JP4777445B2 - VoIP adapter, communication system, and communication method - Google Patents

Description

  The present invention relates to a VoIP adapter for performing communication via an IP network using a fixed telephone, a communication system including the VoIP adapter, and a communication method using the VoIP adapter.

  Due to an increase in demand for data communication, VoIP (Voice over Internet Protocol) service / Internet telephone which performs voice communication via an IP (Internet Protocol) network (hereinafter also referred to as “IPNW”) has become widespread. In particular, in recent years, an outgoing / incoming service via an IPNW with a local telephone number is also provided, and the transition from an existing public switched telephone network (PSTN) to an IPNW is being promoted. In accordance with such a flow, a call control protocol called SIP (Session Initiation Protocol) has become widespread in order to realize a VoIP service with an IPNW.

However, in order to shift from the public telephone network to the IPNW, the user has to replace the conventional telephone for public telephone lines that does not support the VoIP service with a terminal that supports the VoIP service.
In order to solve such a problem, a VoIP (Voice over Internet Protocol) adapter has been developed as a device that makes it possible to use an IPNW by using a telephone for a public telephone line that does not support the VoIP service. (For example, refer to Patent Document 1). By connecting the VoIP adapter to a telephone for a public telephone line, the user can use the IPNW without purchasing a new terminal.

  However, the development of the above VoIP adapter supports limited services that can be used with conventional telephones for public telephone lines, such as converting RTP (Real-time Transport Protocol) to analog and notifying the called party number. In particular, various services using SIP specified by IETF (Internet Engineering Task Force) and 3GPP (3rd Generation Partnership Project) (for example, on-hold transfer, absence, etc.) even if the VoIP adapter is used. I couldn't use tele-transfer, three-party calls, etc.)

For example, Patent Documents 2 and 3 disclose technologies that enable IPNW to be used by non-SIP terminals that do not support SIP.
Patent Document 2 describes that telephones can communicate with each other via an IP network. When an Internet telephone gateway described in Patent Document 2 receives a DTMF (Dual Tone Multi Frequency) signal from a telephone via a public telephone network, the Internet telephone gateway converts the DTMF signal into a digit and includes it in an Internet telephone protocol signal special message. The digit is packaged and transferred via the IP network to the public telephone line network of the other telephone.

  Patent Document 3 discloses a change in contact information by a power ON / OFF switch or a button of a non-SIP terminal (USB device, electronic camera, personal computer, telephone other than an IP telephone, etc.), a non-SIP terminal and a SIP converter. It is described that the SIP converter device establishes or disconnects a SIP session on behalf of a non-SIP terminal when a USB cable connection trigger occurs.

JP 2006-25102 A Special table 2003-502907 JP 2008-311905 A

  The technique described in Patent Document 2 is a technique for transferring a DTMF signal defined by a public telephone line network to the public telephone line network of the other party telephone via the IPNW. Therefore, the signal transmitted from the telephone connected to the public telephone line network is only the signal defined by the public telephone line network, and the service that the telephone can receive is only the public telephone service. Therefore, even if the technique described in Patent Document 2 is used, various services using SIP cannot be used.

Patent Document 3 describes that a session is established or disconnected based on various triggers from a non-SIP terminal, but it is not considered to perform transfer or hold after a session is established. . Therefore, even if the technique described in Patent Document 3 is used, various services using SIP other than transmission and disconnection cannot be used.
The present invention has been made to solve the above-mentioned problems, and a VoIP adapter, a communication system, and a communication method that can use various network services using SIP using a non-SIP terminal. The purpose is to provide.

In order to solve the above problem, the VoIP adapter according to claim 1 is connected to a non-SIP terminal that does not support SIP (Session Initiation Protocol) and can transmit a DTMF (Dual Tone Multi Frequency) signal. A VoIP adapter that receives a DTMF signal for requesting a predetermined SIP service from the non-SIP terminal; a DTMF recognition unit that recognizes a DTMF signal received by the signal transmission / reception unit; Conversion means for converting a DTMF signal recognized by the DTMF recognition means into a SIP message for realizing the predetermined SIP service according to a predetermined conversion rule, and a SIP message generated by the conversion by the conversion means Is transmitted via an IP (Internet Protocol) network. Comprising a sage transmission means, the, wherein the predetermined SIP service to a voice mail transfer, a melody hold for outputting a hold sound stored in the external server while on hold, characterized in that at least one is contained in .

