TWI556616B - A media communication system, an audio terminal, and an acoustic symbol conversion device - Google Patents

Description

Media communication system, audio terminal, and audio symbol conversion device

The present invention relates to a plurality of audio terminals that perform an audio call through a network, and converts an audio symbol conversion device of a media type that communicates between a plurality of audio terminals, and a plurality of audio terminals, an acoustic symbol conversion device, and a SIP server. Media communication system.

With the development of IP (Internet Protocol) networks in recent years, VoIP (Voice Over IP) technology that implements IP packetization and transmission of audio signals has become widespread.

Usually, the call connection protocol between VoIP terminals (audio terminals) uses SIP (Session Initiation Protocol). SIP has the function of negotiating the types of acoustic symbols stored in the audio package that are received by the VoIP terminal. Specifically, the audio symbol type that can be associated with the SDP (Session Description Protocol) is written in the main body of the SIP information, and the VoIP terminal refers to the audio symbol type to be transmitted.

Usually, when the VoIP terminals of the two parties do not include the same audio codec (sound encoder/decoder), they cannot talk, but the device that converts the audio symbols by including the audio packet sent through the VoIP terminal can realize different sounds. A method of talking between VoIP terminals of a codec is disclosed (for example, Patent Document 1).

Patent Document 1 discloses that in a call between VoIP terminals located in different networks, it is necessary to convert audio symbols, and the audio symbol conversion device is configured in a system for performing call control by a SIP server located in each network. In the realm of the Internet, the audio symbol conversion device is used as a relay point function of SIP information and audio packets, and performs techniques for rewriting SIP information and converting audio symbols.

Advanced technical literature Patent literature

Patent Document 1: JP-A-2007-49415

However, in the technique disclosed in Patent Document 1, in a system in which a VoIP terminal including a different audio codec is present in the same network, when the audio symbol is to be converted, the SIP server needs to perform the following actions, and is also to be trusted by the VoIP terminal. The SIP information is not transferred to the original VoIP terminal but is transferred to the acoustic symbol conversion device. After the SIP information is rewritten by the acoustic symbol conversion device, the SIP server is rewritten by the acoustic symbol conversion device. It is then forwarded to the VoIP terminal of the original forwarding end.

However, the operation in conjunction with such an acoustic symbol conversion device is not the original function of the SIP server, and this portion causes a problem that the processing load of the SIP server increases.

The present invention has been made to solve the above problems, and an object of the present invention is to obtain a media communication system capable of performing audio communication between audio terminals even when there is an audio terminal including different audio codecs on the same network, without requiring a SIP server and an audio symbol. The combination of the transformation device can reduce the SIP server The burden.

The media communication system of the present invention connects a plurality of audio terminals, a call control server, and an audio symbol conversion device that converts media types for communication between a plurality of terminals through a network, thereby making a plurality of audio terminals The audio caller includes: a telephone number conversion unit that converts the telephone number of the audio terminal to be transmitted to the telephone number of the audio symbol conversion device according to the set rule; and the terminal SIP control unit The telephone number converted by the telephone number conversion unit generates credit information, and transmits the information to the acoustic symbol conversion device through the call control server. The acoustic symbol conversion device includes an INVITE generation unit, and the audio terminal of the transmission source is trusted by the call control server. When the information of the credit is sent, the destination telephone number in the information of the credited credit is inversely transformed according to the set rule, and is set as the telephone number of the audio terminal to which the sender is to be sent, and the credit information is to be sent. The trusted address of the audio package is set to its own address to generate a new credit Interest; and communication unit for transmitting and receiving an IP packet, the credit line to send information generation portion generates the INVITE signaling messenger to the server object by calling sound control terminal.

According to the present invention, even if there is an audio terminal including different audio codecs on the same network, audio communication between VoIP terminals can be performed, and the SIP server and the acoustic symbol conversion device are not required to be connected, and the SIP server can be alleviated. The processing burden.

1‧‧‧SIP server

21~2N‧‧‧VoIP terminal

3‧‧‧Audio symbol conversion device

4‧‧‧IP network

201‧‧‧Phone number receiving department

202‧‧‧Phone number conversion department

203‧‧‧Login Information Generation Department

204‧‧‧ symbol conversion device registration information generation unit

205‧‧‧ Terminal SIP Control Department

206‧‧‧Audio Receiving Department

207‧‧‧Acoustic encoder

208‧‧‧Audio decoder

209‧‧‧Sound Output Department

210‧‧‧IP packet delivery department

220, 320‧‧‧ CPU

221‧‧‧Monitor

222, 321‧‧‧HDD

223, 322‧‧‧ memory

224‧‧‧ microphone

225, 323‧‧‧Communication I/F devices

301‧‧‧IP packet delivery department

302‧‧‧Transformer SIP Control Department

303‧‧‧INVITE Production Department

304‧‧200200 OK Production Department

305‧‧‧Audio symbol conversion unita

306‧‧‧Audio symbol conversion unit b

307‧‧‧Transferred to REGISTER Production Department

308‧‧‧Self device REGI STER generator

Fig. 1 is a block diagram showing a configuration of a media communication system according to a first embodiment of the present invention.

Fig. 2 is an explanatory diagram showing the internal configuration of a VoIP terminal in the media communication system according to the first embodiment of the present invention.

Fig. 3 is an explanatory diagram showing the hardware configuration of the VoIP terminal according to the first embodiment of the present invention.

Fig. 4 is an explanatory diagram showing the internal configuration of an acoustic symbol conversion device in the media communication system according to the first embodiment of the present invention.

Fig. 5 is an explanatory view showing the hardware configuration of the acoustic sign conversion device according to the first embodiment of the present invention.

Fig. 6 is an explanatory diagram of a call-on sequence in the media communication system according to the first embodiment of the present invention.

Fig. 7 is a flow chart for explaining the operation of step ST1 of Fig. 6 in detail.

Fig. 8 is a flow chart for explaining in detail the operation of step ST3 of Fig. 6.

Fig. 9 is a view showing an example of the input INVITE information of the INVITE generating unit in step ST804 of Fig. 8.

Fig. 10 is a view showing an example of the output INVITE information of the INVITE generating unit in step ST804 of Fig. 8.

11 is an audio symbol conversion device that receives a 200 OK response message from the SIP server 1 (refer to step ST10 of FIG. 6), generates a 200 OK response message to the VoIP terminal, and transmits the message to the SIP server 1 (step ST11 of FIG. 6). The flowchart of the operation up to the end.

Fig. 12 is a diagram showing an example of input 200 OK response information of the 200 OK generating unit in step ST1104 of Fig. 11.

Figure 13 is the output of the 200 OK generating unit in step ST1104 of Figure 11 A diagram showing an example of a 200 OK response message.

