JP3918839B2 - Call accounting method and method, terminal device and program thereof - Google Patents

Description

  The present invention relates to a call billing method and method, a terminal device, and a program thereof, and more particularly, to a call billing method and method, a terminal device and a program thereof that charge a voice by reflecting voice quality in a call time.

  Patent Literature 1 and Patent Literature 2 describe a method of measuring elapsed time and charging according to voice quality during a conventional call.

  In Patent Literature 1, the voice quality monitor includes a monitor device that monitors voice on a telephone line during a call, a timer that measures an elapsed time since the start of the call, and a determination circuit. The determination circuit compares the voice quality output from the monitor device with a preset threshold, and when it is determined that the voice quality is below the threshold, the timer interrupts the measurement of the elapsed time between them and When it is determined that the quality exceeds the threshold, the timer restarts the time measurement. Then, the voice quality monitor notifies the charging server of the measured elapsed time.

  In Patent Document 2, a gateway device arranged at the boundary between a provider IP network and a user access network observes a packet loss rate per unit time for each RTP session, and holds or performs call processing in the gateway device according to a codec used. When the threshold value passed from the server is exceeded, the call processing server is notified. The call processing server takes a reaction so that the call (or user) using the RTP session is not charged or is forcibly disconnected, so that time charging does not occur in a state where the voice quality is deteriorated.

  Further, Patent Document 3 describes a method for measuring end / end quality.

  In Patent Document 3, a telephone set as a terminal that can be connected to a packet network and can communicate by voice transmission / reception is provided with measurement means for measuring voice quality so that end-to-end quality measurement is possible. The voice quality of the network between the terminals incorporating the means can be measured online. In the method of Patent Document 3, end-end quality measurement is performed by generating and transmitting data for quality measurement by a pattern generator when the telephone is not in a call.

JP 2002-27150 A JP 2003-249945 A JP 2003-244235 A

  However, the background art described above has the following problems.

  In Patent Document 1 and Patent Document 2, a method for interrupting the measurement of charging is realized when the sound quality output by a device for measuring sound quality, such as a gateway or a sound quality monitor, falls below a preset threshold value. However, in this method, since there are various degradation factors from the gateway, voice quality monitor, etc. until the sound is finally played through the speaker of the terminal, it does not represent the actual voice quality at the terminal, and appropriate billing processing Is not realized. In addition, a device for measuring voice quality such as a gateway and a voice quality monitor is required separately from the terminal.

  In Patent Document 3, end-to-end quality measurement is performed, but quality measurement using the measurement data cannot be performed during user call.

  SUMMARY OF THE INVENTION An object of the present invention is to solve the above-described problems and provide a call billing method and method, a terminal device, and a program therefor, in which voice quality during a user call is evaluated end to end and reflected in billing.

The call accounting system according to the first aspect of the present invention is based on a call time when a call control server, a calling terminal, and a called terminal are connected via a network and a call is made between the calling terminal and the called terminal. A call charging system for charging based on the call control server, wherein the call control server has means for connecting a call between the calling terminal and the called terminal, and the called terminal is connected to the calling terminal. Means for calculating a first voice quality evaluation value from the voice data received from the calling terminal and transmitting the first voice quality evaluation value to the calling terminal. The calling terminal receives a second voice quality evaluation from the voice data received from the called terminal. Means for calculating a value, and means for subtracting the prepaid frequency by calculating a call time per frequency of the prepaid frequency based on the first audio quality evaluation value and the second audio quality evaluation value. the calling terminal, the sound received from the called terminal Means for calculating a speech rate from a volume of data and a volume of audio data to be transmitted to the called terminal, and the first speech quality evaluation value and the second speech quality evaluation value are weighted by the speech rate. Means for calculating an aggregated voice quality evaluation value and determining a discount rate according to the aggregated voice quality evaluation value; and means for calculating a call time per frequency of the prepaid frequency according to the discount rate and subtracting the prepaid frequency including.

The call accounting method according to the second aspect of the present invention is based on a call time when a call control server, a calling terminal, and a called terminal are connected via a network and a call is made between the calling terminal and the called terminal. A call billing method for performing billing based on the call control server, wherein the call control server connects a call between the calling terminal and the called terminal, and the calling terminal and the called terminal receive voice data The calling terminal calculates a first voice quality evaluation value from the voice data received from the calling terminal and transmits the first voice quality evaluation value to the calling terminal, and the calling terminal The second voice quality evaluation value is calculated from the voice data received from the called terminal, and the call time per frequency of the prepaid frequency is calculated based on the first voice quality evaluation value and the second voice quality evaluation value. calculates, by subtracting the prepaid power, the calling terminal, the adhesive A speech rate is calculated from the volume of audio data received from the terminal and the volume of audio data transmitted to the called terminal, and the first audio quality evaluation value and the second audio quality evaluation value are calculated as the speech rate. To calculate the aggregate voice quality evaluation value by weighting, determine a discount rate according to the aggregate voice quality evaluation value, calculate the call time per frequency of the prepaid frequency according to the discount rate, and subtract the prepaid frequency It is characterized by that.

