JP2006074359A - Transmission reception system of audio data in teleconference system and control method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a transmission reception system of audio data capable of holding a conference with voice of high sound quality at a normal time and holding a conference where sound interruption and delay are not caused at the time of congestion of a communication network in a teleconference system via the communication network for monitoring the audio data transmitted via the communication network from a transmission side to monitor them on the reception side, deciding a congestion state of the communication network, and controlling dynamically a rate of compression of the audio data transmitted from the transmission side in accordance with the congestion state. <P>SOLUTION: In the audio data transmission reception system, the audio data compressed by sound compression means 30b, 30c of one voice transmitter receiver 30-1 are transmitted to the other sound transmitter receiver 30-2 via the communication network 33, and the audio data compressed by the sound compression means of the other sound transmitter receiver are received by the one sound transmitter receiver via the communication network. The system is provided with a congestion monitoring means 30e for monitoring a communication state in the communication network and the rate of compression of the sound compression means 30b and 30c on the transmission side is controlled in accordance with an output of the congestion monitoring means. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

  The present invention relates to an audio data transmission / reception system and control method in a remote conference system such as a video conference system or a remote data conference system.

The following are known as conventional video conference systems. (See Patent Document 1)
FIG. 1 is a block diagram showing a configuration of a conventional video conference terminal.
In FIG. 1, a video conference terminal 1 includes an image A / D (analog / digital) converter 11, an audio A / D converter 12, a marking signal addition circuit 13, an image compression circuit 14, an audio compression circuit 15, and a multiplexing unit. Circuit 16, separation circuit 17, image expansion circuit 18, audio expansion circuit 19, marking signal detection circuit 20, image D / A (digital / analog) converter 21, audio D / A converter 22, And a delay circuit 23.

  On the transmission side, an analog image input signal 101 from a camera or the like (not shown) is quantized by an image A / D converter 11, a marking signal is added by a marking signal adding circuit 13, and the speed of the transmission line 100 is outputted by an image compression circuit 14. After being subjected to compression processing according to the above, the multiplexed signal is multiplexed with the audio signal by the multiplexing circuit 16 and sent to the transmission line 100.

  On the other hand, an analog audio input signal 102 from a microphone (not shown) is quantized by an audio A / D converter 12, a marking signal is added by a marking signal adding circuit 13, compressed by an audio compression circuit 15, and then multiplexed. The signal is multiplexed with the image signal by the circuit 16 and sent to the transmission line 100.

  The marking signal addition circuit 13 converts the quantized signals of the image and the sound immediately after being quantized by the image A / D converter 11 and the sound A / D converter 12, that is, before being compressed. A marking signal is added simultaneously and periodically. The marking signal is added by replacing some bits of the signal with the marking signal.

  The image compression circuit 14 and the audio compression circuit 15 compress the signal to which the marking signal is added by the marking signal addition circuit 13, that is, the quantized image and audio signal using a reversible encoding algorithm.

On the receiving side, a signal received from the other party of communication through the transmission line 100 is separated into an image signal and an audio signal by the separation circuit 17.
The image signal, that is, the compressed signal is decompressed by the image decompression circuit 18 by an algorithm reverse to that on the transmission side, is input to the image D / A converter 21 via the marking signal detection circuit 20, and is analogized by the image D / A converter 21. It is reproduced as an image output signal 103.

  Similar to the image signal, the audio signal is expanded by the audio expansion circuit 19 by an algorithm reverse to that on the transmission side, and is input to the audio D / A converter 22 via the marking signal detection circuit 20. It is reproduced as an analog audio output signal 104.

  In the video conference terminal 1 according to the embodiment of the present invention, the input image and audio analog signals are quantized by the image A / D converter 11 and the audio A / D converter 12, that is, before being compressed. In the stage, marking signals are added simultaneously and periodically to the quantized signals of the image and the sound by the marking signal adding circuit 13.

  The marking signal is added by replacing some bits of the signal with the marking signal. Here, by using a reversible algorithm for signal compression and decompression processing, the signal before compression is completely reproduced on the receiving side.

For this reason, the marking signal added on the transmission side is detected by the marking signal detection circuit 20 on the receiving side, and the arrival time difference t between the marking signal added to the image signal and the marking signal added to the audio signal is known. Can do.
As a result, by delaying the output of the signal arriving first among the image signal and the audio signal by the arrival time difference t in the delay circuit 23, it becomes possible to reproduce the image and the audio at the same timing as at the time of transmission.