Fig. 14 is a flow chart for explaining the operation of the acoustic sign conversion means when receiving an acoustic packet from another VoIP terminal.

Fig. 15 is an explanatory diagram showing an example of REGISTER information generated by the symbol conversion device registration information generating unit of the VoIP terminal.

Figure 16 is an explanatory diagram showing the internal configuration of a VoIP terminal in the media communication system according to the second embodiment of the present invention.

Figure 17 is an explanatory diagram showing the internal configuration of an acoustic symbol conversion device in the media communication system according to the second embodiment of the present invention.

Fig. 18 is an explanatory diagram showing an example of REGISTER information received by the VoIP terminal in the acoustic symbol conversion device in the second embodiment.

Fig. 19 is an explanatory diagram showing an example of information to be transmitted as a basis for creating REGISTER information generated by the REGISTER generating unit in the second embodiment.

Fig. 20 is an explanatory diagram showing an example of REGISTER information generated by the own device REGISTER generating unit when the VoIP terminal receives the REGISTER information shown in Fig. 18 in the audio symbol conversion device according to the second embodiment.

Fig. 21 is an explanatory diagram showing an example of another configuration of the media communication system.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.

Embodiment 1.

Fig. 1 is a block diagram showing a configuration of a media communication system according to a first embodiment of the present invention.

As shown in Fig. 1, the media communication system is formed by a SIP server (call control server) 1, a VoIP terminal (audio terminal) 21 to 2N, and an acoustic symbol conversion device 3 connected via an IP network 4.

The SIP server 1 manages the telephone number and address information of each of the VoIP terminals 21 to 2N, and transfers the SIP information transmitted by each of the VoIP terminals 21 to 2N.

The VoIP terminals 21~2N are VoIP calls between terminals.

The acoustic symbol conversion device 3 converts the acoustic symbols of the acoustic packets transmitted by the respective VoIP terminals 21 to 2N.

Each of the VoIP terminals 21~2N holds one unique telephone number, and the telephone numbers of the VoIP terminals 21, 22, ‧‧‧, and 2N are set to 1001, 1002, ‧‧‧, 1000+N, respectively, in the media communication system The number N of the set VoIP terminals 21 to 2N is set to be less than 1000.

On the other hand, the acoustic symbol conversion device 3 operates as a VoIP terminal having the same number of telephone numbers as the VoIP terminals 21 to 2N provided in the media communication system, and the telephone numbers are set to 2001, 2002, ‧ ‧ ‧ , 2000+N. Here, N is less than 1000, and therefore the telephone numbers of the VoIP terminals 21 to 2N and the telephone number of the acoustic symbol conversion device 3 are not repeated.

Fig. 2 is an explanatory diagram showing the internal configuration of the VoIP terminals 21 to 2N in the media communication system according to the first embodiment of the present invention.

The VoIP terminals 21 to 2N include a telephone number receiving unit 201, a telephone number converting unit 202, a registration information generating unit 203, a symbol conversion device registration information generating unit 204, a terminal SIP control unit 205, an audio receiving unit 206, and an audio encoder 207; The audio decoder 208; the audio output unit 209; and the IP packet transmission and reception unit 210.

The telephone number receiving unit 201 receives the telephone number input from the terminal user.

The telephone number conversion unit 202 converts the telephone number into its own according to a specific rule. His number.

The login information generating unit 203 generates the own terminal information registered in the SIP server 1.

The symbol conversion device registration information generation unit 204 generates registration information of the acoustic symbol conversion device 3.

The terminal SIP control unit 205 generates SIP information, performs interpretation and transmission and reception control, and controls the VoIP terminals 21 to 2N.

The audio receiving unit 206 receives the call audio of the terminal user.

The acoustic encoder 207 encodes the call sound of the end user received by the acoustic receiving unit 206.

The audio decoder 208 decodes the voice of the call that is encoded by the caller.

The audio output unit 209 outputs the call audio that is received by the call terminal and decoded by the audio decoder 208 to the end user.

The IP packet transmission/reception unit 210 transmits and receives an IP packet to and from each VoIP terminal or audio symbol conversion device 3 connected to the IP network 4 via the IP network 4.

Further, in the configuration shown in Fig. 2, the telephone number conversion unit 202 and the symbol conversion device registration information generation unit 204 are configured in the same manner as the VoIP terminal of the general SIP database.

Fig. 3 is an explanatory diagram showing the hardware configuration of the VoIP terminals 21 to 2N according to the first embodiment of the present invention.

In the first embodiment of the present invention, the telephone number conversion unit 202, the registration information generation unit 203, the symbol conversion device registration information generation unit 204, the terminal SIP control unit 205, the acoustic encoder 207, and the acoustic decoder 208 are stored by A processing circuit such as a CPU 220 or a system LSI of a program such as the HDD 222 or the memory 223 is realized.

Moreover, a plurality of processing circuits can perform the above functions in combination.

The telephone number receiving unit 201 uses a monitor 221 having a touch panel or the like. Moreover, this is an example, and the telephone number receiving part 201 can be comprised by another hardware.

The acoustic receiving unit 206 uses the microphone 224, and the acoustic output unit 209 uses the speaker 226. Moreover, as an example, the acoustic receiving unit 206 and the acoustic output unit 209 may be constituted by other hardware such as a headphone.

The IP packet transmission and reception unit 210 uses the communication I/F device 225. Here, as an example, the IP packet transmission/reception unit 210 may be configured by other hardware.

Fig. 4 is an explanatory diagram showing the internal configuration of the acoustic symbol conversion device 3 in the media communication system according to the first embodiment of the present invention.

The acoustic symbol conversion device 3 includes an IP packet transmission/reception unit 301, a conversion device SIP control unit 302, an INVITE generation unit 303, a 200 OK generation unit 304, an acoustic symbol conversion unit a 305, and an acoustic symbol conversion unit b 306.

The IP packet transmission/reception unit 301 transmits and receives the VoIP terminal and the IP packet connected to the IP network 4 via the IP network 4.

The conversion device SIP control unit 302 performs SIP information generation, interpretation, and transmission/reception control, and controls the acoustic symbol conversion device 3.

The INVITE generating unit 303 generates INVITE information when it is transmitted to the VoIP terminals 21 to 2N.

The 200 OK generating unit 304 generates 200 OK response information for the incoming calls of the VoIP terminals 21 to 2N.

The acoustic symbol conversion unit a 305 performs μ specified by ITU-T Recommendation G.711 The law PCM method (hereinafter referred to as the μ rule PCM method) to the audio symbol conversion of the CS-ACELP method (hereinafter referred to as CS-ACELP method) defined by ITU-T Recommendation G.729.