A terminal device according to a third aspect of the present invention is a terminal device that performs a billing process by making a call with a partner terminal device, and means for receiving a first voice quality evaluation value calculated by the partner terminal device; Means for calculating a second voice quality evaluation value from voice data received from the apparatus, and a call time per frequency of the prepaid frequency based on the first voice quality evaluation value and the second voice quality evaluation value Calculating a speech rate from a volume of voice data received from the counterpart terminal apparatus and a volume of voice data transmitted to the counterpart terminal apparatus; and Means for calculating an aggregate speech quality evaluation value by weighting the speech quality evaluation value and the second speech quality evaluation value by the speech rate and determining a discount rate according to the aggregate speech quality evaluation value; According to rate It calculates the call time per frequency Paid power and means for subtracting the prepaid counts to.

A program according to a fourth invention of the present application is a program for a terminal device that performs a call process with a call to a partner terminal device, and receives a first voice quality evaluation value calculated by the partner terminal device. , A function of calculating a second voice quality evaluation value from voice data received from the counterpart terminal device, and per prepaid frequency based on the first voice quality evaluation value and the second voice quality evaluation value A function of calculating a speech rate by subtracting a prepaid frequency, a function of calculating a speech rate from a volume of voice data received from the counterpart terminal apparatus and a volume of voice data transmitted to the counterpart terminal apparatus, A function of calculating an aggregate speech quality evaluation value by weighting the speech quality evaluation value and the second speech quality evaluation value by the speech rate, and determining a discount rate according to the aggregate speech quality evaluation value, the discount Function of subtracting the prepaid power is calculated call time per degree of prepaid power according, to realize.

  The effect of the present invention is that voice quality at the time of a user call can be evaluated end to end and reflected in billing.

  The reason is that the respective voice quality evaluation values are calculated at the calling terminal and the called terminal, and the call time per frequency of the prepaid frequency is calculated based on the respective voice quality evaluation values at the calling terminal to thereby calculate the prepaid frequency. This is because of subtracting.

  The present invention evaluates the voice quality of voice data received at the terminal that has received the voice data, collects the evaluation results at the calling terminal, and grasps the voice quality of the entire system. An appropriate discount rate for the voice quality of the entire system is reflected in the call time per frequency of the prepaid frequency. This makes it possible to realize a prepaid billing system that allows long-time calls as the voice quality deteriorates even with the same call charge, and increases user satisfaction.

Next, the configuration of the best mode for carrying out the present invention will be described in detail with reference to the drawings.
FIG. 1 is a diagram showing the configuration of this system.
FIG. 2 shows the components of the terminal.
FIG. 3 is a diagram showing a configuration example of the accounting table.
FIG. 4 is a diagram illustrating a configuration example of the discount rate table.
FIG. 5 is a diagram showing a configuration example of the voice quality evaluation value table.

  Referring to FIG. 1, an embodiment of the present system includes a terminal 101 to 102 capable of Internet telephone service connected on the Internet 100 or an intranet, and a call control server 103 that connects a telephone call between the terminals 101 to 102. And. In FIG. 1, two terminals 101 and 102 are illustrated as terminals, but the number of terminals is not limited. Further, the constituent elements shown in the terminal 101 and the terminal 102 shown in FIG. 1 are obtained by extracting only the part related to the voice quality evaluation, and the details are as shown in FIG. Since the Internet telephone service implementation method is publicly known, detailed description is omitted.

  The call control server 103 has means for connecting a telephone call between terminals in the Internet telephone service.

  A terminal 101 and a terminal 102 are devices capable of Internet telephone service that operate under program control, such as a personal computer, a personal digital assistant (PDA), or a mobile phone. Further, even if the terminal 101 and the terminal 102 are connected to the Internet wirelessly such as a public wireless LAN 104, they are connected to the Internet 100 via a wired connection such as an ADSL (Asymmetric Digital Subscriber Line) 105. It does not matter. In this embodiment, a terminal that makes a call is called a calling terminal, and a terminal that receives an incoming call is called a called terminal.

  Next, the configuration of the terminal will be described in detail with reference to FIG. The terminal 101 and the terminal 102 have the same configuration.

