JPH10190735A - Call system - Google Patents

Abstract

PROBLEM TO BE SOLVED: To easily secure synchronization between a call origination part and a call termination part by controlling the speed of a prescribed processing in a repeater station based on a data amount measured by a processing standby data amount measurement means. SOLUTION: Once communication start signals have been received, a buffer storage amount measurement part 753 measures the storage amount of the voice data of a reception buffer 752 and sends it out to a clock frequency correction part 754. The clock frequency correction part 754 judges whether or not the storage amount of the voice data exceeds a prescribed value, and in the case of exceeding it, judges that the processing speed of an A/D conversion part provided in a transmission station is faster than the processing speed of a D/A conversion part 745 determined by the clock 746 of a call termination station and controls the clock 746 so as to raise a frequency. On the other hand, in the case of being less than the prescribed value, it is judged that the processing speed of the A/D conversion part is slower than the processing speed of the D/A conversion part 745 determined by the clock 746 of the call termination station and the clock 746 is controlled so as to lower the frequency.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a communication system and, more particularly, to a communication system for transmitting and receiving voice information via a packet communication network such as the Internet, for securing synchronization between a calling station and a called station. About.

[0002]

2. Description of the Related Art In recent years, computers such as personal computers having a voice mail function have been replaced by telephone networks or ISDN (Int.
For example, there is a communication system in which a computer connected to a public network, such as an egrated services digital network, can transmit and receive voices via a computer to the public network. Such a call system,
In particular, when making voice communications overseas, the public line usage fee is limited to local phone charges from the user terminal to the connection provider (provider), and calls can be made cheaper than when using the international telephone network. It is becoming widespread because it can be performed.

[0003] There has also been proposed a telephone communication system that uses the Internet for telephone connection between telephones without using a computer such as a personal computer. FIG. 8 is a diagram showing an example of application of the above-mentioned currently proposed call system to a call connection between telephones. In the communication system shown in FIG. 1, a telephone set 101 is connected to a relay station 103 by a public telephone network 102, while a telephone set 104 is connected to a public telephone network 1
05 is connected to the relay station 106. Also, the relay station 103 and the relay station 106
Is connected for communication.

In the communication system shown in FIG. 1, voice transmitted from the telephone set 101 is encoded by the relay station 103, and is converted into IP packets for each code representing, for example, 30 ms of voice. Then, the IP packet is transmitted to the relay station 106 via the Internet 107 at intervals of 30 ms, where the packet is unpacked and decoded into the original voice. Thus, the sound transmitted from telephone 101 is reproduced on telephone 104.

[0005]

However, although the communication on the Internet 107 is suitable for the communication of discrete data, continuous data such as voice and image data in telephone or broadcasting can be immediately transmitted to the receiving side. However, the above communication system has a problem that it is difficult to secure communication synchronization between the relay station 103 and the relay station 106.

That is, the relay station 103 and the relay station 10
6, there is a variation in the information processing performance. For example, a CODEC that converts analog audio data into digital audio data that can be transmitted through the Internet 107 or converts digital audio data transmitted from the Internet 107 into analog audio data The processing speed may be different.

In such a case, for example, if the processing speed of the relay station 103 on the transmitting side is faster than the processing speed of the relay station 106 on the receiving side, the digital voice data is transmitted to the relay station 106 on the receiving side. Since the speed of conversion into analog voice data is slower than the input speed, a delay occurs in voice reproduction and a delay time increases in proportion to the talk time. The same applies to the case where the station 103 is the receiving side.) If such a situation is left unchecked, the receiving-side relay station 1
At 06, a buffer overflow may occur.

On the other hand, if the processing speed of the relay station 106 on the receiving side is faster than the processing speed of the relay station 103 on the transmitting side, the analog voice signal is input faster than the speed at which digital voice data is input to the relay station 106 on the receiving side. Since the speed of conversion into data is increased, there is a problem that a buffer underflow occurs and a blank time is generated in sound reproduction.

Further, when the amount of packets arriving at the relay station 106 on the receiving side sharply increases or decreases due to fluctuations in the traffic on the Internet 107, the relay station 106 on the receiving side may change.
May cause buffer overflow or underflow.

[0010] The present invention has been made in view of the above problems, and has as its object to provide a communication system capable of easily ensuring synchronization between a calling side and a called side.