The acoustic symbol conversion unit b 306 performs conversion from the CS-ACELP method to the μ-law PCM method.

Fig. 5 is an explanatory view showing the hardware configuration of the acoustic sign conversion device 3 according to the first embodiment of the present invention.

The INVITE generation unit 303, the 200 OK generation unit 304, the conversion device SIP control unit 302, the acoustic symbol conversion unit a 305, and the acoustic symbol conversion unit b 306 are CPUs 320 and systems that execute programs stored in the HDD 321, the memory 322, and the like. It is realized by a processing circuit such as LSI.

Moreover, a plurality of processing circuits can perform the above functions in combination.

The IP packet transmission and reception unit 301 uses the communication I/F device 323. Here, as an example, the IP packet transmission/reception unit 301 can be configured by other hardware.

Fig. 6 is an explanatory diagram of a call-on sequence in the media communication system according to the first embodiment of the present invention.

Hereinafter, the operation of the VoIP terminals 21 to 2N and the acoustic symbol conversion device 3 shown in Figs. 2 and 4 will be described based on the call-on sequence of Fig. 6.

Further, Fig. 6 is a sequence in which the VoIP terminal 21 (telephone number 1001) transmits a VoIP terminal 22 (telephone number 1002) before the start of the call.

First, when the VoIP terminal 21 transmits a message to the VoIP terminal 22, the VoIP terminal 21 transmits INVITE information (step ST1 of Fig. 6).

Fig. 7 is a flow chart showing the detailed description of the operation of step ST1 of Fig. 6.

Hereinafter, the operation of step ST1 of Fig. 6 will be described with reference to Figs. 7 and 2 .

The telephone number receiving unit 201 of the VoIP terminal 21 receives the telephone number 1002 of the transmission target by the user, and outputs the received telephone number 1002 to the telephone number conversion unit 202 (step ST701).

The telephone number conversion unit 202 inputs the telephone number 1002 outputted by the telephone number receiving unit 201 in step ST701, converts it according to a specific and previously set rule, and outputs the converted telephone number to the terminal SIP control unit 205 (step ST702). ). Further, in the first embodiment, an example of the telephone number conversion rule to be set is a numerical value obtained by adding the input telephone number to 1000 as the converted number. Therefore, the telephone number converted by the telephone number conversion unit 202 becomes 2002, and the number is output to the terminal SIP control unit 205.

Here, as described above, the set telephone number conversion rule sets the value obtained by adding the telephone number to 1000 as the converted number. However, this rule is only an example, and other phone number change rules can also be used.

The terminal SIP control unit 205 generates the INVITE information of the credit according to the telephone number converted by the telephone number conversion unit 202 in step ST702, and outputs the INVITE information to the IP packet transmission/reception unit 210 (step ST703).

The IP packet transmission/reception unit 210 performs IP packetization on the INVITE information of the credit outputted by the terminal SIP control unit 205 in step ST703, and transmits the INVITE information to the SIP server 1 via the IP network 4 (step ST704).

Moreover, according to the above telephone number conversion rule, the telephone numbers of the VoIP terminals 21, 22, ‧‧‧, 2N are 1001, 1002, ‧ ‧ ‧, 1000 + N, so the converted number becomes 2001, 2002, ‧ ‧‧, 2000+N, and the telephone number included in the acoustic symbol conversion device 3 is identical.

Go back to the call-on sequence in Figure 6.

As shown in Fig. 6, the INVITE information sent by the VoIP terminal 21 is received by the SIP server 1.

Since the destination telephone number based on the received INVITE information is 2002, the SIP server 1 transfers the INVITE information to the acoustic symbol conversion device 3 holding the telephone number 2002 (step ST2 of Fig. 6).

Next, the acoustic symbol conversion device 3 receives the INVITE information from the SIP server 1, generates new INVITE information, and transmits it (step ST3 in Fig. 6).

Fig. 8 is a flow chart for explaining in detail the operation of step ST3 of Fig. 6.

Hereinafter, the operation of step ST3 of Fig. 6 will be described with reference to Figs. 8 and 3 .

When the IP packet transmission/reception unit 301 receives the IP packet in which the SIP information is stored, the IP packet is judged to be the SIP information by the other number, and the SIP information is output to the conversion device SIP control unit 302 (step ST801). Also, in general, the SIP information is transmitted using the UDP packet of the number 5060.

The conversion device SIP control unit 302 determines whether or not the information input by the IP packet transmission/reception unit 301 in step ST801 is INVITE information (step ST802). Also, whether the input information is INVITE information can be determined by the first line of the information. Specifically, the conversion device SIP control unit 302 determines whether or not the first header of the input information is "INVITE", and if it is "INVITE", it determines INVITE information (see FIG. 9 to be described later).

When the non-INVITE information is determined in step ST802 ("NO" in step ST802), the processing is terminated by skipping the following processing.

When it is determined as INVITE information in step ST802 ("YES" in step ST802), the conversion device SIP control unit 302 will input the INVITE. The information is output to the INVITE generating unit 303 (step ST803).

After the INVITE information is input by the conversion device SIP control unit 302, the INVITE generation unit 303 generates new INVITE information (step ST804).

The destination telephone number of the new INVITE information generated in the step ST804 is reversed to the destination telephone number of the INVITE information input by the INVITE generating unit 303 using the telephone number conversion rule set in advance similar to the VoIP terminals 21 to 2N. Obtained by transformation. Therefore, the value of 1000 is reduced by the destination telephone number in the input INVITE message to become the destination telephone number.

In addition, the INVITE generating unit 303 generates an SDP main unit that has been added and changed as described below based on the SDP (Session Description Protocol) main unit in the input INVITE information.

First, the SDP main unit in the input INVITE information describes the symbolization type information of the audio codec included in the VoIP terminal 21, and further adds the symbolization type information of the audio codec included in the other VoIP terminal.

Moreover, the audio packet trusted address and the 埠 information of the SDP main unit are changed to the 埠 number that the device can trust.

Here, Fig. 9 is a view showing an example of input INVITE information of the INVITE generating unit 303 in step ST804 of Fig. 8.

Fig. 10 is a view showing an example of the output INVITE information of the INVITE generating unit 303 in step ST804 of Fig. 8.

Further, in Fig. 9 and Fig. 10, the IP address of the SIP server is "192.168.0.1", the IP address of the VoIP terminal 21 is "192.168.0.21", and the IP address of the acoustic symbol conversion device 3 is "192.168". Example of .0.2".

Moreover, the audio codec included in the VoIP terminal 21 is set to the μ rule PCM In addition, it is assumed that there is a VoIP terminal including a CS-ACELP type audio codec in the media communication system.

The output INVITE information of the INVITE generating unit 303 shown in Fig. 10 is generated in accordance with the following guidelines.

