JP2005045568A - Voice communication system and voice communication method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a voice communication system and a voice communication method which allow voice communication to be performed by sacrificing the real-time property to secure a minimum voice quality even where there are many data errors. <P>SOLUTION: In a voice communication system 10 wherein voice data is transmitted between transmission/reception terminals through a network, each of transmission/reception terminals is provided with a variable-sized reception buffer for temporarily storing the received voice data therein, a data error detection part for detecting the errors of the voice data stored in the reception buffer, and a control part for transmitting a retransmission request to the other terminal. The control part sets the size of the reception buffer to the minimum in the case of the data error quantity equal to or smaller than a prescribed quality and increases the size of the reception buffer in the case of the data error quantity larger than prescribed and then transmits the retransmission request to the transmission-side transmission/reception terminals. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to a technique for enabling minimum voice communication even when a transmission situation deteriorates in a voice communication system via a network such as a digital mobile phone or a so-called IP phone.

2. Description of the Related Art Conventionally, for example, in a digital cellular phone service, interactive voice communication is realized by transmitting audio data to each other via a network by digital audio bidirectional communication.
By placing the highest priority on real-time performance as a quality factor of a call, even if errors, omissions or delays occur in the audio data received at the receiving terminal, no data retransmission request is made to the transmitting side. The voice data received as it is is converted into voice by a speaker or the like.

  Therefore, if the effective transmission data rate between terminals from the transmitting terminal to the receiving terminal via the network decreases due to the deterioration of the transmission state or the increase in traffic volume in the network, the noise emitted from the speaker etc. There will be interruptions. As a result, it may be difficult to hear the content of a voice call, or it may be impossible to hear at all.

  For this reason, there is a method in which the received audio data is temporarily stored in the buffer and the audio data is read from the buffer after a predetermined time. In this case, depending on the buffer capacity, an overflow occurs, a delay occurs, and The delay will continue, and in the case of underflow, there will be no audio data to be read out in the buffer, causing interruptions.

  In addition, in a so-called IP phone using VoIP (Voice over IP) on an IP (Internet Protocol) network that is not controlled by QoS (Service Priority Control), when the network is congested, the communication band becomes very narrow and the sound is cut off. And delays, and in some cases, communication interruptions may occur.

On the other hand, for example, in Patent Document 1, a received voice packet is temporarily stored in a buffer memory at a receiving terminal, and a data block number is detected from the voice packet to determine whether a voice packet is missing. Thus, there has been disclosed a voice packet interpolating apparatus that compensates for a missing voice packet by generating and replacing interpolation data for the missing voice packet.
According to this apparatus, when a voice packet is lost, interpolation data is generated and inserted into the missing part, thereby eliminating the voice packet loss and suppressing voice interruption. Will be.

For example, in Patent Document 2, similarly, reception data is stored in a buffer at a receiving terminal, transmission time information is acquired from the received data, and fluctuations in delay time of the received data are detected. Thus, an audio fluctuation correction control method is disclosed in which an optimal waiting time is calculated from the fluctuation, and audio data is read based on the waiting time.
According to such a method, when a delay occurs in the received data, the audio data is read out from the buffer on the basis of the optimum waiting time calculated so as to match the delay time. It is possible to realize a simple conversation.
Japanese Patent Laid-Open No. 08-256131 JP 2002-300316 A

However, each of Patent Documents 1 and 2 described above has the following problems.
That is, in Patent Document 1, when a voice packet is lost, it is possible to generate and insert interpolation data into the missing part, thereby eliminating apparent voice packet loss and interrupting voice. However, since the interpolated data is different from the original voice packet, it is difficult to maintain the call quality.

  Further, in Patent Document 2, the fluctuation of the delay time of the received data is detected and the optimum delay time is calculated. Based on this delay time, the received data is read from the buffer and output as sound. Apparently uninterrupted conversation can be performed, but since the received data is read from the buffer with the optimal delay time based on the fluctuation of the delay time, it always has a delay time and lacks real-time characteristics. Become.

  The present invention has been made to solve the above problem, and even when data errors occur frequently, voice communication can be performed by ensuring a minimum call quality at the expense of real-time performance. An object of the present invention is to provide a voice communication system and a voice communication method.