According to the present invention, the VoIP adapter realizes the predetermined SIP service by converting the DTMF signal received from the non-SIP terminal for requesting the predetermined SIP service according to a predetermined conversion rule. SIP message is generated and transmitted through the IP network. Therefore, by defining a SIP service corresponding to each DTMF signal that can be output from a non-SIP terminal, and defining a conversion rule for converting the DTMF signal into a SIP message for realizing the SIP service, It is possible to use various network services realized by SIP using a non-SIP terminal.
In particular, the predetermined SIP service includes at least one of answering machine transfer and melody hold that outputs a hold sound stored in an external server while the user is on hold. Can be used.

The VoIP adapter according to claim 2 is the VoIP adapter according to claim 1, wherein the DTMF signal for requesting the predetermined SIP service is transmitted from the non-SIP terminal by a user operation on the non-SIP terminal. It is characterized by that.
According to the present invention, a user can use a SIP service desired by the user by performing an operation for outputting a DTMF signal corresponding to the SIP service desired by the user using a non-SIP terminal. It becomes.

The communication system according to claim 3 is a non-SIP terminal that does not support SIP (Session Initiation Protocol) and can transmit a DTMF (Dual Tone Multi Frequency) signal; and a VoIP adapter connected to the non-SIP terminal; The VoIP adapter recognizes the DTMF signal received by the signal transmitting / receiving means and the signal transmitting / receiving means for receiving a DTMF signal for requesting a predetermined SIP service from the non-SIP terminal. DTMF recognizing means, a converting means for converting a DTMF signal recognized by the DTMF recognizing means into a SIP message for realizing the predetermined SIP service according to a predetermined conversion rule, and conversion by the converting means SIP messages generated by the Internet Protocol (IP) Comprising a SIP message transmission means for transmitting via a click, and wherein the predetermined SIP services, and voice mail transfer, a melody hold for outputting a hold sound stored in the external server to hold, at least one of It is included .

According to the present invention, the VoIP adapter realizes the predetermined SIP service by converting the DTMF signal received from the non-SIP terminal for requesting the predetermined SIP service according to a predetermined conversion rule. SIP message is generated and transmitted through the IP network. Therefore, by defining a SIP service corresponding to each DTMF signal that can be output from a non-SIP terminal, and defining a conversion rule for converting the DTMF signal into a SIP message for realizing the SIP service, It is possible to use various network services realized by SIP using a non-SIP terminal.
In particular, the predetermined SIP service includes at least one of answering machine transfer and melody hold that outputs a hold sound stored in an external server while the user is on hold. Can be used.

The communication method according to claim 4 is a non-SIP terminal that does not support SIP (Session Initiation Protocol) and can transmit a DTMF (Dual Tone Multi Frequency) signal; and a VoIP adapter connected to the non-SIP terminal; A DTMF signal transmitting step in which the non-SIP terminal transmits a DTMF signal for requesting a predetermined SIP service; and a VoIP adapter DTMF recognition step for recognizing the DTMF signal transmitted by the DTMF signal transmission means, and the VoIP adapter converts the DTMF signal recognized in the DTMF recognition step into the predetermined SIP according to a predetermined conversion rule. Convert to SIP message to realize service And conversion step, the VoIP adapter, the SIP message generated by the conversion in the conversion step, the IP and a SIP message transmission step of transmitting via the (Internet Protocol) network, wherein the predetermined SIP service, It is characterized in that at least one of answering machine transfer and melody hold for outputting the hold sound stored in the external server during hold is included .

According to the present invention, the VoIP adapter realizes the predetermined SIP service by converting the DTMF signal received from the non-SIP terminal for requesting the predetermined SIP service according to a predetermined conversion rule. SIP message is generated and transmitted through the IP network. Therefore, by defining a SIP service corresponding to each DTMF signal that can be output from a non-SIP terminal, and defining a conversion rule for converting the DTMF signal into a SIP message for realizing the SIP service, It is possible to use various network services realized by SIP using a non-SIP terminal.
In particular, the predetermined SIP service includes at least one of answering machine transfer and melody hold that outputs a hold sound stored in an external server while the user is on hold. Can be used.

