JP3992799B2 - Voice communication device - Google Patents

Description

[0001]
BACKGROUND OF THE INVENTION
The present invention relates to a voice communication apparatus used in a voice relay system in which voice data is exchanged between voice communication apparatuses, which are terminals, and conversation is possible.
[0002]
[Prior art]
As a voice relay system for sending and receiving voice data between personal computers (personal computers) connected to a LAN (Local Area Network), for example, Nikkei Data Pro Datacom, 1995, p. Talkware of Hitachi, Ltd. described in 681 is known. An example of a voice relay system for making a call using the Internet is, for example, Internetworking, 1996.69, p. A Cool Talk of Netscape Communications, described in 14-15, is known.
[0003]
When audio data is transmitted / received over a communication network, the audio data is delayed and reaches the other party due to the load on the communication network. If playback is performed immediately when audio data is received, sound interruption will occur if the audio data arrives late. Therefore, in Talkware and Cool Talk, sound interruption is prevented by accumulating and reproducing a little received audio data.
[0004]
In particular, in Talkware, the amount of audio data stored is automatically calculated and audio reproduction processing is performed.
[0005]
[Problems to be solved by the invention]
However, if the network transmission rate is slow, the amount of voice data stored increases, so the voice delay increases. When the voice delay increases, some received operators may feel uncomfortable with the received voice. In addition, since there is an upper limit for the amount of audio data stored, if a transmission delay of the audio data that exceeds this upper limit occurs, sound interruption will occur. In the sound interruption, since the received voice is intermittent, the received voice may be uncomfortable depending on the receiving operator.
[0006]
When the user feels uncomfortable with the sound delay or feels uncomfortable with the sound interruption, there are individual differences depending on the operator. Therefore, the conventional method of automatically calculating the amount of audio data stored and performing the audio reproduction process has a problem in that it is difficult to provide audio with better sound quality according to the sensitivity of the operator. It was. Depending on the operator, there may be a case where it is desired to reduce the delay of the sound because the sound may be interrupted to some extent, or a case where it is desired to eliminate the sound because the delay may be increased.
[0007]
An object of the present invention is to provide an audio communication apparatus that can provide audio with better sound quality.
[0008]
[Means for Solving the Problems]
(1) In order to achieve the above object, the present invention provides a voice communication device that relays voice data, a storage unit that temporarily stores voice data transmitted from another voice communication device after receiving the voice data; Reproducing means for reproducing the audio data sent from the accumulating means by the audio output means, setting value varying means for varying the setting value of the amount of accumulated audio data accumulated in the accumulating means during voice relay, and A display unit that displays at least a part of the set value of the audio data; and an input unit that is instructed by the user to input the set value of the audio data. The display unit includes the set value of the audio data. A setting value of a voice data storage amount representing the amount of voice data stored in the storage means, and the input means receives the voice data storage amount by a user, and the setting value variable It is a set of the audio data accumulation amount from the input means is set as a new audio data accumulation amount In addition, the set value of the audio data is further input to the storage unit at once, or an audio data length representing the length of audio data to be output from the storage unit at once, or the audio data Including at least one of the encoding methods, the audio data length or at least one of the audio data encoding methods is displayed by the display means, and new information is input via the input means, and the set value The input new information is set by the variable means. It is what I did. With this configuration, the set value of the accumulated amount of voice data can be changed during voice relay according to the user's judgment, so that it is possible to provide a voice that is easy to hear for the user, that is, a voice with better sound quality.
[0010]
( 2 ) In the above (1), preferably, control means for controlling an accumulation amount of audio data accumulated in the accumulation means is provided, and the control means has an accumulation amount of audio data accumulated in the accumulation means as a predetermined value. When the number is less, the voice data sent from another voice communication device is stored in the storage means, and then a part of the same voice data sent as the voice data is stored in the storage means again. is there. With such a configuration, sound interruption can be prevented, and sound quality of sound reproduced for preventing sound interruption can be improved.
[0011]
DETAILED DESCRIPTION OF THE INVENTION
Hereinafter, a voice relay system using a voice communication apparatus according to an embodiment of the present invention will be described with reference to FIGS.
