Abstract:
A telephone apparatus of a telephone system, including at least one server for connecting a plurality of telephone apparatuses connected to a computer network, comprises: 
     a communication circuit means for processing data to and from said computer network; an audio input means for generating a digital audio signal from input voice; an audio compression means for compressing the audio signal from the audio input means and for supplying the compressed audio signal to the communication circuit means; and a control means for determining a compression rate of the audio compression means in response to information on the ratio of use of the computer network obtained from the server.

Description:
BACKGROUND OF THE INVENTION 
     1. Field of the Invention 
     This invention relates to a telephone system suitable for use as an internet telephone for exchanging audio data through the internet that is a worldwide computer network system. 
     2. Related Art 
     The internet is a worldwide computer network system connecting computer networks in corporations or universities beyond countries. Increasingly provided are various services using the internet, such as e-mail service, file transfer service, and information search service. 
     FIG. 1 schematically shows a general aspect of the internet. In FIG. 1, each of computer networks NET 101 , NET 102 , NET 103 , . . . has a plurality of terminals T which are connected together by LAN (Local Area Network) in form of Ethernet or a token ring. 
     These computer networks NET 101 , NET 102 , NET 103 , . . . are connected together through routers R 101 , R 102 , R 103 , . . . that route data from a computer network to another, depending on the destination of the data. 
     Computer networks NET 101 , NET 102 , NET 103 , . . . connected through the routers R 101 , R 102 , R 103 , . . . form a computer network system. The computer network system is called internet. The internet enables exchanges of data among computer networks NET 101 , NET 102 , NET 103 , . . . 
     The internet uses IP (Internet Protocol) as the protocol of its network layer. IP assigns an IP address to each terminal to identify a destination terminal of data. Each IP address is made up of four numerals each of which can be expressed by decimal 8 bits, such as 43.3.25.246. 
     As the internet is extended, the number of IP addresses possibly becomes insufficient. In some networks in which a large number of terminals are registered but only a small number of terminals are connected simultaneously, for example, it is possible to use a server on the network to allot currently available IP addresses to actually connected terminals alone in order to minimize the number of IP addresses used. In this manner, the network need not prepare IP addresses in the number corresponding to its terminals, but can effectively use a limited number of IP addresses. 
     The internet uses TCP (Transmission Control Protocol) and UDP (User Datagram Protocol) as protocols of its transport layer. TCP permits communication after establishing a connection-type transmission connection, and deals with packet sequence control, re-transmission, flow control and congestion control. UDP is a connectionless-type protocol that is used in lieu of TCP in networks requiring real-time transmission. In digital audio transmission, for example, re-transmission is not requested even when a part of packets drops, but audio data is sent successively. In such audio transmission, UDP is used. 
     Thus, the internet basically uses TCP/IP protocol. That is, IP addresses are assigned to terminals of a computer network to identify individual terminals, and packets are transferred by TCP or UDP. 
     However, personal computers are not always connected by LAN, and there are some without IP addresses. Therefore, some individuals participating the internet use internet service providers. Through internet service providers, personal computers can be connected to computer networks and can participate the internet by, for example, PPP (Point to Point Protocol) or SLIP (Serial Line IP) through telephone lines. 
     FIG. 2 shows a construction of an internet service provider. The computer network NET 151  of the internet service provider includes a server S 151  and a router R 151 . The server S 151  is connected to a public telephone line network TEL 151  via modems M 151 , M 152 , M 153 , . . . 
     Terminals T 151 , T 152 , T 153 , . . . are those of individuals personally participating the internet. 
     Terminals T 151 , T 152 , T 153 , . . . are connected to the public telephone line-network TEL 151  through modems (not shown). Individual terminals T 151 , T 152 , T 153 , . . . may be personal computers having serial ports. 
     For participation in the internet through an internet service provider, users previously make a contract with an internet service provider in most cases. When a contract is concluded between a user and an internet service provider, an account code and a password are sent to the user. 
