Terminal equipment for speaking system

[0001] The present invention is a terminal device of a communication system suitable for an Internet telephone that transmits and receives voice data using the Internet, which is a global computer network.<u style="single">And transmission method</u>Regarding.

[0002] The Internet is a computer network that connects computer networks existing in companies, universities, and the like via a wide area line, and is spread all over the world. Various services using the Internet, such as e-mail services, file transfer services, and information retrieval services, are already being actively carried out.

[0003] FIG. 11 shows an overview of the Internet. In FIG. 11, NET101, NET102, NET103, ... Are computer networks. The computer networks NET101, NET102, NET103, ... each have a plurality of terminals T, T, T, .... The terminals T, T, T ... of each computer network NET101, NET102, NET103, ... Are connected by LAN (Local Area Network), respectively. As a form of LAN, Ethernet, token ring, or the like is used.

[0004] Each computer network NET101, NET102, NET103, ... Is connected to each other via routers R101, R102, R103, .... Routers R101, R102, R103, ... Perform routing processing such as delivering data on a computer network to another computer network according to its destination.

[0005] As described above, the computer networks NET101, NET102, NET103, ... Are connected via the routers R101, R102, R103, ..., And the computer network network is constructed. Such a computer network is called the Internet. The Internet makes it possible to transfer data between computer networks NET1, NET2, NET3, ... all over the world.

[0006] On the Internet, IP (Internet Protocol) is used as a network layer protocol. In IP, an IP address is assigned to each terminal, and the destination to which data is transferred is identified by the IP address. An IP address consists of a collection of four numbers that can be represented by 8-bit in binary, and is expressed as, for example, "43.3.25.246".

[0007] As the Internet spreads, there is concern about a shortage of IP addresses. Therefore, when the number of terminals registered on a certain network is large and the number of terminals actually connected is small, in order to save IP addresses, a server that assigns IP addresses on the network is provided and actually connected. A free IP address may be assigned only when the terminal is connected. In this way, it is not necessary to prepare IP addresses for the number of terminals, and a limited number of IP addresses can be effectively used.

[0008] On the Internet, TCP (Transmission Control Protocol) and UDP (User Datagram Protocol) are used as transport layer protocols. TCP communicates after making a so-called connection-type communication connection, and has functions of packet order control, retransmission, flow control, and congestion control. UDP is a connectionless protocol and is used in place of TCP when real-time performance is required. For example, in digital voice communication or the like, even if a part of a packet is dropped, the packet is not requested to be retransmitted, and the voice is transmitted without interruption. In the case of such voice communication, UDP is used.

[0009] As described above, the TCP / IP protocol is basically used on the Internet. That is, an IP address is assigned to a terminal of the computer network, and this IP address is used to identify the terminal. Then, the packet is transferred by TCP or UDP.

[0010] However, personal computers are not always connected by LAN and may not have an IP address. Therefore, when an individual participates in the Internet, a company called an Internet service provider is used. By using an Internet service provider, an individual computer can join the Internet by connecting to the computer net using a telephone line, for example, PPP (Point to Point Protocol) or SLIP (Serial Line IP).

That is, FIG. 12 is an example of an Internet service provider. The computer network NET151 of the Internet service provider has a server S151 and a router R151. server<u style="single">S</u>The 151 is connected to the public telephone network TEL151 via modems M151, M152, M153, ....

[0012] Terminals T151, T152, T153, ... Are terminals of individuals who participate in the Internet. The terminals T151, T152, T153, ... Are connected to the public telephone network TEL151 via a modem (not shown). As the personal terminals T151, T152, T153, ..., A personal computer having a serial port can be used.

[0013] When participating in the Internet using an Internet service provider, it is usual that a contract is concluded between the user and the company of the Internet service provider in advance. When a contract is signed between the user and the Internet service provider, the account name and password will be sent to the user.

[0014] When joining the Internet from personal terminals T151, T152, T153, ..., The user dials the Internet service provider and calls the computer network NET151 of the Internet service provider. When the server S151 receives the call, it prompts for the account name and password, and authenticates whether or not a contract has been concluded with the Internet service provider. If the entered account name and password are correct and it is authenticated that a contract has been signed with the Internet service provider, the server S151 searches for a free IP address, and if there is a free IP address, , Assign this IP address to terminals T151, T152, T153, ... as a temporary IP address. As a result, terminals T151, T152, T153, ... will be given temporary IP addresses and will be able to connect to the Internet.