  In order to achieve this object, a voice communication system according to claim 1 of the present invention is a voice communication system in which voice data is mutually transmitted from a transmission unit of one transmission / reception terminal to a reception unit of another transmission / reception terminal via a network. Each transmitting / receiving terminal has a variable-size reception buffer for temporarily storing the audio data received by the reception unit, a data error detection unit for detecting a data error in the audio data stored in the reception buffer, and a transmission unit And a control unit that transmits a retransmission request to the other transmission / reception terminal via the network. When the data error amount detected by the data error detection unit is equal to or less than a predetermined amount, the reception unit When the buffer size is set to the minimum and the data error amount exceeds a predetermined amount, the buffer size is increased and the transmission unit receives and transmits the data on the transmission side. It is constituted to transmit a retransmission request to the terminal.

  When the voice communication system has such a configuration, when the amount of data error is less than or equal to a predetermined value, usually, the receiving unit of the receiving side transmitting / receiving terminal minimizes the size of the receiving buffer and detects the data error. Even if it is done, voice communication is performed by receiving voice data without retransmitting data. As a result, since the size of the reception buffer is minimum, even if there is a data error such as some data loss, voice communication can be performed without substantially impairing real-time performance.

  On the other hand, when the data error amount exceeds a predetermined value, the receiving unit of the receiving side transmitting / receiving terminal increases the size of the receiving buffer and transmits a retransmission request to the transmitting side transmitting / receiving terminal. Thus, by performing data retransmission, the bit error rate is reduced, the data error amount is reduced, and voice communication is performed. As a result, a delay occurs due to the accumulation of data based on the size of the reception buffer, and real-time performance in voice communication is somewhat impaired, but data errors such as data errors and data loss are reduced due to data retransmission. The call quality in voice communication will be improved. Therefore, although real-time performance is somewhat impaired, sound interruption and noise in voice communication are reduced, and a minimum call can be ensured.

  The audio communication system according to claim 2, wherein the network includes a receiving unit and a transmitting unit in both directions, and a variable-size receiving buffer that temporarily stores audio data received by the receiving unit in each direction; A data error detection unit that detects a data error in the audio data stored in the reception buffer, and a control unit that transmits a retransmission request to the transmission / reception terminal on the transmission side by the transmission unit in each direction. When the data error amount detected by the data error detection unit is less than or equal to a predetermined amount, the control unit sets the size of the reception buffer to a minimum, and when the data error amount exceeds the predetermined amount, And a transmission unit transmits a retransmission request to a transmission / reception terminal.

  When the voice communication system has such a configuration, normally, when the data error amount is equal to or less than a predetermined value, even if a data error is detected at the reception unit of the network by minimizing the size of the reception buffer. The voice communication is performed by receiving the voice data without transferring the data and transferring the voice data to the transmitting / receiving terminal on the receiving side. As a result, the size of the reception buffer in the reception unit of the network is the smallest, so that even if there are some data errors such as missing data, voice communication can be performed without substantially impairing real-time performance. it can.

  On the other hand, when the data error amount exceeds a predetermined value, the reception unit of the network increases the size of the reception buffer and transmits a retransmission request to the transmission / reception terminal on the transmission side to retransmit the data. By performing the above, the bit error rate is reduced, the data error amount is reduced, the voice data is received, and the voice communication is performed by transferring the voice data from the transmitter to the transmitting / receiving terminal. As a result, a delay occurs due to the accumulation of data based on the size of the reception buffer, and real-time performance in voice communication is somewhat impaired, but data errors such as data errors and data loss are reduced due to data retransmission. The call quality in voice communication will be improved. Therefore, although real-time performance is somewhat impaired, sound interruption and noise in voice communication are reduced, and a minimum call can be ensured.

The voice communication system according to claim 3 is configured such that the network is a digital cellular phone network and each transmitting / receiving terminal is a digital cellular phone terminal.
When the voice communication system has such a configuration, when voice communication is performed between digital mobile phone terminals in a digital mobile phone network, even when the transmission / reception environment of the digital mobile phone network deteriorates, real-time characteristics are somewhat sacrificed. By doing so, a minimum call can be secured.