First, the overall configuration of a voice relay system according to an embodiment of the present invention will be described with reference to FIG.
[0012]
The PBX gateway 200 is connected to the telephone line 10 and can accept calls received from the telephone line 10. The PBX gateway 200 is connected via a LAN (Local Area Network) 20 to personal computers (personal computers) 300 and 400 constituting the voice communication apparatus according to the present embodiment, and can exchange information via the LAN 20. The PBX gateway 200 can send voices heard from the telephone to the personal computers 300 and 400 and reproduce voice data sent from the personal computers 300 and 400 on the telephone line.
[0013]
Handsets 390 and 490 are connected to the personal computers 300 and 400, and voice input / output is possible using the handsets 390 and 490.
A setting screen 300A can be displayed on the display unit of the personal computer 300, and the setting value of the personal computer 300 can be changed while the setting screen 300A is displayed. The contents of the setting screen 300A and the setting change method will be described later with reference to FIG.
[0014]
Next, the hardware configuration of the PBX gateway in the voice relay system according to the embodiment of the present invention will be described with reference to FIG.
[0015]
A telephone interface 205 in the PBX gateway 200 is connected to the telephone line 10 and can recognize incoming calls. The telephone interface 205 is controlled by the telephone interface driver 210. The telephone interface driver 210 has an internal buffer (not shown), accumulates voice data sent from the telephone line 10 via the telephone interface 205, and sends it to the telephone line 10 via the telephone interface 205. Accumulate audio data.
[0016]
The keyboard driver 215 accepts input from the keyboard 220. The mouse driver 225 receives input from the mouse 230. Display on the display 235 is controlled by the display controller 240. A program or the like executed by the PBX gateway 200 is read from the disk 250 into the main memory 255 by the disk controller 245 and executed by the CPU 260. The PBX gateway 200 is connected to the LAN 20 that is a communication network by a communication interface 265.
[0017]
Next, the hardware configuration of a personal computer that is a voice communication device in the voice relay system according to the embodiment of the present invention will be described with reference to FIG. Here, the personal computer 300 will be described as an example, but the personal computer 400 has the same configuration.
[0018]
The voice input / output interface 305 is connected to the handset 390 and can input / output voice from the handset 390. The voice input / output interface 310 is controlled by the voice input / output interface driver 310. The audio interface driver 310 has an internal buffer (not shown), accumulates audio data sent from the handset 390 via the audio interface 305, and transmits audio data sent to the handset 390 via the audio interface 305. Accumulate.
[0019]
The keyboard driver 315 accepts input from the keyboard 320. The mouse driver 325 accepts input from the mouse 330. The display on the display 335 is controlled by the display controller 340. A program or the like executed on the personal computer 300 is read from the disk 350 into the main memory 355 by the disk controller 345 and executed by the CPU 360. The personal computer 300 is connected to the LAN 20, which is a communication network, via a communication interface 365.
[0020]
Next, the voice relay processing of the PBX gateway 200 in the voice relay system according to the embodiment of the present invention will be described using FIG. 4 and FIG.
[0021]
The voice relay program executed by the PBX gateway 200 is read from the disk 250 into the main memory 255 by the disk controller 245 and executed by the CPU 260. The voice relay program is installed in the disk 250 from a storage medium such as an FD or a CD-ROM.
[0022]
In step 410 of FIG. 4, the CPU 260 of the PBX gateway 200 normally enters a state waiting for an incoming call. When waiting for an incoming call, the voice relay program periodically checks whether there is an incoming call to the telephone interface 205 using the telephone interface driver 210 shown in FIG.
[0023]
When there is an incoming call, in step 420, the CPU 260 opens the audio file stored in the disk 250 and sends the audio data recorded in the file to the audio interface 205 using the telephone interface driver 210. Make a voice response. The content of this voice response is, for example, a content that announces that the extension number is to be pressed, such as “This is a PBX gateway. Please press the extension number”.
[0024]
In step 430, CPU 260 accepts an extension number. When the extension number is received, the voice relay program periodically checks the number input to the telephone interface 205 with the DTMF signal using the telephone interface driver 210 shown in FIG. 2, and the DTMF signal recognized by the telephone interface 205 is received. Is obtained from the telephone interface driver 210.
