JP2008211670A - Internet telephone system, call control server, and communication terminal - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To enable a user to grasp the situation of communication easily by informing the user of transmission quality of a communication line by an Internet telephone in real time. <P>SOLUTION: The Internet telephone system 100 has a communication terminal 130 and a call control server 110. The call control server 110 includes a communication establishing part 154 for establishing a communication line from a communication terminal 130 to another communication terminal 132 complying with a request from the communication terminal 130, a quality deriving part 156 for deriving a transmission quality value of the communication line, and a quality transmitting part 160 for transmitting the transmission quality value to the communication terminal 130. The communication terminal 130 has a communication requesting part 152 for requesting voice communication to another communication terminal 132 from the call control server 110, a quality receiving part 162 for receiving the transmission quality value from the quality transmitting part 160, and an indicator 164 for indicating the received transmission quality value in real time. <P>COPYRIGHT: (C)2008,JPO&INPIT

Description

  The present invention relates to an Internet telephone system, a call control server, and a communication terminal that perform voice communication between communication terminals via the Internet.

  The Internet telephone is a service that performs two-way voice communication with a partner communication terminal using the Internet protocol (IP). Such voice communication data may include not only voice data but also videophone image data. In the Internet telephone, an IP address is assigned to each communication terminal, and a partner communication terminal is specified by an IP telephone number associated with the IP address. In Internet telephones, voice is transmitted in compression-encoded packets, so even long-distance or long-distance calls can be enjoyed at lower costs than ordinary landline telephones. .

  However, while there is an advantage that it is possible to make a call at such a low cost, on the Internet telephone, the sound quality may be deteriorated according to the transmission quality, or the call may be interrupted. This is affected by packet loss due to traffic congestion, packet transmission delay, and arrival time interval (jitter) between packets, and there is no method for completely solving such a problem.

Therefore, a call quality guarantee method (for example, Patent Document 1) that guarantees a transmission quality level of an Internet telephone for a specific user registered in advance is disclosed.
JP 2001-308913 A

  However, in this technique, instead of assigning a port having a high transmission quality to a specific registered user, the transmission quality of a non-registered user is significantly reduced. In addition, a port with high transmission quality is assigned to a specific user, but this technology does not guarantee the subsequent transmission quality, and the high transmission quality at that time also depends on the usage status of the communication line of the transmission path. Depending on the situation, it may change from moment to moment, making it difficult to talk on the phone.

  In addition, the Internet communication capacity is increasing year by year, but such Internet users and usage are also increasing, packet loss due to traffic congestion, packet transmission delay, transmission due to the arrival time interval between packets. Quality degradation has not been resolved yet.

  Although prevention of such deterioration of transmission quality is a problem to be solved, the user wants to grasp at least that transmission quality is deteriorated or is likely to deteriorate. This is to minimize the damage caused by sudden inability to communicate by grasping the deterioration of the transmission quality and terminating the call early.

  In addition, it is difficult to predict the time when traffic is concentrated on the Internet, and if the Internet phone is used carelessly, the transmission quality may deteriorate immediately and the call may become impossible. However, with the conventional technology, it has been impossible to grasp the timing when it is better to avoid telephone calls by Internet telephone.

  The present invention has been made in view of the above-mentioned problems of conventional Internet telephones, and an object of the present invention is to inform the user of the transmission quality of the communication line by the Internet telephone in real time, thereby facilitating the communication situation. The present invention provides a new and improved Internet telephone system, call control server, and communication terminal that can be grasped by the user.

  In order to solve the above problems, according to an aspect of the present invention, a communication terminal, a call control server connected to the communication terminal via the Internet and controlling voice communication between the communication terminal and another communication terminal, In the Internet telephone system comprising: the call control server, in response to a request from the communication terminal, derives a communication establishment unit that establishes a communication line from the communication terminal to another communication terminal, and a transmission quality value of the communication line A quality derivation unit, and a quality transmission unit that transmits a transmission quality value to the communication terminal. The communication terminal includes: a communication request unit that requests voice communication to another communication terminal from the call control server; and the quality transmission unit. There is provided an Internet telephone system comprising: a quality receiving unit that receives the transmission quality value of the first and a display unit that displays the received transmission quality value in real time.