The VoIP adapter according to claim 5 is connected to a communication terminal capable of transmitting a DTMF (Dual Tone Multi Frequency) signal that does not support advanced voice services such as transfer and hold other than session establishment and disconnection. A VoIP adapter that receives a DTMF signal for requesting a predetermined advanced voice service from the communication terminal, and a DTMF recognition unit that recognizes a DTMF signal received by the signal transmission / reception unit; Conversion means for converting the DTMF signal recognized by the DTMF recognition means into a message for realizing the predetermined advanced voice service according to a predetermined conversion rule; and a message generated by the conversion by the conversion means Sending message over an IP (Internet Protocol) network When, wherein the the predetermined altitude voice services, to a voice mail transfer, a melody hold for outputting a hold sound stored in the external server while on hold, characterized in that at least one is contained in.

According to the present invention, the VoIP adapter converts the DTMF signal received from the communication terminal for requesting a predetermined advanced voice service according to a predetermined conversion rule, thereby converting the predetermined advanced voice service. A message for realization is generated and transmitted via the IP network. Therefore, by defining a service corresponding to each DTMF signal that can be output from the communication terminal, and defining a conversion rule for converting the DTMF signal into a message for realizing the service, the communication terminal It is possible to use various advanced voice services by using.
In particular, since the predetermined advanced voice service includes at least one of the answering machine transfer and the melody hold that outputs the hold sound stored in the external server during the hold, the user uses the advanced voice service. can do.

The VoIP adapter according to claim 6 is the VoIP adapter according to claim 5 , wherein the DTMF signal for requesting the predetermined advanced voice service is transmitted from the communication terminal by a user operation on the communication terminal. It is characterized by.
According to the present invention, the user can use the advanced voice service desired by the user by performing an operation for outputting a DTMF signal corresponding to the service desired by the user using the communication terminal. Become.

The communication method according to claim 7 is a communication terminal that does not support advanced voice services such as transfer and hold other than session establishment and disconnection and can transmit a DTMF (Dual Tone Multi Frequency) signal; A communication method via an IP (Internet Protocol) network performed by a communication system including a VoIP adapter connected to a communication terminal, wherein the communication terminal transmits a DTMF signal for requesting a predetermined advanced voice service. A DTMF signal transmission step, a DTMF recognition step in which the VoIP adapter recognizes a DTMF signal transmitted by the DTMF signal transmission means, and a DTMF signal recognized in the DTMF recognition step by the VoIP adapter. In order to realize the predetermined advanced voice service according to the conversion rule A conversion step of converting the message, the VoIP adapter, the messages generated by the conversion in the conversion step, IP (Internet Protocol) and a message transmission step of transmitting over a network, said predetermined altitude Voice Services Includes at least one of voice mail transfer and melody hold that outputs a hold tone stored in an external server during hold .

  According to the present invention, the VoIP adapter realizes the predetermined SIP service by converting the DTMF signal received from the non-SIP terminal for requesting the predetermined SIP service according to a predetermined conversion rule. SIP message is generated and transmitted through the IP network. Therefore, by defining a SIP service corresponding to each DTMF signal that can be output from a non-SIP terminal, and defining a conversion rule for converting the DTMF signal into a SIP message for realizing the SIP service, It is possible to use various network services realized by SIP using a non-SIP terminal.

1 is a diagram illustrating an overall configuration of a communication network according to an embodiment of the present invention. It is a block diagram which shows the function structure of the VoIP adapter 10 which concerns on the same embodiment. It is a figure which shows an example of the conversion rule which concerns on the same embodiment. It is a sequence diagram which shows the procedure of the answering machine transfer process based on the embodiment. It is a sequence diagram which shows the procedure of the melody holding | maintenance process which concerns on the same embodiment.

Embodiments of the present invention will be described below with reference to the drawings. In each drawing referred to in the following description, components equivalent to those in the other drawings are denoted by the same reference numerals.
FIG. 1 is a diagram showing an overall configuration of a communication network according to an embodiment of the present invention. The communication network according to the present embodiment includes a SIP-AS (Session Initiation Protocol-Application Server) 50 and a CSCF (Call Session Control Function) 40 based on the IMS (IP Multimedia Subsystem) core NW architecture defined by 3GPP. An IP network (hereinafter referred to as “IPNW”) 30, a telephone 20 installed in a house, and a VoIP adapter 10 connected to the telephone 20 and the IPNW 30 are configured.