[0015] In the above example, the PPP connection is made using a telephone line, but ISDN (Integrated Service Digital Network) may be used. ISDN64 consists of two 64kbps B channels and one 16kbps D channel on one line, for a total of three channels. When ISDN is used, it can be used as a 64kbps line by sending an IP packet over the B channel.

[0016] [Problems to be Solved by the Invention] The development of an Internet telephone for making a call with a other party using the Internet is underway. Since the Internet is basically free to use, when making a call using the Internet, you can make a call only with the charges incurred by the contract with the Internet service provider, the telephone charges to the Internet service provider, and the ISDN usage charges. It will be possible to make long-distance calls and international calls at a very low rate.

[0017] When making a call using the Internet, voice data is compressed and transmitted. Various compression methods have been proposed, and various sampling frequencies of audio data such as 8 kHz, 10 kHz, and 16 kHz are used. As the sump link frequency increases, the amount of data decreases, but the sound quality deteriorates. Also, in general, the higher the compression rate, the worse the sound quality.

[0018] In a computer network such as the Internet, when the network becomes congested, it takes time to transfer data. For this reason, when the network is congested, the problem is that the audio data cannot be transferred in time and the sound is interrupted unless the sampling frequency of the audio data is lowered and the audio data is compressed and transmitted by a method with a high compression rate. Will occur. On the other hand, when the network is free, if the sampling frequency of the audio data is lowered and compressed by a high compression method for transmission, good sound quality cannot be obtained.

[0019] Therefore, an object of the present invention is a terminal device of a communication system that has good sound quality in an Internet telephone and can make a call without interruption even when the computer network is congested.<u style="single">And transmission method</u>To provide<u style="single">Ah</u>To.

[Means for Solving Problems] The present invention relates to a network.<u style="single">First server included in</u>Other terminal devices via<u style="single">When</u>Connected<u style="single">For other terminal devices</u>A terminal device that transmits data including at least voice data.<u style="single">From the data delay time between the first server and the second server connected to the network containing other terminals</u>A means to determine the network usage rate when transmitting data to the network, and an audio compression circuit that compresses audio data<u style="single">When,</u> Audio compression circuit according to network usage<u style="single">Compression rate</u>It is a terminal device provided with an audio compression circuit control means for controlling the above.

[0021] The present invention describes this invention.<u style="single">A terminal device that is connected to another terminal device via a first server included in a network and transmits data including at least voice data to the other terminal device.</u><u style="single">A means for determining the network usage rate when transmitting data to the network from the data delay time between the first server and the second server connected to the network including other terminal devices.</u><u style="single">An audio compression circuit that compresses audio data,</u><u style="single">With voice data control means that controls the sampling frequency of voice data according to the usage rate of the network</u><u style="single">It is a terminal device characterized by being provided with.</u>[0022] The present invention is a network.<u style="single">First server included in</u>Other terminal devices via<u style="single">When</u>Connected<u style="single">For other terminal devices</u>It is a transmission method that transmits data including at least audio data.<u style="single">From the data delay time between the first server and the second server connected to the network containing other terminals</u>Judge the network usage rate when sending data to the network<u style="single">Steps to do and</u> Audio data compression processing according to network usage<u style="single">Compression rate in</u>To<u style="single">With steps to control</u><u style="single">Consists of</u>This is a transmission method characterized by the above. This invention<u style="single">It is a transmission method that is connected to another terminal device via the first server included in the network and transmits data including at least voice data to the other terminal device.</u><u style="single">The step of determining the network usage rate when transmitting data to the network from the data delay time between the first server and the second server connected to the network including other terminal devices, and</u><u style="single">Steps to control the sampling frequency of audio data according to the network usage rate</u><u style="single">It is a transmission method characterized by consisting of.</u>[0023] The sampling frequency of the audio compression circuit and the compression method of the audio compression circuit are optimally set according to the usage rate of the computer network. Therefore, the sound quality is good, and even when the computer network is congested, the sound is not interrupted and a good call can be made.

[Embodiment of the Invention] The present invention is applied to an Internet telephone that communicates voice data using the Internet, and in particular, when an Internet telephone is used in a PPP connection using a public telephone network. It is suitable for use in.

