JP2002354537A - Communication system - Google Patents

Abstract

PROBLEM TO BE SOLVED: To stabilize speech communication quality on the occurrence of congestion control in a wired network. SOLUTION: When a radio base station 203 on a reception side receives a compacted speech signal via a wired network 40 from a radio base station 202 on a transmission side, the radio base station 203 monitors congestion control. When it has determined it as congestion control, it designates a change of a compact rate of the speech signal to the radio base station on the transmission side, and the radio base station on the transmission side changes the compact rate of the speech signal based on this designation and transmits it to the radio base station on the reception side.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

[0001] The present invention relates to a communication system for monitoring congestion on a wired network.

[0002]

2. Description of the Related Art Conventionally, as a system for making a voice call using a LAN or a WAN as a communication path as a wired network, a terminal device for transmitting and receiving a voice packet as disclosed in, for example, (1) Japanese Patent No. 3075246. As a countermeasure against delay fluctuations caused by the congestion state of the communication channel, received voice packets are stored in a buffer to some extent and absorbed / reproduced to continuously reproduce voices, and the occurrence of excessive delay fluctuations is stored in the storage buffer. A method has been proposed in which overflow or underflow is detected and the amount of buffer to be stored is increased or decreased as a control method therefor.

As a conventional example of performing congestion control of the entire system, as disclosed in, for example, (2) Japanese Patent No. 3067727, the transmitting side and the receiving side terminals performing communication are provided with reliable communication. Two communications are connected between the transmitting and receiving terminals with unreliable communication, and the transfer rate of the unreliable communication is changed by judging congestion by a reception acknowledgment in a retransmission sequence of a communication protocol by the reliable communication. ing. Further, when the transfer rate is changed, congestion is reduced by changing the time interval of the unreliable communication.

[0004] Further, as a conventional example of an Internet telephone or the like, a plurality of voice compression systems are provided, and a transfer rate including a voice compression ratio between communication terminals at the start of communication is confirmed according to a H323 standard protocol, and a telephone conversation is performed. There are also things. In particular, for sound information that is not related to a call, such as music on hold, (3) as described in Japanese Patent No. 3014366, in order to reduce the load on the apparatus itself, data of a plurality of voice compression schemes is stored and necessary. When you're going to play.

As a conventional example of the conversion of the audio compression ratio, (4) as described in Japanese Patent No. 2959627, the audio data from the communication channel is once returned to an analog signal, and then compressed again with high efficiency to store the storage device. There is also a method of transferring and storing it to

[0006]

However, the above (1)
In a system using only the buffer control method described above, the packet discard rate at the time of occurrence of congestion can be reduced, but when the occurrence of congestion becomes larger than the threshold for determining the amount of storage, the communication quality is significantly deteriorated. In addition, since the occurrence of congestion fluctuates and the data underflow or data overflow of the amount of data accumulated in the absorption buffer is used as a base point for detection of congestion detection,
When congestion occurs frequently and exceeds the control capacity at that time, data discard always occurs. Such control of the accumulation buffer when congestion occurs, when the congestion exceeds the buffer control amount accumulation amount, and occurs intermittently,
Data in the accumulation buffer may be discarded continuously.

Also, as described in (2) above, TCP / IP communication as reliable communication and UD as unreliable communication
In a system that has P / IP communication, monitors the congestion state by TCP / IP communication, and controls the transmission interval of UDP / IP communication based on the monitoring result, the UDP / IP communication is performed based on the result of TCP / IP communication. U to determine the communication speed
If congestion increases during DP / IP communication, packet discarding occurs.

Similarly, in (3) above, in a method having a plurality of voice compression schemes in an Internet telephone or the like, the voice compression rate is determined by exchanging between terminals at the start of a call. This merely synchronizes the transfer speed (including the voice compression ratio) between the communication terminals, and is not a measure for the occurrence of congestion during a call.

In the voice compression ratio conversion as described in (4), the input compression code data is once returned to the analog signal level and then compressed again. It is difficult to apply to real-time communication. Therefore, line termination equipment and PBX
For example, it is only applied as a service such as voice message recording that does not require real time.