  In such an Internet telephone system, the call control server can derive the transmission quality value, and the transmission quality value derived at the communication terminal can be referred to by text display or graphic display. Therefore, the user can easily and in real time understand the transmission quality of the communication line to the other party by Internet telephone, and can determine whether to continue or stop the call according to the current transmission quality. When the user determines that the call should be stopped, the user can be notified of that, or the requirement can be quickly communicated to avoid making the other party unpleasant due to sudden inability to call.

  The call control server further includes a quality estimation unit that estimates a transmission quality value after a predetermined time based on a current value of the transmission quality value and a past transition, and the quality transmission unit also communicates the transmission quality value after the predetermined time. It may be transmitted to the terminal.

  With this configuration, the user can also grasp the transmission quality after a predetermined time and predict the situation where communication that will come in the future will be difficult, so it is more appropriate to decide whether to continue or cancel the current call. It becomes possible to do.

  The quality deriving unit derives the transmission quality value of the communication line between the communication terminal and the call control server when voice communication from the communication terminal is not executed, and the quality transmitting unit communicates the derived transmission quality value You may transmit to a terminal.

  In the present invention, the transmission quality value from the own communication terminal to the call control server is derived in real time even in a state where no telephone call is made. Therefore, the user can grasp the transmission quality value before starting the call, and can select the timing for starting the call proactively, such as starting communication after the transmission quality is high.

  The transmission quality value is ITU-T recommendation G.264. An R value that is an evaluation value of the total transmission quality between communication terminals defined in 107 may be used.

  By using the R value as an evaluation criterion for transmission quality, it is possible to easily grasp the transmission quality from standard numerical values without separately accepting a troublesome new index.

  The display unit may display a warning if the R value is 50 or less. The R value is also provided with a class. If the R value is greater than 80, it is the same as a fixed telephone, and if it is greater than 70, it is the same as a mobile phone. A value larger than 50 is a requirement for obtaining the “050” number in the Internet telephone. Therefore, in the present invention, the threshold value at which the transmission quality is determined to be degraded is set to 50, and a warning is displayed instead of or in addition to the numerical value (R value). With this configuration, the user can easily notice the change even during a call, and can be used as a trigger for determining whether to continue or stop the call.

  In order to solve the above-described problem, according to another aspect of the present invention, there is provided a call control server that is connected to a communication terminal via the Internet and controls voice communication between the communication terminal and another communication terminal. The control server, in response to a request from the communication terminal, establishes a communication line from the communication terminal to another communication terminal, a quality derivation unit that derives a transmission quality value of the communication line, and displays the communication terminal And a quality transmission unit for transmitting the transmission quality value to the communication terminal in order to display the transmission quality value on the unit in real time, a call control server is provided.

  In order to solve the above problems, according to still another aspect of the present invention, there is provided a communication terminal that performs voice communication with another communication terminal via the Internet by a call control server connected via the Internet. Requesting unit for requesting voice communication to the communication terminal from the call control server, a quality receiving unit for receiving the transmission quality value of the communication line from the communication terminal to the other communication terminal, and the received transmission There is provided a communication terminal comprising a display unit that displays a quality value in real time.

  The components corresponding to the technical idea of the Internet telephone system described above and the description thereof can be applied to a call control server and a communication terminal.

  As described above, in the Internet telephone system of the present invention, the transmission quality of the communication line by the Internet telephone can be notified to the user in real time, and the communication status can be easily grasped.

  Hereinafter, preferred embodiments of the present invention will be described in detail with reference to the drawings. In addition, in this specification and drawing, about the component which has the substantially same function structure, duplication description is abbreviate | omitted by attaching | subjecting the same code | symbol.

  In the Internet, transmission quality may deteriorate due to packet loss due to traffic congestion, packet transmission delay, transmission quality deterioration due to arrival time interval between packets, and the like. Further, such transmission quality changes every moment according to the use situation of the communication line of the transmission path, and it is difficult to grasp the transmission quality by regular calculation or objective judgment.

  In order to solve such a problem, fundamentally, it is only necessary to prevent the transmission quality from being deteriorated. However, at present, it is best to suppress the significant deterioration in the transmission quality. Therefore, the user wants to grasp at least that the transmission quality has deteriorated or is likely to deteriorate. This is to minimize the damage caused by sudden inability to communicate by grasping the deterioration of transmission quality and rounding up the call early.

(Internet telephone system)
In the present embodiment, the purpose is to easily grasp the current communication status by deriving the transmission quality of the communication line by Internet telephone and notifying the user of the result in real time. Hereinafter, the configuration of an Internet telephone system that can be used as an Internet telephone will be described, and then the detailed configuration and operation of each component of the Internet telephone system will be described.