[0011]

In order to solve the above-mentioned problems, the present invention provides processed audio information obtained by performing a first predetermined process on audio information transmitted from a transmitting terminal. A first relay station that transmits the processed audio information transmitted from the first relay station, and performs the second predetermined processing on the processed audio information, A second relay station for transmitting to the receiving terminal, wherein the second relay station measures a data amount of processed voice information waiting for the second predetermined processing at the second relay station. Means, and a processing speed control means for controlling the speed of the second predetermined processing in the second relay station based on the data amount measured by the processing waiting data amount measuring means, I do.

In the present invention, voice information transmitted from the transmitting terminal is first received by the first relay station. Then, the first relay station performs first predetermined processing such as A / D conversion and data compression on the received voice information, and sends the processed voice information generated in this way to the second relay station. Send. Then, the second relay station performs second predetermined processing such as D / A conversion and data decompression on the received processed voice information, and transmits the voice information thus generated to the receiving terminal. In this way, the transmission terminal and the reception terminal are connected to each other.

At the time of this call connection, the processing waiting data amount measuring means measures the data amount waiting for the second predetermined processing in the second relay station. And
The processing speed control means controls the speed of the second predetermined processing in the second relay station based on the data amount measured by the processing waiting data amount measuring means.

With this configuration, synchronization of data communication with the first relay station can be easily achieved only on the side of the second relay station. That is, the delay of the sound reproduction at the second relay station and the occurrence of the blank time of the reproduced sound at the second relay station due to the difference in the data processing speed between the first relay station and the second relay station are prevented. Quality degradation can be effectively prevented.

The transmitting terminal and the receiving terminal mean a terminal on the transmitting side and a terminal on the receiving side of voice information in one-way communication or two-way communication, respectively. Therefore, the two-way communication is not limited to the calling terminal and the called terminal.

Further, the processing waiting data amount measuring means of the present invention detects the amount of data stored in a buffer provided in the second relay station, so that the processed voice waiting for the second predetermined processing is detected. It can also be configured to measure the data amount of information. This makes it possible to easily measure the amount of data waiting to be processed by the second relay station.

Further, the processing speed control means controls the operation clock of the information processing means for performing the second predetermined processing in the second relay station, so that the second relay station performs the second predetermined processing. The speed of the predetermined processing may be controlled. This makes it possible to easily control the speed of the second predetermined processing in the second relay station.

[0018]

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

FIG. 1 is a schematic diagram showing a communication system according to the present embodiment. Hereinafter, the configuration of the telephone system will be described with reference to FIG.

Reference numeral 1 denotes a calling telephone connected to a telephone network A, which is a kind of public network. Reference numeral 2 denotes a transmitting station connected to both the telephone network A and the Internet B, and a client unit 3 and a server. Part 4 is included. 5 is Internet B
And a receiving station connected to both the telephone network C and includes a server unit 6 and a client unit 7. Reference numeral 8 denotes a called telephone connected to the telephone network C. R1 is a router on the transmitting station side that determines a communication path on the Internet B, and R2 is a router on the receiving station side that determines a communication path on the Internet B.

In this embodiment, the telephone network A and the telephone network C, which are public networks, are shown as examples, but a CATV network or the Internet, which is a dedicated network, may be used instead. Also, B indicates the Internet which is a private network, but a telephone network which is a public network or a CATV network which is a private network may be used instead.

Hereinafter, each component of the communication system will be described in more detail.

First, the client section 3 of the transmitting station 2 will be described. FIG. 2 is a diagram illustrating the configuration of the client unit 3. The client unit 3 shown in FIG. 1 includes an NCU (Network Control Unit) 31 having an automatic calling / receiving function,
NCU control unit 32 having a control program for controlling CU 31, client unit 3 and server unit 4 using a communication interface such as RS232C interface
A communication interface control unit 33 for controlling communication with the
COD for encoding and decoding data such as audio
It comprises an EC 34 and a CODEC control unit 35 for controlling the CODEC 34.