Each of these devices constituting the communication network includes, as hardware, a CPU (Central Processing Unit) (not shown) that controls the entire device, a RAM (Random Access Memory) that stores software such as programs and data, and a ROM (Read Only Memory), a storage device such as a hard disk, and a communication interface for controlling communication.
The CSCF 40 manages an identifier of a terminal that performs communication via the IPNW 30 and relays transmission / reception of the SIP message.

The SIP-AS 50 transmits / receives a SIP message to / from the CSCF 40 in order to realize various application services.
The telephone 20 is a push-type fixed telephone capable of performing a voice call via a public telephone network (PSTN: Public Switched Telephone Networks) (not shown). The telephone 20 is a non-SIP terminal that does not support SIP (Session Initiation Protocol) that enables communication via the IPNW 30.

The telephone 20 includes a circuit that generates a DTMF (Dual Tone Multi Frequency) signal corresponding to a plurality of types of push buttons. The DTMF signal is composed of signal sounds of two voice frequency bands, low group and high group, and a total of 16 types of numbers from 0 to 9 and symbols of *, #, A, B, C, and D Can be expressed. The telephone 20 includes a DTMF signal transmitting unit that generates a DTMF signal corresponding to the pressed push button and transmits the DTMF signal to the VoIP adapter 10 when the push button is pressed.
The VoIP adapter 10 is an adapter that enables communication by the telephone 20 via the IPNW 30.

  FIG. 2 shows a functional configuration of the VoIP adapter 10. As shown in the figure, the VoIP adapter 10 includes a signal transmission / reception unit 11, a DTMF (Dual Tone Multi Frequency) recognition unit 12, a conversion unit 13, a SIP message transmission unit 14, and a signal transmission / reception unit 15. Yes. These functions are functions realized by the hardware and software included in the VoIP adapter 10 operating in cooperation.

The signal transmission / reception unit 11 controls transmission / reception of signals with the telephone 20. The signal received from the telephone 20 includes an audio signal representing the voice uttered by the user, a DTMF signal generated when the push button of the telephone 20 is pressed, and the like.
The signal transmission / reception unit 15 controls transmission / reception of packet signals via the IPNW 30.
The DTMF recognition unit 12 includes various frequency filters, a detection circuit, a code converter, and the like, and analyzes the DTMF signal included in the signal received by the signal transmission / reception unit 11 and converts it into a digital signal, thereby the DTMF. Recognize numbers and symbols indicated by signals.

The conversion unit 13 converts the DTMF signal recognized by the DTMF recognition unit 12 into a SIP message for realizing a predetermined SIP service according to a predetermined conversion rule.
The predetermined conversion rule may be stored in the storage device of the VoIP adapter 10 as a conversion table, or may be described in a program stored in the storage device. When the conversion rule is stored as a conversion table in the storage device, the CPU of the VoIP adapter 10 performs conversion processing according to the conversion table stored in the storage device. When the conversion rule is described in the program, the CPU of the VoIP adapter 10 performs conversion processing according to the program in which the conversion rule is described.

FIG. 3 is a diagram illustrating an example of the conversion rule. As shown in the figure, in the conversion rule, the type of DTMF signal and the SIP message to be generated when the DTMF signal is received from the telephone 20 are stored in association with each other.
For example, when the user off-hooks the handset when a ringing tone is being output from the telephone 20 and presses the push button “1” of the telephone 20 to answer the call, the conversion unit 13 determines that the conversion rule is based on the conversion rule. A SIP message “SIP_Invite_OK” is generated.

Also, when the user presses the push button “2” to hold the melody while talking on the telephone 20, the conversion unit 13 uses the SIP for transmission to the melody server based on the conversion rule. The message “SIP_Invite no sdp” and the SIP message “SIP_Invite” to be transmitted to the partner terminal are generated.
In addition, when the user presses the push button “*” to end the call while talking on the telephone 20 or after being on-hook, the conversion unit 13 determines that the SIP message “SIP_Bye” is based on the conversion rule. Is generated.