     When an individual participates in the internet from one of the terminals T 151 , T 152 , T 153 , . . . the user dials into the internet service provider to call up the server S 151  of the computer network NET 151  of the provider. The server S 151  responsively requests entry of the account code and the password for authentication whether the user is a contractor. When the server S 151  authenticates that the entered account code and password are those of a contractor, it searches for an available IP address. If there is any IP address available, it temporarily assigns it to the terminal T 151 , T 152 , T 153 , or any other. Thus, the terminal obtaining the temporary IP address can connect to the internet. 
     In the above example, terminals are connected by PPP using telephone lines. However, ISDN (Integrated Service Digital Network) may be used alternatively. ISDN  64  includes three channels, namely, two B channels of 64 kbps and one D channel of 16 kbps. When ISDN is used, it can be used as a line of 64 kbps by sending IP packets on the B channels. 
     Internet telephones for effecting telephone communication using the internet are now being developed. Since the internet is basically free of charge, what is to be paid by the user for internet telephone communication through the internet is the charge based on the contract with the internet service provider and the charge for the call between the user and the internet service provider or the charge for the use of ISDN. Thus, users can enjoy long-distance telephone calls and international telephone calls very economically. 
     For a telephone call using the internet, audio data is transmitted in a compressed form. There are various systems for compression, and various sampling frequencies of audio data, such as 8 kHz, 10 kHz, 16 kHz, and so on, are used. As the sampling frequency becomes high, the data amount decreases, but the quality of sound becomes better. Systems using high compression rates are liable to deteriorate the quality of sound. 
     In a computer network system like the internet, the time for data transmission increases as the network becomes crowded. Therefore, while the network is crowded, it is necessary to use a lower sampling frequency for audio data and to employ a compression system with a higher compression rate in order to prevent that voice is interrupted due to a delay of transmission of the audio data. On the other hand, it is not wise to send audio data with a low sampling frequency and a high compression rate even when the network is not crowded because the quality of sound is not good. 
     OBJECTS AND SUMMARY OF THE INVENTION 
     It is therefore an object of the invention to provide a terminal apparatus of a telephone system used as an internet telephone apparatus, which ensures telephone conversation with a high quality of sound with no interruption of voice even when a computer network system is crowded. 
     According to the invention, there is provide a telephone apparatus of a telephone system, including at least one server for connecting a plurality of telephone apparatuses connected to a computer network, comprising: 
     a communication circuit means for processing data to and from said computer network; an audio input means for generating a digital audio signal from input voice; an audio compression means for compressing the audio signal from the audio input means and for supplying the compressed audio signal to the communication circuit means; and a control means for determining a compression rate of the audio compression means in response to information on the ratio of use of the computer network obtained from the server. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a block diagram for use in explanation of the internet; 
     FIG. 2 is a block diagram for use in explanation of PPP connection. 
     FIG. 3 is a block diagram for use in explanation of an internet telephone system to which the invention is applicable; 
     FIG. 4 is a schematic diagram for use in explanation of an internet telephone system to which the invention is applicable; 
     FIG. 5 is a flow chart for use in explanation of an internet telephone system to which the invention is applicable; 
     FIG. 6 is a block diagram for use in explanation of another internet telephone system to which the invention is applied; 
     FIG. 7 is a flow chart for use in explanation of another internet telephone system to which the invention is applicable; 
     FIG. 8 is a perspective view of a telephone set in an internet telephone system to which the invention is applied; 
     FIG. 9 is a cross-sectional view for use in explanation of a telephone set in an internet telephone system to which the invention is applied; 
     FIG. 10 is a block diagram of a telephone set in an internet telephone system to which the invention is applied; 
     FIG. 11 is a flow chart for use in explanation of a telephone set in an internet telephone system to which the invention is applied; and 
     FIG. 12 is a flow chart for use in explanation of a telephone set in an internet telephone system to which the invention is applied. 