The voice communication system according to claim 4 is an Internet connection service in which the network provides VoIP, and each transmitting / receiving terminal is an IP telephone terminal.
When the voice communication system has such a configuration, when voice communication is performed between IP telephone terminals in an Internet connection service that provides VoIP, even if the communication band becomes very narrow due to network congestion, real-time performance is ensured. By sacrificing a little, it is possible to secure a minimum number of calls.

5. The voice communication system according to claim 5, wherein each transmitting / receiving terminal transmits voice data by a protocol not normally performing retransmission control, and a retransmission request is received from a transmitting / receiving terminal on the receiving side when a data error amount exceeds a predetermined amount. In this case, the audio data is transmitted by a protocol for performing retransmission control.
When the voice communication system has such a configuration, the voice data is transmitted by a protocol that does not normally perform retransmission control, thereby reducing the amount of data transfer and realizing a higher data transmission speed. A good call can be secured, and when the amount of data error exceeds a predetermined amount, the amount of data error can be reduced by transmitting voice data using a protocol for performing retransmission control.

In order to achieve the above object, according to the voice communication method of claim 6 of the present invention, voice data is mutually transmitted from the transmitter of one transmitter / receiver terminal to the receiver of another transmitter / receiver terminal via the network A voice communication method in which the receiving unit of each transmitting / receiving terminal temporarily stores received voice data in a variable-size receiving buffer and detects a data error in the voice data stored in the receiving buffer to detect When the amount of data error is less than or equal to a predetermined amount, the size of the reception buffer is set to the minimum, and when the amount of data error exceeds the predetermined amount, the size of the buffer is increased and the transmitting side transmission / reception terminal Is a method for transmitting a retransmission request to the user.
Thus, the present invention can be realized as a voice communication method.

  As described above, according to the voice communication system according to the present invention, the transmission situation in the network deteriorates, or the transmission speed between terminals decreases due to the congestion of the network, and it becomes the worst environment of extremely low reliability / ultra narrow band. Even if the transmission / reception terminal or network on the receiving side sacrifices some real-time performance, the audio data is retransmitted so that the data error amount becomes a predetermined value or less. It becomes possible to secure a minimum telephone call without obscuring.

  Embodiments of the present invention will be described below with reference to the drawings.

A first embodiment in which the voice communication system of the present invention is applied to a digital mobile phone will be described with reference to FIGS.
FIG. 1 is a block diagram showing the configuration of the voice communication system of the present embodiment.

As shown in FIG. 1, the voice communication system 10 includes a first digital mobile phone terminal 100, a second digital mobile phone terminal 200, and a network 300.
In FIG. 1, only two digital mobile phone terminals 100 and 200 are shown for transmission and reception, but the number is not limited to two. Voice communication can be performed.

Since the first digital mobile phone terminal 100 and the second digital mobile phone terminal 200 have the same configuration, the first digital mobile phone terminal 100 will be described below.
A first digital mobile phone terminal (hereinafter referred to as a digital mobile phone terminal) 100 is configured as shown in FIG.
In addition, although description is abbreviate | omitted, as shown in the parenthesis in FIG. 2, the 2nd digital cellular phone terminal 200 changed the 100s of the code | symbol of each part of the 1st digital cellular phone terminal 100 to the 200s. The component which attached | subjected the code | symbol is provided.

In FIG. 2, the digital mobile phone terminal 100 includes a transmission unit 110 and a reception unit 120.
The transmission unit 110 has a known configuration, and converts an audio signal (analog signal) taken from a microphone (not shown) or the like into audio data that is a digital signal by the audio encoding unit 111, and encodes the audio signal. After the voice data (digital signal) from the unit 111 is temporarily stored in the transmission buffer 112, the wireless transmission unit 113 reads the voice data (digital signal) from the transmission buffer 112, and the base station of the network 300 via an antenna (not shown). (Not shown).