  That is, even when there is a difference in arrival time between the image and the audio signal, the video conference terminal 1 on the receiving side can always reproduce the image and the audio at the same timing as when transmitting.

  In the conventional video conference terminal 1 of FIG. 1, the marking signal is processed immediately after the image signal and the audio signal are quantized or immediately before being analogized. Therefore, if it is a reversible algorithm, it depends on the compression / decompression method. In addition, it is possible to provide a synchronized playback function between sound and image.

  Further, in the conventional video conference terminal 1 of FIG. 1, since a short marking signal is directly embedded in an image signal and an audio signal instead of using a long time stamp, synchronous reproduction of an image and audio is performed with a smaller amount of information. be able to.

Next, a voice data transmission / reception system applied to the conventional remote conference system will be described in detail with reference to FIG. 2 while focusing on the voice portion in the remote conference system of FIG.
FIG. 2 is a block diagram showing an outline of an audio data transmission / reception system applied to a conventional remote conference system.
In FIG. 2, one voice transmission / reception device 30-1 and the other voice transmission / reception device 30-2 are connected via a communication network (for example, an IP network) 33.
The voice transmitting / receiving apparatuses 30-1 and 30-2 include a communication function (communication control means) 30a, a voice compression means 30b, a voice capturing means 30c, a voice decompression means 30f, and a voice output means 30g. .

  With the configuration of FIG. 2, a voice input from a voice input device (such as a microphone) connected to one voice transmission / reception device is sent from a voice output device (such as a speaker) 32 connected to the other voice transmission / reception device 30-2. While outputting, the audio | voice input from the audio | voice input apparatus (microphone etc.) connected to the other audio | voice transmission / reception apparatus is output from the audio | voice output apparatus (speaker etc.) 32 connected to one audio | voice transmission / reception apparatus 30-2. .

The voice capturing means 30c corresponds to the voice A / D converter of FIG. 1, and an analog voice signal from a voice input device 31 such as a microphone is quantized by a predetermined algorithm to be converted into a digital signal, and the next stage. To the audio compression means 30b.
The audio compression unit 30b compresses the audio signal converted into a digital signal by the audio capturing unit with a predetermined compression method and compression rate, and adds information related to the compression to add the communication function 30a at the next stage. Output to.
The communication function 30a adds the control information to the voice data compressed by the voice compression unit 30b and adds a serial number to each packet at the time of transmission using a predetermined protocol via the communication network 33. To do.
Further, the voice decompression means 30f decompresses the voice signal before compression based on the information added with the received voice signal to the voice signal before compression by the reverse process of the compression process in the voice compression means. Output to.
The audio output means 30g corresponds to the audio D / A converter of FIG. 1, converts a digital signal into an analog signal by a predetermined algorithm, and outputs it from the audio output device at the next stage.
In addition, the structure and operation | movement of one audio | voice transmission / reception apparatus 30-1 and the other audio | voice transmission / reception apparatus are the same.

In the teleconference system using the audio data transmission / reception system described above, priority is given to keeping the sound quality uniform, so the sampling frequency for audio data acquisition and the compression rate for audio compression are fixed in the system. Or it is fixed at the start of the remote conference and is fixed during the conference.
JP 2001-136503 A (FIG. 1, (0027) to (0039))

The teleconferencing system using the audio data transmission / reception system of FIG. 2 has the advantage that the sound quality can be kept uniform, but the integrated teleconferencing system that integrates video conferencing and remote data conferencing, which have recently been increasing in demand. As described above, when various data transmission / reception occurs in the same network, as described above, the band used for audio transmission / reception in which the sampling frequency for capturing audio data and the compression rate for audio compression are fixed by the system is constant. When other devices execute transmission / reception of large-capacity data in the same communication network (for example, IP network), burst load is generated in the communication network, and voice interruptions and delays are caused by the load. There is a problem that it may occur.
Note that transmission of large-capacity data files used in the conference in the remote conference system using the voice data transmission / reception system of FIG. Etc.

  An object of the present invention is to monitor voice data transmitted from a transmission side via a communication network on the reception side, determine a congestion state of the communication network, and compress voice data transmitted from the transmission side according to the congestion state. By controlling the rate dynamically, in a teleconferencing system that intervenes in a communication network, it is possible to achieve a conference with high-quality audio during normal times, and sound interruptions and delays occur even when the communication network is congested, although the sound quality is low. An object of the present invention is to provide a voice data transmission / reception system capable of realizing a conference that is not present.