[0010] In any of the conventional examples (1) to (4), there is a possibility that voice interruption occurs frequently due to the occurrence of buffer underflow at the time of occurrence of congestion, and the length of delay during a call due to a sudden increase or decrease of congestion. May occur.

An object of the present invention is to provide a communication system capable of stabilizing call quality when congestion occurs, in view of the problems of the conventional example.

[0012]

SUMMARY OF THE INVENTION In order to achieve the above object, the present invention provides a transmitting radio base station when a receiving radio base station determines that congestion has occurred when receiving a compressed voice signal via a wired network. Is instructed to change the compression ratio of the audio signal to prevent congestion.

That is, according to the present invention, in a communication system including a plurality of wireless base stations for performing wired communication via a wired network between a plurality of wireless terminals for transmitting and receiving data, The receiving-side radio base station monitors congestion when receiving compressed data from the transmitting-side radio base station via the wired network, and when the congestion is determined, the compression rate of the compressed data is transmitted to the transmitting-side radio base station. A communication system instructing the change, and the transmission-side radio base station changes the compression rate of the compressed data based on the compression-rate change instruction and transmits the data to the reception-side radio base station. Is done.

[0014]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is a block diagram showing an embodiment of a communication system according to the present invention, FIG. 2 is a block diagram showing a wireless base station of FIG. 1 in detail, and FIG. 3 is an explanatory diagram showing a sequence of a communication packet of the communication system of FIG. It is.

In FIG. 1, a line controller 10 and a plurality of radio base stations (CS1 to CSN) 20 to 23 are connected via a LAN 40 as a wired network. The wireless base stations 20 to 23 have wireless terminals 30 to 3 respectively.
7 are connected wirelessly. The line controller 10 establishes a communication path connection from any of the radio base stations 20 to 23 that has received a call from any of the radio terminals 30 to 37 with the LAN 40 and the radio base stations 20 to 23. Through any of them, incoming call control to any one of the wireless terminals 30 to 37 is performed.

The radio base stations 20 to 23 are radio terminals 30 to 3
7 has a wireless interface with three channels, and performs conversion between a wireless communication path from the wireless terminals 30 to 37 and a wired communication path such as the LAN 40.
Is controlled. The wireless base stations 20 to 23 also convert voice data from the wireless terminals 30 to 37 into IP packets and perform the reverse conversion, and further perform voice compression ratio conversion according to the congestion degree of the LAN 40. The wireless terminals 30 to 37 are in accordance with ITU-T Recommendation G. Wireless communication with the wireless base stations 20 to 23 is performed by 726 ADPCM 32 kbits / s unrestricted digital communication. The LAN 40 is a general private network, and corresponds to a transmission speed of 10 Mbits / s or 100 Mbits / s. 41 and 42 are wireless terminals 30 to 37 and wireless base station 2
This is a communication path in a wireless space between 0 and 23.

Next, with reference to FIG.
Will be described in detail. The wireless interface (I / F) unit 100 receives AD signals from the wireless terminals 30 to 37.
It performs extraction and transmission of voice data and wireless channel control for PCM unrestricted digital communication data. The audio compression / decompression unit 101 performs IPCM inverse conversion on the ADPCM data
TU-T Recommendation G. 8kbits of 729 CS-ACELP
Performs compression / expansion to / s data. Accumulation buffer 102
Accumulates data in order to stably perform data communication between the LAN 40 and the wireless spatial communication path 114 (41, 42).
The traffic monitoring unit 103 monitors the traffic amount and controls each function unit.

The transmission control unit 104 sets the transmission interval from the 32 kbits / s transmission rate of the unrestricted digital communication to the LAN 40 to about 36 ms, adds control information for each transmission packet, and sets L
Send to AN40. The reception control unit 105 performs reception processing from the LAN 40 and extracts connection information. The accumulation buffer transfer unit 106 sends the packet data received from the LAN 40 to the audio compression / decompression unit 101 at regular time intervals to the accumulation buffer 102, and stores the audio data transmitted to the LAN 40 in the accumulation buffer 102. However,
The transfer of the received audio data to the storage buffer 102 is started at a certain threshold or more. The call control unit 107 is connected to the LAN 40
And call control communication with the line controller 10.