[0025] FIG. 1 shows an example of an Internet telephone system to which the present invention can be applied. In FIG. 1, the computer network NET1 is, for example, a computer network of an Internet service provider. The computer network NET1 has a server S1 and a router R1.

[0026] The server S1 is connected to the public telephone line network TEL1 via modems M1, M2, M3, .... It is now possible to transmit data at 28.8 Kbps with a high-speed modem using the public telephone network TEL1.

[0027] The computer network NET1 is connected to another computer network constituting the Internet via the router R1. Router R1 performs a routing process that delivers data on a computer network to another computer network depending on its destination.

[0028] Terminals T1, T2, T3, ... Are, for example, terminals of a person who participates in the Internet individually. As terminals T1, T2, T3, ..., A personal computer in which an Internet telephone program is installed can be used, or a dedicated Internet telephone can be used. The dedicated Internet telephone is a terminal dedicated to the Internet telephone, which makes it easy to make a call using the Internet, as will be described later.

[0029] The server S1 has a database DB1. As shown in Fig. 2, the database DB1 contains "terminal name", "terminal Japanese name", "notation name on the Internet", "connection format", "PPP phone number", "user name", etc. Etc. are recorded. The database DB1 is constructed using the information obtained from the contract contents when a contract is concluded between the Internet service provider and the user, for example. This database DB1 contains information on the PPP telephone number of the person who has a contract with the Internet service provider for the PPP connection when the connection format is PPP.

[0030] In this example, the server S1 and the terminals T1, T2, T3, ... Are PPP-connected by a public telephone line network, but they are connected by using a digital network such as ISDN. Is also good.

Next, call control in the call system will be described. For example, in FIG. 1, suppose that terminal T1 calls terminal T2 to make a call. On the Internet, the other party is identified using the IP address, but in this case, the other party's terminal T2 is a terminal connected by PPP, so it is not always connected to the computer network NET1. Therefore, it is not possible to call the other party's terminal T2 using the IP address. Therefore, database DB1 is used.

That is, FIG. 3 is a flowchart showing the procedure at that time. First, the calling terminal T1 dials the Internet service provider NET1 to call the Internet service provider NET1. When the server S1 of the Internet service provider NET1 receives the call, it requests the terminal T1 to input the account name and password, and makes an authentication request as to whether or not a contract has been concluded with the Internet service provider. In response to this authentication request, the user of the calling terminal T1 enters the account name and password. If the entered account name and password are correct and it is authenticated that a contract has been signed with the Internet service provider, the server S1 assigns a temporary IP address to the terminal T1. As a result, the PPP connection with the terminal T1 is started (step ST1).

Next, the terminal T1 specifies the address of the other party to talk to (for example, the terminal T2) (step ST2).

[0034] When the address of the other party is specified, the server S1 searches the information of the terminal T2 using the database DB1. From the information in database DB1, you can find the telephone number for PPP connection of terminal T2 (step ST3).

[0035] The server S1 determines the IP address for designating the remote terminal T2 in the server, prepares for the PPP connection, and notifies the calling terminal T1 of the IP address of the remote terminal (step ST4). ).

Then, the server S1 dials the telephone number of the terminal T2 searched from the database DB1 and calls the terminal T2. When the telephone line connection with the other party terminal T2 is confirmed, the server S1 authenticates and assigns an IP address to the other party terminal (step ST5).

This initiates a PPP connection (step ST6). Then, voice data is transmitted and received between the terminal T1 and the terminal T2, and a call is made (step ST7). The voice data is compressed and transmitted. When sending and receiving voice data, UDP is used as the protocol of the transport layer.

[0038] When the call is terminated, all connections such as the PPP connection and the telephone line connection between the terminal T1 and the server S1 and the PPP connection and the telephone line connection between the terminal T2 and the server S1 are terminated (step ST8).

[0039] In the above example, the IP address of the terminal T2 is determined before the call of the terminal T2 is completed, but the IP address of the terminal T2 is determined after the call of the terminal T2 is completed. You may do it. At this time, if necessary, the determined IP address may be communicated to the calling terminal T1. When the server determines the IP address of the other party's terminal, the calling party can also be notified of the IP address, so that the calling party can prepare for communication with the other party's terminal in advance. It is possible to smoothly start the call.