First, the INVITE generating unit 303 extracts the "2002" of the destination telephone number set as part of the Request-URI (Uniform Resource Identifier) of the first line in the input INVITE information of the ninth figure, and subtracts "2002". After the 1000 is dropped, "1002" is set as the destination phone number.

Further, the INVITE generating unit 303 sets the source telephone number set as one of the SIP-URIs of the From header to any telephone number of the acoustic symbol conversion device 3, and the INVITE information is input from FIG. In the 5th line, the Request-URI of the From header is extracted as "1001", and it is converted according to the preset telephone number conversion rule of the VoIP terminal 21~2N, that is, the "1001" is added to 1000 and the "2001" is set. Is the source phone number. Further, the SDP main unit that outputs the INVITE information (that is, each line including v=0 and below) is set by the SDP main unit of the input INVITE information according to FIG. 9, and the INVITE generating unit 303 adds the following change to generate the 10 figure output INVITE information.

First, the INVITE generating unit 303 sets the address of the second line (that is, the line of o=) and the line of the fourth line (that is, the line of c=) of the SDP body unit indicating the trusted IP address of the audio packet by VoIP. The IP address "192.168.0.21" of the terminal 21 is changed to the IP address "192.168.0.2" of the acoustic symbol conversion device 3. Further, the INVITE generation unit 303 sets the sixth line of the SDP main unit to "m=audio 42102 RTP/AVP 0 18", and the sound is The symbol conversion device 3 adds the code of the "18" of the CS-ACELP method to the code type information of the corresponding audio codec, and the code number of the received audio packet is "42102". Further, the former "0" means that the μ-law PCM method is used as the code of the coding type information of the corresponding audio codec.

Further, the INVITE generating unit 303 adds "a=rtpmap:18 G729/8000" to the eighth line of the SDP main unit, and describes the attributes of the CS-ACELP encoding method.

The other parts of the SDP body portion are the same as those in Figs. 9 and 10. Further, regarding the above-mentioned part of the INVITE information shown in FIG. 10 (Via Header, Max-Forwards Header, Call-ID Header, CSeq Header, Contact Header, Content-Type Header, Content-Length Header) The INVITE generating unit 303 generates the normal rule for generating the INVITE information according to the new rule.

Go back to the flowchart in Figure 8.

As described above, the INVITE information generated by the INVITE generation unit 303 is output to the conversion device SIP control unit 302, and the IP packet transmission/reception unit 301 IP-packages the INVITE information and transmits it to the IP network 4 (step ST805). This concludes the operation of step ST3 of Fig. 6 using Fig. 8 .

When the INVITE information is transmitted to the IP network 4 in step ST3 of FIG. 6, as shown in FIG. 6, the INVITE information is received on the SIP server 1, and the destination telephone number based on the information is 1002 (refer to FIG. 10). Therefore, the SIP server 1 transfers the INVITE information to the VoIP terminal 22 holding the telephone number of 1002 (step ST4 of Fig. 6).

After the INVITE message is received by the SIP server 1, the VoIP terminal 22 transmits a tentative response indicating the end user's call, that is, 180 Ringing response information, to the SIP server 1 (step ST5 of FIG. 6), and the SIP server 1 responds. Transfer to the acoustic symbol conversion device 3 (step ST6 of Fig. 6). Further, each SIP entity (SIPentity), that is, in this case, when the VoIP terminal 22 and the SIP server 1 operate in accordance with the SIP, the response is forwarded to the request source, that is, the acoustic symbol conversion device 3.

The acoustic symbol converting means 3, after receiving the 180 Ringing response message, generates 180 Ringing response information as a response corresponding to the INVITE information (step ST2 of FIG. 6) received by the VoIP terminal 21 side, and transmits the response to the SIP server 1 (Step ST7 in Fig. 6), the SIP server 1 transfers the 180 Ringing response information received by the acoustic symbol conversion device 3 to the VoIP terminal 21 (step ST8 in Fig. 6). Steps ST7 and ST8 of Fig. 6 of the drawings also operate in accordance with the provisions of SIP.

Next, when the user of the VoIP terminal 22 responds to the call, the VoIP terminal 22 transmits a 200 OK response message to the SIP server 1 (step ST9 of FIG. 6), and the response is forwarded to the acoustic symbol conversion device 3 (step of FIG. 6) ST10). When the SIP server 1 receives the 200 OK response message, the acoustic symbol conversion device 3 generates a 200 OK response message as a response to the INVITE information (step ST2 of FIG. 6) received by the VoIP terminal 21 side, and transmits the message to The SIP server 1 (step ST11 of Fig. 6), the SIP server 1 transfers the response to the VoIP terminal 21 (step ST12 of Fig. 6).

Here, the audio signal conversion device 3 of FIG. 11 receives the 200 OK response message from the SIP server 1 (refer to step ST10 of FIG. 6), and generates A flow chart for explaining the operation until the 200 OK response message of the VoIP terminal 21 is sent to the SIP server 1 (step ST11 in FIG. 6).

The operation of steps ST10 to ST11 of Fig. 6 will be described below with reference to Figs. 11 and 4 .

When the IP packet transmission/reception unit 301 receives the IP packet in which the SIP information is stored, the IP packet is judged to be the SIP information by the UI number, and the SIP information is output to the conversion device SIP control unit 302 (step ST1101).

The conversion device SIP control unit 302 determines whether or not the information input by the IP packet transmission/reception unit 301 in step ST1101 is 200 OK response information (step ST1102). Also, whether the input information is 200 OK response information can be judged by the first line of the information. Specifically, when the value after the "SIP/2.0" of the first header of the input information is "200", the conversion device SIP control unit 302 determines 200 OK response information (refer to FIG. 10).

When the non-200 OK response information is judged in step ST1102 (NO at step ST1102), the following processing is skipped and the processing is terminated.

When it is determined in step ST1102 that the 200 OK response information is present (YES in step ST1102), the conversion device SIP control unit 302 outputs the input 200 OK response information to the 200 OK generation unit 304 (step ST1103).

When the 200 OK response information is input by the conversion device SIP control unit 302, the 200 OK generation unit 304 generates new 200 OK response information (step ST1104).

In step ST1104, the 200 OK generating unit 304 generates a portion of the newly generated 200 OK response information in addition to the SDP body portion in accordance with the normal SIP rule. Regarding the SDP body part, the SDP in the input 200 OK response message The main body unit is added with the following changes to generate the SDP main unit.

