instruction to the path control unit 36 so as to request the SIP conversion device 14 to play the hold sound (step). S114).

  In response to the hold instruction, the path control unit 36 generates a transmission stop signal for instructing the first line termination device 16 to stop transmission of the RTP packet, and transmits the signal to the first line termination device 16 (step). S116).

  The first line termination device 16 stops the transmission of the RTP packet to the IP telephone terminal 22 in response to the transmission stop signal from the path control unit 36 of the call control server 12 (step S118). The stop of RTP packet transmission may be realized by the voice / IP converter 72 or the channel switch 74. In the first line termination device 16, a response signal indicating stop of RTP packet transmission may be transmitted to the call control server 12.

  Further, in the call control server 12, when the RTP packet transmission in the first line termination device 16 is stopped, the path control unit 36 indicates a path indicating a request for stopping the path connection between the analog telephone terminal 26 and the IP telephone terminal 22. A connection signal, that is, a path connection signal for instructing closing of the RTP port of the IP telephone terminal 22 is generated and transmitted to the SIP conversion device 14 via the IP communication interface 42 (step S120).

  In the SIP converter 14, the call control unit 50 detects a path connection signal indicating a path connection stop request as described above. In the call control unit 50, for example, based on terminal information such as a telephone number indicated by this path connection signal, it is searched whether or not the IP telephone terminal 22 corresponding to this terminal information is accommodated in the SIP conversion device 14. The When such an IP telephone terminal 22 is searched, the voice control unit 56 closes the RTP port of the IP telephone terminal 22, that is, the path connection between the analog telephone terminal 26 and the IP telephone terminal 22 is stopped. (Step S122). Further, in the SIP conversion device 14, a response signal indicating the stop of the path connection may be transmitted to the call control server 12.

  Further, when the path connection on the SIP conversion device 14 side is stopped as described above, the call control server 12 generates a transmission stop signal that instructs the first line termination device 16 to perform the hold connection, and the first line It is transmitted to the terminating device 16 (step S124).

  In the first line termination device 16, the channel resource of the connection path between the analog telephone terminal 26 and the IP telephone terminal 22 is put on hold by the channel switch 74 (step S126).

  Further, in the call control server 12, when the path connection on the SIP conversion device 14 side is stopped as described above, the path control unit 36 instructs the IP telephone terminal 22 to reproduce the hold tone by using the hold tone playback request signal. Is generated and transmitted to the SIP converter 14 via the IP communication interface 42 (step S128). The hold sound reproduction request signal includes terminal information of the IP telephone terminal 22, and may include sound source information such as the type of hold sound and the sound level.

  In the SIP conversion device 14, the hold tone request signal is detected by the call control unit 50. The call control unit 50 searches for the IP telephone terminal 22 based on the terminal information indicated by the hold tone request signal. Further, the call control unit 50 searches the sound source storage unit 52 for a hold sound to be played based on the sound source information indicated by the hold sound request signal (step S130).

  Then, a SIP message including the searched hold tone is generated by the voice control unit 56, and this SIP message is transmitted to the searched IP telephone terminal 22 via the IP communication interface 76 (step). S132).

  When the IP telephone terminal 22 receives the SIP message including the hold tone from the SIP conversion device 14, the hold tone is reproduced (step S134).

  Next, in the IP-PBX system 10 of the present embodiment, an operation example when a hold release operation is performed at the analog telephone terminal 26 will be described.

  First, the path connection between the IP telephone terminal 22 and the analog telephone terminal 26 is stopped by the hold operation at the analog telephone terminal 26 as described above. In such a case, when the user performs a hold release operation on the analog telephone terminal 26, the analog telephone terminal 26 generates a hold release notification signal and transmits it to the first line terminating device 16.

  In the first line termination device 16, the hold release notification signal from the analog telephone terminal 26 is received by the line package 70 and further transmitted to the call control server 12.

  In the call control server 12, the hold release notification signal from the analog telephone terminal 26 is detected by the call control unit 30. The call control unit 30 issues a hold release instruction to the path control unit 36 based on the hold release notification signal.

  In response to the hold release instruction, the path control unit 36 generates a hold sound stop signal instructing to stop transmission of the hold sound to the IP telephone terminal 22, and transmits it to the SIP converter 14 via the IP communication interface 42. Is done. The music on hold stop signal includes the terminal information of the IP telephone terminal 22.