[0040] As described above, when the database DB1 having the telephone number information for PPP connection is provided and the other party of the call is a PPP connection terminal (for example, T2), the telephone number information of this database DB1 is used as a base. , The other party's terminal is called. Then, the terminal T2 on the other side is connected to the server S1 by PPP. As a result, even when the other party of the call is a PPP-connected terminal, the other party can be called to make a call.

[0041] In the above example, the case where a call is made between terminals in the same computer net has been described, but it is also possible to make a call with a terminal in another computer net. FIG. 4 shows an example of making a call between terminals between different computer networks.

[0042] In FIG. 4, the computer network NET11 is, for example, a computer network of an Internet service provider. The computer network NET11 has a server S11 and a router R11. The server S11 is connected to the public telephone network TEL11 via modems M11, M12, M13, .... The server S11 has a database DB11. The database DB11 stores information including the telephone numbers of terminals connected to the computer network NET11 by PPP. The computer network NET11 is connected to other computer networks constituting the Internet via the router R11. The router R11 performs a routing process that delivers data on the network to the network according to its destination. Terminals T11, T12, T13, ... are, for example, terminals of individuals who participate in the Internet.

[0043] The computer network NET21 is, for example, a computer network of another Internet service provider. The computer network NET21 has a server S21 and a router R21. The server S21 is connected to the public telephone network TEL21 via modems M21, M22, M23, .... The server S21 has a database DB21. The database DB21 stores information including the telephone numbers of terminals connected to the computer network NET21 by PPP. The computer network NET21 is connected to other computer networks constituting the Internet via the router R21. The router R21 performs a routing process that delivers data on the network to the network according to its destination. Terminals T21, T22, T23, ... are, for example, terminals of individuals who participate in the Internet.

[0044] For example, suppose that a call is made from the terminal T11 of the computer network NET11 to the terminal T21 of the computer network NET21. In this case, the processing shown in FIG. 5 is performed.

[0045] First, the calling terminal T11 dials the network NET11 of the Internet service provider, and the server S11 of the computer network NET11 is called.

[0046] When the server S11 receives the call, it requests the input of the account name and the password, and makes an authentication request as to whether or not a contract has been concluded with the Internet service provider.

[0047] In response to this authentication request, the user of the calling terminal T11 inputs an account name and a password. If the entered account name and password are correct and it is authenticated that a contract has been concluded with the Internet service provider, the server S11 assigns a temporary IP address to the terminal T11. As a result, the PPP connection of the terminal T11 is started.

Next, the terminal T11 sends a call request to the server S11, and the server S11 returns a call answer to the terminal T11. When the call answer is returned, the address of the other party you want to call (for example, terminal T21) is specified.

When the address of the call destination is specified, the server S11 including the calling terminal makes a call request to the server S21 of the computer network including the other party (for example, computer network NET21), for example, to the terminal T21. send. When the server S21 receives a call request to the terminal T21, the server S21 returns a call answer to the server S11. When the server S11 of the network NET11 including the terminal of the calling side receives a call answer, it sends the address of the called party and also sends the information of the calling side.

[0050] The server S21 of the computer network NET21 including the other party searches the information of the terminal T21 using the database DB21. From the information in database DB21, you can find the telephone number for PPP connection of terminal T21. The server S21 of the computer network NET21 dials the telephone number of the terminal T21 searched from the database DB21 and calls the terminal T21.

[0051] When the terminal T21 on the other side receives the call from the server S21, the terminal T21 returns the call response to the server S21. When the server S21 receives the call response, it makes a PPP connection request, and when the terminal T21 receives the PPP connection request, it notifies that it will make a PPP connection.

[0052] Then, the server S21 requests the input of the account name and the password, and authenticates. The user on the other side enters the account name and password in response to this authentication. If it is confirmed that the entered account name and password are correct and a contract has been concluded with the Internet service provider, the IP address will be assigned to the terminal T21. As a result, the PPP connection of the terminal T21 is started.

[0053] When the PPP connection is started, the call request is sent from the server S21 to the terminal T21, and the call answer is returned from the terminal T21 to the server S21. Then, the call completion is sent from the server S21 of the computer network NET21 to the server S11 of the computer network NET11, and the call completion is sent from the server S11 to the terminal T11. Then, voice data is transmitted and received between the terminal T11 and the terminal T21, and a call is made.