[0025]
When the recognized extension number is received, in step 440, the CPU 260 obtains an IP address using the extension number-IP address correspondence table 500 shown in FIG. 5, and sends a call message to the personal computer 300 connected by the LAN 20 or the like. send.
[0026]
Here, the configuration of the extension number-IP address correspondence table 500 will be described with reference to FIG.
As shown in FIG. 5, the extension number / IP address correspondence table 500 includes an extension number 510, a name 520, and an IP address 530. For example, the operator of the personal computer corresponding to the extension number 510 of “1234” is “taro-yamamoto” as described in the name 520 column, and the IP address 530 of the personal computer being used is “192”. .10.12 ". That is, when the extension number received in step 430 is “1234”, the corresponding IP address “192.168.1.12.” Is acquired. The extension number / IP address correspondence table 500 is stored in the main memory 255.
Further, in the personal computer that has received a call from the PBX gateway 200 in step 440, a call bell or a message is displayed to notify the user of the personal computer of the incoming call. When the user of this personal computer presses a button for accepting a call, a reception OK message is sent from the personal computer to the PBX gateway 200.
[0027]
In step 450, the CPU 260 determines whether or not the personal computer is receiving OK, and when the PBX gateway receives the receiving OK message, in step 460, the CPU 260 starts voice relay. If the reception is not OK, the CPU 260 hangs up the telephone in step 490.
[0028]
In step 460, the CPU 260 starts the voice relay. Details of the voice relay process will be described later with reference to FIGS.
[0029]
After starting the voice relay, the voice relay program of the CPU 260 uses the telephone interface driver 210 to periodically check the telephone disconnection.
In step 470, the CPU 260 determines whether the telephone disconnection is detected or whether a disconnect message is received from the personal computer 300. If the telephone disconnection is detected or a disconnection message is received from the personal computer 300, the CPU 260 ends the voice relay at step 480. In step 490, CPU 260 disconnects the telephone and returns to the incoming call waiting state in step 410.
[0030]
Next, voice relay processing when the personal computer 400 is called from the personal computer 300 which is a voice communication device in the voice relay system according to the embodiment of the present invention will be described with reference to FIGS.
[0031]
The voice relay program executed on the personal computer 300 is read from the disk 350 into the main memory 355 by the disk controller 345 shown in FIG. 3 and executed by the CPU 360. The voice relay program is installed on the disk 350 from a storage medium such as an FD or a CD-ROM.
[0032]
When starting the voice relay, the user inputs the “IP address” on the outgoing screen 300B shown in FIG. 7 and designates the other party to call, and then presses the “outgoing” button.
In step 610, CPU 360 confirms that the call button has been pressed.
Next, in step 620, the CPU 360 sends the outgoing message to the other personal computer 400.
[0033]
Next, in step 630, the CPU 360 sends a call message to the personal computer 400 connected via the LAN 20.
The personal computer 400 that received the call from the personal computer 300 in step 630 notifies the user of the personal computer of the incoming call by displaying a call bell or a message. Then, when the user of this personal computer presses a button for accepting a call, a reception OK message is sent from the personal computer 400 to the personal computer 300.
[0034]
In step 640, the CPU 360 determines whether or not the personal computer is reception OK. When the personal computer 300 receives the reception OK message, in step 650, the CPU 360 starts voice relay. On the other hand, if the reception is not OK, the process returns to the incoming call waiting state in step 610.
[0035]
In step 650, the CPU 360 starts voice relay. When the voice relay is started, the personal computer 300, 400 that is voice relaying displays the screen 300C for voice relay shown in FIG.
[0036]
In step 660, CPU 360 determines whether a disconnect button on screen 300C during voice relay has been pressed or whether a disconnect message from personal computer 340 has been received. If it is detected that the disconnect button has been pressed or if a disconnect message is received from the personal computer 400, in step 670, the CPU 360 ends the voice relay. Then, CPU 360 disconnects the phone and returns to the incoming call waiting state in step 610.
[0037]
Next, processing at the time of data transmission and data reception in the voice relay processing in the voice relay system according to the embodiment of the present invention will be described with reference to FIGS.