The reception unit 120 has a well-known configuration, and a reception signal received by an antenna (not shown) is converted into audio data (digital signal) by the wireless reception unit 121, temporarily stored in the reception buffer 122, and then decoded. The audio data (digital signal) is read from the reception buffer 122 by the conversion unit 123, decoded into an audio signal (analog signal), and output to a speaker (not shown) or the like.
Here, the reception buffer 122 is configured so that its size (storage capacity) can be changed, and its size can be set by the control unit 140 described later.

  Thereby, in the digital cellular phone terminal 100, the audio signal is digitized by the transmission unit 110 and transmitted through the transmission buffer 112, and the audio data (digital signal) received by the reception unit 120 is transmitted through the reception buffer 122. Thus, it is converted into an audio signal (analog signal).

Further, the digital mobile phone terminal 100 includes a data error detection unit 130 and a control unit 140.
The data error detection unit 130 checks the audio data (digital signal) temporarily stored in the reception buffer 122 of the reception unit 120 and detects data errors such as data loss, data corruption, and data loss. Then, the data error detection unit 130 sends the detected data error amount to the control unit 140 described later.

Based on the data error amount from the data error detection unit 130, the control unit 140 sets the size of the reception buffer 122 to the minimum when the data error amount is a predetermined value or less.
In addition, when the data error amount exceeds a predetermined value, the control unit 140 increases the size of the reception buffer 122, and transmits from the wireless transmission unit 113 via the network 300 via the transmission buffer 112. A retransmission request is transmitted to the second digital mobile phone terminal 200 that is the other party of the call.

Furthermore, when the receiving unit 120 receives a retransmission request from the second digital mobile phone terminal 200 that is the other party of the call, the data error detecting unit 130 detects the retransmission request, whereby the control unit 140 However, the audio data (digital signal) temporarily stored in the transmission buffer 112 is retransmitted from the wireless transmission unit 113 to the second digital mobile phone terminal 200 via the network 300.
Here, when the data error amount from the data error detection unit 130 becomes equal to or smaller than the predetermined value, the control unit 140 returns the size of the reception buffer 122 to the minimum again and stops transmission of the retransmission request.

The network 300 is a digital mobile phone network, and provides a voice communication service between these digital mobile phone terminals by connecting the digital mobile phone terminals.
Although only one network 300 is shown in the drawing, the present invention is not limited to this, and a plurality of networks are provided, and digital cellular phone terminals that can be connected to individual networks by being connected to each other. It may be possible to perform voice communication.

Here, the network 300 is configured as shown in FIG.
In FIG. 3, the network 300 has receiving units 310 and 320, backbones 330 and 340, and transmitting units 350 and 360, respectively, in both directions regarding data transmission.
That is, with respect to the first transmission direction A from the first digital mobile phone terminal 100 to the second digital mobile phone terminal 200, the network 300 includes a reception unit 310, a backbone 330, and a transmission unit 350, and vice versa. With respect to the second transmission direction B, a receiving unit 320, a backbone 340, and a transmitting unit 360 are provided.

The receivers 310 and 320 receive voice data (digital signals) from the digital mobile phone terminals 100 and 200 on the transmission side by the wireless receivers 311 and 321 and temporarily store them in the reception buffers 312 and 322, respectively. The data is transmitted to the backbones 330 and 340.
Also in this case, the reception buffers 312 and 322 are configured so that the size can be changed in the same manner.

  Further, when the audio data (digital signals) from the backbones 330 and 340 are input to the transmission units 350 and 360, the transmission units 350 and 360 temporarily store the transmission data in the transmission buffers 351 and 361, and then transmit the data from the wireless transmission units 352 and 362. It has become.

Further, the network 300 includes data error detection units 370 and 380 and control units 371 and 381 in both directions regarding data transmission.
Data error detection units 370 and 380 check audio data (digital signals) temporarily stored in the reception buffers 312 and 322 of the reception units 310 and 320, and detect data errors such as data loss, data corruption, and data loss. To do. Then, the data error detection units 370 and 380 send the detected data error amount to the control units 371 and 381 described later.