  In the SIP conversion device 14, the hold tone stop signal is detected by the call control unit 50. The call control unit 50 searches for the IP telephone terminal 22 based on the hold tone stop signal. Then, the voice control unit 56 stops the transmission of the SIP message including the hold sound to the IP telephone terminal 22. In addition, the SIP conversion device 14 may transmit a response signal indicating the stop of the hold sound to the call control server 12.

  Further, in the call control server 12, when the holding tone transmission in the SIP conversion device 14 is stopped, the path control unit 36 causes a path connection signal indicating a path connection start request between the analog telephone terminal 26 and the IP telephone terminal 22, That is, a path connection signal instructing opening of the RTP port of the IP telephone terminal 22 is generated and transmitted to the SIP converter 14 via the IP communication interface 42.

  In the SIP converter 14, the call control unit 50 detects a path connection signal indicating a path connection start request as described above. In the call control unit 50, for example, based on terminal information such as a telephone number indicated by this path connection signal, it is searched whether or not the IP telephone terminal 22 corresponding to this terminal information is accommodated in the SIP conversion device 14. The When such an IP telephone terminal 22 is searched, the voice control unit 56 opens the RTP port of the IP telephone terminal 22, that is, the path connection between the analog telephone terminal 26 and the IP telephone terminal 22 is started. Is done. In addition, the SIP conversion device 14 may transmit a response signal indicating the start of this path connection to the call control server 12.

  Further, in the call control server 12, when the path connection on the SIP conversion device 14 side is started as described above, the path control unit 36 instructs the first line termination device 16 to start transmission of the RTP packet. A signal is generated and transmitted to the first line terminator 16.

  The first line termination device 16 starts transmission of an RTP packet to the IP telephone terminal 22 in response to a transmission start signal from the path control unit 36 of the call control server 12. The start of RTP packet transmission may be realized by the voice / IP converter 72 or the channel switch 74. In the first line termination device 16, a response signal indicating the start of RTP packet transmission may be transmitted to the call control server 12.

  In this manner, a path connection between the analog telephone terminal 26 and the IP telephone terminal 22 is established by the hold release operation at the analog telephone terminal 26, and a call can be established between the two.

  In this embodiment, the operation example in which the sound source information such as the holding tone is specified by the path control unit 36 of the call control server 12 has been described. However, the specification of the sound source information is performed by the IP communication network communication control unit 40 of the call control server 12. It may be configured to be performed by.

  As described above, the IP-PBX system 10 of this embodiment is compatible with the SIP and the IP extension telephone terminal 24 or the analog telephone terminal 26 that does not support SIP, that is, the call control server 12 that accommodates the SIP non-compliant telephone terminal. And a SIP conversion device 14 that converts a communication signal between the non-SIP compatible telephone terminal and the IP telephone terminal 22 into a SIP message.

  When the call control server 12 detects call information of a non-SIP compatible telephone terminal and the sound source data is to be reproduced by the IP telephone terminal 22 in accordance with the call information, the other party telephone terminal of the call information is the IP telephone terminal 22 A call control unit 30 for determining whether or not the call is received by the call control unit 30 and when the other party's telephone terminal is the IP telephone terminal 22, And a path control unit 36 for the SIP conversion device 14.

  In addition, the SIP conversion device 14 includes a sound source storage unit 52 that stores sound source data to be played back by the IP telephone terminal 22 in accordance with call information of a telephone terminal that does not support SIP.

  As a result, the IP-PBX system 10 does not need a dedicated line termination device to provide sound source data such as music on hold, guidance, and DTMF, but implements a dedicated voice / IP converter and dedicated channel switch. It is not necessary to do so, so that the equipment cost can be reduced. In addition, in the IP-PBX system 10, a space for arranging hardware devices for providing sound source data is not required, so that other component devices can be efficiently arranged.

  As described above, in the IP-PBX system 10, the configuration for providing sound source data such as music on hold, guidance, and DTMF is independent of the first line terminator 16 used for the VoIP call. 14 is provided. Therefore, in the channel switch 74 of the first line termination device 16, shortage of channel resources is suppressed, channel resources are easily secured, and occurrence of call loss can be suppressed.

  In another embodiment, the IP-PBX system 10 of the present invention includes a specific destination database (specific destination storage unit) 90 that stores terminal information such as a telephone number of a specific destination such as a customer.

  The specific destination database 90 is associated with the telephone number of the analog telephone terminal 26 of the predetermined user, and the sound source information of the sound source data such as the hold tone desired to be provided to the specific destination Remember. The sound source data desired to be provided to a specific destination includes, for example, a hold sound for commercial or other advertising purposes. The specific destination database 90 can also store the telephone numbers and sound source information of specific destinations for each of a plurality of users, and can also store the phone numbers and sound source information of a plurality of specific destinations for one user. You can also.