[0038]
When voice relay is started in step 460 of FIG. 4 or step 650 of FIG. 6, the PBX gateway 200 or the personal computer 300 performs data transmission processing for transmitting voice data to the other party (the personal computer 300 or the personal computer 400) shown in FIG. Data reception processing when the voice data sent from the other party shown in FIG. 10 is received is started. In the following description, a case where data is transmitted from the personal computer 300 to the personal computer 400 will be described.
[0039]
First, the data transmission process will be described with reference to FIG.
In step 910, the CPU 360 confirms the amount of audio data to be transmitted using the audio input / output interface driver 310.
Here, setting values for voice relay will be described with reference to FIG. As shown in FIG. 11, the setting contents and setting values are configured, and these setting contents and setting values are stored in the main memory 355. In the example shown in FIG. 11, “length of audio data” is set to “120” bytes (Bytes). The audio data is transmitted collectively for each predetermined number of bytes. Here, “the length of the audio data” indicates the length of the audio data to be transmitted at one time. An audio data length of 120 bytes corresponds to an audio signal having a length of about 30 mS. Further, the “audio data accumulation amount” is set to “400 to 500” bytes. Here, “sound data storage amount” indicates the amount of sound data stored in order to prevent sound interruption. This will be described in detail in steps 1030 and 1040 in FIG. The “encoding method” is set to “ADPCM”. “Volume” is set to “10” (arbitrary unit).
[0040]
In step 920, the CPU 360 determines that the amount of audio data confirmed in step 910 is larger than the setting value “120” bytes (length of audio data to be sent to the other party at a time) of “length of audio data” shown in FIG. If it does not satisfy the set value, the process returns to step 910 to repeat the confirmation of the audio data amount.
[0041]
When the audio data to be transmitted exceeds the set value of “length of audio data”, in step 930, the CPU 360 uses the audio input / output interface driver 310 to extract the audio data from the audio input / output interface 310.
[0042]
In step 940, the CPU 360 sends the extracted audio data to the partner computer.
[0043]
In step 950, the CPU 360 determines whether or not the voice relay is finished, and if not finished, repeats the steps 910 to 940.
[0044]
Next, processing at the time of data reception will be described with reference to FIG. The personal computer 400 performs data reception processing. Since the personal computer 300 and the personal computer 400 have the same configuration, the description will be made based on the configuration of the personal computer 300 shown in FIG.
[0045]
In step 1010, the CPU 360 receives the audio data sent in step 740 of FIG.
[0046]
Next, in step 1020, the CPU 360 confirms the amount of audio data stored in the audio input / output interface 310.
In the configuration of the personal computer 300 shown in FIG. 3, audio data received via the LAN 20 and the communication interface 365 is once taken into an internal buffer in the CPU 360. Thereafter, the audio data in the internal buffer of the CPU 360 is input to the internal buffer of the audio input / output interface driver 310. The capacity of the internal buffer of the audio input / output interface driver 310 is, for example, 4 KB, but audio data of about 400 to 500 bytes is always stored in the internal buffer. The voice input / output interface driver 310 sends voice data to the voice input / output interface 305 at regular time intervals, for example, every 30 ms, and reproduces sound from the handset 390. When there is no audio data stored in the internal buffer of the audio input / output interface driver 310, no sound is played from the handset 390, and the sound is cut off.
Therefore, in step 1020, the CPU 360 checks the amount of audio data stored in the audio input / output interface driver 310.
[0047]
Here, assuming that “400 to 500” KB is set as “sound data storage amount” shown in FIG. 11, in steps 1030 and 1040, the storage amount of the sound data in the sound input / output interface 310 is the same. Then, it is determined whether it is larger or smaller than the set maximum value of the voice data storage amount.
[0048]
In step 1030, the CPU 360 determines that the amount of audio data stored in the audio input / output interface driver 310 is less than the set maximum value of the audio data storage amount. In step 1040, the CPU 360 If it is determined that the amount of audio data stored in the output interface driver 310 is greater than the set minimum value of the audio data storage amount, in step 1060, one audio data is input from the internal buffer of the CPU 360 into the audio input. Input to the internal buffer of the output interface driver 310.