    
    
     DESCRIPTION OF THE PREFERRED EMBODIMENTS 
     The present invention is applied to an internet telephone for transmitting audio data through the internet, and is especially suitable for use of internet telephones connected by PPP through public telephone networks. 
     FIG. 3 shows an internet telephone system to which the invention is applicable. In FIG. 3, a computer network NET 1  is, for example, a computer network prepared by an internet service provider. The computer network NET 1  contains a server S 1  and a router R 1 . 
     The server S 1  is connected to a public telephone network TEL 1  through modems M 1 , M 2 , M 3 , . . . Currently, data can be transmitted at the rate of 28.8 kbps through the public telephone network TEL 1  by using a high-speed modem. 
     The computer network NET 1  is connected to other computer networks forming the internet through the router R 1 . The router R 1  routes data on the computer network to another computer network containing a destination terminal. 
     Terminals T 1 , T 2 , T 3  . . . are those of individuals personally participating the internet. Individual terminals T 1 , T 2 , T 3 , . . . may be personal computers installed with an internet telephone program or exclusive internet telephone sets. Exclusive internet telephone sets are terminals exclusive to internet telephones facilitating telephone communication using the internet as explained later. 
     The server S 1  has a data base DB 1 . As shown in FIG. 4, the data base DB 1  stores “terminal names”, “internet names”, “connection types”, “public phone numbers for PPP”, “users&#39; names”, and others. The data base DB 1  may be established using information obtained from contents of contracts concluded between the internet service provider and users. When the connection type is PPP, the data base DB 1  contains telephone numbers for PPP of users contracting with the internet service provider by PPP connection. 
     Although the terminals T 1 , T 2 , T 3 , . . . are connected to the server S 1  by PPP through the public telephone network in the above example, they may be connected through a digital network such as ISDN. 
     Next explained is a telephone call control in a telephone system. Assume here that a telephone call from the terminal T 1  to the terminal T 2  is desired in FIG.  3 . The internet requires an IP address to specify a destination terminal. In this case, it is possible that the destination terminal T 2  to be connected by PPP is not currently connected to the computer network NET 1  and cannot be accessed to by using its IP address. To cope with the matter, the data base DB 1  is used. 
     FIG. 5 is a flow chart showing the accessing process using the data base DB 1 . First, the source terminal T 1  dials the computer network NET 1  of the internet service provider to call up the server S 1  of the computer network NET 1 . Responsively, the server S 1  requests the terminal T 1  to enter its account code and the password to authenticate whether the source terminal T 1  is one of contractors of the internet service provider. The user of the source terminal T 1  answers the request by entering its account code and the password. When the server S 1  authenticates that the entered account code and password are those of a contractor, it assigns a temporary IP address to the terminal T 1 . Thus, PPP connection with the terminal T 1  is started (step ST 1 ). 
     After that, the terminal T 1  designates a desired destination address (for example, terminal T 2 ) (step ST 2 ). 
     Responsively, the server S 1  searches into the data base DB 1  to find out information on the terminal T 2  corresponding to the requested destination address. The telephone number of the terminal T 2  for PPP connection can be known from information in the data base DB 1  (step ST 3 ). 
     The server S 1  subsequently determines an IP address for specifying the destination terminal T 2  within the server to prepare for PPP connection, and gives a notice on the destination terminal&#39;s IP address to the source terminal T 1  (step ST 4 ). 
     Then, the server S 1  dials the telephone number of the terminal T 2  found out from the data base DB 1  to call up the terminal T 2 . When connection of the telephone line to the destination terminal T 2  is confirmed, the server S 1 , after authentication, assigns the IP address to the destination terminal (step ST 5 ). 
     PPP connection is thus started (step ST 6 ). As a result, audio data is exchanged for communication between the terminals T 1  and T 2  (step ST 7 ). The audio data is transmitted in a compressed form. For exchanging audio data, UDP is used as the protocol of the transport layer. 
     When the communication ends, all connection including PPP connection and telephone line connection between the terminal T 1  and the server S 1 , those between the terminal T 2  and the server S 1  is disconnected (step ST 8 ). 