  FIG. 1 is a block diagram showing a schematic configuration of the Internet telephone system 100. The Internet telephone system 100 includes a call control server 110, a communication network 120 such as the Internet or a dedicated line, a communication terminal 130, and another communication terminal 132.

  The above-mentioned call control (SIP: Session Initiation Protocol) server 110 is also called a gatekeeper, and works in conjunction with a VoIP (Voice over IP) adapter or a DNS (Domain Name System) server in accordance with an IP telephone number issued by the communication terminal 130. Control connection or disconnection between communication terminals. The call control server 110 also executes management of personal information such as a user identifier in the Internet telephone system 100 and conversion of an IP telephone number and an IP address. Further, in the call control server 110, as an evaluation value of transmission quality described later, ITU-T recommendation G. An R value that is an evaluation value of the total transmission quality between communication terminals defined in 107 is derived. Here, SIP refers to an IETF (Internet Engineering Task Force) standard communication protocol for establishing a session between two or more communication terminals.

  The communication terminal 130 connects to the Internet directly or through a general telephone line using the Internet protocol (IP), and performs voice communication (call) with another communication terminal 132 via the call control server 110.

  In the Internet telephone system 100 described above, when a user attempts to call another communication terminal 132 that is a partner using the communication terminal 130, the user operates his / her communication terminal 130 via the communication network 120. Then, the IP telephone number of the other communication terminal 132 is transmitted to the call control server 110. The call control server 110 authenticates the communication terminal 130, derives an IP address associated with the IP telephone number of another communication terminal 132, and transmits the IP address to the communication terminal 130.

  The communication terminal 130 directly accesses another communication terminal 132 based on the IP address received from the call control server 110. In another communication terminal 132, in order to start voice communication with the communication terminal 130, authentication with the call control server 110 is performed, and when a connection is permitted, a communication line between the communication terminal 130 and the other communication terminal 132 is established. After that, voice communication is performed only between the communication terminal 130 and the other communication terminal 132 without going through the call control server 110. In such a hybrid network topology configuration, each communication terminal can be centrally managed by the call control server 110 and the call itself is performed independently by each communication terminal, so that the burden on the call control server 110 can be reduced. .

  FIG. 2 is a block diagram showing another configuration of the Internet telephone system 100. The Internet telephone system 100 includes a call control server 110, a communication network 120 such as the Internet or a dedicated line, a base station 140, a communication terminal 130, and another communication terminal 132. Since the call control server 110, the communication network 120, the communication terminal 130, and the other communication terminals 132 have already been described with reference to FIG. 1, the configuration of the base station 140 is confirmed here.

  The base station 140 is connected to the communication terminal 130 by wireless communication, and can execute high-speed data communication with the communication terminal 130. As such wireless communication, there is a time division multiplexing method in which a plurality of time slots obtained by time division of frames in the base station 140 are assigned to channels of the communication terminal 130, respectively. In addition, adaptive modulation that can adapt an optimum arbitrary modulation scheme from a plurality of coded modulation schemes can be used in accordance with the radio wave condition with the communication terminal 130.

  By using such adaptive modulation, it is possible to apply a modulation scheme capable of high-speed data rate transfer when the radio wave quality is high, and a low-speed modulation scheme that is difficult to disconnect the communication line when the radio wave quality is low.

  FIG. 3 is a functional block diagram showing the function of each component in the Internet telephone system 100. Here, in order to facilitate understanding, the transmission quality evaluation in the present embodiment has been described, but it goes without saying that other components related to voice communication are included.

  The call control server 110 constituting the Internet telephone system 100 includes a communication establishing unit 154, a quality deriving unit 156, a quality estimating unit 158, and a quality transmitting unit 160, and the communication terminal 130 includes an input unit. 150, a communication request unit 152, a quality receiving unit 162, and a display unit 164.

  The input unit 150 of the communication terminal 130 includes switches such as a keyboard, a cross key, and a joystick, and accepts user operation input such as number transmission.

  The communication request unit 152 of the communication terminal 130 transmits the IP telephone number of the other communication terminal 132 input by the user through the input unit 150 to the call control server 110 and requests voice communication with the other communication terminal 132. .

  The communication establishment unit 154 of the call control server 110 establishes a communication line from the communication terminal 130 to another communication terminal 132 in response to a request from the communication terminal 130.