Reference numeral 110 denotes a transmission / reception data path (unrestricted digital data transfer path) between the wireless I / F section 100 and the audio compression / storage section 101. Reference numeral 111 denotes a data path between the audio compression / expansion unit 101 and the storage buffer 102 (compression / compression).
(A data transfer path after transmission through the extension unit). 112 is LA
This is the path of the received packet from N40 (the received packet data transfer path from the LAN). Reference numeral 113 denotes a path for transmitting the transmission voice data in the storage buffer 102 to the LAN 40 (transmission packet data transfer path from the storage buffer). 114 is a wireless channel (wireless channel route)
It is. Reference numeral 115 denotes a connection path (call control communication data path) through which the call control unit 107 communicates with the LAN 40. 12
“0” is a reception timing signal from the reception control unit 105 to the traffic monitoring unit 103.

Reference numeral 121 denotes a packet transmission timing signal from the traffic monitoring unit 103 to the transmission control unit 104. Reference numeral 122 denotes a compression / decompression control communication path between the traffic monitoring unit 103 and the audio compression / decompression unit 101. 1
Reference numeral 23 indicates that the traffic monitoring unit 103 is
This is a communication path for controlling a wireless channel. 1
Reference numeral 24 denotes a communication path between the accumulation buffer transfer unit 106 and the traffic monitoring unit 103. Reference numeral 125 denotes a path for notifying the call control unit 107 of the suspension of a new call connection simultaneously with the suspension of the wireless channel from the traffic monitoring unit 103.

In the present invention, when monitoring and determining congestion, UDP / IP communication for voice communication is used instead of receiving TCP / IP communication used as call control communication. After the start of the voice call, voice packets are communicated between the wireless terminals 30 to 37 via the wired network (LAN 40). At this time, communication is performed between the wireless terminals 30 to 37 and the wireless base stations 20 to 23 at a fixed transfer rate, but the wireless base stations 20 to 2 through the wired network 40 are communicated.
The communication intervals among the three are not always constant.

For this reason, the radio base stations 20 to 23 have LA
An accumulation buffer 102 for absorbing fluctuations is provided on the packet reception side from N40, and the radio terminal 30
To 37. The receiving side of the radio base stations 20 to 23 measures the reception interval of the received packet from the transmitting side of the radio base stations 20 to 23 by the internal clock of the terminal itself.
Also, the number of remaining data in the accumulation buffer 102 for fluctuation absorption at the time of reception is measured.

Further, a difference value between the reception interval time and the previous reception interval time, the number of transfer remaining voice packet data for each reception, the number of data added to the previous reception voice packet data, and the number of data obtained by adding the reception voice packet data are held. The average of the difference value and the difference value set at the previous reception is set as a new difference value. In addition, the system standard transmission interval time,
Based on the data transfer speed of 37, the number of data transferred to the wireless terminals 30 to 37 until the next reception is predicted. The occurrence of buffer underflow, that is, the occurrence of congestion, is predicted from the predicted value and the current number of remaining data. Here, the buffer underflow refers to a state in which data is transferred to the wireless terminals 30 to 37 in a state where there is no received data in the fluctuation absorbing storage buffer 102. Then, the presence / absence of congestion is determined by comparing the current number of remaining data in the fluctuation absorbing accumulation buffer 102 with the predicted number of transfer data until the next reception.

Here, the determination method is based on the following method. Assuming that the reception interval time = T and the standard transfer time T0, the difference value is ΔT. Further, it holds the difference value ΔTs of the previous reception interval time and the number Ds of remaining transfer data for each reception. Next, the average of the difference value ΔT and the previous difference value ΔTn−1 is set as a new difference value ΔTn−1. Also, the wireless terminals 30 to
The following equation is provided when the transfer time interval between T.37 and Ts is Ts, the data size at that time is Dss, and the compression ratio coefficients are r1 and r2, respectively. {([Delta] Ts- [Delta] Tn-1) + [Delta] Ts + (T0 * r2)} * (Dss / r1) / Ts = Dnn (1) Equation (1) indicates that the mobile radio terminals 30 to 3 are predicted by the predicted time.
7 indicates the amount of data to be transferred to the receiving side. In the result of the above equation (1), if s = 2 and Ds / s> Dnn, it is determined that no congestion has occurred. If Ds / s <Dnn or Ds / 2 = Dnn,
It is determined that congestion has occurred.