  The audio data transmission / reception system according to claim 1 transmits the audio data compressed by the audio compression unit of one audio transmission / reception device to the other audio transmission / reception device via the communication network, and the audio of the other audio transmission / reception device. In the voice data transmission / reception system for receiving the voice data compressed and transmitted by the compression means by the one voice transmission / reception device via the communication network, the congestion monitoring means for monitoring the communication status in the communication network is provided, and the congestion monitoring The compression rate of the audio compression means on the transmission side is controlled according to the output of the means.

  The audio data transmission / reception system according to claim 2 is an audio data transmission / reception system for transmitting audio data compressed by the audio compression means of the audio transmission device to the audio reception device via a communication network. Congestion monitoring means for monitoring the communication status to the voice receiving apparatus is provided, and the compression rate of the voice compression means of the voice transmitting apparatus is controlled according to the output of the congestion monitoring means.

  An audio data transmission / reception method according to claim 9 is an audio data transmission / reception control method for transmitting audio data compressed by audio compression means of an audio transmission device to an audio reception device via a communication network, Monitoring the communication status from the transmission device to the voice reception device; transmitting the output monitoring the communication status to the voice transmission device; and controlling the compression rate of the voice compression means of the voice transmission device; It is characterized by including.

  An audio data transmission / reception method according to claim 10 is an audio data transmission / reception control method for transmitting audio data compressed by audio compression means of an audio transmission device to an audio reception device via a communication network, wherein Transmitting a bandwidth check command from the transmission device to the voice reception device via a communication network, monitoring a response thereof to calculate an effective speed of the communication network, and voice compression of the voice transmission device according to the effective speed Controlling the compression rate of the means.

The voice data transmission / reception system of the present invention applied to the remote conference system will be described in detail with reference to FIG.
FIG. 3 is a block diagram showing an outline of a voice data transmission / reception system according to the first embodiment of the present invention applied to the remote conference system.

In FIG. 3, one voice transmission / reception device 30-1 and the other voice transmission / reception device 30-2 are connected via a communication network (for example, an IP network) 33.
The voice transmitting / receiving apparatuses 30-1 and 30-2 include a communication function (communication control means) 30a, a voice compression means 30b, a voice capturing means 30c, a voice decompression means 30f, and a voice output means 30g. .

  With the configuration of FIG. 3, a voice input from a voice input device (such as a microphone) connected to one voice transmitter / receiver is sent from a voice output device (such as a speaker) 32 connected to the other voice transmitter / receiver 30-2. While outputting, the audio | voice input from the audio | voice input apparatus (microphone etc.) connected to the other audio | voice transmission / reception apparatus is output from the audio | voice output apparatus (speaker etc.) 32 connected to one audio | voice transmission / reception apparatus 30-2. This is the same as the conventional audio data transmission / reception system of FIG.

  The feature of the voice data transmission / reception system according to the first embodiment of the present invention is that the voice data transmitted from the transmission side via the communication network is monitored on the reception side, the congestion state of the communication network is determined, and the congestion is detected. The purpose is to dynamically control the compression rate of the audio data transmitted from the transmission side according to the state. As a configuration for that purpose, the received audio monitoring means 30e and the audio compression method control means 30d are provided in each audio transmission / reception apparatus. I have.

  The received voice monitoring means (function) 30e monitors the loss of the packet No. of the voice data received via the communication network, calculates the packet loss rate of the received voice data, and sends the calculation result via the communication network. To the transmitting side.

The compression method control means (function) 30d determines an optimum compression method according to the packet loss rate transmitted from the receiving side, and compresses the sampling frequency of the sound capturing means 30c and / or the sound compression means 30b. Control the rate dynamically.
In addition, although the audio | voice data transmission / reception system of FIG. 3 transmits and receives an audio | voice bi-directionally, the structure which transmits an audio | voice only to one direction may be sufficient.

The following method is suitable as a form of controlling the compression rate of the audio data in the first embodiment.
The audio compression method control means 30d is provided with a “determination table” and a “compression method management table” shown in Table 1.