  The quality deriving unit 156 of the call control server 110 derives a transmission quality value in the communication line after the communication between the communication terminal 130 and the other communication terminal 132 is started.

  The transmission quality value derived by the quality deriving unit 156 is displayed on the display unit 164 of the communication terminal 130 described later. Therefore, the user can easily and in real time understand the transmission quality of the communication line to the other party by Internet telephone, and can determine whether to continue or stop the call according to the current transmission quality. When the user determines that the call should be stopped, the user can be notified of that, or the requirement can be quickly communicated to avoid making the other party unpleasant due to sudden inability to call.

  The quality deriving unit 156 derives the transmission quality value of the communication line between the communication terminal 130 and the call control server 110 when the voice communication from the communication terminal 130 is not executed.

  In this way, even in a state in which a call is not being performed, by deriving the transmission quality value from the own communication terminal 130 to the call control server 110 in real time, the user can set the transmission quality value before starting the call. Thus, it is possible to select the timing for starting a call, such as starting communication after waiting for a high transmission quality.

  Further, the transmission quality value derived to the quality deriving unit 156 is the ITU-T recommendation G.264 as described above. An R value that is an evaluation value of the total transmission quality between communication terminals defined in 107 may be used. By using the R value as a transmission quality evaluation criterion, it is possible to easily grasp the transmission quality from standard numerical values without separately accepting a troublesome new index. The specific contents of the R value will be described in detail later.

  The quality estimation unit 158 of the call control server 110 estimates a transmission quality value (R value) after a predetermined time based on the current value of the R value that is the transmission quality value and the past transition. As such a predetermined time, any one of a general number of minutes to a few minutes may be used as the total time per call.

  FIG. 4 is an explanatory diagram showing an example of the estimation process of the quality estimation unit 158. The quality estimation unit 158 generates an extension curve that passes the current value from past transitions, and uses a value after a predetermined time of the curve as an estimated value of the transmission quality value. In FIG. 4, it is expected that the transmission quality will deteriorate in the future.

  With this configuration, the user can also grasp the transmission quality after a predetermined time and predict the situation where communication that will come in the future will be difficult, so it is more appropriate to decide whether to continue or cancel the current call. It becomes possible to do. For example, in FIG. 4 described above, since the deterioration of transmission quality is expected, the user can quickly round up the call with the communication partner to prepare for a sudden call failure.

  The quality transmitting unit 160 of the call control server 110 transmits the transmission quality derived by the quality deriving unit 156 and the transmission quality value after a predetermined time estimated by the quality estimating unit 158 to the communication terminal 130.

  The quality receiving unit 162 of the communication terminal 130 receives the transmission quality value from the quality transmitting unit 160 and the transmission quality value after a predetermined time from the quality estimating unit 158.

  The display unit 164 of the communication terminal 130 is configured with a color or monochrome display, and the IP telephone number of another communication terminal 132 input by a user operation or a call when an incoming call is received from another communication terminal 132. The IP telephone number of the call communication terminal is displayed. Further, the transmission quality value received by the communication terminal 130 and the estimated value of the transmission quality value are displayed in real time as text or graphics.

  FIG. 5 is an explanatory diagram showing a display example of the display unit 164. In FIG. 5A, a status bar 170 is provided under the display of the IP telephone number and the like, and the current transmission quality value (TQ: Transmission Quality) is displayed. When the estimated value (EV) estimated by the quality estimation unit 158 is also displayed, the transmission quality is arranged in parallel with the status bar 170 of the current transmission quality value as shown in FIG. An estimated value status bar 172 is provided so that the user can intuitively grasp the increase or decrease in transmission quality. For example, in FIG. 5, since it is estimated that the transmission quality value (R value) drops to 50 after a predetermined time, the user can be prepared when the call is interrupted.

  FIG. 6 is an explanatory diagram showing another display example of the display unit 164. In FIG. 6A, an antenna picture 180 is provided at the upper left of the display of the IP telephone number and the like, and the current transmission quality value (R value) is displayed. Further, when the estimated value of the transmission quality value estimated by the quality estimation unit 158 is also displayed, the estimated value of the transmission quality value is juxtaposed to each of the antenna pictures 180 of the current transmission quality value as shown in FIG. An antenna picture 182 is provided, and the user can intuitively grasp the increase / decrease in transmission quality as in FIG.