     Although the above example is configured to determine the IP address of the terminal T 2  prior to completing access to the terminal T 2 , the IP address of the terminal T 2  may be determined after the access to the terminal T 2  is completed. It is also possible to inform the source terminal T 1  of the assigned IP address, if necessary. When the source terminal is informed of the IP address at the time when the server determines the IP address of the destination terminal, the source terminal can prepare for communication with the destination terminal such that the terminals can smoothly proceed to communication. 
     In this manner, the data base DB 1  is provided which stores information on telephone number for PPP connection, and a destination terminal is accessed to through the telephone number obtained from the data base DB 1  when the destination terminal is a PPP-connected terminal (T 2 , for example). Then, the destination terminal T 2  is connected to the server S 1  by PPP. Therefore, even when the destination terminal is a PPP-connected terminal, the destination terminal can be called up for communication. 
     In the above example, a terminal is connected for communication with another terminal in a common computer network. However, a terminal in a computer network can be connected for communication also with a terminal in a different computer network. FIG. 6 shows an example where terminals in different computer networks are connected for communication. 
     In FIG. 6, a computer network NET 11  is, for example, a computer network prepared by an internet service provider. The computer network NET 11  contains a server S 11  and a router R 11 . The server S 11  is connected to a public telephone network TEL 11  through modems M 11 , M 12 , M 13 , . . . The server S 11  has a data base DB 11 . The data base DB 11  stores information containing telephone numbers of terminals connected by PPP to the computer network NET 11 . The computer network NET 11  is connected to other computer networks forming the internet through the router R 11 . The router R 11  routes data on the computer network to an appropriate computer network containing a destination terminal. Terminals T 11 , T 12 , T 13  . . . are those of individuals personally participating the internet. 
     A computer network NET 21  is, for example, a computer network prepared by another internet service provider. The computer network NET 21  contains a server S 21  and a router R 21 . The server S 21  is connected to a public telephone network TEL 21  through modems M 21 , M 22 , M 23 , . . . The server S 21  has a data base DB 21 . The data base DB 21  stores information containing telephone numbers of terminals connected by PPP to the computer network NET 21 . The computer network NET 21  is connected to other computer networks forming the internet through the router R 21 . The router R 21  routes data on the computer network to an appropriate computer network containing a destination terminal. Terminals T 21 , T 22 , T 23  . . . are those of individuals personally participating the internet. 
     Assume here that the terminal T 11  desires a telephone call to the terminal T 12 . In this case, a process is progressed as shown in FIG.  7 . 
     First, the source terminal T 11  dials the computer network NET 11  of the internet service provider to call up the server S 11  of the computer network NET 11 . Responsively, the server S 11  requests the terminal T 11  to enter its account code and the password for authentication whether the source terminal T 11  is one of contractors of the internet service provider. 
     The user of the source terminal T 11  answers the authentication request by entering its account code and the password. When the server S 11  confirms that the entered account code and password are those of a contractor, it assigns a temporary IP address to the terminal T 11 . Thus, PPP connection of the terminal T 11  is started. 
     After that, the terminal T 11  sends a call request to the server S 11 , and the server S 11  sends back a call approval. In receipt of the call approval, the terminal T 11  gives a desired destination address (terminal T 21 , for example). 
     The server S 11  connected to the source terminal in receipt of the destination address sends a call request for communication with the terminal T 21 , for example, to the server S 21  of the computer network (NET 21 , for example) containing the destination terminal. In receipt of the call request for communication with the terminal  21 , the server S 21  sends back a call approval to the server S 11 . In receipt of the call approval, the server S 11  of the network NET 11  containing the source terminal sends the destination address and information on the source terminal. 
     The server S 21  of the computer network NET 21  containing the destination terminal searches into the data base DB 21  to find out information on the terminal T 21 . The telephone number of the terminal T 21  for PPP connection can be known from information of the data base DB 21 . The server S 21  of the computer network NET 21  dials the telephone number of the terminal T 21  obtained from the data base DB 21  to call up the terminal T 21 . 