(A) of Table 1 is an example of the “determination table”, and the “packet loss rate” transmitted from the receiving side is divided into three stages and the following control signals are output.
Packet loss rate less than 3% → 1 stage UP signal Packet loss ratio 3 to 7% → Packet loss ratio 7% or more as it is → 1 stage DOUN signal

(B) in Table 1 is an example of a “compression method management table”. In this table, sampling is performed in each case corresponding to a required bandwidth (1 to 9) of 16 to 320 Kbps with 8-bit quantization. (Sampling) rate and the presence or absence of compression are shown.
In this table, No5 is set as the default (Default), and according to the determination in the determination table, the pointer is moved up by one step from the current position, and the control instruction for the compression ratio of the one-step DOWN is voice compression. Given to the means 30b, the compression rate is dynamically controlled in accordance with the voice packet loss rate monitored on the receiving side.
This process is repeated every certain time or every time a certain number of packets are received.
In the present invention, the compression rate control may be performed independently or in combination with the sampling frequency control by the voice capturing unit 30c and the compression rate control by the compression unit 30b.

  By executing the control as described above, in a teleconferencing system via a communication network, it is possible to realize a conference with high-quality sound during normal times, and when the communication network is congested, the sound is interrupted or delayed although the sound quality is low. Meeting that does not occur can be realized.

In the example of Table 1 (a), the determination level of the packet loss rate is set to less than 3%, 3 to 7%, 7% or more, but the setting value is not limited to this value, It may be set to a numerical value.
In the example of Table 1 (b), the sampling frequency value is set to 8 KHz, 16 KHz, 24 KHz, 32 KHz, and 40 KHz. However, the sampling frequency value is not limited to this value, and may be set to another value. good.
Further, although the compression rate is selected depending on whether or not compression is performed, compression methods having different compression rates may be selected.

FIG. 4 is a block diagram showing an outline of an audio data transmission / reception system according to the second embodiment of the present invention applied to the remote conference system.
In FIG. 4, one voice transmission / reception device 30-1 and the other voice transmission / reception device 30-2 are connected via a communication network (for example, an IP network) 33.
The voice transmitting / receiving apparatuses 30-1 and 30-2 include a communication function (communication control means) 30a, a voice compression means 30b, a voice capturing means 30c, a voice decompression means 30f, and a voice output means 30g. .
With the configuration of FIG. 4, a voice input from a voice input device (such as a microphone) connected to one voice transmitter / receiver is sent from a voice output device (such as a speaker) 32 connected to the other voice transmitter / receiver 30-2. While outputting, the audio | voice input from the audio | voice input apparatus (microphone etc.) connected to the other audio | voice transmission / reception apparatus is output from the audio | voice output apparatus (speaker etc.) 32 connected to one audio | voice transmission / reception apparatus 30-2. This is the same as the conventional audio data transmission / reception system of FIG.

  The voice data transmission / reception system according to the second embodiment of the present invention is characterized by determining the congestion state of the communication network and dynamically controlling the compression rate of the voice data transmitted from the transmission side according to the congestion state. However, the method of determining the congestion state of the communication network is different from that of the first embodiment.

In the second embodiment, in order to determine the congestion state of the communication network, a band survey command transmission / reception unit 30h and a band calculation unit 30i are provided.
The bandwidth check command transmission / reception means 30h calculates a bandwidth from the bandwidth check command transmission / reception means provided in one voice transmission / reception device to the bandwidth check command transmission / reception means provided in the other voice transmission / reception device via the communication network. Send data and monitor the response.
The bandwidth calculation means 30i calculates the execution speed of the communication network based on the execution result of the bandwidth check command received by the bandwidth check command transmission / reception means.

The following method is suitable as a form of controlling the compression rate of the audio data in the second embodiment.
The audio compression method control means 30d is provided with a “determination table” and a “compression method management table” shown in Table 2 in association with each other.
Table 2 shows two examples of pattern A and pattern B according to the voice band occupancy rate.

(A) of Table 2 is an example in which the pattern A has a voice band occupancy rate of 30%.
In this determination table, the relationship between the execution speed calculation result and the voice bandwidth available bandwidth is defined in 11 stages.
Of the 11 stages, the top 9 stages are uniquely associated with 9 stages similar to the “compression method management table” (b) in the first embodiment.
In this case, in the execution speed calculation results of the lower two stages, 32 to 63 and 31 to 0 have no corresponding appropriate values.