  The display unit 164 can display a warning by graphic display, and displays this warning if the R value is 50 or less. The R value is also provided with a class. If the R value is greater than 80, it is the same as a fixed telephone, and if it is greater than 70, it is the same as a mobile phone. A value larger than 50 is a requirement for obtaining the “050” number in the Internet telephone. Accordingly, the threshold value is set to 50, and a warning is displayed instead of or in addition to the numerical value. With this configuration, the user can easily notice the change even during a call, and can be used as a trigger for determining whether to continue or stop the call.

  In the Internet telephone system 100 described above, since the transmission quality of the communication line by the Internet telephone is notified to the user in real time, the user can easily grasp the communication status.

(R value)
The R value, which is an example of the transmission quality value used in this embodiment, will be described in detail below. The R value (Rating Factor) is an index that can evaluate the transmission quality of the communication lines between the communication terminals 130, and considers not only the call quality but also the propagation delay of the communication terminal 130 and the call control server between them. .

  The R value is specifically obtained from the parameters “Ro”, “Is”, “Id”, “Ie”, and “A”, and the calculation formula is R value = Ro−Is−Id−Ie + A. Here, Ro is the magnitude of the signal taking into account the influence of noise, Is is the degradation due to loudness and side sounds (sound transmitted from the mouthpiece to the earpiece through the telephone), and Id is the degradation due to delay and echo. , Ie represents deterioration in sound quality. Furthermore, a positive factor A is defined as a complementary value that takes into account convenience and the like when the communication terminal is a mobile phone.

  In derivation of the above-described elements, a total of 20 parameters such as loudness, room noise, delay, echo, and noise level affect the R value. In particular, calculation of Ro, Is, and Id has to substitute each parameter into a complicated expression. The maximum R value is 100, and the higher the R value, the better the call quality. In addition, although the R value includes many parameters, it may be difficult to measure all of them. Therefore, ITU-T Recommendation G. Reference numeral 107 defines standard values for parameters that are difficult to measure. Hereinafter, each parameter “Ro”, “Is”, “Id”, “Ie”, and “A” of the R value will be described in detail.

  “Ro (Basic signa 1-to-noise ratio)” expresses subjective quality deterioration related to noise due to line noise, transmitting / receiving room noise, and subscriber line noise. Such parameters are items resulting from terminal characteristics and usage environments, and do not directly affect network quality. The range of values that Ro can take is 0-80.

  “Is (Simultaneous impairment factor)” expresses subjective quality deterioration related to volume feeling due to OLR (volume), sidetone, and quantization distortion. Such parameters are items resulting from encoding and terminal characteristics, and network quality does not directly affect the parameters. The range of values that Is can take is 0-20.

  “Id (Delay impairment factor)” expresses subjective quality deterioration related to echo / delay caused by a transmitter echo, a receiver echo, and an absolute delay. Further, an IP packet transfer delay and an IP packet loss rate are also used for deriving Id. Also, the value of Id varies depending on the codec (encoding) method in voice communication. The range of values that Id can take is 0-20.

  FIG. 7 is an explanatory diagram showing the relationship between the codec and the delay degradation (Id). As shown in FIG. 7, depending on the codec, processing for channel failure and error mitigation ability are different, so that delay degradation is also different. Referring to FIG. From 723.1 729A has less degradation of the R value, and G. It can be confirmed that 711 is less deteriorated. Even if the delay time (standard value) required for the codec is simply compared, the G.G. 729A 25.0 msec is shorter, G. 711's 1.0 msec is shorter.

  “Ie (Equipment impairment factor)” expresses subjective quality degradation related to distortion / discontinuity due to a low bit rate codec, packet loss, and the like. Further, IP packet transfer delay fluctuation is also used for deriving Ie. Further, like the above-mentioned Id, the value of Ie varies depending on the codec method in voice communication. The range of values that Ie can take is 0-40.

  FIG. 8 is an explanatory diagram showing the relationship between the codec and the packet loss (Id). Packet loss (Id) occurs due to random channel errors, bursts or IP network route changes. For example, when two or more packets are lost in succession, the voice quality is fatally deteriorated rather than the loss of one packet. Therefore, it can be understood that the degradation of the R value is different depending on the codec method with respect to the packet loss (Ie).

  “A (Advantage Factor)” interpolates the influence of convenience such as mobile communication on subjective quality (satisfaction). Such parameters are a factor of convenience and do not directly affect network quality. For example, the value of A changes even when the mobile phone is used indoors or during high-speed movement. The range of values that A can take is 0-20.