Based on the data error amount from the data error detection units 370 and 380, the control units 371 and 381 set the sizes of the reception buffers 312 and 322 to the minimum when the data error amount is a predetermined value or less. .
In addition, when the data error amount exceeds a predetermined value, the control units 371 and 381 increase the size of the reception buffers 312 and 322 and transmit the transmission units 360 and 351 via the transmission buffers 361 and 351, respectively. A retransmission request is transmitted from the 350 wireless transmission units 362 and 352 to the digital mobile phone terminals 100 and 200 that are the transmission source of the call.

Furthermore, when the receiving unit 310, 320 receives a retransmission request from the digital mobile phone terminal 100, 200 that is the other party of the call, the data error detection unit 370, 380 detects this retransmission request, The control units 371 and 381 retransmit the audio data (digital signals) temporarily stored in the transmission buffers 361 and 351 from the wireless transmission units 362 and 352 to the digital mobile phone terminals 100 and 200. .
Here, when the amount of data error from the data error detection units 370 and 380 becomes equal to or smaller than the predetermined value, the control units 371 and 381 again reduce the sizes of the reception buffers 312 and 322 to the minimum, and send a retransmission request. Cancel transmission.

Next, a voice communication method using the voice communication system of this embodiment will be described.
First, audio data is transmitted from the transmission unit 110 of the first digital mobile phone terminal 100 to the second digital mobile phone terminal 200 that is the other party of the call via the network 300.

At this time, the audio data transmitted from the first digital cellular phone terminal 100 is temporarily stored in the reception buffer 312 on the network 300, and a data error is detected by the data error detection unit 370.
Here, when the data error amount is equal to or smaller than the predetermined value, the audio data is transmitted to the second digital mobile phone terminal 200 as it is.
When the data error amount exceeds a predetermined value, the control unit 371 enlarges the size of the reception buffer 312 and sends the transmission unit 360 to the first digital mobile phone terminal 100 that is the transmission source. Send a resend request.
As a result, the first digital cellular phone terminal 100 transmits again the audio data temporarily stored in the transmission buffer 112.

  In this way, the audio data transmitted from the first digital cellular phone terminal 100 as the transmission source to the network 300 is temporarily stored in the reception buffer 312 and the transmission buffer 351 in the network 300, and then transmitted to the transmission destination. To the second digital mobile phone terminal 200.

At this time, in the second digital cellular phone terminal 200, the audio data transmitted from the first digital cellular phone terminal 100 via the network 300 is temporarily stored in the reception buffer 222, and the data error detection unit 230 causes a data error. Is detected.
Here, when the data error amount is equal to or smaller than the predetermined value, the audio data is converted into an audio signal (analog signal) by the audio decoding unit 123 as it is and output to a speaker or the like.

When the data error amount exceeds a predetermined value, the control unit 240 enlarges the size of the reception buffer 222 and causes the transmission unit 210 to respond to the first digital mobile phone terminal 100 that is the transmission source. Send a resend request.
As a result, the first digital cellular phone terminal 100 transmits again the audio data temporarily stored in the transmission buffer 112.
In this way, in the second digital mobile phone terminal 200 as the transmission destination, when audio data for a predetermined time is stored in the reception buffer 222, the audio is decoded from the reception buffer 222 to the audio decoding unit 223. Data is output, converted into an audio signal (analog signal), and output as audio from a speaker or the like.

Then, the reception buffer 222 waits for the audio data for a predetermined time to be accumulated again, and outputs the audio data to the audio decoding unit 223.
Thereby, in the second digital cellular phone terminal 200, when the data error amount is equal to or smaller than the predetermined value, the size of the reception buffer 222 is set to the minimum by the control unit 240, so that the reception buffer 222 is substantially reduced. The audio data is output to the audio decoder 223 in real time, converted into an audio signal (analog signal), and output from a speaker or the like.
As a result, when the data error amount is less than or equal to a predetermined value, there is little influence on the call (sound interruption etc.) due to the data error. To avoid.

  When the data error amount exceeds a predetermined value, the size of the reception buffer 222 is enlarged by the control unit 240, and audio data for a predetermined time (for example, about 5 to 10 seconds) is accumulated and transmitted. A retransmission request is transmitted to the original first digital mobile phone terminal 100. Then, when audio data for a certain time is accumulated in the reception buffer 222, the audio data is output to the audio decoding unit 223, converted into an audio signal (analog signal), and output from a speaker or the like.