[0054] When the call is completed and, for example, a disconnection request is issued from the calling side terminal T11, this disconnection request is sent to the other party terminal T21. Upon receiving the disconnection request, the terminal T21 returns a disconnection response to the terminal T11 and disconnects all connections.

By the way, when making a call using the Internet in this way, the voice data is compressed and transmitted. Various compression methods have been proposed, and various sampling frequencies of audio data such as 8 kHz, 10 kHz, and 16 kHz are used. If the sampling frequency is low, the amount of data is reduced, but the sound quality is deteriorated. Also, in general, the higher the compression rate, the worse the sound quality. When the network becomes congested, it takes time to transfer data. For this reason, when the network becomes congested, there arises a problem that the audio data cannot be transferred in time and the sound is interrupted unless the sampling frequency of the audio data is lowered and the audio data is compressed and transmitted by a method having a high compression rate. Come on. On the other hand, when the network is free, if the sampling frequency of the audio data is low and the audio data is compressed and transmitted by a high compression method, good sound quality cannot be obtained.

[0056] Therefore, it is conceivable to determine the network usage rate and optimally set the sampling frequency and the compression method according to the network usage rate. That is, the usage rate of the network is judged, and when the network is not congested, the sampling frequency is increased and transmission is performed by a compression method with high sound quality, and when the network is congested, the sampling frequency is decreased and the compression rate is high. Transmit by method.

[0057] FIGS. 6 to 8 show an example of an Internet telephone to which the present invention is applied. This Internet phone determines the usage rate of the computer network, raises the sampling frequency when the network is not congested, transmits with a compression method with a low compression rate, and lowers the sampling frequency when the network is congested. It is transmitted by a compression method with a high compression rate.

[0058] In FIG. 6, reference numeral 1 denotes a telephone body. A display operation unit 2 is provided on the upper surface of the telephone body 1. As shown in FIG. 7, the display operation unit 2 has a structure in which the touch panel 4 is laminated on the display panel 3. An icon indicating a number key, an operation key, or the like is displayed on the display operation unit 2, and when the touch panel 4 on this icon is pressed, necessary input can be performed. In addition to the icons indicating the keys and the like, the display operation unit 2 displays help for providing operation guidance and a setting state. Further, various information is displayed on the display operation unit 2. Further, the handset 5 is connected to the telephone body 1. The telephone body 1 is connected to a public telephone line.

[0059] FIG. 8 shows an internal configuration of an Internet telephone to which the present invention is applied. In this example, the controller 10, the communication control circuit 14, the voice compression circuit 13, and the voice decompression circuit 17 are represented by separate blocks, or these can be realized by software.

[0060] In FIG. 8, the audio signal from the microphone 11 of the handset 5 is supplied to the A / D converter 12. A sampling clock is supplied to the A / D converter 12 from the clock generation circuit 20. The clock generation circuit 20 can generate clocks of, for example, 4kHz, 6kHz, 8kHz, 10kHz, 12kHz, and 16kHz. The clock frequency from the clock generation circuit 20 is controlled by the controller 10.

The A / D converter 12 digitizes the audio signal from the microphone 11. The output of the A / D converter 12 is supplied to the audio compression circuit 13. This audio data is compressed by the audio compression circuit 13. The audio compression circuit 13 can set a plurality of compression / decompression methods such as a CVSD method and a GSM method. The compression method of the audio compression circuit 13 is controlled by the controller 10.

[0062] The output of the audio compression circuit 13 is supplied to the communication control circuit 14. The communication control circuit 14 performs communication control such as packetization of data. The output of the communication control circuit 14 is supplied to the telephone line via the modem 15 and the NCU (Node Control Unit) 16 and sent to the server side.

[0063] The voice data sent from the server side is supplied to the communication control circuit 14 via the NCU 16 and the modem 15. The communication control circuit 14 controls communication such as packet decomposition. The output of the communication control circuit 14 is supplied to the voice extension circuit 17. The audio decompression circuit 17 can set a plurality of compression / decompression methods such as a CVSD method and a GSM method. voice<u style="single">Stretch</u>Of circuit 17<u style="single">Stretch</u>The method is controlled by the controller 10.