  Referring to FIG. 2, the terminal 101 and the terminal 102 are a call control unit 201, a microphone 202, an audio data transmission unit 203, an audio data reception unit 204, a speaker 205, an audio quality measurement unit 206, and an audio. A quality transmission unit 207, a voice quality reception unit 208, a voice quality aggregation unit 209, a billing table 210, a discount rate table 211, a call duration determination unit 212, a prepaid frequency table 213, and a prepaid frequency subtraction unit 214 A voice quality evaluation value table 215.

  The call control unit 201 communicates with the call control server 103 to connect / disconnect a call to / from a call partner terminal. Further, the call control unit 201 has a prepaid frequency subtraction timer that measures a call time per frequency of the prepaid frequency.

  The microphone 202 records the user's voice.

  The audio data transmission unit 203 transmits the audio data recorded by the microphone 202 to the terminal of the call partner.

  The voice data receiving unit 204 receives voice data from the terminal of the other party and sends it to the speaker 205. Also, the parameter value related to the voice quality is transferred to the voice quality measuring unit 206. Parameters include packet loss rates, delays, and jitters that occur on the network, delays due to jitter buffers generated at the terminal, jitter buffer overflow, deterioration due to voice codec conversion, and basic elements of the entire system. There are transmission volume, echo size and attenuation. Note that a method of collecting parameter values related to voice quality is a known method.

  The speaker 205 outputs a sound based on the sound data received by the sound data receiving unit 204, and sounds a ring tone when receiving an incoming call from the partner terminal.

  The voice quality measuring unit 206 measures (calculates) a voice quality evaluation value (individual voice quality evaluation value) from the voice data (parameter related to voice quality) received by the voice data receiving unit 204. The voice quality evaluation value of this embodiment is determined by ITU-T (International Telecommunication Union-Telecommunication Standardization sector), which is a standardization organization, as G.D. 107 and G.I. Description will be made using the R value recommended in 109. The R value is calculated from a large number of parameters relating to voice quality, and ranges from 0 to 100. The larger the value, the better the voice quality is defined. The method for calculating the voice quality evaluation value is a known method.

  The voice quality transmitting unit 207 periodically transmits the voice quality evaluation value measured by the voice quality measuring unit 206 to the terminal of the calling terminal (for example, every 15 seconds).

  The voice quality receiving unit 208 receives the voice quality evaluation value transmitted by the voice quality transmitting unit 207.

  The audio quality aggregating unit 209 stores the audio quality evaluation value received by the audio quality receiving unit 208 in the audio quality evaluation value table 215 and performs an aggregation operation from the plurality of audio quality evaluation values stored in the audio quality evaluation value table 215. An aggregate voice quality evaluation value is calculated. The voice quality aggregating unit 209 sets a minimum value as an aggregation calculation result (aggregated voice quality evaluation value) from the plurality of stored voice quality evaluation values. Further, the average value of the voice quality evaluation values may be used as the aggregation calculation result (aggregated voice quality evaluation value).

  The billing table 210 stores the basic call time per frequency of the prepaid frequency for each system of the Internet telephone address of the call partner. The Internet telephone of this embodiment will be described by taking SIP (Session Initiation Protocol) recommended by the standardization organization IETF (Internet Engineering Task Force) in RFC (Request for Comments) 3261 as an example. Internet telephone addresses instead of general telephone numbers are called SIP-URI (SIP-Uniform Resource Identifiers) in SIP. Referring to FIG. 3, the accounting table 210 includes items of an evaluation order 301, an address system 302, and a basic call time 303. The value of each item in the billing table 210 is registered in advance by an appropriate method.

  The evaluation order 301 stores the order in which the Internet telephone address of the actual call partner is compared with the address system 302.

  The address system 302 stores an Internet telephone address or an element obtained by grouping Internet telephone addresses for matching with the actual telephone telephone address of the other party. In this embodiment, the Internet telephone address grouping method is stored in the regular expression format of SIP-URI.

  The basic call time 303 stores the basic call time (call time per prepaid frequency) corresponding to the address system 302. When the Internet telephone address of the other party matches the address system 302, the basic call time corresponding to the address system is applied.

  The discount rate table 211 stores the discount rate of the billing fee according to the voice quality evaluation value calculated by the voice quality aggregating unit 209. Referring to FIG. 4, the discount rate table 211 includes items of a voice quality evaluation value range 401 and a discount rate 402. The value of each item in the discount rate table 211 is registered in advance by an appropriate method.

  The audio quality evaluation value range 401 stores a range of audio quality evaluation values (aggregated audio quality evaluation values). The voice quality evaluation value obtained by the voice quality aggregation unit 209 is compared with the voice quality evaluation value range 401.

  In the discount rate 402, the discount rate for the voice quality evaluation value that matches the voice quality evaluation value range 401 is stored as a percentage.