Next, the server section 4 of the transmitting station 2 will be described. FIG. 3 is a diagram showing a configuration of the server unit 4. A server unit 4 shown in FIG. 1 includes a communication interface control unit 41 that controls communication between the client unit 3 and the server unit 4 using a communication interface such as an RS232C interface, and a buffer that temporarily stores transmission information, voice information, and the like. A packet communication unit 42 for packetizing the voice information and the like and transferring the packet to the receiving side;
It is configured to include a network control unit 43 that controls communication with the network.

Next, the server section 6 of the receiving station 5 will be described. FIG. 4 is a diagram showing a configuration of the server unit 6. The server unit 6 shown in FIG. 1 includes a network control unit 61 that controls communication with the Internet B, a buffer that receives a packet transmitted from the transmitting station 2 and temporarily stores the packet.
A packet communication unit 62 for transmitting voice information to the receiving terminal side;
It includes a communication interface control unit 63 that controls communication between the client unit 7 and the server unit 6 using a communication interface such as an RS232C interface.

Further, the client section 7 of the receiving station 5 will be described. FIG. 5 is a diagram showing a configuration of the client unit 7. The client unit 7 shown in the figure controls a communication interface control unit 71 that controls communication between the client unit 7 and the server unit 6 using a communication interface such as an RS232C interface, and controls an NCU 72 having an automatic transmission / reception function and an NCU 72. NCU control unit 73 having a control program to perform encoding, CODEC 74 for encoding and decoding data such as audio, CODEC 7
4 and a CODEC control unit 75 for controlling the control unit 4. Then, the CODEC control unit 75 further measures the amount of data waiting for processing in the receiving station 5 and controls the data processing speed in the receiving station 5 based on the data amount.

FIG. 6 is a diagram showing a more detailed configuration of the CODEC 74 and the CODEC controller 75. As shown in the figure, the CODEC 74 includes a compression processing unit 741, an A / D conversion unit 742, a clock 743, an expansion processing unit 744, a D /
An A conversion unit 745 and a clock 746 are included. Further, the CODEC control unit 75 includes a transmission buffer 75
1, reception buffer 752, buffer accumulation amount measurement unit 75
3 and a clock frequency correction unit 754.

The A / D converter 742 converts analog audio data input from the telephone network C into digital audio data at the operation timing of the clock 743. The compression processing unit 741 compresses the audio data digitized by the A / D conversion unit 742 and converts the compressed audio data into a CO.
Output to the DEC control unit 75. Conversely, the decompression processing unit 744 decompresses the digital audio data input from the CODEC control unit 75 and compressed. D / A converter 745
Converts the digital audio data decompressed by the decompression processing unit 744 into analog audio data at the operation timing of the clock 746. Here, the clock 746 operates at a sampling frequency of 8 KHz in an initial state, and is controlled by a switching control by a clock frequency correcting unit 754 described later.
The operating frequency can be switched in units of 100 Hz.

The transmission buffer 751 is a buffer for temporarily storing digital audio data encoded by the CODEC 74 for output to the communication interface control unit 71. The reception buffer 752 is a buffer that temporarily stores digital audio data input from the communication interface control unit 71. The buffer accumulation amount measurement unit 753
Based on the input data amount and the output data amount of the reception buffer 752, the data storage amount of the reception buffer 752 is measured, and the stored data is sent to the clock frequency correction unit 754. The clock frequency correction unit 754 controls to increase the frequency of the clock 746 by 100 Hz when the accumulation amount data input from the buffer accumulation amount measurement unit 753 indicates that the accumulation amount of the reception buffer 752 exceeds 120 BYTE. When the accumulated amount is 48BYTE or less,
Control is performed so that the clock frequency of the clock 746 is reduced by 100 Hz.

Next, the operation of the telephone system having the above-described configuration will be described.

First, when the caller goes off-hook on the calling telephone 1 and dials the telephone number of the calling station 2, a call signal is sent to the calling station 2 by an exchange (not shown) of the telephone network A. Then, the NCU control unit 3 of the client unit 3
2 receives the calling signal via the NCU 31, the calling telephone 1 and the calling station 2 are connected by the telephone network A.

Next, when the caller dials the telephone number of the called party from the calling telephone 1, the telephone number is stored in the telephone network A, the NCU 31 of the client unit 3, and the NCU.
It is transmitted to the server unit 4 via the control unit 32 and the communication interface control unit 33. The telephone number data of the other party is transmitted to the packet communication unit 42 via the communication interface control unit 41, and the telephone number data of the other party is packetized in the packet communication unit 42. Then, the packetized telephone number data of the other party is transmitted from the network control unit 43 to the router R1,
It is transmitted to the receiving station 5 via the Internet B and the router R2.