That is, when the process of step 1060 is performed, the amount of audio data stored in the internal buffer of the audio input / output interface driver 310 is within an appropriate range of 400 to 500 bytes. Therefore, as shown in FIG. 11, since the “length of audio data” is set to “120” bytes, 120 bytes of audio data for one time of audio data is sent from the internal buffer of the CPU 360 to the audio data. Input to the internal buffer of the input / output interface driver 310. On the other hand, since the voice input / output interface driver 310 sends 120 bytes of voice data to the voice input / output interface 305 at a constant cycle, the amount of voice data stored in the internal buffer of the voice input / output interface driver 310 is 400 to 400. An appropriate range of 500 bytes can be maintained. Therefore, the sound reproduced from the handset 390 will not be interrupted.
[0049]
Next, in step 1030, the CPU 360 determines that the accumulated amount of audio data in the audio input / output interface driver 310 is smaller than the set maximum value of the audio data accumulated amount. In step 1040, the CPU 360 If it is determined that the amount of sound data stored in the sound input / output interface driver 310 is less than the minimum value of the set amount of sound data, in step 1050, 1.5 times of sound data is stored in the CPU 360. Input from the internal buffer to the internal buffer of the audio input / output interface driver 310.
In other words, the processing in step 1050 is performed when the amount of audio data stored in the internal buffer of the audio input / output interface driver 310 is less than 400 bytes. Therefore, as shown in FIG. 11, since the “length of audio data” is set to “120” bytes, the audio data for 180 bytes in 1.5 times of the audio data is stored in the internal buffer of the CPU 360. To the internal buffer of the voice input / output interface driver 310. On the other hand, since the voice input / output interface driver 310 sends 120 bytes of voice data to the voice input / output interface 305 at regular intervals, the amount of voice data stored in the internal buffer of the voice input / output interface driver 310 gradually increases. Thus, it is possible to return to the proper range of 400 to 500 bytes.
In addition, in order to input audio data of 180 bytes in 1.5 times of audio data from the internal buffer of the CPU 360 to the internal buffer of the audio input / output interface driver 310, 120 bytes of data for the same time and the same 120 are used. Sends 60 bytes of audio data, half of the byte data. That is, the same data is sent for 60-byte audio data. As a result, the sound reproduced from the handset 390 by the voice input / output interface 305 is repeated for 60 bytes of the same sound, but this does not cause discomfort to those who use the handset 390. The first reason is that the 60-byte audio data has an extremely short time of about 15 mS. Further, since the sound reproduced at this time is the same as the previous sound, no unnaturalness occurs in human hearing. Here, it is conceivable to input pseudo audio data to the audio input / output interface 305. However, since the pseudo audio data is different from the sound reproduced before that, it is rather uncomfortable on human hearing. You will feel.
[0050]
Accordingly, the amount of audio data stored in the internal buffer of the audio input / output interface driver 310 can be gradually increased and restored to the appropriate range of 400 to 500 bytes. No cuts will occur.
[0051]
If the CPU 360 determines in step 1030 that the accumulated amount of audio data stored in the audio input / output interface driver 310 is greater than the maximum value of the set audio data accumulated amount, step 1010 is performed without doing anything. Return to receiving voice data.
That is, in this case, since the amount of audio data stored in the audio input / output interface driver 310 is 500 bytes or more, if the audio data is sent to the audio input / output interface driver 310 here, the audio data increases, It takes time to reproduce the data. That is, the delay of the voice data becomes large, and the other party's voice can be heard with a delay.
[0052]
As described above, in the audio data reproduction process, by setting the accumulated amount of audio data within a certain set value, even when the audio data arrives late, the audio can be reproduced without interruption, The delay exceeding the set amount can be eliminated.
[0053]
In the above description of FIGS. 9 and 10, the sound reproduction process in the personal computer has been described. However, in the case of the sound reproduction process in the PBX gateway 200, the sound input / output interface 305 described above is used as the telephone interface 205. The input / output interface driver 310 may be replaced with the telephone interface driver 210, and the PBX gateway 200 can perform the same audio reproduction process.