  The call time determination unit 212 determines the call time per frequency of the prepaid frequency using the audio quality evaluation value aggregated by the audio quality aggregation unit 209, the accounting table 210, and the discount rate table 211. That is, the call time per frequency is determined from the basic call time 303 and the discount rate 402. In this embodiment, the calculation is performed so that the call time per frequency of the prepaid frequency is doubled when the discount rate is 50%. Calculation is performed so that the call time is 5 times when the discount rate is 80% and the call time is 1.25 times when the discount rate is 20%.

  The prepaid frequency table 213 stores the prepaid frequency. The method for acquiring the initial value of the prepaid frequency and the method for storing the acquired prepaid frequency in the prepaid frequency table 213 are based on known methods. For example, the prepaid frequency (initial value) is acquired via the web from a center or the like that manages the prepaid frequency. Further, the prepaid frequency (initial value) may be acquired by infrared communication or the like. Moreover, you may make it acquire a prepaid frequency (initial value) by setting a prepaid card in a terminal.

  The prepaid frequency subtraction unit 214 subtracts the prepaid frequency from the prepaid frequency table 213 according to the call time determined by the call time determination unit 212.

  The voice quality evaluation value table 215 stores a voice quality evaluation value for each voice data transmission / reception direction. Referring to FIG. 5, the voice quality evaluation value table 215 includes items of a voice data transmission direction 501 and a voice quality evaluation value 502.

  The voice data transmission direction 501 stores the Internet telephone addresses of the voice data transmission terminal 503 and the reception terminal 504 for identifying the voice data transmission direction.

  The voice quality evaluation value 502 stores a voice quality evaluation value for each voice direction. The voice quality evaluation value of the present embodiment is determined by G.I. 107 and G.I. Description will be made using the R value recommended in 109. The R value is calculated from a large number of parameters relating to voice quality, and ranges from 0 to 100. The larger the value, the better the voice quality is defined.

Next, the operation of the best mode for carrying out the present invention will be described in detail with reference to FIGS.
FIG. 6 shows an operation sequence.
FIG. 7 shows a flowchart of the operation.

  The SIP transmits a delivery confirmation protocol in order to confirm that information is correctly transmitted to the transmission partner in the call control protocol, but is omitted in this embodiment for the sake of easy explanation. Further, it is assumed that the billing table 210 is registered in advance as shown in FIG. 3, and the discount rate table 211 is registered in advance as shown in FIG.

  The sequence in FIG. 6 is assumed to be the calling terminal 101, the called terminal 102, and the call control server 103 from the left. Further, the SIP-URI of the terminal 101 is “sip: taro@aaa.corp.com”, and the SIP-URI of the terminal 102 is “sip: hanako@bbb.corp.com”.

  Referring to FIG. 6, the user of the terminal 101 performs an operation of making a call to the terminal 102 (S601).

  The call control unit 201 of the terminal 101 checks the prepaid frequency table 213 to confirm that the prepaid frequency is not “0” (S602).

  If the prepaid frequency is not “0”, the call control unit 201 transmits a call signal to the call control server 103 to call the terminal 102 (S603).

  When the prepaid frequency is “0”, the call control unit 201 cancels the user call operation (S604).

  When receiving the call signal from the terminal 101, the call control server 103 transfers the call signal to the terminal 102 (S605).

  The terminal 102 that has received the call signal sounds a ringing tone (RINGING) from the speaker 205 to notify the user of the incoming call (S606).

  When the user of the terminal 102 responds to the call, the terminal 102 stops the ringing tone (RINGING) (S607) and transmits a call response to the call control server 103 (S608).

  Receiving the call response, the call control server 103 transfers the call response to the terminal 101 that is the caller (S609).

  When the terminal 101 receives the call response, each of the terminal 101 and the terminal 102 enters a call state (S610).

  The terminal 101 and the terminal 102 that are in a call state transmit the voice data recorded from the microphone 202 to the other party of call from the voice data transmission unit 203.

  The transmitted voice data is received by the voice data receiving unit 204 of the other party of the call, and is played from the speaker 205, and parameters related to the voice quality are transferred to the voice quality measuring unit 206. Parameters include packet loss rates, delays, and jitters that occur on the network, delays due to jitter buffers generated at the terminal, jitter buffer overflow, deterioration due to voice codec conversion, and basic elements of the entire system. There are transmission volume, echo size and attenuation.

  The voice quality measurement unit 206 obtains a voice quality evaluation value based on the transferred parameters and transfers the voice quality evaluation value to the voice quality transmission unit 207.

  The voice quality transmitting unit 207 periodically transmits a voice quality evaluation value to the terminal 101 that is a calling terminal (S611). The terminal 101 which is a calling terminal also transmits the voice quality evaluation value to itself, but since it is not necessary to increase the network load by transmission to itself, the voice quality evaluation value is transferred by local loopback.