  In the IP-PBX system 10 of this other embodiment, the call control server 12 can issue a hold instruction using the sound source information stored in the specific destination database 90.

  The call control unit 30 of the call control server 12 acquires the telephone number of the source analog telephone terminal 26 and the telephone number of the destination IP telephone terminal 22 indicated in the hold notification signal from the analog telephone terminal 26. . Further, the call control unit 30 refers to the specific partner database 90 based on the caller telephone number and the callee telephone number, and acquires the sound source information of the sound source data to be provided to the callee. The call control unit 30 designates the sound source information acquired from the specific partner database 90 and instructs the path control unit 36 to hold.

  When the sound source information is designated by the hold instruction from the call control unit 30, the path control unit 36 designates this sound source information, generates a hold sound request signal, and transmits it to the SIP conversion device 14.

  Next, in the IP-PBX system 10 of this other embodiment, an operation example when a hold operation is performed at the analog telephone terminal 26 connected to the IP telephone terminal 22 is referred to the sequence chart of FIG. While explaining.

  In this other embodiment, the operations from the path connection establishment (step S102) to the hold notification (step S112) are the same as those in the above-described embodiment, and the description thereof is omitted.

  When the call control unit 30 of the call control server 12 detects a hold notification signal from the analog telephone terminal 26, the call control unit 30 stores the destination telephone terminal indicated by the hold notification signal in the specific destination database 90. It is determined whether or not it has been done. When the destination telephone terminal is stored in the specific destination database 90, the call control unit 30 designates terminal information such as the telephone number of the destination telephone terminal and stores the sound source information in the specific destination database 90. An acquisition request is made (step S140).

  The specific destination database 90 acquires sound source information based on the terminal information of the destination telephone terminal (step S142) and transmits it to the call control server 12 (step S144).

  When the call control unit 30 of the call control server 12 acquires the sound source information from the specific destination database 90, the call control unit 30 designates the sound source information and gives a hold instruction to the path control unit 36 (step S146).

  Since the operation from the transmission stop instruction (step S116) to the pending connection (step S126) in the path control unit 36 is the same as that in the above-described embodiment, the description thereof is omitted.

  In the path control unit 36, when the path connection on the SIP conversion device 14 side is stopped and the hold connection of the first line termination device 16 is performed, the sound source information acquired from the specific destination database 90 is designated and the IP telephone terminal A holding tone reproduction request signal for instructing reproduction of the holding tone to 22 is generated and transmitted to the SIP converter 14 via the IP communication interface 42 (step S148).

  When the hold control request signal is detected, the call control unit 50 of the SIP conversion device 14 desires to provide the hold sound to be reproduced based on the sound source information indicated by the hold sound request signal, that is, to the destination telephone terminal. The hold tone is retrieved from the sound source storage unit 52 (step S150).

  Since the operation from the holding sound transmission (step S132) to the holding sound reproduction (step S134) in the SIP converter 14 is the same as that in the above-described embodiment, the description thereof is omitted.

  According to another embodiment, it is possible to specify a destination of a call and store desired sound source data in the specified destination database 90, and to provide sound source data corresponding to the destination. For example, when the user's destination is a customer, the commercial or advertisement including the user's company information or product information is stored in the specific destination database 90 as sound source data such as music on hold or guidance, and this commercial is stored. And advertisements can be provided to the other party, the media for advertisements and promotions can be diversified, and the opportunities for advertisements and promotions can be increased.

  In the above-described embodiment, the case where the call connection between the IP phone terminal 22 and the analog phone terminal 26 is suspended is illustrated, but the IP-PBX system 10 of the present invention is not limited to this, and the IP phone terminal 22, The present invention can also be applied to a case where a call connection with any combination of the IP extension telephone terminal 24 and the analog telephone terminal 26 is put on hold.