In addition, when the user presses the push button “3” to perform transfer to a predetermined transfer destination during a call or hold on the telephone 20, the conversion unit 13 is input by the user based on the conversion rule. The SIP message “SIP_refer / replace” is generated by designating the transfer destination telephone number or the telephone number registered in advance as “Refer to”.
In addition, after the user has off-hooked the handset when a ringing tone is being output from the telephone 20, or during a call or on hold, a push to activate the answering service and connect the other terminal to the answering service center. When the button “4” is pressed, the conversion unit 13 describes, for example, the SIP URI of the answering machine service center in the header and describes a predetermined parameter indicating activation of the answering machine in the body part based on the conversion rule. , The SIP message “SIP_refer” is generated.

Further, when the user presses the push button “5” of the telephone 20 and inputs “A” as the additional telephone number in order to make a three-party call while the telephone 20 is talking or on hold, the conversion unit 13 Generates a SIP message “SIP_Invite to: telephone number (A)” based on the conversion rule.
In addition, when a ringing tone is output from the telephone set 20, the user presses the push button “6” to off-hook the handset and activate the nuisance call stop service to connect the other terminal to the nuisance call stop service center. In this case, the conversion unit 13 generates a SIP message “Use Proxy 305” based on the conversion rule.

The melody hold tone, hold transmission, answering machine activation, three-party call, and nuisance call stop service shown in FIG. 3 are examples of services realized by SIP. By converting the DTMF signal for requesting the SIP service into a SIP message for realizing the SIP service, the user can use the public telephone line telephone 20 that does not support SIP to use the public telephone line. In this case, it is possible to receive a SIP service that cannot be used.
The SIP message transmission unit 14 transmits the SIP message generated by the conversion by the conversion unit 13 from the signal transmission / reception unit 15 to the IPNW 30.

(Operation example)
Next, an operation example when using the SIP service using the telephone 20 connected to the VoIP adapter 10 will be described.
(Voice mail forwarding)
First, referring to the sequence diagram shown in FIG. 4, the procedure of the answering machine transfer process performed when the user presses the push button “4” off-hooking the handset while the telephone 20 is ringing. explain.

First, a partner terminal (not shown) makes a call to the telephone 20. Thereby, the IPNW 30 transmits SIP_INVITE to the VoIP adapter 10 (step S101).
The VoIP adapter 10 that has received the SIP_INVITE calls the telephone 20 (step S102). As a result, the telephone 20 outputs a ringing tone, and the user goes off-hook of the handset of the telephone 20 (step S103).

The telephone 20 detects an off-hook and transmits an off-hook detection signal to the VoIP adapter 10 (step S104). Thereby, the VoIP adapter 10 recognizes the off-hook of the telephone 20 (step S105).
After the off-hook, the user presses the push button “4” on the telephone 20 to activate the answering machine service and connect the other terminal to the answering service center (step S106). Thereby, the telephone set 20 transmits a DTMF signal representing “4” to the VoIP adapter 10 (step S107).

When the VoIP adapter 10 receives the DTMF signal via the signal transmission / reception unit 11, the DTMF recognition unit 12 recognizes that the DTMF signal represents “4” (step S108).
Based on the conversion rule, the conversion unit 13 of the VoIP adapter 10 describes the SIP URI of the answering machine service center in the header, and sets a parameter indicating activation of the answering machine in a predetermined part (header or body part) defined by the NW. The described SIP message “SIP_refer” is generated, and the caller / IPNW is instructed to transfer to the answering machine service center (step S109).
The SIP message transmission unit 14 transmits the SIP message from the signal transmission / reception unit 15 to the IPNW 30 (step S110).

  The SIP message is transferred to the partner terminal via the SIP-AS 50 and CSCF 40 of the IPNW 30. Thereafter, according to the SIP procedure, the session between the VoIP adapter 10 and the partner terminal is disconnected, and the session between the partner terminal and the answering machine service center is established (step S111). As a result, the communication partner records the message in the answering machine service center. After the message recording is completed, the answering machine service center notifies a pre-registered communication terminal that the answering machine message has been recorded.

As described above, even when the telephone 20 does not have the answering machine function, the incoming call to the telephone 20 of the partner terminal can be transferred to the answering service center.
By using such an answering service, for example, when an incoming call arrives at the telephone 20 just before the user goes out, the user can off-hook the telephone 20 and press the push button “4” to The incoming call can be transferred to the answering service center and the other party can record the message. Then, the user can connect to the answering machine service center using, for example, a mobile phone while walking to the station and listen to the message recorded by the other party.
The partner terminal may be a SIP terminal that supports SIP or a telephone for a public telephone line to which the VoIP adapter 10 is connected.