[0054]
Next, the voice relay setting process will be described with reference to FIGS.
[0055]
When the “Setting” button on the voice relay setting screen 300A shown in FIG. 1 is pressed, the voice relay setting process shown in FIG. 12 is performed.
[0056]
In step 1210, when recognizing that the “setting” button on the setting screen 300A has been pressed, the CPU 360 checks which setting has been changed and performs each process.
[0057]
In step 1220, the CPU 360 determines whether or not the value of the reproduction buffer has been changed. If the value of the reproduction buffer has been changed, in step 1230, the CPU 360 performs the voice relay shown in FIG. The “audio data storage amount” in the set value is changed to the set value. If the “audio data accumulation amount” has a set range, the set range is determined in advance. In the example shown in FIG. 11, the set range is set to “120” bytes. In the voice relay setting screen 300A, the lower limit value of the set value of “voice data storage amount” is set, so that “voice data storage amount” is 400 to 500 bytes.
In the present embodiment, the amount of audio data accumulated in the audio input / output interface driver 310 can be changed by simply changing the set value of “audio data accumulation amount” according to the method described with reference to FIG.
[0058]
When the operator of the personal computer 300 feels that the sound delay has increased while receiving a call from the other party, the operator displays a voice relay setting screen 300A on the display screen of the personal computer 300 as shown in FIG. Decrease the setting value of “Audio data accumulation amount”. As a result, the “sound data storage amount” can be reduced, and the delay amount of the reproduced sound can be reduced. Further, when sound interruption occurs, as shown in FIG. 1, the voice relay setting screen 300A is displayed on the display screen of the personal computer 300, and the setting value of “voice data accumulation amount” is increased. As a result, the “sound data accumulation amount” can be increased, so that it is possible to prevent interruption of the reproduced sound.
[0059]
If you feel uncomfortable with the sound delay or feel uncomfortable with the sound interruption, there are individual differences depending on the operator, and some operators may want to reduce the sound delay because sound interruption may occur. In some cases, it may be necessary to eliminate the sound interruption because the delay may be large, so it is possible to obtain a sound with better sound quality according to the sensitivity of the operator.
[0060]
Next, in step 1240, the CPU 360 determines whether or not the encoding method has been changed. If the encoding method has changed, in step 1250, the CPU 360 changes the encoding method.
Here, the encoding method is a method of changing the voice input from the handset 390 into a form that can be transmitted and received by the LAN 20. The encoding method is changed by changing the setting of the voice input / output interface 310 using the voice input / output interface driver 310.
[0061]
In step 1260, the CPU 360 determines whether or not the audio data length has been changed. If the audio data length has been changed, the CPU 360 changes “length of audio data” in step 1270. To do. By simply changing this set value, the length of the audio data to be transmitted can be changed by the processing at the time of audio data transmission described with reference to FIG.
[0062]
As described above, in this embodiment, the user can change the voice relay setting so that the sound can be cut off, or the person who wants to delay as much as possible or the person who does not like to cut off the sound because it can be delayed. Can be fulfilled.
[0063]
Next, with reference to FIG. 13, a case where the setting value of voice relay is changed using a terminal such as a telephone will be described.
[0064]
In the example shown in FIG. 12, since a personal computer is used as the voice communication device, the setting value can be changed using a display. However, in the case of a voice communication device such as a telephone without a display, the setting value is changed. A setting button or the like can be used, and a setting can be performed using a DTMF (Dual Tone Multi Frequency) signal generated when the telephone button is pressed.
[0065]
FIG. 13 is a flowchart illustrating a process in which the PBX gateway 200 detects a DTMF signal transmitted from a telephone connected to the telephone network 10 and changes the setting.
[0066]
In step 1310, the CPU 260 of the PBX gateway 200 repeatedly inquires the telephone interface driver 210 for a DTMF signal during voice relay.
[0067]
In step 1320, the CPU 260 determines whether or not a DTMF signal has been detected. When the DTMF signal is detected, the CPU 260 acquires the received DTMF signal from the telephone interface driver 210 in step 1330.