In the receiving station 5 receiving the telephone number data of the other party, the telephone number data of the other party is transmitted from the network control unit 61 of the server unit 6 to the packet communication unit 6.
Send to 2. The packet communication unit 62 unpacks the packetized telephone number data of the other party and transmits it to the client unit 7 via the communication interface control unit 63. In the client unit 7, the telephone number of the called party is transmitted via the communication interface control unit 71 to the NCU control unit 73.
Send to Then, the NCU control unit 73 automatically transmits the received telephone number of the communication partner from the NCU 72 to an exchange (not shown) of the telephone network C. Thus, when the other party (recipient) goes off-hook on the called telephone, the response signal is sent to the NCU controller 73 of the receiving station 5 and the NCU controller 3 of the calling station 2.
2 and the NCU controllers 73 and 32 transmit a call start signal to the CODEC controllers 75 and 35, respectively. In this way, the CODEC voice connection by the CODEC controller 35 of the transmitting station 2 and the CODEC controller 75 of the receiving station 5 is started.

When the voice connection is started, the voice of the caller is transmitted to the telephone network A and the CODEC 3 of the client unit 3 of the calling station 2.
4. The packet is transmitted to the packet communication unit 42 via the communication interface control unit 33 and the communication interface control unit 41. Then, the 30 ms voice is converted into one packet by the packet communication unit 42. Then, the voice packet is sent to the network control unit 43, the router R
1, transmitted to the receiving station 5 via the Internet B and the router R2. It should be noted that when the audio is encoded by the CODEC 34, a CO
The DEC control unit 35 displays a busy signal “SPEE
“CH”. Thus, one voice packet is transmitted from the transmitting station 2 to the receiving station 5 every 30 ms.

Further, the voice data is transmitted to the network control unit 61, the packet communication unit 62,
It is transmitted to the called telephone 8 via the communication interface control units 63 and 71, the CODEC control unit 75, the CODEC 74, and the telephone network C. Here, the packet communication unit 62 unpacks the audio packet in the subsequent or previous stage of the FIFO buffer (not shown), and the CODEC 74 decodes the encoded audio data. Similarly, for the voice of the other party (recipient), data transmission reverse to that described above is performed.

At the time of such a call connection, in particular, in the present call system, the CODEC 74 is used in order to prevent a delay in data processing in the terminating station 5 and the resulting stagnation of unprocessed data, and the occurrence of the above-described blank time of reproduced sound. The processing speed is controlled. FIG. 7 is a flowchart illustrating the operation of the CODEC control unit 75 that performs such processing speed control. Hereinafter, the processing will be described with reference to FIG.

As shown in FIG.
Waits for the control process until receiving the above-described call start signal from the NCU control unit 73 (S1). Upon receiving the call start signal, the buffer storage amount measurement unit 753 measures the storage amount of audio data in the reception buffer 752 and sends it to the clock frequency correction unit 754.

Then, the clock frequency correction unit 754
It is determined whether or not the amount of input audio data exceeds 120 BYTE (S2).
Is determined to be faster than the processing speed of the D / A converter 745 determined by the clock 746 of the terminating station 5 and the frequency is set to 10
The clock 746 is controlled to increase by 0 Hz (S
3).

On the other hand, when it is less than 120 BYTE, 48
It is determined whether or not BYTE or less (S4). First, 48
If not more than BYTE, A (not shown) provided in the transmitting station 2
The processing speed of the / D conversion unit is better than the clock 746 of the called station 5.
It is determined that the processing speed is lower than the processing speed of the D / A converter 745 determined by the formula (1), and the clock 746 is controlled so that the frequency is reduced by 100 Hz (S5).

If it exceeds 48 BYTE, the transmitting station 2
A D / A conversion unit 7 determined by the processing speed of an A / D conversion unit (not shown) and a clock 746 of the receiving station 5
It recognizes that the processing speed of 45 is almost the same, and cancels the control of the clock frequency by the clock frequency correction unit 754.