  From here, processing specialized for the terminal 101 which is the calling terminal will be described.

  The voice quality reception unit 208 that has received the voice quality evaluation value from the voice quality transmission unit 207 of the terminal 101 and the terminal 102 transfers the voice quality evaluation value to the voice quality aggregation unit 209.

  The voice quality aggregation unit 209 stores the voice quality evaluation value received from the voice quality reception unit 208 in the voice quality evaluation value table 215 for each voice data transmission direction 501. Here, the voice quality evaluation value for the voice data from the terminal 101 to the terminal 102, that is, the voice quality evaluation value transmitted from the voice quality transmitting unit 207 of the terminal 102 and received by the voice quality receiving unit 208 of the terminal 101 is “72. ”. Further, it is assumed that the voice quality evaluation value for the voice data from the terminal 102 to the terminal 101, that is, the voice quality evaluation value measured by the voice quality measuring unit 206 of the terminal 101 is “85”. As shown in FIG. 5, the voice quality evaluation value table 215 is stored.

  The voice quality aggregating unit 209 performs an aggregation operation on the plurality of voice quality evaluation values stored in the voice quality evaluation value table 215 and transfers the result to the call time determination unit 212. In the example shown in FIG. 5, the minimum value “72” is transferred to the call duration determination unit 212 as the aggregation calculation result.

  In addition, the call control unit 201 activates a prepaid frequency subtraction timer for periodically subtracting the prepaid frequency during a call state. The first time is started with a timer interval of 0 seconds (S612).

  As a result, the prepaid frequency subtraction timer times out immediately (S613). Then, a process for subtracting the prepaid frequency is performed (S614).

  The process at this time (the process of S614) will be described with reference to the flowchart of FIG. The flowchart of FIG. 7 includes processing of the call control unit 201 (S701 and S707 to S709), processing of the call duration determination unit 212 (S702 to S704), and processing of the prepaid frequency subtraction unit 214 (S705 to S706). It is out.

  Referring to FIG. 7, when the prepaid frequency subtraction timer times out, the call control unit 201 transfers “sip: hanako@bbb.corp.com”, which is the SIP-URI of the other party, to the call time determination unit 212 (S701). ).

  Upon receiving the call partner's SIP-URI, the call duration determination unit 212 performs a matching process using a regular expression between the call partner's SIP-URI and the address system 302 of the charge table 210 in accordance with the evaluation order 301 of the charge table 210. To do. If it matches the regular expression, the corresponding basic call time is taken out (S702). Here, since the SIP-URI of the terminal 102 of the calling party is “sip: hanako@bbb.corp.com”, the basic calling time of 30 seconds can be obtained by matching the second evaluation order.

  If there is no match, matching processing is performed again with the address system 302 in the next evaluation order 301. If there is no match up to the end, a call disconnect request is made from the call time determination unit 212 to the call control unit 201 (“failure” in S703).

  Then, the call duration determination unit 212 refers to the discount rate table in FIG. 4 in order to obtain the discount rate from the voice quality evaluation value transferred from the voice quality aggregation unit 209. Here, if the voice quality evaluation value transferred from the voice quality aggregating unit 209 is “72”, the discount rate is 20%. Therefore, since a 20% discount is given for 30 seconds of the basic call time, the call time per frequency of the actual prepaid frequency is 30 (seconds) × 100 ÷ (100−20%) = 37.5. Calculated as seconds≈37 seconds (S704). If the voice quality evaluation value has not been transferred from the voice quality aggregation unit 209 immediately after the start of the call, the discount rate is calculated as 0% and 30 seconds is used as it is.

  Then, the prepaid frequency subtraction unit 214 is notified of the call time determined by the call time determination unit 212. The prepaid frequency subtraction unit 214 determines whether the prepaid frequency stored in the prepaid frequency table 213 is “0” (S705).

  If the prepaid frequency is not “0”, the prepaid frequency stored in the prepaid frequency table 213 is subtracted by 1 frequency, and then the call control unit 201 is notified of 37 seconds, which is a new call time (S706).

  The call control unit 201 notified of the call time updates the prepaid frequency subtraction timer with the new call time and continues the call (S707).

  When the prepaid frequency is “0”, the call control unit 201 is notified of a disconnection request (frequency “0” in S705).

  The call control unit 201 notified of the disconnection request stops the prepaid frequency subtraction timer (S708) and disconnects the call (S709).

As described above, the voice quality evaluation values measured at each call terminal are collected at the calling terminal to be charged, and a plurality of voice quality evaluation values are aggregated to obtain the voice quality of the entire system. There is an effect that an appropriate billing process can be performed according to the quality.
realizable.