As a result, in the case where data errors frequently occur, real-time performance is somewhat impaired due to the enlargement of the size of the reception buffer 222. However, since the amount of data errors is reduced by retransmission of audio data, Since a limited call quality can be ensured, voice communication becomes possible.
Note that due to the delay of the audio signal by the reception buffer 222, the time for data retransmission and accumulation of the audio data in the reception buffer 222 becomes silent without the audio signal being output from a speaker or the like.

Next, a second embodiment in which the voice communication system of the present invention is applied to a so-called IP telephone will be described with reference to FIGS.
FIG. 4 is a block diagram showing the configuration of the voice communication system of the present embodiment.

As shown in FIG. 4, the voice communication system 20 includes a first IP telephone terminal 100, a second IP telephone terminal 500, and a network 600.
In FIG. 1, only two IP telephone terminals 400 and 200 are shown for transmission / reception, but the number is not limited to two, and the voice communication between the digital portable telephone terminals is mutually limited. Communication can be performed.

Since first IP telephone terminal 400 and second IP telephone terminal 500 have the same configuration, first IP telephone terminal 400 will be described below.
A first IP telephone terminal (hereinafter referred to as an IP telephone terminal) 100 is configured as shown in FIG.
Although description is omitted, the second IP phone terminal 500 uses a code obtained by changing the code number 400 of each part of the first IP phone terminal 400 to the 500 number as shown in parentheses in FIG. It has the attached components.

In FIG. 5, the IP telephone terminal 400 includes a transmission unit 410 and a reception unit 420.
The transmission unit 410 has a known configuration, and converts an audio signal (analog signal) captured from a microphone (not shown) or the like into audio data that is a digital signal by the audio encoding unit 411, and performs audio encoding. After the voice data (digital signal) from the unit 411 is temporarily stored in the transmission buffer 412, the IP packet transmission unit 413 reads the voice data (digital signal) from the transmission buffer 412, and a connection cable or a wireless LAN (not shown) ) To the network 600 and to the second IP telephone terminal 500 as the transmission destination.

The receiving unit 420 has a known configuration, and converts the received signal from the network 600 into audio data (digital signal) by the IP packet receiving unit 421, temporarily stores it in the receiving buffer 422, and then decodes the audio. The unit 423 reads out audio data (digital signal) from the reception buffer 422, decodes it into an audio signal (analog signal), and outputs it to a speaker (not shown) or the like.
Here, the reception buffer 422 is configured so that its size (storage capacity) can be changed, and its size can be set by the control unit 440 described later.

  As a result, in IP telephone terminal 400, the voice signal is digitized by transmission section 410 and transmitted via transmission buffer 412, and the voice data (digital signal) received by reception section 420 is received via reception buffer 422. The sound signal is converted into an analog signal.

Further, the IP telephone terminal 400 includes a data error detection unit 430 and a control unit 440.
The data error detection unit 430 checks the audio data (digital signal) temporarily stored in the reception buffer 422 of the reception unit 420, and detects data loss and data based on transmission delay when the network 600 is congested and timeout due to extremely narrow bandwidth. Detects data errors such as corruption and data loss. Then, the data error detection unit 430 sends the detected data error amount to the control unit 440 described later.

Based on the data error amount from the data error detection unit 430, the control unit 440 sets the size of the reception buffer 422 to the minimum when the data error amount is equal to or smaller than a predetermined value, for example, UDP (User The IP packet transmission unit 412 is controlled to use a Layer 4 transport layer protocol that does not perform retransmission control such as Datagram Protocol).
In addition, when the data error amount exceeds a predetermined value, the control unit 440 enlarges the size of the reception buffer 422 and transmits the IP packet transmission unit 413 via the network 600 via the transmission buffer 412. Then, the IP packet transmitting unit 412 is controlled so as to transmit a retransmission request to the second IP telephone terminal 500 that is the other party of the call and to use a protocol for performing retransmission control.