The reason for setting s = 2 in the above is the determined value of the buffer remaining data ratio for preventing the occurrence of buffer underflow, and is also the threshold value of the fluctuation absorbing storage buffer 102. If this value is large, it means that the storage capacity is large, and this is for determining the call delay in advance. As described above, the next reception interval is statistically predicted from the difference between the reception interval difference time and the standard reception interval time, and the transfer remaining data of the accumulation buffer 102 to the reception side of the wireless terminals 30 to 37 is obtained beforehand. The feature of the present invention is that a judgment is made to prevent a buffer underflow.

If it is determined that congestion has occurred in the above determination,
It notifies the voice compression / expansion unit 101 of a change in the compression ratio, and instructs the wireless base station on the transmitting side to change the compression ratio of the voice packet. In the transmission packet processing, compression information is added to the header information to the transmission packet so that the compression method of the voice packet can be confirmed on the receiving side, and the data size for transmission per unit time is reduced by changing the compression ratio, Extending the transmission interval has the effect of suppressing the occurrence of congestion.

When changing the audio compression ratio, data of 32 kbps ADPCM, which is usually unrestricted digital data, is communicated as audio packets, and when congestion is detected, expansion / recompression processing is performed. At the same time, a notification is sent to the line controller 10 through the call control unit 107 of the radio base stations 20 to 23, and the line controller 10 synchronizes the audio compression ratios of the radio base stations 20 to 23 of the entire system to thereby make a call for the entire system. Aim for uniform and stable quality.

As described above, the voice compression ratio and the transmission interval are changed, and the traffic volume of voice packets on the LAN 40 is reduced to suppress the increase in traffic on the LAN 40. In normal traffic, the sound quality and the delay amount are generally reduced. When the traffic of the LAN 40 is increased, the communication quality according to the traffic volume can be ensured while providing the communication quality according to the telephone system.

Next, in the system shown in FIG.
The operation of the radio base station 20 when making a call with 0 and 34 will be described as an example. As an initial condition of the connection, the wireless terminal 3
0 and the wireless terminals 31 and 32 are also under the management of the wireless base station 20. Calling side wireless terminal 3 as an initial operation
0 performs off-hook, a communication path 41 is established between the calling wireless terminal 30 and the wireless base station 20, and the wireless base station 2
0 starts a call control protocol based on an IP packet with the line controller 10 through the LAN 40, and the called radio terminal 34 is connected to the radio space communication path 42 with the radio base station 22 to start a call. .

At the start of a call, the reception control unit 105
The packet received from N40 is stored in the accumulation buffer 102. The voice compression / decompression unit 101 performs data transfer without changing the voice compression. Further, the reception control unit 105 sets the traffic monitoring unit 10 every time a packet is received.
3 is notified of the reception timing. The storage buffer transfer unit 106 operates at regular intervals, and when the number of stored received packets reaches the transfer start number, the transfer start is determined by the communication path 12.
4 to the traffic monitoring unit 103. When the traffic monitoring unit 103 receives the transfer start,
Start traffic monitoring.

FIG. 3 shows an example of traffic monitoring. In FIG. 3, reference numeral 200 denotes a time when no congestion occurs on the LAN 40, and reference numeral 201 denotes a time after the congestion has occurred. A wireless base station 202 transmits a voice packet, and a wireless base station 203 determines congestion from a received packet. Here, the radio base stations 202 and 203 both transmit and receive voice packets and also perform call control. The radio base station 202 will be described as a transmitting side and the radio base station 203 will be described as a receiving side. 204 is the wireless base station 20 on the receiving side
3 is a wireless mobile terminal that receives 32 Kbps data of ADPCM, taking PHS (Personal Handy Phone System) as an example. 211, 212, and 213 are LAN 40
The upper ADPMC 32 kbps voice packet is shown. Also,
Reference numeral 205 denotes a time at which congestion is detected, and reference numeral 206 denotes a communication state after the compression ratio is changed. Also, 221 to 224 are 8 kb
Indicates an audio packet compressed by ps.