  In addition, since the voice quality of the entire system is ascertained by aggregating multiple voice quality evaluation values at the calling terminal, there is no need for a new hardware device that measures the packet loss rate, only the terminal software. The system can be constructed at a low cost because it can be realized by the above.

  Next, a second embodiment of the present invention will be described.

First, the structure of Example 2 of this invention is demonstrated using drawing.
FIG. 8 is a diagram illustrating the components of the terminal according to the second embodiment of the present invention.
FIG. 9 is a diagram showing a configuration example of the voice quality evaluation value table.

  Although the system configuration of the second embodiment of the present invention is not shown, the system configuration of the first embodiment shown in FIG. 1 is obtained by replacing the terminal 101 and the terminal 102 with a terminal 800a and a terminal 800b. . In this embodiment, it is assumed that the terminal 800a is a terminal that makes a call (calling terminal), and the terminal 800b is a terminal that receives an incoming call (calling terminal). Further, the SIP-URI of the terminal 800a is “sip: taro@aaa.corp.com”, and the SIP-URI of the terminal 800b is “sip: hanako@bbb.corp.com”.

  Referring to FIG. 8, a terminal 800 (terminals 800a to 800b) according to the second embodiment of the present invention adds a speech time measuring unit 801 to the terminal configuration according to the first embodiment illustrated in FIG. 203 and the voice data reception unit 204 are replaced with a voice data transmission unit 803 and a voice data reception unit 804, the voice quality aggregation unit 209 is replaced with a voice quality aggregation unit 809, and the voice quality evaluation value table 215 is replaced with a voice quality evaluation value table. It is replaced with 815.

  The speaking time measuring unit 801 calculates the speaking time of each terminal from the volume transferred from the voice data receiving unit 204 and the voice data transmitting unit 203 and the sound volume threshold to be held, and the speaking rate of each terminal is calculated based on the speaking time. Calculated and stored in the voice quality evaluation value table 815. It is assumed that the sound production volume threshold is stored in advance.

  In addition to the function of the voice data transmission unit 203 shown in the first embodiment, the voice data transmission unit 803 has a function of measuring the volume of voice data to be transmitted at regular intervals, quantifying it, and transferring it to the speech time measurement unit 801. . The method for measuring and quantifying the volume of audio data at regular intervals is a known method.

  In addition to the function of the audio data receiving unit 204 shown in the first embodiment, the audio data receiving unit 804 has a function of measuring the volume of received audio data at regular intervals, quantifying it, and transferring it to the speech time measuring unit 801. . The method for measuring and quantifying the volume of audio data at regular intervals is a known method.

  The voice quality aggregating unit 809 transfers a value obtained by weighting each voice quality evaluation value stored in the voice quality evaluation value table 815 with each speech rate to the call duration determination unit 212 as a result of the aggregation calculation.

  The voice quality evaluation value table 815 includes a voice transmission direction 901, a voice quality evaluation value 902, and a speech rate 903, as shown in FIG.

  The voice transmission direction 901 stores the Internet telephone addresses of the voice data transmission terminal 904 and the reception terminal 905 for identifying the voice data transmission direction.

  The audio quality evaluation value 902 stores an audio quality evaluation value for each audio transmission direction.

  The speech rate 903 stores the speech rate for each audio data transmission direction.

  Next, the operation of the second embodiment of the present invention will be described.

  During the call state, the terminal transfers the volume of the voice data received by the voice data reception unit 204 and the voice data transmission unit 203 to the speech time measurement unit 801.

  The speech time measurement unit 801 stores a speech volume threshold value in advance, and determines that the user is speaking when the volume exceeds the speech volume threshold. Then, the measured speech rate for each voice transmission direction is stored in the voice quality evaluation value table 815. At this time, the speech rate 903 of the voice quality evaluation value table 815 is the speech rate of each terminal during the total speech time, and is measured except for the silent time when the volume of the calling terminal and the called terminal does not reach the speech volume threshold. To do. For example, of the volume of the terminal 800a transferred from the voice data transmission unit 203, the time that has exceeded the voice volume threshold is 40 seconds, and the message of the volume of the terminal 800b transferred from the voice data transmission unit 203 It is assumed that the time when the volume threshold is exceeded is 360 seconds. The speech rate of the terminal 800 a is calculated as 40 seconds ÷ (40 seconds + 360 seconds) × 100 = 10% and stored in the voice quality evaluation value table 815. The speech rate of the terminal 800 b is calculated as 360% ÷ (40 seconds + 360 seconds) × 100 = 90% and stored in the voice quality evaluation value table 815.