Furthermore, when receiving section 420 receives a retransmission request from second IP telephone terminal 500 that is the other party of the call, data error detecting section 430 detects this retransmission request, so that control section 440 The voice data (digital signal) temporarily stored in the transmission buffer 412 is retransmitted from the IP packet transmission unit 413 to the second IP telephone terminal 500 via the network 600.
Here, when the data error amount from the data error detection unit 430 becomes equal to or smaller than the predetermined value, the control unit 440 returns the size of the reception buffer 422 to the minimum again and stops transmission of the retransmission request.

  The network 600 is an IP network that provides an IP telephone service based on VoIP, and provides a telephone service between IP telephone terminals connected to the network.

Next, a voice communication method by the voice communication system 20 of this embodiment will be described.
First, voice data is transmitted from the transmission unit 410 of the first IP telephone terminal 400 to the second IP telephone terminal 500 that is the other party of the call via the network 600.

At this time, in the second IP telephone terminal 500, the voice data transmitted from the first IP telephone terminal 400 via the network 600 is temporarily stored in the reception buffer 522, and a data error is detected by the data error detection unit 530. Is done.
Here, when the data error amount is equal to or smaller than a predetermined value, the audio data is converted as it is into an audio signal (analog signal) by the audio decoding unit 523 and output to a speaker or the like.

When the data error amount exceeds a predetermined value, the control unit 540 increases the size of the reception buffer 522 and retransmits the transmission unit 510 to the first IP telephone terminal 400 that is the transmission source. Send a request.
As a result, the first IP telephone terminal 400 transmits the voice data temporarily stored in the transmission buffer 412 again. At that time, first IP telephone terminal 400 retransmits the voice data using a protocol for performing retransmission control.
In this way, in the second IP telephone terminal 500 that is the transmission destination, when audio data for a predetermined time is stored in the reception buffer 522, the audio data is received from the reception buffer 522 to the audio decoding unit 523. Is converted into an audio signal (analog signal) and output as a sound from a speaker or the like.

Then, the reception buffer 522 waits until audio data for a predetermined time is accumulated again, and outputs the audio data to the audio decoder 523.
Thereby, in the second IP telephone terminal 500, when the data error amount is equal to or smaller than the predetermined value, the size of the reception buffer 522 is set to the minimum by the control unit 540, so that the reception buffer 522 is substantially reduced. Audio data is output to the audio decoding unit 523 in real time, converted into an audio signal (analog signal), and output from a speaker or the like.
As a result, when the data error amount is less than or equal to a predetermined value, there is little influence on the call (sound interruption etc.) due to the data error. To avoid.

  When the data error amount exceeds a predetermined value, the size of the reception buffer 522 is enlarged by the control unit 540, and audio data for a certain time (for example, about 5 to 10 seconds) is accumulated and transmitted. A retransmission request is transmitted to the original first IP telephone terminal 400. Then, when audio data for a certain time is accumulated in the reception buffer 522, the audio data is output to the audio decoding unit 523, converted into an audio signal (analog signal), and output from a speaker or the like.

As a result, audio data is transmitted using a protocol that normally does not perform retransmission control with a high transmission rate, thereby improving real-time performance. In the case where data errors frequently occur, the real-time property is somewhat impaired due to the enlargement of the size of the reception buffer 522. However, the amount of data errors due to retransmission of audio data using a protocol for performing retransmission control. Since the minimum call quality can be ensured by reducing the above, voice communication becomes possible.
Note that due to the delay of the audio signal by the reception buffer 522, the time for data retransmission and accumulation of the audio data in the reception buffer 222 becomes silent without the audio signal being output from a speaker or the like.

  In the above-described embodiment, the voice communication system using the digital mobile phone terminal or the IP phone terminal has been described as the voice communication system. However, the present invention is not limited to this, and the present invention is applied to the voice communication system using other terminals. It is clear that can be applied.

  Thus, according to the present invention, in various voice communication systems such as a digital mobile phone, a so-called IP phone, or a digital voice communication system in which data loss frequently occurs between terminals, network congestion and transmission conditions are poor. Even in this case, the voice communication can be performed by reducing the amount of data error and ensuring the minimum call quality by sacrificing the real-time performance to some extent.