(Melody hold)
Next, with reference to the sequence diagram shown in FIG. 5, the procedure of the melody hold process that is performed when the user presses the push button “2” while talking on the telephone 20 will be described. It is assumed that the melody hold sound is stored in a music server (not shown).
The processing from step S201 to S205 shown in FIG. 5 is the same as the processing from step S101 to S105 shown in FIG.
The user presses the push button “2” on the telephone 20 during a call (step S206). As a result, the telephone 20 transmits a DTMF signal representing “2” to the VoIP adapter 10 (step S207).

When the VoIP adapter 10 receives the DTMF signal via the signal transmission / reception unit 11, the DTMF recognition unit 12 analyzes the DTMF signal and recognizes that the DTMF signal represents “2” (step S208).
The conversion unit 13 of the VoIP adapter 10 generates a SIP message for performing melody hold based on the conversion rule (step S209). In the present embodiment, the session control is held by the caller, but for the media stream, it is necessary to directly connect the music server and the other terminal, so that the mechanism of 3PCC (3rd Party Call Control) (RFC3725) should be used. Is assumed.

The SIP message transmission unit 14 of the VoIP adapter 10 transmits the SIP message “SIP_INVITE no sdp” to the music server (not shown) via the signal transmission / reception unit 15 and the IPNW 30 (steps S210 and S211).
The music server returns a response message “200 OK” to the VoIP adapter 10 via the IPNW 30 (steps S212 and S213).

  When receiving the response message “200 OK” from the music server, the VoIP adapter 10 transmits the SIP message “SIP_INVITE” to the counterpart terminal via the IPNW 30 (steps S214 and S215). Thereafter, according to the SIP procedure, a media session between the music server and the partner terminal is established while the session between the VoIP adapter 10 and the partner terminal is held, and a melody hold sound is transmitted from the music server to the partner terminal.

Thereby, it is possible to use the melody hold sound stored in the music server without using the tasteless hold sound built in the telephone 20.
In addition to the above-described operation example, for example, when using the hold transfer service corresponding to the DTMF signal “3” shown in FIG. Can also be transferred to the mobile terminal by the family on the go.

  As described above, the VoIP adapter 10 converts the DTMF signal transmitted from the telephone set 20 when the user pushes the push button according to a predetermined conversion rule, thereby realizing a SIP for realizing a predetermined SIP service. A message is generated and the SIP message is transmitted via the IPNW 30. Accordingly, a SIP service corresponding to each DTMF signal that can be output from the telephone 20 is defined, and a conversion rule for converting the DTMF signal into a SIP message for realizing the SIP service is defined. 20 can use various network services realized by SIP.

Therefore, the user can use the advanced voice service using SIP without replacing the public telephone line telephone 20 that has been used conventionally.
In addition to the conversion rule from the DTMF signal to the SIP message described in the above-described embodiment, by preparing conversion rules for SIP service such as incoming call transfer during a call, incoming call transfer when there is no response, automatic redialing, etc. Various SIP services can be used.

The telephone 20 is not limited to the push-type telephone 20 and may be any non-SIP terminal that can output different DTMF signals according to different operations by the user.
In the above-described embodiments, SIP has been described. However, the present invention is not limited to SIP, and the present invention is a communication protocol that enables advanced voice services such as transfer and hold other than session establishment and disconnection. Applicable.

  Furthermore, in this embodiment, the VoIP adapter 10 performs SIP control. In IMS, the architecture assumes that the SIP-AS 50 performs call control after receiving a SIP trigger from a user, but can be realized by applying the procedure described in this embodiment. Therefore, the present invention is not limited to this embodiment, but rather is a wide-valued invention for users who want to use various services realized by SIP without replacing existing non-SIP terminals.