  Based on the above description, for example, when the R value is derived with Ro = 80, Is = 10, Id = 10, Ie = 10, and A = 5, the R value is 80-10-10-10 + 5 = 55.

  Among “Ro”, “Is”, “Id”, “Ie”, and “A” described above, the absolute delay and packet loss particularly in “Id” and “Ie” have a great influence on the R value. Also, as shown in FIGS. 7 and 8, the codec is also important in evaluating the transmission quality. Therefore, if it is difficult to refer to a plurality of parameters, it is possible to estimate the transmission quality of a certain communication line even if only these three items are evaluated.

  As mentioned above, although preferred embodiment of this invention was described referring an accompanying drawing, it cannot be overemphasized that this invention is not limited to the example which concerns. It will be apparent to those skilled in the art that various changes and modifications can be made within the scope of the claims, and these are naturally within the technical scope of the present invention. Understood.

  For example, in the above-described embodiment, the configuration in which the R value is displayed on the display unit is shown. However, the present invention is not limited to this, and each parameter of the R value is displayed side by side or three items such as the absolute delay are displayed. Or various items can be displayed.

  The present invention can be used for an Internet telephone system, a call control server, and a communication terminal that perform voice communication between communication terminals via the Internet.

1 is a block diagram showing a schematic configuration of an Internet telephone system. It is the block diagram which showed the other structure of the internet telephone system. It is the functional block diagram which showed the function of each component in an internet telephone system. It is explanatory drawing which showed the example of the estimation process of a quality estimation part. It is explanatory drawing which showed the example of a display of a display part. It is explanatory drawing which showed the other example of a display of a display part. It is explanatory drawing which showed the relationship between a codec and delay degradation (Id). It is explanatory drawing which showed the relationship between a codec and packet loss (Id).

Explanation of symbols

100 Internet telephone system 110 call control server 130 communication terminal 152 communication request unit 154 communication establishment unit 156 quality derivation unit 158 quality estimation unit 160 quality transmission unit 162 quality reception unit 164 display unit

Claims (7)

In an Internet telephone system comprising a communication terminal and a call control server connected to the communication terminal via the Internet and controlling voice communication between the communication terminal and another communication terminal,
The call control server
In response to a request from the communication terminal, a communication establishment unit that establishes a communication line from the communication terminal to the other communication terminal;
A quality deriving unit for deriving a transmission quality value of the communication line;
A quality transmitter for transmitting the transmission quality value to the communication terminal;
Including
The communication terminal is
A communication requesting unit that requests the call control server to perform voice communication to the other communication terminal;
A quality receiver for receiving a transmission quality value from the quality transmitter;
A display unit for displaying the received transmission quality value in real time;
An internet telephone system comprising: The call control server
A quality estimation unit for estimating a transmission quality value after a predetermined time based on a current value and a past transition of the transmission quality value;
The Internet telephone system according to claim 1, wherein the quality transmission unit also transmits a transmission quality value after the predetermined time to the communication terminal. The quality deriving unit derives a transmission quality value of a communication line between the communication terminal and the call control server when voice communication from the communication terminal is not performed,
The Internet telephone system according to claim 1, wherein the quality transmission unit transmits the derived transmission quality value to a communication terminal.   The transmission quality value is ITU-T recommendation G.264. The Internet telephone system according to claim 1, wherein the Internet telephone system is an R value that is an evaluation value of total transmission quality between communication terminals defined in 107.   The Internet telephone system according to claim 4, wherein the display unit displays a warning if the R value is 50 or less. A call control server connected to a communication terminal via the Internet and controlling voice communication between the communication terminal and another communication terminal,
The call control server
In response to a request from the communication terminal, a communication establishment unit that establishes a communication line from the communication terminal to the other communication terminal;
A quality deriving unit for deriving a transmission quality value of the communication line;
In order to display the transmission quality value in real time on the display unit of the communication terminal, a quality transmission unit that transmits the transmission quality value to the communication terminal;
A call control server comprising: A communication terminal that performs voice communication with another communication terminal via the Internet by a call control server connected via the Internet,
A communication requesting unit that requests the call control server to perform voice communication to the other communication terminal;
A quality receiving unit for receiving a transmission quality value of a communication line from the communication terminal to the other communication terminal from the call control server;
A display unit for displaying the received transmission quality value in real time;
A communication terminal comprising:

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