In the second embodiment of FIG. 4, based on the command execution result received by the bandwidth check command transmission / reception means, the communication network execution speed calculated by the bandwidth check means is compressed based on the execution speed shown in Table 2. Determine the table compression rate (sampling frequency and presence / absence of compression).
For example, if the execution speed of the communication network calculated by the bandwidth checking means is 500, the execution speed is within the range of 384 to 511, so the available bandwidth of the voice band is 80 Kbps and the sampling of No5 in the compression method management table. The frequency is 40 Khz and the necessary band with compression is uniquely selected as 80 Kbps, and the sampling frequency of the voice capturing means 30c and / or the compression rate of the voice compression means 30b is controlled.

(B) of Table 2 is an example in which the pattern A has a voice band occupancy rate of 50%.
In this determination table, the relationship between the execution speed calculation result and the voice band available band is defined in nine stages.
Of the nine stages, the upper eight stages are uniquely associated with the eight stages of the “compression method management table”.
In this case, the lowest execution speed calculation result is 31 to 0, and the corresponding appropriate value is none.

Also in this case, based on the command execution result received by the bandwidth check command transmission / reception means, the execution speed of the communication network calculated by the bandwidth check means is calculated based on the compression rate (sampling) of the compression method management table based on the execution speed of Table 2. Frequency and compression).
For example, if the execution speed of the communication network calculated by the bandwidth investigation means is 500, the execution speed is within the range of 384 to 511, so the available bandwidth of the voice band is 192 Kbps and the sampling of No3 in the compression method management table The frequency is 24 Khz, no compression, and the required bandwidth is 192 Kbps, and the sampling frequency of the voice capturing unit 30 c and / or the compression rate of the voice compression unit 30 b is controlled.
It should be understood that when the voice band occupancy rate is 50%, an appropriate compression rate can be selected even when the execution speed is as low as 63 to 32.

In the examples of Tables 2 (a) and 2 (b), the sampling frequency values are 8 KHz, 16 KHz, 24 KHz, 32 KHz, and 40 KHz. However, the sampling frequency value is not limited to this value, and may be set to another value. You may do it.
Further, although the compression rate is selected depending on whether or not compression is performed, compression methods having different compression rates may be selected.

Next, an image when the data amount (packet amount) is congested in the communication network (network) according to the present invention will be described with reference to the conceptual diagram of FIG.
FIG. 5A shows a state in which only audio data is transmitted / received to / from a communication network (line) connecting audio data transmitting / receiving apparatuses.
In this state, since there is a sufficient margin in the bandwidth of the line, for example, transmission of voice with high sound quality and uninterrupted sound can be executed under the condition of compression at the No. 5 sampling frequency of 40 kHz, which is the default of Table 1. .

FIG. 5B shows a control state in the first embodiment of FIG.
In this figure, when distribution of a large-capacity file is started, it is detected that the packet loss rate has become 7% or more, a change of compression method (1 rank down) is instructed, and the default (Default ) No. 5 is changed to No. 6 to reduce the necessary bandwidth.
After that, even after the compression method is changed (down one rank), if it is detected that the packet loss rate is 7% or more, the change of the compression method (down one rank) is instructed, and No7 is given. As a result, the necessary bandwidth is further reduced.

As long as the packet loss rate recovers and falls within the range of 3 to 7% due to the change of the compression method, the No. 7 compression method is continued.
After that, when the delivery of a large-capacity file is completed, it is detected that the packet loss rate has become less than 3%, and a change of the compression method (up one rank) is instructed, and the change from No7 to No6 is necessary. Bandwidth is expanded.
After that, even after the compression method is changed (up one rank), if it is detected that the packet loss rate is less than 3%, the change of the compression method (up one rank) is instructed and No5 As a result of the change, the necessary bandwidth is further expanded to return to the state of FIG.

FIG. 5C shows the state of control in the second embodiment of FIG.
In this figure, the compression method management table is based on the execution speed of the communication network calculated by the bandwidth investigation means based on the command execution result received by the bandwidth investigation command transmission / reception means after the delivery of the large-capacity file is started. The optimum compression rate (sampling frequency and presence / absence of compression) is determined, and this state continues until the delivery of the large file is completed.
After the delivery of the large-capacity file is completed, the compression method management table is optimized based on the execution speed of the communication network calculated by the bandwidth survey means based on the command execution result received by the bandwidth survey command transmission / reception means. The compression rate (sampling frequency and presence / absence of compression) is determined and the state continues.