DESCRIPTION OF SYMBOLS 10 VoIP adapter 11 Signal transmission / reception part 12 DTMF recognition part 13 Conversion part 14 SIP message transmission part 15 Signal transmission / reception part 20 Telephone

Claims (7)

A VoIP adapter connected to a non-SIP terminal that does not support SIP (Session Initiation Protocol) and can transmit a DTMF (Dual Tone Multi Frequency) signal,
Signal transmitting and receiving means for receiving a DTMF signal for requesting a predetermined SIP service from the non-SIP terminal;
DTMF recognition means for recognizing the DTMF signal received by the signal transmission / reception means;
Conversion means for converting a DTMF signal recognized by the DTMF recognition means into a SIP message for realizing the predetermined SIP service according to a predetermined conversion rule;
SIP message transmission means for transmitting a SIP message generated by the conversion by the conversion means via an IP (Internet Protocol) network ,
The VoIP adapter, wherein the predetermined SIP service includes at least one of an answering machine transfer and a melody hold that outputs a hold tone stored in an external server during the hold .   The VoIP adapter according to claim 1, wherein the DTMF signal for requesting the predetermined SIP service is transmitted from the non-SIP terminal by a user operation on the non-SIP terminal. A communication system including a non-SIP terminal that does not support SIP (Session Initiation Protocol) and can transmit a DTMF (Dual Tone Multi Frequency) signal, and a VoIP adapter connected to the non-SIP terminal,
The VoIP adapter is
Signal transmitting and receiving means for receiving a DTMF signal for requesting a predetermined SIP service from the non-SIP terminal;
DTMF recognition means for recognizing the DTMF signal received by the signal transmission / reception means;
Conversion means for converting a DTMF signal recognized by the DTMF recognition means into a SIP message for realizing the predetermined SIP service according to a predetermined conversion rule;
SIP message transmission means for transmitting a SIP message generated by the conversion by the conversion means via an IP (Internet Protocol) network ,
The predetermined SIP service includes at least one of an answering machine transfer and a melody hold that outputs a hold tone stored in an external server during the hold . An IP (Internet Protocol) performed by a communication system including a non-SIP terminal that does not support SIP (Session Initiation Protocol) and can transmit a DTMF (Dual Tone Multi Frequency) signal and a VoIP adapter connected to the non-SIP terminal ) A communication method via a network,
A DTMF signal transmission step in which the non-SIP terminal transmits a DTMF signal for requesting a predetermined SIP service;
A DTMF recognition step in which the VoIP adapter recognizes a DTMF signal transmitted by the DTMF signal transmission means;
A conversion step in which the VoIP adapter converts the DTMF signal recognized in the DTMF recognition step into a SIP message for realizing the predetermined SIP service according to a predetermined conversion rule;
A SIP message transmitting step in which the VoIP adapter transmits a SIP message generated by the conversion in the converting step via an IP (Internet Protocol) network ;
The predetermined SIP service includes at least one of an answering machine transfer and a melody hold that outputs a hold tone stored in an external server during the hold . A VoIP adapter connected to a communication terminal that does not support advanced voice services such as transfer and hold other than session establishment and disconnection and can transmit a DTMF (Dual Tone Multi Frequency) signal,
A signal transmitting / receiving means for receiving a DTMF signal for requesting a predetermined advanced voice service from the communication terminal;
DTMF recognition means for recognizing the DTMF signal received by the signal transmission / reception means;
Conversion means for converting a DTMF signal recognized by the DTMF recognition means into a message for realizing the predetermined advanced voice service according to a predetermined conversion rule;
A message transmission means for transmitting a message generated by the conversion by the conversion means via an IP (Internet Protocol) network ,
The predetermined advanced voice service includes at least one of an answering machine transfer and a melody hold that outputs a hold tone stored in an external server while the VoIP adapter is on hold . 6. The VoIP adapter according to claim 5 , wherein the DTMF signal for requesting the predetermined advanced voice service is transmitted from the communication terminal by a user operation on the communication terminal. A communication terminal that does not support advanced voice services such as transfer and hold other than session establishment and disconnection and can transmit a DTMF (Dual Tone Multi Frequency) signal, and a VoIP adapter connected to the communication terminal A communication method via an IP (Internet Protocol) network performed by a communication system including:
A DTMF signal transmission step in which the communication terminal transmits a DTMF signal for requesting a predetermined advanced voice service;
A DTMF recognition step in which the VoIP adapter recognizes a DTMF signal transmitted by the DTMF signal transmission means;
A conversion step in which the VoIP adapter converts the DTMF signal recognized in the DTMF recognition step into a message for realizing the predetermined advanced voice service according to a predetermined conversion rule;
A message transmission step in which the VoIP adapter transmits a message generated by the conversion in the conversion step via an IP (Internet Protocol) network ;
The predetermined advanced voice service includes at least one of an answering machine transfer and a melody hold that outputs a hold tone stored in an external server during the hold .

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