     The destination terminal T 21  accessed by the server S 21  sends back an acknowledgement. The server S 21  in receipt of the acknowledgement requests PPP connection, and the terminal T 21  in receipt of the request for PPP connection gives confirmation of PPP connection. 
     The server S 21  then requests entry of the account code and the password for authentication. In response to the authentication, the user of the destination terminal enters the account code and the password. When the entered account code and password are confirmed to be those of a proper contractor, an IP address is assigned to the terminal T 21 . Thus, PPP connection of the terminal T 21  is started. 
     When the PPP connection is started, a call request is sent from the server S 21  to the terminal T 21 , and a call approval is sent back from the terminal T 21  to the server S 21 . Then, the server S 21  of the computer network NET 21  sends a call connection completion notice to the server S 11  of the computer network NET 11 , and the server S 11  sends a call connection completion notice to the terminal T 11 . As a result, audio data is exchanged for communication between the terminals T 11  and T 21 . 
     When a disconnection request is issued from the source terminal T 11 , for example, after the communication ends, the disconnection request is sent to the destination terminal T 21 . In receipt of the disconnection request, the terminal T 21  sends back a disconnection agreement to the terminal T 11 , and all connection is disconnected. 
     For a telephone call using the internet, audio data is transmitted in a compressed form. There are various systems for compression, and various sampling frequencies of audio data, such as 8 kHz, 10 kHz, 16 kHz, and so on, are used. As the sampling frequency becomes high, the data amount decreases, but the quality of sound becomes better. Systems using high compression rates are liable to deteriorate the quality of sound. A network takes more time to transmit data as the network becomes crowded. Therefore, while the network is crowded, it is necessary to use a lower sampling frequency for audio data and to employ a compression system with a higher compression rate in order to prevent that voice is interrupted due to a delay of transmission of the audio data. On the other hand, it is not wise to send audio data with a low sampling frequency and a high compression rate even when the network is not crowded because the quality of sound is not good. 
     Taking it into account, it is advisable to flexibly selecting an optimum sampling frequency and an optimum compression system, depending on the current traffic density of the network. That is, by accounting the current traffic density of the network, data is transmitted using a higher sampling frequency and a compression system promising a higher quality of sound when the network is not crowded, and using a lower sampling frequency and a compression system of a high compression rate when the network is crowded. 
     FIGS. 8 through 10 show an internet telephone apparatus to which the invention is applied. The internet telephone apparatus accounts the traffic density of the computer network to use a high sampling frequency of data and transmit the data by a compression system using a low compression rate when the network is not crowded and to use a low sampling frequency and transmit the data by a compression system using a high compression rate. 
     In FIG. 8, numeral  1  denotes a telephone main body. The telephone main body  1  has a display/operator  2  on its upper surface. The display/operator  2  is a multi-layered panel including a touch panel  4  stacked on a display panel  3  as shown in FIG.  9 . The display/operator  2  displays icons of numerical keys, operational keys, and so forth, which permit a user to enter a desired instruction by pressing the touch panel  4  at the portion of a corresponding icon. The display/operator  2  also displays help messages explaining how to operate the keys and the current modes of setting in addition to key icons, etc. Other various information is also displayed on the display/operator  2 . A handset  5  is connected to the telephone main body  1 , and the telephone main body  1  is connected to a public telephone circuit. 
     FIG. 10 shows the interior construction of the internet telephone apparatus to which the invention is applied. Although a controller  10 , communication control circuit  14 , audio compression circuit  13  and audio expansion circuit  17  are shown in separate blocks, they can be realized as software. 
     In FIG. 10, audio signals from a microphone of the handset  5  are supplied to an A/D converter  12 . The A/D converter  12  is supplied with sampling clocks from a clock generating circuit  20 . The clock generating circuit  20  can generate clocks of 4 kHz, 6 kHz, 8 kHz, 10 kHz, 12 kHz, and 16 kHz, for example. The clock frequency from the clock generating circuit  20  is controlled by the control circuit  10 . 