  Then, the voice quality aggregating unit 209 transfers a value obtained by weighting each voice quality evaluation value stored in the voice quality evaluation value table 815 with each speech rate to the call time determination unit 212 as a result of the aggregation calculation. Here, the voice quality evaluation value (aggregated voice quality evaluation value) which is the result of the aggregation calculation is 72 × 10% × 0.01 + 85 × 90% × 0.01 = 83.7.

  Since the aggregate voice quality evaluation value is 83.7, the call time determination unit 212 refers to the discount rate table 211 and sets the discount rate to 0%.

  As a result, even when the voice quality in one direction is poor, it is possible to suppress the discount rate by determining that the degree of influence on the entire call is small when the ratio of the speech of the user having the poor voice quality is small.

  Next, Embodiment 3 of the present invention will be described.

First, the structure of Example 3 of this invention is demonstrated using drawing.
FIG. 10 is a diagram showing a system configuration of Embodiment 3 of the present invention.
FIG. 11 is a diagram illustrating components of a terminal according to the third embodiment of the present invention.

  Referring to FIG. 10, in the system configuration in the third embodiment of the present invention, a service directory server 1001 and a terminal setting server 1002 are added to the system configuration in the first embodiment shown in FIG. Are replaced with a terminal 1003 and a terminal 1004.

  The service directory server 1001 provides the terminal with the types of services that can be provided on the connected network. The service directory server 1001 may be SLP (Service Location Protocol) recommended by the standardization organization IETF.

  The terminal setting server 1002 holds terminal setting information and transmits setting information to the terminal in response to a download request from the terminal. The setting information is, for example, information on a billing table or a discount rate table.

  Referring to FIG. 11, a terminal 1003 and a terminal 1004 according to the third embodiment of the present invention add a service detection unit 1101 and a table download unit 1102 to the terminal configuration according to the first embodiment shown in FIG. 210 and the discount rate table 211 are replaced with a billing table 1110 and a discount rate table 1111.

  The service detection unit 1101 searches the service directory server 1001 for services available on the connected network.

  The table download unit 1102 downloads a table in the terminal from the terminal setting server 1002.

  The item configurations of the charging table 1110 and the discount rate table 1111 are the same as the item configurations of the charging table 210 and the discount rate table 211.

  Subsequently, the operation of the third embodiment of the present invention will be described.

  The terminal 1003 and the terminal 1004 are connected to a network (public wireless LAN 104 or ADSL 105). Then, the service detection unit 1101 of the terminal 1003 and the terminal 1004 inquires about a service that can be provided to the service directory server 1001 installed on the network.

  When determining that the Internet telephone service can be provided, the service detection unit 1101 requests the table download unit 1102 to update the table information.

  The table download unit 1102 requested to update the table information requests the terminal setting server 1002 for new table information together with the network attribute information. The network attribute information is information that can identify a connected network. For example, a DNS (domain name server) name is used.

  Then, the table download unit 1102 updates the charging table 1110 and the discount rate table 1111 with the table information downloaded from the terminal setting server 1002.

  Note that the prepaid frequency table 213 is not subject to updating because the stored prepaid frequency corresponds to money. Also, the voice quality evaluation value table 215 is not subject to update.

  With these operations, it is possible to provide the terminal with a call service having different conditions for each connected network.

  In the above-described embodiment according to the present invention, a program or the like for executing a processing operation of the terminal device is recorded as data on a recording medium (not shown) such as a magnetic disk or a ROM of the terminal device. The data is read out and used to operate the terminal device. As described above, data for operating the terminal device according to the present invention is recorded on the recording medium, and the function of the terminal device can be realized by installing the recording medium.

Diagram showing the configuration of this system Diagram showing the components of the terminal A diagram showing a configuration example of a billing table A diagram showing a configuration example of a discount rate table The figure which shows the structural example of an audio | voice quality evaluation value table Diagram showing sequence of operation The figure which shows the flowchart of operation The figure which shows the component of the terminal in Example 2 of this invention. The figure which shows the structural example of an audio | voice quality evaluation value table The figure which shows the system configuration | structure of Example 3 of this invention. The figure which shows the component of the terminal in Example 3 of this invention.