10 IP-PBX system
12 Call control server
14 SIP converter
16 First line terminator
18 Second line terminator
20 IP communication network
22 IP phone terminals
24 IP extension telephone terminals
26 Analog telephone terminals
30 Call controller
36 Path controller
52 Sound source memory
90 Specific destination database

Claims (9)

In an IP-PBX system in which a non-SIP compatible telephone terminal that does not support SIP (Session Initiation Protocol) and an IP (Internet Protocol) telephone terminal that supports SIP are connected via an IP communication network, the system includes:
A call control server that accommodates the SIP non-compliant telephone terminals;
A SIP converter that accommodates the IP telephone terminal and converts a communication signal between the non-SIP-compliant telephone terminal and the IP telephone terminal into a SIP message;
The call control server detects call information of the non-SIP compatible telephone terminal, and when the call information is a sound source reproduction call that causes the IP telephone terminal to reproduce sound source data, the other party telephone terminal of the call information A call control unit for determining whether the terminal is a telephone terminal;
Path control that is controlled by the call control unit to transmit a reproduction request to the SIP conversion device in order to reproduce the sound source data on the IP telephone terminal when the counterpart telephone terminal is the IP telephone terminal With
The SIP converter includes an audio source storage unit that stores the audio source data reproduced by the IP telephone terminal in accordance with the call information of the SIP non-compliant telephone terminal. The IP-PBX system according to claim 1, wherein the system is:
Further comprising a line terminator storing other sound source data to be played by the non-SIP compatible telephone terminal,
The path control unit makes a request for reproduction of the other sound source data to be reproduced by the non-SIP compatible telephone terminal to the line terminating device when the counterpart telephone terminal is the non-SIP compatible telephone terminal. IP-PBX system characterized by In the IP-PBX system according to claim 1,
The IP-PBX system, wherein the non-SIP compatible telephone terminal is an analog telephone terminal connected to a public switched telephone network.   2. The IP-PBX system according to claim 1, wherein the sound source storage unit stores, as the sound source data, a plurality of types of hold sounds that are played back on the IP telephone terminal in response to a hold operation on the SIP non-compliant telephone terminal. IP-PBX system characterized by memorizing.   2. The IP-PBX system according to claim 1, wherein the sound source storage unit stores, as the sound source data, a plurality of types of guidance voices that are reproduced on the IP telephone terminal in response to an operation on the SIP non-compliant telephone terminal. An IP-PBX system characterized by   2. The IP-PBX system according to claim 1, wherein the sound source storage unit stores, as the sound source data, a plurality of types of DTMFs (Dual-Tone Multi-Frequency) corresponding to operations at the non-SIP compatible telephone terminal. IP-PBX system characterized by this. In the IP-PBX system according to claim 1,
The sound source storage unit stores a plurality of types of sound source data,
The call control server stores a plurality of sound source information for identifying the plurality of types of sound source data,
The path control unit specifies one sound source information among the plurality of sound source information and transmits the reproduction request to the SIP conversion device,
The SIP conversion apparatus acquires the sound source data corresponding to the sound source information specified by the reproduction request from the path control unit from the sound source storage unit, and sends a SIP message including the sound source data to the IP telephone terminal. An IP-PBX system characterized by transmitting. The IP-PBX system according to claim 7, wherein the system stores specific partner terminal information in association with the sound source information of one sound source data among the plurality of types of sound source data. Further comprising
The call control unit determines whether or not the terminal information of the counterparty telephone terminal is stored in the specific destination storage unit when the call information of the SIP non-compliant telephone terminal is the sound source reproduction call If the terminal information is stored, the sound source information associated with the terminal information is acquired from the specific partner storage unit,
The IP-PBX system, wherein the path control unit specifies the sound source information acquired by the call control unit from the specific destination storage unit and transmits the reproduction request to the SIP conversion device. In a sound source data providing method of an IP-PBX system in which a non-SIP compatible telephone terminal that does not support SIP and an IP telephone terminal that supports SIP are connected via an IP communication network, the method includes:
In a SIP converter that accommodates the IP telephone terminal and converts a communication signal between the SIP non-compliant telephone terminal and the IP telephone terminal into a SIP message, according to call information of the non-SIP-compliant telephone terminal A storage step of storing sound source data reproduced by the IP telephone terminal in a sound source storage unit;
When the call control server that accommodates the SIP non-compliant telephone terminal detects the call information of the SIP non-compliant telephone terminal, and the call information is a sound source reproduction call that causes the IP telephone terminal to reproduce the sound source data, A call control step of determining whether the other party telephone terminal of the call information is the IP telephone terminal;
As a result of the determination by the call control step by the call control server, when the counterpart telephone terminal is the IP telephone terminal, a reproduction request is transmitted to the SIP conversion apparatus in order to reproduce the sound source data on the IP telephone terminal. A path control process to transmit to
Including a sound source data acquisition step of acquiring, from the sound source storage unit, the sound source data to be reproduced by the IP telephone terminal according to the call information of the SIP non-compliant telephone terminal by the SIP converter. To provide sound source data.

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