     The A/D converter  12  converts the audio signal from the microphone  11  into a digital signal and supplies it to the audio compression circuit  13 . The audio compression circuit  13  compresses the received signal. The audio compression circuit  13  is configured to select any one of a plurality of compression/expansion systems such as CVSD system, GSM system and others. The compression system of the audio compression circuit  13  is controlled by the controller  10 . 
     Output from the audio compression circuit  13  is supplied to the communication control circuit  14  for communication control such as processing of data into packets. Output from the communication control circuit  14  is put on the telephone line through a modem  15  and NCU (Node Control Unit)  16  and sent to a server. 
     Audio data sent from the server is supplied to the communication control circuit  14  through NCU  16  and the modem  15 . The communication control circuit  14  executes communication control such as decomposition of packets. Output from the communication control circuit  14  is supplied to the audio expansion circuit  17  that can employ any one of a plurality of compression/expansion systems such as CVSD system, GSM system and others under control of the controller  10 . 
     Output form the audio expansion circuit  17  is supplied to a D/A converter  18  that is also supplied with sampling clocks from the clock generating circuit  20 . As explained above, the clock generating circuit  20  can generate clocks of 4 kHz, 6 kHz, 8 kHz, 10 kHz, 12 kHz, and 16 kHz, for example under control of the controller  10 . Output from the D/A converter  18  is supplied to the speaker  19  of the handset  5 . 
     Inputs from the touch panel  4  are given to the CPU  11 , and outputs from the CPU  11  are displayed on the display panel  3 . 
     The controller  10  executes dial connection processing, processing for switching the audio compression system, and processing for switching the sampling frequency. 
     That is, when the controller  10  receives a destination address and other materials entered through the touch panel  4 , it controls NCU  16  to dial the telephone number of the internet service provider to perform the telephone connection processing. When the telephone connection is completed, it proceeds to PPP connection processing. In some cases, a call may arrive from the internet service provider, and PPP connection is requested. Also in such cases, the controller  10  executes telephone connection processing and, after completion of telephone connection, connection processing by PPP. 
     Then, as shown in FIG. 11, it is judged whether a predetermined time has passed or not (step ST 11 ). If it has passed, the server is requested to inform the traffic density of the telephone circuit (step ST 12 ). It is reviewed whether the traffic density has been informed from the server (step ST 13 ). If it has been informed, an optimum clock frequency and an optimum audio compression system are chosen in accordance with the traffic density (step ST 14 ), and they are sent to the destination terminal (step ST 15 ). According to the determined clock frequency and audio compression system, the clock frequency of the clock generating circuit  20  and the compression system of the audio compression circuit  13  and the audio expansion circuit  17  are established (step ST 16 ). 
     The server judges the traffic density of the telephone circuit in a process as shown in FIG. 12, for example. 
     In FIG. 12, a delay measuring timer is reset (step ST 21 ). After an echo request is sent to the server of a destination terminal (step ST 22 ), the delay measuring timer restart its counting action (step ST 23 ) to measure the delay time required until an echo reply answering the echo request reaches (step ST 24 ), and then stops the measurement (step ST 25 ). 
     The delay time required after despatch of the echo request until arrival of the echo reply reflects the traffic density of the telephone circuit, namely, the ratio of use of the telephone circuit. A long delay time represents that the telephone circuit is crowded and the ratio of use of the telephone circuit is high. 
     The traffic density of the telephone circuit can be known also by using a control packet and measuring the delay time between servers in lieu of measuring the delay time from despatch of the echo request to arrival of the echo reply. It is also possible to use a Ping command between connected terminals to measure the delay time. 
     According to the invention, the sampling frequency of the audio compression circuit and the compression system of the audio compression circuit can be flexibly set to optimum values according to the traffic density of the computer network system. Therefore, a good quality of sound is ensured, and voice is not interrupted during telephone conversation even when the computer network system is crowded.