Explanation of symbols

100 Internet 101 Terminal 102 Terminal 103 Call Control Server 104 Public Wireless LAN
105 ADSL
DESCRIPTION OF SYMBOLS 201 Call control part 202 Microphone 203 Voice data transmission part 204 Voice data reception part 205 Speaker 206 Voice quality measurement part 207 Voice quality transmission part 208 Voice quality reception part 209 Voice quality aggregation part 210 Charge table 211 Discount rate table 212 Call time determination part 213 Prepaid frequency table 214 Prepaid frequency subtraction unit 215 Voice quality evaluation value table 301 Evaluation order 302 Address system 303 Basic call time 401 Voice quality evaluation value range 402 Discount rate 501 Voice data transmission direction 502 Voice quality evaluation value 503 Transmission terminal 504 Reception terminal 800 terminal 801 speech time measurement unit 803 audio data transmission unit 804 audio data reception unit 809 audio quality aggregation unit 815 audio quality evaluation value table 901 audio transmission direction 902 audio quality evaluation value 903 Speech rate 904 Transmission terminal 905 Reception terminal 1001 Service directory server 1002 Terminal setting server 1003 Terminal 1004 Terminal 1101 Service detection unit 1102 Table download unit 1110 Billing table 1111 Discount rate table

Claims (4)

A call charging system for connecting a call control server, a calling terminal, and a called terminal via a network and charging based on a call time during which a call is made between the calling terminal and the called terminal,
The call control server
Means for connecting a telephone call between the calling terminal and the called terminal;
The called terminal is
Means for calculating a first voice quality evaluation value from voice data received from the calling terminal and transmitting the first voice quality evaluation value to the calling terminal;
The calling terminal is
Means for calculating a second voice quality evaluation value from voice data received from the called terminal;
Means for calculating a call duration per frequency of the prepaid frequency based on the first audio quality evaluation value and the second audio quality evaluation value, and subtracting the prepaid frequency;
The calling terminal is
Means for calculating a speech rate from a volume of audio data received from the called terminal and a volume of audio data transmitted to the called terminal;
Means for calculating an aggregate speech quality evaluation value by weighting the first speech quality evaluation value and the second speech quality evaluation value by the speech rate, and determining a discount rate according to the aggregate speech quality evaluation value; ,
Means for calculating a call duration per frequency of the prepaid frequency according to the discount rate and subtracting the prepaid frequency.
Through story billing method, characterized in that. A call billing method for performing billing based on a call time in which a call control server, a calling terminal, and a called terminal are connected via a network and a call is made between the calling terminal and the called terminal,
The call control server
Connecting a call between the calling terminal and the called terminal;
The calling terminal and the called terminal are:
Send and receive audio data to make calls,
The called terminal is
Calculating a first voice quality evaluation value from the voice data received from the calling terminal and transmitting to the calling terminal;
The calling terminal is
Calculating a second voice quality evaluation value from the voice data received from the called terminal;
Based on the first voice quality evaluation value and the second voice quality evaluation value, a call time per frequency of the prepaid frequency is calculated,
Subtract the prepaid frequency,
The calling terminal is
A speech rate is calculated from the volume of audio data received from the called terminal and the volume of audio data transmitted to the called terminal,
An aggregate speech quality evaluation value is calculated by weighting the first speech quality evaluation value and the second speech quality evaluation value by the speech rate;
Determining a discount rate according to the aggregate voice quality evaluation value;
According to the discount rate, the call time per frequency of the prepaid frequency is calculated,
Subtract prepaid frequency,
Through story billing how to, characterized in that. A terminal device that makes a call with a partner terminal device and performs billing processing,
Means for receiving a first voice quality evaluation value calculated by the counterpart terminal device;
Means for calculating a second voice quality evaluation value from voice data received from the counterpart terminal device;
Means for calculating a call duration per frequency of the prepaid frequency based on the first audio quality evaluation value and the second audio quality evaluation value, and subtracting the prepaid frequency;
Means for calculating a speech rate from the volume of voice data received from the counterpart terminal apparatus and the volume of voice data transmitted to the counterpart terminal apparatus;
Means for calculating an aggregate speech quality evaluation value by weighting the first speech quality evaluation value and the second speech quality evaluation value by the speech rate, and determining a discount rate according to the aggregate speech quality evaluation value; ,
Means for calculating a call duration per frequency of the prepaid frequency according to the discount rate and subtracting the prepaid frequency.
End terminal device you wherein a. A program for a terminal device that makes a call with a partner terminal device and performs billing processing,
On the terminal device,
A function of receiving the first voice quality evaluation value calculated by the counterpart terminal device;
A function of calculating a second voice quality evaluation value from voice data received from the counterpart terminal device;
A function of calculating a call duration per frequency of the prepaid frequency based on the first audio quality evaluation value and the second audio quality evaluation value and subtracting the prepaid frequency;
A function of calculating a speech rate from the volume of audio data received from the counterpart terminal apparatus and the volume of audio data transmitted to the counterpart terminal apparatus;
A function of calculating an aggregate speech quality evaluation value by weighting the first speech quality evaluation value and the second speech quality evaluation value by the speech rate and determining a discount rate according to the aggregate speech quality evaluation value;
A function of calculating a call time per prepaid frequency according to the discount rate and subtracting the prepaid frequency;
Program for realizing.

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