The traffic monitoring unit 103 of the wireless base station 203 on the receiving side firstly outputs a voice packet 211 of 32 kbps.
, When the received data size is 5 × 32 bytes (byte), the previous reception interval is 41 ms, and the standard reception interval is 40 m
s, the current reception interval time is 42 ms. It is assumed that the number of remaining transfer data is 6 × 32 bytes. Further, the average value of the reception interval difference up to the previous time is 2 ms. The number of received data is 4 × 3
2 byte, wireless base station 203 for wireless mobile terminal 204
Assuming that the transfer interval is 8 ms, the expected transfer data size is Dnn = {(42-40) -2+ (42-41) +40} × 32/8 = 164 bytes from equation (1). Then, transfer remaining data Ds = 6 × 32 = 192b
yte.

Here, assuming that the delay coefficient is s = 1 and the minimum delay mode is set, Ds> Dnn, and congestion determination is not detected. Then, the average difference value is set to (2 + 2) / 2 = 2. Next, assume that the reception data = 4 × 32 bytes, the previous reception interval is 42 ms, and the current reception interval is 48 ms when a 32 kbps voice packet 212 is received. The remaining data at the time of reception is 5 × 32by
As te, Dnn = ((48−40) −2 + 7 + 40) × 32/8 = 212 bytes.

Then, the remaining transfer data Ds = 5 × 32 = 1
Since it is 60 bytes, Ds <Dnn, and the next 32 kbp
When receiving the voice packet 213 of s, the prediction is 53 ms later,
It is determined that there is no more data in the accumulation buffer 102, and it is determined that congestion has occurred in the LAN 40.

Based on this determination, the traffic monitoring unit 10
No. 3 issues a “compression rate change notification” to the call control unit 107. Further, a “compression rate change notification” is also issued to the audio compression / expansion unit 101 and the transmission control unit 104. Upon receiving the “compression ratio change notification”, the call control unit 107 sets a compression ratio change flag in the status response to the line controller 10 and transmits the status response. Also,
When the compression rate change flag is set in the status response polling flag from the line controller 10, the call control unit 107 of the transmission-side radio base station 202 transmits a compression rate change notification to the traffic monitoring unit 103.

When the voice compression / decompression unit 101 of the transmission-side radio base station 202 receives the "compression rate change notification" from the traffic monitoring unit 103, it converts the data of the unrestricted digital communication from ADPCM 32 kbps to, for example, CS-ACE.
Convert to a transfer rate of LP 8 kbps. At this time, the wireless I /
A communication path 110 between the F section 100 and the audio compression / decompression section 101
By decompressing the data once to the PCM signal and compressing it again, the processing time is shortened as compared to returning the data to the analog signal. Also, the wireless base station 203 on the receiving side
From the storage buffer transfer unit 106 to the audio compression / decompression unit 10
The data of the communication path 111 to the PCM
After converting the signal, the signal is converted to ADPCM again and transmitted. This processing includes hardware processing and software processing. In this system, the processing is performed by software calculation processing in a DSP to simplify the hardware configuration.

The storage buffer transfer unit 106 of the wireless base station 203 on the receiving side transfers the received data of the storage buffer 102 and the compression ratio header information to the audio compression / decompression unit 101.
At this time, the number of read data is set to 1/4 of 32 kbits / s.
And adjust the playback time. In this system,
Normally, data is transferred in units of 32 bytes, but is transferred in units of 8 bytes by changing the compression ratio.

When the transmission control unit 104 of the transmission-side radio base station 202 receives the “compression rate change notification”, it transmits all the transmission data of the accumulation buffer 102 to the LAN 40. The compression mode flag is set in the control value of the transmission data added to the transmission packet. The transmission interval is normally about 40 ms and the transmission data size is about 160 bytes, but when the compression mode is changed, the transmission interval is extended to transmit.
Thus, traffic on the LAN 40 is reduced. In the present embodiment, the transmission interval is changed from 40 ms to 80 ms.
ms is set. The reception control unit 105 of the wireless base station 203 on the receiving side stores the control information of the received packet in the accumulation buffer 102 including the normal processing. After the completion of the control of each functional unit after the determination, a call is performed in the compression ratio mode whose setting has been changed.