[0068]
Next, it is determined whether or not the acquired DTMF signal is a setting change command. Here, it is assumed that the setting change command is determined in advance. For example, when “400” is pressed after “#” is pressed, “audio data accumulation amount” is set to “400”. .
[0069]
If the DTMF signal is a setting change command, in step 1350, the CPU 260 changes the voice relay setting value shown in FIG.
[0070]
As described above, in the present embodiment, the setting of the voice data storage amount for voice relay can be set not only from the personal computer but also from the telephone side.
[0071]
【The invention's effect】
According to the present invention, it is possible to provide a sound that is easy for the user to hear, that is, a voice with better sound quality, based on the judgment of the user.
[Brief description of the drawings]
FIG. 1 is a system configuration diagram showing the overall configuration of a voice relay system according to an embodiment of the present invention.
FIG. 2 is a block diagram of a hardware configuration of a PBX gateway in a voice relay system according to an embodiment of the present invention.
FIG. 3 is a block diagram of a hardware configuration of a personal computer that is a voice communication device in the voice relay system according to the embodiment of the present invention.
FIG. 4 is a flowchart showing voice relay processing of a PBX gateway in the voice relay system according to the embodiment of the present invention.
FIG. 5 is an explanatory diagram of an extension number-IP address correspondence table possessed by a PBX gateway in a voice relay system according to an embodiment of the present invention.
FIG. 6 is a flowchart showing voice relay processing of a personal computer which is a voice communication device in the voice relay system according to the embodiment of the present invention.
FIG. 7 is an explanatory diagram of a call origination screen when making a call from a personal computer in a voice relay process of a personal computer that is a voice communication device in a voice relay system according to an embodiment of the present invention.
FIG. 8 is an explanatory diagram of a screen during voice relay in voice relay processing of a personal computer that is a voice communication device in the voice relay system according to the embodiment of the present invention.
FIG. 9 is a flowchart illustrating processing at the time of data transmission in voice relay processing in the voice relay system according to the embodiment of the present invention.
FIG. 10 is a flowchart for explaining processing at the time of data reception in voice relay processing in the voice relay system according to the embodiment of the present invention;
FIG. 11 is an explanatory diagram illustrating an example of setting values used in the voice relay processing in the voice relay system according to the embodiment of the invention.
FIG. 12 is a flowchart for explaining processing at the time of voice relay setting in voice relay processing in the voice relay system according to the embodiment of the present invention;
FIG. 13 is a flowchart illustrating processing when voice relay is set from a terminal such as a telephone in voice relay processing in the voice relay system according to the embodiment of the present invention.
[Explanation of symbols]
10 ... Telephone network
20 ... LAN (Local Area Network)
200 ... PBX gateway device
205 ... Telephone interface
210 ... Telephone interface driver
260, 360 ... CPU
300, 400 ... PC
305 ... Voice input / output interface
310 ... Voice input / output interface driver
390, 490 ... Handset

Claims (2)

In a voice communication device that relays voice data,
Storage means for temporarily storing after receiving the voice data sent from another voice communication device;
Reproduction means for reproducing the audio data sent from the storage means by means of an audio output means;
Set value variable means for changing the set value of the amount of voice data stored in the storage means during voice relay;
Display means for displaying at least a part of the set value of the audio data to the user;
An input means for instructing input of a set value of the audio data by a user;
The display means displays a set value of an audio data accumulation amount representing an amount of audio data accumulated in the accumulation means as the audio data set value,
The input means receives the voice data accumulation amount by a user,
The set value variable means sets the setting of the audio data accumulation amount from the input means as a new audio data accumulation amount ,
The set value of the audio data is further input to the storage unit at once, or an audio data length representing the length of audio data to be output from the storage unit at once, or the audio data encoding Including at least one of the methods,
At least one of the audio data length and the audio data encoding method is displayed by the display means, new information is input via the input means, and the new information input by the set value variable means Is set . The voice communication device according to claim 1,
Control means for controlling the amount of voice data stored in the storage means;
When the storage amount of the voice data stored in the storage unit is less than a predetermined value, the control unit stores the voice data sent from another voice communication device and stores the voice data in the storage unit. A voice communication apparatus, wherein a part of the same voice data as the data is stored again in the storage means.

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