[0064] The output of the audio expansion circuit 17 is supplied to the D / A converter 18. A sampling clock is supplied to the D / A converter 18 from the clock generation circuit 20. As described above, the clock generation circuit 20 can generate clocks of, for example, 4kHz, 6kHz, 8kHz, 10kHz, 12kHz, and 16kHz, and the clock frequency is controlled by the controller 10. The output of the D / A converter 18 is supplied to the speaker 19 of the handset 5.

[0065] Input is given to the controller 10 from the touch panel 4. In addition, the display output from the controller 10 is displayed on the display panel 3.

[0066] The controller 10 performs overall control, dial connection processing, audio compression method switching processing, and sampling frequency switching processing.

That is, when the address of the other party is input from the touch panel 4, the controller 10 controls the NCU 16 and dials the telephone number of the Internet service provider to process the telephone connection. Then, when the telephone connection is completed, the connection process by PPP is performed. In addition, you may be required to make a PPP connection by receiving a call from your Internet service provider. In this case as well, the controller 10 processes the telephone connection, and when the telephone connection is completed, performs the connection process by PPP.

Then, as shown in FIG. 9, it is determined whether or not the predetermined time has elapsed (step ST11), and when the predetermined time has elapsed, the server is requested to send the line usage rate (step ST12). Whether or not the line usage rate has been sent from the server is determined (step ST13), and when the line usage rate is sent, the optimum clock frequency and audio compression method are determined according to the line usage rate (step ST14). ). Then, the determined clock frequency and audio compression method are sent to the other terminal (step ST15), and based on the determined clock frequency and audio compression method, the clock frequency of the clock generation circuit 20 and the audio compression circuit 13 , The compression method of the audio decompression circuit 17 is set (step ST16).

[0069] On the server side, for example, as shown in FIG. 10, the line usage rate can be determined.

[0070] In FIG. 10, the timer for measuring the delay time is reset (step ST21), and an echo request is transmitted to the server on the other side (step ST22). When the echo request is sent, the timer for measuring the delay time is started to be measured (step ST23), and the delay time until the echo reply for this echo request is returned is measured (step ST24). When the echo reply is returned, the measurement of the delay time is completed (step ST25).

The delay time from sending this echo request to returning the echo reply reflects the degree of line congestion, that is, the line usage rate. If the delay time is long, it means that the line is congested and the line usage rate is high.

[0072] In addition to measuring the delay time from sending the echo request to returning the echo reply, the congestion degree of the line is congested by the delay time between the servers using the control packet. You may try to measure the degree. Further, the measurement may be performed by using the Ping command between the connected terminals.

[Effectiveness of the Invention] According to the present invention, the sampling frequency of the audio compression circuit and the compression method of the audio compression circuit are optimally set according to the usage rate of the computer network. Therefore, the sound quality is good, and even when the computer network is congested, the call can be made without interruption of the sound.

BRIEF DESCRIPTION OF THE DRAWINGS [FIG. 1] FIG. 1 is a block diagram used for explaining an example of an Internet telephone system to which the present invention can be applied.

FIG. 2 is a schematic diagram used for explaining an example of an Internet telephone system to which the present invention can be applied.

FIG. 3 is a flowchart used for explaining an example of an Internet telephone system to which the present invention can be applied.

FIG. 4 is a block diagram used to explain another example of an Internet telephone system to which the present invention has been applied.

FIG. 5 is a flow chart used to explain another example of an Internet telephone system to which the present invention can be applied.

FIG. 6 is a perspective view of an example of a telephone of an Internet telephone system to which the present invention is applied.

FIG. 7 is a cross-sectional view used for explaining an example of a telephone of an Internet telephone system to which the present invention is applied.

FIG. 8 is a block diagram of an example telephone of an Internet telephone system to which the present invention has been applied.

FIG. 9 is a flowchart used for explaining a telephone of an Internet telephone system to which the present invention is applied.

FIG. 10 is a flowchart used for explaining a telephone of an Internet telephone system to which the present invention is applied.

FIG. 11 is a block diagram used for explaining the Internet.

FIG. 12 is a block diagram used for explaining a PPP connection.

[Code description] 10 Controller 12 A / D converter 13 Voice compression circuit 14 Call control circuit 17 Voice extension circuit 18 D / A converter