Specifically, first, the main body unit SDP in the input 200 OK response information describes the coding type information of the audio codec included in the VoIP terminal 22, and the category and the INVITE information received by the VoIP terminal 21 ( When the coding type information of step ST2) of FIG. 6 is different, the 200 OK generation unit 304 rewrites the coding type information of the generated audio codec of the 200 OK response message to the INVITE trusted by the VoIP terminal 21. The coding category information of the information is in the same category. On the other hand, the coding type information in the input 200 OK response message is not rewritten when it is the same as the coding type information of the INVITE information received by the VoIP terminal 21.

Further, the 200 OK generating unit 304 changes the audio packet trusted address and the UI information of the SDP main unit to the IP address and the UI number to which the device is trusted.

Here, Fig. 12 is a view showing an example of the input 200 OK response information of the 200 OK generating unit 304 in step ST1104 of Fig. 11.

Fig. 13 is a diagram showing an example of the output 200 OK response information of the 200 OK generating unit 304 in step ST1104 of Fig. 11.

Further, Fig. 12 and Fig. 13 show an example in which the IP address of the VoIP terminal 22 is "192.168.0.22", and other addresses such as the address of the SIP server 1, the IP address of the VoIP terminal 21, and the acoustic symbol conversion device 3 The IP addresses are shown in Figure 9 and Figure 10, respectively, as "192.168.0.1", "192.168.0.21", and "192.168.0.2".

Further, the audio codec included in the VoIP terminal 21 is also set to the μ-law PCM method as shown in Fig. 9. The audio codec included in the VoIP terminal 22 is set to the CS-ACELP mode.

In the output 200 OK response information of the 200 OK generating unit 304 shown in Fig. 13, except for the SDP main unit, the respective heads from the first line to the eleventh line are generated according to the usual SIP regulations. In the SDP main unit, the following changes are added to the SDP main unit that inputs the 200 OK response information in FIG.

First, the 200 OK generating unit 304 sets the address of the second line (that is, the line of o=) of the SDP body unit for indicating the trusted IP address of the audio packet and the line of the fourth line (that is, the line of c=). The IP address "192.168.0.22" of the VoIP terminal 22 is changed to the IP address "192.168.0.2" of the acoustic symbol conversion device 3. In addition, the 200 OK generation unit 304 sets the sixth line of the SDP main unit to "m=audio 42104 RTP/AVP 0", and designates the number of the audio symbol conversion device 3 to be "42104", and uses the communication. The coding type information of the audio codec is changed to mean "0" of the μ rule PCM method. Further, the 200 OK generation unit 304 sets the seventh line of the SDP main unit to "a=rtpmap:0 PCMU/8000", and sets the attribute of the μ rule PCM method. The 200 OK generation unit 304 does not change the input 200 OK response information for other descriptions, and is similar to Fig. 12 .

Go back to the flowchart in Figure 11.

The 200 OK generation unit 304 transfers the 200 OK response information generated as described above to the IP packet transmission/reception unit 301 via the conversion device SIP control unit 302 (step ST1105).

Here, the conversion device SIP control unit 302 outputs the audio packet relay information to the IP packet transmission/reception unit 301 (step ST1106). The audio packet relay information is based on the 200 OK response information input by the 200 OK generating unit 304 (see According to Fig. 12), the information of the SDP main unit is outputted by outputting 200 OK response information (see Fig. 13), input INVITE information of the INVITE generating unit 303 (see Fig. 9), and outputting INVITE information (see Fig. 10). The information of the SDP main body is specifically an IP address and a 埠 number (transmission IP address = 192.168.0.21, transmission 埠 number: 39642, trusted 埠 number: 42102) for performing audio packet transmission and reception with the VoIP terminal 21 and The coding scheme type (μ rule PCM method), the IP address and the 埠 number (transmission IP address: 192.168.0.22, transmission 埠 number: 65520, trusted 埠 number: 42104) and coding method for transmitting and receiving the audio packet with the VoIP terminal 22. Category (CS-ACELP method).

The IP packet transmission/reception unit 301 stores the audio packet relay information (step ST1107), and performs IP packetization on the 200 OK response information input by the conversion device SIP control unit 302 in step ST1105, and transmits the packet to the IP network 4 to SIP server 1 (step ST1108).

As described above, the operation of step ST10 and step ST11 of Fig. 6 will be described using Fig. 11 .

In step ST11 of Fig. 6, when the acoustic symbol converting means 3 generates 200 OK response information and transmits it to the SIP server 1, the SIP server 1 transfers the 200 OK response information received by the acoustic symbol converting means 3 to the VoIP terminal 21 ( Step ST12) of Fig. 6.

When the VoIP terminal 21 receives the 200 OK response message sent by the SIP server 1 in step ST12 of FIG. 6, the ACK message is sent to the SIP server 1 (step ST13 of FIG. 6). ). Further, each SIPentity operates in accordance with the SIP, and thus the ACK information and the INVITE information are transferred through the same path. Therefore, the SIP server 1 is trusted to The ACK information is transferred to the acoustic symbol conversion device 3 (step ST14 of Fig. 6).

When the audible symbol converting means 3 receives the ACK information from the VoIP terminal 21, it generates ACK information with respect to the 200 OK response message of the VoIP terminal 22 and transmits it to the SIP server 1 (step ST15 of Fig. 6), and the SIP server 1 This ACK information is transferred to the VoIP terminal 21 (step ST16 of Fig. 6).

By the above sequence of completing the call of the SIP, the VoIP terminal 21 and the VoIP terminal 22 start the transmission of the acoustic packet (step ST17, step ST18 of Fig. 6).

As shown in FIGS. 10 and 13, the SDP main unit that transmits the INVITE information and the 200 OK response information to the VoIP terminals 21 and 22 by the acoustic symbol conversion device 3 specifies the acoustic symbol conversion as the acoustic packet received IP address. Since the IP address of the device 3, the VoIP terminal 21 and the VoIP terminal 22 transmit the audio packet to the acoustic symbol conversion device 3.

In this manner, after the sequence of call completion is completed, the audio packet communication between the VoIP terminal 21 and the VoIP terminal 22 (steps ST17 and ST18 in FIG. 6), the VoIP terminals communicate with each other without passing through the SIP server 1.

Fig. 14 is a flow chart for explaining the operation of the acoustic symbol conversion device 3 when receiving an acoustic packet from the VoIP terminal 21 and the VoIP terminal 22.

Hereinafter, description will be made using Figs. 14 and 4.

First, the IP packet transmission/reception unit 301 receives the audio packet received by the VoIP terminal 21 and the VoIP terminal 22 (step ST1401).