  In the above description, the audio data transmission / reception system is described as being applied to a remote conference system such as a video conference system or a remote data conference system, but the audio data transmission / reception system of the present invention is not necessarily applied to the remote conference system. The present invention is not limited and can be applied to other uses.

It is a block diagram which shows the structure of the conventional video conference terminal. It is a block diagram which shows the outline | summary of the transmission / reception system of the audio | voice data applied to the conventional remote conference system. It is a block diagram which shows the outline | summary of the transmission / reception system of the audio | voice data which is the 1st Embodiment of this invention. It is a block diagram which shows the outline | summary of the transmission / reception system of the audio | voice data which is the 2nd Embodiment of this invention. It is a conceptual diagram which shows the image at the time of data amount congestion in the communication network in this invention.

Explanation of symbols

30-1, 30-2 Voice transmitting / receiving device 31 Voice input device 32 Voice output device 33 Communication network (IP network)
30a Communication control means (communication function)
30b Audio compression means 30c Audio capture means 30d Audio compression method control means 30e Received audio monitoring means 30f Audio expansion means 30g Audio output means

Claims (10)

The voice data compressed by the voice compression unit of one voice transmission / reception device is transmitted to the other voice transmission / reception device via the communication network, and the voice data compressed by the voice compression unit of the other voice transmission / reception device is transmitted. In the voice data transmission / reception system that is received by the one voice transmission / reception device via a communication network,
Provide congestion monitoring means for monitoring the communication status in the communication network,
An audio data transmission / reception system that controls a compression rate of the audio compression means on the transmission side according to the output of the congestion monitoring means. In an audio data transmission / reception system for transmitting audio data compressed by audio compression means of an audio transmission device to an audio reception device via a communication network,
Providing congestion monitoring means for monitoring the communication status from the voice transmitting device to the voice receiving device;
An audio data transmission / reception system, wherein a compression rate of an audio compression unit of the audio transmission device is controlled according to an output of the congestion monitoring unit. The congestion monitoring means includes
The reception side of the compressed audio data is constituted by reception audio monitoring means for monitoring the reception state of the reception audio data,
3. The audio data transmission / reception system according to claim 1, wherein an output of the reception voice monitoring unit is transmitted to a transmission side of the audio data to control a compression rate of the voice compression unit on the transmission side.    The audio according to any one of claims 1 to 3, wherein the compression rate of the audio compression means is controlled by controlling a sampling frequency for sampling audio data and / or controlling a compression rate. Data transmission / reception system.   The determination of the congestion status by the received voice monitoring means is determined by the packet loss rate of the received voice data. When the packet loss rate is equal to or higher than a predetermined value or lower than a predetermined value, the compression rate of the voice compression means is sequentially increased. The voice data transmission / reception system according to claim 1, wherein the voice data transmission / reception system is lowered. The congestion monitoring means includes
Send the bandwidth survey command to the destination voice data transmitter / receiver via the communication network and calculate the effective speed by monitoring the response,
6. The voice data transmission / reception system according to claim 1, wherein a compression rate of the voice compression unit on the transmission side is controlled according to the calculated execution speed.    7. The audio data transmission / reception system according to claim 6, wherein the compression rate of the audio compression unit is executed by referring to a table in which a sampling frequency and / or compression rate corresponding to the effective speed is defined. .   The audio data transmission / reception system according to claim 1, wherein the audio data transmission / reception system is applied to a remote conference system such as a video conference system or a remote data conference system. An audio data transmission / reception control method for transmitting audio data compressed by audio compression means of an audio transmission device to an audio reception device via a communication network,
Monitoring the communication status from the voice transmitting device to the voice receiving device;
Transmitting the output monitoring the communication status to the voice transmitting device, and controlling the compression rate of the voice compression means of the voice transmitting device;
A voice data transmission / reception control method comprising: An audio data transmission / reception control method for transmitting audio data compressed by audio compression means of an audio transmission device to an audio reception device via a communication network,
Transmitting a bandwidth survey command from the voice transmitting device to the voice receiving device via a communication network, monitoring a response thereof and calculating an effective speed of the communication network;
Controlling the compression rate of the voice compression means of the voice transmitting device according to the effective speed;
A voice data transmission / reception control method comprising:

Images