It is a block diagram which shows the structure of the audio | voice communication system of 1st embodiment of this invention. It is a block diagram which shows the structure regarding transmission / reception of the digital mobile telephone terminal in the audio | voice communication system of FIG. It is a block diagram which shows the structure of the network in the audio | voice communication system of FIG. It is a block diagram which shows the structure of the audio | voice communication system of 2nd embodiment of this invention. It is a block diagram which shows the structure regarding transmission / reception of the IP telephone terminal in the audio | voice communication system of FIG.

Explanation of symbols

10, 20 Voice communication system 100, 200 Digital mobile phone terminal 110, 210 Transmitter 111, 211 Voice encoder 112, 212 Transmit buffer 113, 213 Radio transmitter 120, 220 Receiver 121, 221 Radio receiver 122, 222 Reception buffer 123, 223 Speech decoding unit 130, 230 Data error detection unit 140, 240 Control unit 300 Network 310, 320 Reception unit 311, 321 Radio reception unit 312, 322 Reception buffer 330, 340 Backbone 350, 360 Transmission unit 351 362 Transmission buffer 352, 362 Wireless transmission unit 370, 380 Data error detection unit 371, 381 Control unit 400, 500 IP telephone terminal 410, 510 Transmission unit 411, 511 Speech encoding unit 412, 512 Transmission buffer 413,513 IP packet transmission unit 420, 520 wireless reception unit 421 and 521 IP packet receiver 422,522 receive buffer 423,523 speech decoder 430, 530 data error detection unit 440, 540 control unit 600 network

Claims (6)

An audio communication system that transmits audio data to each other from a transmitting unit of one transmitting / receiving terminal to a receiving unit of another transmitting / receiving terminal via a network,
Each transmitting / receiving terminal temporarily stores the audio data received by the receiving unit, a variable-size receiving buffer, a data error detecting unit for detecting a data error in the audio data stored in the receiving buffer, and a retransmission request by the transmitting unit. And a control unit that transmits to the other party's transmission / reception terminal via a network,
When the data error amount detected by the data error detection unit is less than or equal to a predetermined amount, the control unit sets the size of the reception buffer to a minimum, and when the data error amount exceeds a predetermined amount, the buffer size is increased. A voice communication system characterized in that the resending request is transmitted to the transmitting side transmitting / receiving terminal by the transmitting unit. The network includes a reception unit and a transmission unit in both directions, a variable-size reception buffer that temporarily stores audio data received by the reception unit in each direction, and audio data data stored in the reception buffer A data error detection unit that detects an error, and a control unit that transmits a retransmission request to a transmission / reception terminal by a transmission unit in each direction via a network,
When the data error amount detected by the data error detection unit is less than or equal to a predetermined amount, the control unit sets the size of the reception buffer to a minimum, and when the data error amount exceeds a predetermined amount, the buffer size is increased. The voice communication system according to claim 1, wherein the resending request is transmitted to the transmitting side transmitting / receiving terminal by the transmitting unit. 3. The voice communication system according to claim 1, wherein the network is a digital mobile phone network, and each transmitting / receiving terminal is a digital mobile phone terminal. 3. The voice communication system according to claim 1, wherein the network is an Internet connection service that provides VoIP, and each transmitting / receiving terminal is an IP telephone terminal. When each transmitting / receiving terminal transmits audio data using a protocol that normally does not perform retransmission control and the amount of data error exceeds a predetermined amount and receives a retransmission request from the transmitting / receiving terminal on the receiving side, the audio is transmitted using the protocol that performs retransmission control. The voice communication system according to claim 4, wherein data is transmitted. A voice communication method for mutually transmitting voice data from a transmission unit of one transmission / reception terminal to a reception unit of another transmission / reception terminal via a network,
At the receiving unit of each transmitting / receiving terminal, the received audio data is temporarily stored in a variable-size receiving buffer,
Detects data errors in audio data stored in the reception buffer,
When the detected data error amount is less than or equal to the predetermined amount, the size of the reception buffer is set to the minimum,
When the data error amount exceeds a predetermined amount, the size of the buffer is enlarged, and a retransmission request is transmitted to the transmitting / receiving terminal.

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