As described above, the IP packet transmission/reception unit 301 is previously used by the conversion device SIP control unit 302 as the audio packet relay information, and is stored with the VoIP terminal. 21 IP address and transmission number for sending and receiving audio packets (transmission IP address: 192.168.0.21, transmission number: 39642, trusted number: 42102) and coding mode category (μ law PCM method), and VoIP terminal 22 The IP address, the 埠 number (the IP address of the delivery IP address: 192.168.0.22, the delivery number: 65520, the trusted number: 42104) and the coding mode category (CS-ACELP mode) for the transmission and reception of the audio package. Based on the audio packet relay information, the IP packet transmission/reception unit 301 determines whether or not the audio packet is received by the VoIP terminal 21 (step ST1402). Specifically, the IP packet transmission and reception unit 301 confirms the IP address of the IP address of the IP header of the trusted audio packet. When the IP address of the VoIP terminal 21 is 192.168.0.21, it is determined to be from VoIP. The envelope packet of the terminal 21 is then checked for the destination number of the destination in the UDP header of the trusted audio packet, which is 42102 and is determined to be an audio packet. Further, the VoIP terminal 21 sets the destination number of the transmission audio packet to 42102. As described above, the audio symbol conversion device 3 rewrites the description in the SIP information to the device itself to be trusted in the call control sequence. number.

When it is determined in step ST1402 that the audio packet is received by the VoIP terminal 21 (YES in step ST1402), the IP packet transmission/reception unit 301 extracts the acoustic symbol in the packet, and outputs the acoustic symbol in the extracted packet to the acoustic symbol conversion. Part a 305 (step ST1403).

The acoustic symbol conversion unit a 305 changes the μ-law PCM symbol to the CS-ACELP symbol and sends it back to the IP packet transmission/reception unit 301. The IP packet transmission/reception unit 301 IP-packs and transmits the acoustic symbol sent back from the acoustic symbol conversion unit a 305. The VoIP terminal 22 is reached (step ST1404).

When it is determined in step ST1402 that the audio packet is not received by the VoIP terminal 21 (NO at step ST1402), steps ST1403 and ST1404 are skipped.

The IP packet transmission/reception unit 301 determines whether or not the audio packet is received by the VoIP terminal 22 based on the audio packet relay information (step ST1405). Specifically, in the same manner as step ST1402, the IP packet transmission/reception unit 301 confirms the IP address of the IP address of the IP header of the trusted audio packet, which is the transmission IP address of the VoIP terminal 22, that is, 192.168.0.22. It is determined that the envelope packet from the VoIP terminal 22 is confirmed by the destination number of the destination in the UDP header of the trusted audio packet. When it is 42104, it is determined to be an audio packet.

When it is determined in step ST1405 that the audio packet is received by the VoIP terminal 22 (YES in step ST1405), the IP packet transmission/reception unit 301 extracts the acoustic symbol in the packet, and outputs the acoustic symbol in the extracted packet to The acoustic symbol conversion unit b 306 (step ST1406).

The acoustic symbol conversion unit b 306 changes the CS-ACELP symbol to the μ-law PCM symbol and sends it back to the IP packet transmission/reception unit 301. The IP packet transmission/reception unit 301 IP-packs and transmits the acoustic symbol sent back by the acoustic symbol conversion unit b 306. The VoIP terminal 21 is reached (step ST1407).

When it is determined in step ST1405 that the audio packet is not received by the VoIP terminal 22 (NO at step ST1405), step ST1406 and step ST1407 are skipped.

As explained above, a call between the VoIP terminal 21 including the audio codec of different coding classes and the VoIP terminal 22 is implemented.

Moreover, the VoIP terminal 21 and the VoIP terminal 22 include the same coding category In the codec, the IP packet transmission/reception unit 301 can recognize the audio packet relay information supplied from the conversion device SIP control unit 302. Therefore, the received audio packet is not output to the acoustic symbol conversion unit a 305 or the acoustic symbol conversion unit b 306. The audio packet received by the VoIP terminal 21 is forwarded to the VoIP terminal 22, and the audio packet received by the VoIP terminal 22 is forwarded to the VoIP terminal 21.

Further, when the SIP server 1 wants to transfer the SIP information received by the VoIP terminals 21 to 2N and the acoustic symbol conversion device 3, it is necessary to maintain the telephone number information and the address information of each terminal.

Usually, the VoIP terminals 21 to 2N notify the SIP server 1 of the information of the own terminal by receiving the REGISTER information.

The VoIP terminals 21 to 2N are generated by the registration information generating unit 203 shown in Fig. 2 to generate REGISTER information for normal terminal information notification. The symbol conversion device registration information generation unit 204 generates REGISTER information for information notification by the acoustic symbol conversion device 3.

Fig. 15 is an explanatory diagram showing an example of REGISTER information generated by the symbol conversion device registration information generating unit 204 of the VoIP terminal 21.

The To header of the 5th line in Fig. 15 contains a SIP-URI called Address-of-Record. When the SIP terminal sends a message to another terminal, the address-of-Record of the specified destination terminal is sent. Usually, the Address-of-Record includes the telephone number of the terminal, and becomes "the telephone number of the registration target terminal @SIP server area" or "the telephone number of the login target terminal @SIP server IP address".

Moreover, the Contact header of the 8th line contains the address information of the terminal, which is also called the SIP-URI of the contact address. Usually, the connection bit The address includes the IP address of the terminal, and becomes the "IP address of the terminal to be registered @ the IP address of the terminal to be registered". As shown in Fig. 15, the telephone number of the To header at the fifth line and the telephone number of the Contact header set in the eighth line are converted to "1001" which is the telephone number of the VoIP terminal 21. 2001". Further, the UP address set in the Contact header of the eighth line is the IP address of the acoustic symbol conversion device 3. The SIP server 1 that has received the REGISTER information associates the Address-of-Record in the information with the connection address setting, and recognizes the telephone number "2001" as the telephone number of the acoustic symbol conversion device 3. Further, the parts other than the To header and the Contact header are generated according to the usual SIP regulations.

As described above, according to the first embodiment, in the audio communication system including the SIP server 1, the plurality of VoIP terminals 21 to 2N, and the acoustic symbol conversion device 3 for converting the audio medium between the VoIP terminals 21 and 2N, The audio symbol conversion device 3 includes a plurality of telephone numbers. When the VoIP terminals 21 to 2N transmit to other VoIP terminals, the telephone number of the transmission target is converted into the telephone number held by the acoustic symbol conversion device 3 according to a certain setting rule, and the telephone number is sent at the same time. The telephone number converted from the telephone number of the terminal itself according to the same rule as above is registered as the telephone number of the acoustic symbol conversion device 3, and the acoustic symbol conversion device 3 is notified by the same rule as the VoIP terminal 21~2N. The telephone number of the incoming call is inversely transformed and the reverse converted telephone number is sent as the originating telephone number, and media information including the coding category included in the VoIP terminals 21~2N is generated, and the SIP information including the media information is sent to VoIP terminals 21~2N, so even if VoIP terminals including different audio codecs exist on the same network, VoIP terminals Audio communication can also In the media communication system, the operation of the SIP server 1 and the acoustic symbol conversion device 3 is not required, and the processing load of the SIP server can be reduced.