Next, upon recognizing the disconnection of all call connections, the call control unit 107 of the transmission-side radio base station 202 sends an initialization notification to the traffic monitoring unit 103 via the notification path 125. Upon receiving the initialization notification, the traffic monitoring unit 103 receives the voice compression / decompression unit 101 and the transmission control unit 1 via the communication paths 122, 121, and 120, respectively.
04, Send an initialization notification to all the reception control units 105. Wireless I / F unit 100, audio compression / decompression unit 101, accumulation buffer transfer unit 106, transmission control unit 104, and reception control unit 1
05 receives the initialization notification from each notification path, and returns the compression mode to the initial state before the start of the communication, that is, the setting corresponding to the transfer rate of the unrestricted digital communication ADPCM 32 kbps.

When the receiving-side radio base station 203 receives a voice packet set in the compression mode from the transmitting-side radio base station 202, the reception control unit 105 notifies the traffic monitoring unit 103. Traffic monitoring unit 103
Receives the compression mode setting notification,
The same operation as the processing procedure is started after the traffic increase of No. 2 to synchronize the compression ratio of the transmission data between the radio base stations 202 and 203. Also, after the compression mode is changed, congestion monitoring is performed in the same manner. If congestion is not detected, the voice compression ratio is changed in the same manner and transmitted to the line controller 10. Alternatively, a change in the audio compression ratio is received from the line controller 10 and a compression ratio change process in the wireless terminal is performed.

[0041]

As described above, it is possible to reduce the traffic volume of the entire LAN by monitoring the traffic volume of, for example, the LAN connected to the private telephone system and changing the compression rate of voice packets. It can provide stable call quality. Also, in the private telephone system, the maximum number of wireless terminals and the number of wireless base stations that can be used in the dedicated communication network is the maximum system setting, but this function can be used even if the private telephone network can not be set up for the convenience of the user. It is possible to provide stable call quality.

[Brief description of the drawings]

FIG. 1 is a block diagram showing one embodiment of a communication system according to the present invention.

FIG. 2 is a block diagram illustrating a wireless base station of FIG. 1 in detail.

FIG. 3 is an explanatory diagram showing a sequence of a call packet in the communication system of FIG. 1;

[Explanation of symbols]

DESCRIPTION OF SYMBOLS 10 Line control apparatus (line controller) 20-23 Wireless base station 30-37 Wireless terminal 40 Wired network (LAN / WAN) 41, 42 Wireless spatial communication path 100 Wireless I / F section 101 Voice compression / expansion section 102 Storage buffer 103 Traffic monitoring unit 104 Transmission control unit 105 Reception control unit 106 Storage buffer transfer unit 107 Call control unit 110 Unrestricted digital data transfer path 111 Data transfer path after transmission through compression / decompression unit 112 Received packet data transfer path from LAN 113 Storage buffer Transmission packet data transmission route 114 Radio channel route 115 Call control communication data route 200 Congestion non-occurrence time 201 Congestion occurrence time 202, 203 Radio base station 204 Wireless mobile terminal 205 Congestion detection time point 206 Compression rate change data transmission / reception Between 221 and 224 8kbps compressed voice packet transmission after changing compression ratio

Claims (3)

[Claims] 1. A communication system comprising a plurality of wireless base stations for performing wired communication via a wired network between a plurality of wireless terminals for transmitting and receiving data, the receiving wireless base station comprising: Monitors congestion when receiving compressed data from the transmitting wireless base station via the wired network, instructs the transmitting wireless base station to change the compression ratio of the compressed data when it is determined that the congestion, The communication system according to claim 1, wherein the transmission-side radio base station changes the compression ratio of the compressed data based on the compression ratio change instruction and transmits the data to the reception-side radio base station. 2. The reception-side radio base station includes: a reception time interval for packets that are the compressed data to be received via the wired network; a storage amount remaining in a packet storage unit that stores the received packets; 2. The communication system according to claim 1, wherein a congestion is determined by predicting a next reception time based on the data, and congestion is detected before a data underflow of the packet storage means occurs. 3. The transmission-side radio base station changes a data compression ratio and a transmission interval based on the compression ratio change instruction, and adds the data compression ratio change information and the transmission interval change information. The communication system according to claim 1 or 2, wherein the transfer rate of the packet to the reception-side radio base station is synchronized by transferring the packet to the reception-side radio base station.