Then, the CODEC control unit 75
The control of the clock 746 is repeated until a signal indicating the end of the call is received from the control unit 73 (S6).

By the above control processing, the processing speed of the A / D converter (not shown) of the transmitting station 2 does not match the processing speed of the D / A converter 745 of the receiving station 5 determined by the clock 746. Also, synchronization of data communication with the transmitting station 2 can be easily achieved only on the receiving station 5 side. In other words, it is possible to prevent a delay in voice reproduction at the receiving station 5 or a blank time of reproduced voice at the receiving station 5 due to a difference in data processing speed between the transmitting station 2 and the receiving station 5, and to actively prevent deterioration of voice quality. Can be.

The communication system described above can be modified in various ways. For example, in the above embodiment, the processing speed of the CODEC 74 is controlled in the client unit 7 of the receiving station 5, but the processing speed of any processing of the client unit 7 or the server unit 6 may be controlled. .

Further, in the present embodiment, the determination whether to control the processing speed is made by the reception buffer 75 of the client unit 7.
The processing speed is controlled based on the number of stored voice data, but the processing speed may be controlled based on the number of voice packets temporarily stored for unpacking in the packet communication unit 62 of the server unit 6.

In this embodiment, the clock 746 for providing the operation timing of the CODEC 74 is controlled as the processing speed of the receiving station 5.
The processing speed may be controlled by making the accuracy of the D / A conversion by the D. 45 or the data decompression by the decompression processing unit 744 variable.

Furthermore, in the above-described embodiment, an example has been described in which the calling telephone 1 is used as a transmitting terminal and the called telephone 8 is used as a receiving terminal. However, the above-described communication system can perform two-way communication. Therefore, it is possible to transmit or make a call from the receiving telephone 8 side. In this case, if a configuration for correcting the data processing speed provided in the receiving station 5 is also added to the transmitting station 2, the delay of the sound reproduction and the occurrence of the blank time are also prevented on the transmitting station 2 side. be able to.

[Brief description of the drawings]

FIG. 1 is a diagram showing an outline of a call system.

FIG. 2 is a diagram showing a configuration of a client unit of a transmitting station.

FIG. 3 is a diagram showing a configuration of a server unit of a transmitting station.

FIG. 4 is a diagram showing a configuration of a server unit of a receiving station.

FIG. 5 is a diagram showing a configuration of a client unit of a receiving station.

FIG. 6 is a diagram illustrating a configuration of a CODEC and a CODEC control unit.

FIG. 7 is a flowchart illustrating processing by a CODEC control unit;

FIG. 8 is an explanatory diagram of a currently proposed telephone connection between telephones.

[Explanation of symbols]

B Internet, 1 calling telephone (transmitting terminal, receiving terminal), 2 transmitting station (first relay station), 5 receiving station (second relay station), 8 called telephone (receiving terminal, talk terminal) Terminal), 74 CODEC, 75 CODEC controller,
746 clocks, 753 buffer accumulation amount measurement unit (processing waiting data amount measurement unit), 754 clock frequency correction unit (processing speed control unit).

Continued on the front page (72) Inventor Mitsuo Furumura 8-10-16 Shimorenjaku, Mitaka City, Tokyo Inside Secom Co., Ltd.

Claims (3)

[Claims] 1. A first relay station for transmitting, to a receiving terminal, processed voice information obtained by performing a first predetermined process on voice information transmitted from a transmitting terminal, and the first relay station. A second relay station for receiving the processed voice information transmitted from the second terminal and transmitting voice information obtained by performing a second predetermined process on the processed voice information to the receiving terminal. A waiting data amount measuring means for measuring a data amount of processed voice information waiting for the second predetermined processing in the second relay station; and a data amount measured by the processing waiting data amount measuring means. And a processing speed control means for controlling a speed of the second predetermined processing in the second relay station based on the received information. 2. The communication system according to claim 1, wherein the processing waiting data amount measuring unit detects the amount of data stored in a buffer provided in the second relay station, thereby obtaining the second data amount. A communication system for measuring a data amount of processed voice information waiting for a predetermined process. 3. The communication system according to claim 1, wherein the processing speed control unit controls an operation clock of an information processing unit that performs the second predetermined processing in the second relay station. Controlling a speed of the second predetermined processing in the second relay station.