Embodiment 2.

In the first embodiment, the telephone number of the acoustic symbol conversion device 3 is registered in the SIP server 1 by the VoIP terminals 21 to 2N. In the second embodiment, the embodiment in which the acoustic symbol conversion device 3 itself registers the telephone number is described. .

Figure 16 is an explanatory diagram showing the internal configuration of the VoIP terminals 21 to 2N in the media communication system according to the second embodiment of the present invention.

The VoIP terminals 21 to 2N shown in Fig. 16 are different from the VoIP terminals 21 to 2N described in the second embodiment of the first embodiment, except that the symbol conversion device registration information generating unit 204 is not included, and other configurations are the same as those of the second figure. Since the VoIP terminals 21 to 2N are the same, the same components are denoted by the same reference numerals, and the filtering is repeated.

In the second embodiment, the VoIP terminals 21 to 2N do not perform the registration information generation of the acoustic symbol conversion device 3 and the transmission of the REGISTER information. Further, in the second embodiment, the VoIP terminals 21 to 2N transmit the REGISTER information to the acoustic symbol conversion device 3 instead of the SIP server 1.

The other operations of the VoIP terminals 21 to 2N are the same as those of the VoIP terminals 21 to 2N described in the first embodiment, and thus the overlapping description will be omitted.

Figure 17 is an explanatory diagram showing the internal configuration of the acoustic symbol conversion device 3 in the media communication system according to the second embodiment of the present invention.

The acoustic symbol conversion device 3 shown in Fig. 17 is different from the acoustic symbol conversion device 3 described in Fig. 4 in the first embodiment, and is different in that it includes a transfer REGISTER generating unit 307 and a device REGISTER generating unit 308. The other configurations are the same as those of the acoustic symbol conversion device 3 of Fig. 4, and therefore, the same components are denoted by the same reference numerals, and the description thereof will not be repeated.

The transfer REGISTER generation unit 307 generates REGISTER information when the REGISTER information from the VoIP terminals 21 to 2N is transferred.

The own device REGISTER generating unit 308 generates REGISTER information for its own device.

The operation is the same as the acoustic symbol conversion device 3 described in the first embodiment except that the VoIP terminals 21 to 2N receive the REGISTER information.

When the VoIP terminals 21 to 2N receive the REGISTER information, the IP packet transmission/reception unit 301 outputs the information to the conversion device SIP control unit 302, and the conversion device SIP control unit 302 outputs the information to the transfer REGISTER generation unit 307 and the own device REGISTER generation unit 308. .

The transfer REGISTER generation unit 307 generates a new header information based on the REGISTER information input by the conversion device SIP control unit 302, and generates a new header information, and outputs the new REGISTER information to the conversion device SIP control unit 302.

The device REGISTER generating unit 308 generates a telephone number by the same rule as the VoIP terminals 21 to 2N based on the REGISTER information input by the conversion device SIP control unit 302, based on the telephone number included in the information, by the number. The new device REGISTER information is registered as the own device telephone number, and is output to the conversion device SIP control unit 302.

The conversion device SIP control unit 302 outputs the REGISTER information input by the transfer REGISTER generation unit 307 and the own device REGISTER generation unit 308 to the IP packet transmission/reception unit 301, and the IP packet transmission/reception unit 301 transmits them. The REGISTER message is sent to the SIP server 1.

Fig. 18 is a view showing an example of REGISTER information received by the VoIP terminal 21 in the acoustic symbol conversion device 3 in the second embodiment.

Fig. 19 is a view showing an example of the REGISTER information generated by the transfer REGISTER generating unit 307 in the second embodiment.

Compared with Fig. 18, the information in Fig. 19 is added to the Via first head of the second line and the Record-Route first head of the fourth line, and the field value of Max-Forwards of the fifth line is reduced from 70 to 69. They are added and changed according to the SIP regulations when relaying information.

Fig. 20 is an explanatory diagram showing an example of REGISTER information generated by the own device REGISTER generating unit 308 when the VoIP terminal 21 receives the REGISTER information shown in Fig. 18 by the VoIP terminal 21.

In the 20th figure, the telephone number "1000" after the telephone number "1001" to be registered in the 18th figure is set as the telephone number to be registered, and is set to the first head and the 5th line of the 4th line. To first head, the eighth line of Contact first head.

Further, the IP address "192.168.0.2" of the acoustic symbol conversion device 3 to be registered is set in the Contact header of the eighth line.

The SIP server 1 can recognize "1001" as the telephone number of the VoIP terminal 21 and recognize "2001" as the telephone number of the acoustic symbol conversion device 3 by receiving the REGISTER information shown in Figs. 19 and 20, respectively. number.

As described above, according to the second embodiment, the SIP server 1 can convert the VoIP terminals 21 to 2N and the acoustic symbols. The telephone number and IP address of the device 3 are registered, so even if the VoIP terminals 21~2N including different audio codecs exist on the same network, the audio communication between the VoIP terminals 21~2N can be performed, and the media communication system does not need to The operation of the SIP server 1 and the acoustic symbol conversion device 3 can reduce the processing load of the SIP server.

Further, in the first embodiment and the second embodiment, the REGISUTER information is transmitted from the VOIP terminals 21 to 2N or the acoustic symbol conversion device 3, but the VoIP terminals 21 to 2N and the acoustic symbol conversion device 3 do not transmit the REGISTER information. It is also possible to manually set the registration information of the VoIP terminals 21 to 2N and the acoustic symbol conversion device 3 to the SIP server 1 by hand.

Further, in the telephone number conversion rule set in the first embodiment and the second embodiment, the telephone numbers of the VoIP terminals 21 to 2N are set to 1001, 1002, ‧‧‧, 1000+N, and the telephone number of the acoustic symbol conversion device 3 is set. It is assumed that 2001, 2002, ‧‧‧, and 200N, and the telephone number of the VoIP terminals 21 to 2N plus 1000 is set as the telephone number of the acoustic symbol conversion device 3, but other conversion rules may be used. The acoustic symbol conversion device 3 includes N telephone numbers that correspond one-to-one with the telephone numbers of the VoIP terminals 21 to 2N, and the N telephone numbers and other numbers may not be repeated.

Further, for example, the VoIP terminal including the μ rule PCM method is set to the 1000 series, and the telephone number of the VoIP terminal including the CS-ACELP method is set to the 2000 series, and it is possible to distinguish by the audio codec type included in the VoIP terminal. Audio call system with telephone number.

At this time, when the VoIP terminal 21 to 2N and the VoIP terminal to be transmitted include the same type of audio codec as the own terminal, the telephone number conversion is not performed. Send a letter too.

In this manner, the SIP server 1 transfers the information transmitted by the VoIP terminals 21 to 2N to the terminal of the transmission target instead of the acoustic symbol conversion device 3. Therefore, the establishment of the audio communication between the usual VoIP terminals can reduce the burden on the acoustic symbol conversion device 3.

Further, in the first and second embodiments, the audio codecs included in the VoIP terminals 21 to 2N are set to the μ-law PCM method and the CS-ACELP method. However, the present invention is not affected by the audio coding type and may include different audio codes. Audio communication is performed between the VoIP terminals of the decoder.

Further, the media communication system of the present invention, as shown in Fig. 21, is also applicable to an audio communication system that requires conversion of an acoustic symbol in a conversation between VoIP terminals 21 to 2N of different networks.

As shown in Fig. 21, the router 7 performs routing control of transmitting IP packets between the devices connected to the IP network 4 and the devices connected to the IP network 6.

The other components perform the same operation as the audio call system shown in Fig. 1, and the call between the VoIP terminals 21 to 2N including the different audio codecs can be realized.

Further, the prior art discloses a method disclosed in Patent Document 1, an audio coding method of an audio codec included in the VoIP terminals 21 to 2M to which the IP network 4 is connected, and a VoIP terminal 2M+1 connected to the IP network 6. The audio encoding of the audio codec included in ~2N can also be used for calls.

However, according to the audio communication system of the present invention, the voice codec included in the VoIP terminals 21 to 2M connected to the IP network 4 can also implement a call when the coding modes of the audio codecs are different, which has the advantage.

Further, the hardware configurations of the VoIP terminals 21 to 2N and the acoustic symbol conversion device 3 according to the second embodiment are the same as those described in the third and fifth embodiments of the first embodiment.

The transfer REGISTER generation unit 307 and the own device REGISTER generation unit 308 are realized by executing processing circuits such as the CPU 220 and the system LSI stored in the programs such as the HDD 222 and the memory 223.

Further, a plurality of processing circuit combinations may perform the above functions.

Further, the VoIP terminals 21 to 2N according to the first embodiment of the present invention are configured as shown in Fig. 2, and the VoIP terminals 21 to 2N include the telephone number conversion unit 202 and the terminal SIP control unit 205, and the above effects can be obtained.

Further, within the scope of the invention, the present invention can be freely combined with the respective embodiments, or the modifications of any of the constituent elements of the respective embodiments, or the omission of any constituent elements in the respective embodiments.

[Industrial Availability]

According to the configuration of the audio terminal of the present invention, even when an audio terminal including different audio codecs exists on the same network, audio communication between VoIP terminals can be performed, and there is no need to connect the call control server and the acoustic symbol conversion device. In the mobile communication system, the processing load of the call control server can be reduced, so that an audio terminal that performs an audio call through the network can be applied.

1‧‧‧SIP server

21~2N‧‧‧VoIP terminal

3‧‧‧Audio symbol conversion device

4‧‧‧IP network

Claims (7)

A media communication system that connects a plurality of audio terminals, a call control server, and an audio symbol conversion device that converts media types for communication between the plurality of terminals via a network, and causes the plurality of audio terminals to perform The audio caller includes: a telephone number conversion unit that converts a telephone number of the audio terminal to be transmitted to a telephone number of the audio symbol conversion device according to a set rule; and a terminal SIP The control unit generates the credit information by the telephone number converted by the telephone number conversion unit, and transmits the information to the acoustic symbol conversion device via the call control server; the acoustic symbol conversion device includes an INVITE generation unit, and the transmission source When the audio terminal receives the information of the credit transmission through the call control server, inversely transforms the destination telephone number in the received credit information according to the set rule and sets the destination telephone number as the sender. Telephone number of the audio terminal, which will be credited as above The trusted address of the audio packet in the interest is set to its own address to generate a new credit information; and the IP packet sending and receiving unit transmits the credit information generated by the INVITE generating unit to the call control server through the call control server. To the audio terminal of the above-mentioned sender. For example, the media communication system of claim 1 of the patent scope, wherein the above sound ends Further, the symbol conversion device registration information generation unit converts the telephone number of the own terminal into a telephone number of the audio symbol conversion device according to the rule set above, and generates an information notification of the acoustic symbol conversion device in which the telephone number is set. Use the REGISTER information and log in to the above call control server. The media communication system according to claim 1, wherein the audio terminal further includes: a registration information generating unit that generates REGISTER information for information notification in which a telephone number of the terminal is set, and notifies the acoustic symbol conversion device; The acoustic symbol conversion device further includes: a self-device REGISTER generating unit that generates a new REGISTER for the telephone number included in the REGISTER information received by the audio terminal according to the set rule and converts the telephone number contained in the device as its own device Information, registered to the above call control server. An audio terminal of a media communication system, wherein a plurality of audio terminals, a call control server, and an audio symbol conversion device for converting a media type for communicating between the plurality of terminals are connected via a network, and the plurality of audio signals are connected The audio terminal includes an audio caller, and the audio terminal includes a telephone number conversion unit that converts a telephone number of the audio terminal to be transmitted to a telephone number of the audio symbol conversion device according to a set rule; The terminal SIP control unit generates the credit information by the telephone number converted by the telephone number conversion unit, and transmits the information to the acoustic symbol conversion device via the network. The audio terminal of claim 4, wherein the symbol conversion device registration information generation unit converts the telephone number of the terminal to the telephone number of the audio symbol conversion device according to the rule set above, and generates the setting The information of the above-mentioned acoustic symbol conversion device of the telephone number is notified of the REGISTER information, and is registered in the call control server connected through the above network. An acoustic symbol conversion device for a media communication system, wherein a plurality of audio terminals, a call control server, and an acoustic symbol conversion device for converting a media type for communicating between the plurality of terminals are connected via a network, The plurality of audio terminals perform an audio caller; the audio symbol conversion device includes: an INVITE generating unit, wherein when the audio terminal of the transmitting source receives the information of the credit through the call control server, the trusted message is received according to the set rule The destination telephone number in the credit information is inversely transformed and is set as the telephone number of the audio terminal to be transmitted, and the trusted address of the audio packet in the credit information is set as its own address. The IP packet sending and receiving unit transmits the credit information generated by the INVITE generating unit to the audio terminal of the transmitting target through the call control server. For example, the acoustic symbol conversion device of claim 6 of the patent scope includes: The REGISTER generating unit itself generates a new REGISTER message by converting the telephone number included in the REGISTER information received by the audio terminal through the network and generating the new REGISTER information according to the rules set above. The above call control server connected through the above network.