JP2009124320A - Communication terminal device and computer program - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To transmit quality information at timing that can suppress deterioration in service quality as much as possible when notifying a quality management device of the service quality of an end-to-end service in a service session such as during the speech communication of an IP telephone. <P>SOLUTION: This communication terminal device is provided with: a quality information calculating part 14 for calculating quality information on the basis of a reception signal received from the opposite communication party through a communication network; a quality information transmission buffer for storing the quality information; a signal reproducing part 13 for reproducing a signal on the basis of the reception signal; a transmission timing calculating part 16 for making a reproduction section where a fluctuation in service quality is within an allowable range regardless of the existence of a fault in reproducing the signal transmission timing of the quality information; and a quality packet transmitting part 17 for transmitting the quality information to the quality management device at the transmission timing through the communication network. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to a communication terminal device and a computer program.

  Conventionally, in an IP (Internet Protocol) telephone system or a multimedia distribution system via an IP network, a quality control operation can be performed when transmission delay or transmission delay fluctuation larger than the quality standard occurs due to the influence of network congestion or the like. In order to maintain service quality, it is important to monitor end-to-end service quality for each service session and perform quality management.

In the prior art described in Patent Document 1, a threshold for voice quality is set in advance for GateWay (GW) or Operation Support System (OSS), and quality degradation is detected when the voice quality falls below the threshold. .
In the prior art described in Patent Document 2, the telephone itself has an end-to-end voice quality measurement function and a function of transmitting the measurement result to the management server, and generates specific pattern data when the telephone is on-hook to perform measurement. Audio quality monitoring.
In the prior art described in Patent Document 3, outgoing / incoming voice packet data during a call is stored in a measurement target, and the voice packet data is transmitted to the voice evaluation system after the call ends. Then, the voice evaluation system calculates quality information based on the voice packet data, receives a subjective evaluation value from the measurement subject, and performs maintenance operation of the voice network.
JP 2002-271392 A JP 2003-244235 A JP 2007-150774 A

However, the above-described conventional technology has the following problems.
In the prior art described in Patent Document 1, since quality degradation is detected only when the voice quality falls below the threshold, the voice quality during a call cannot be constantly monitored.
In the prior art described in Patent Document 2, since specific pattern data is generated and measured when the telephone is on-hook, the voice quality during a call cannot be measured.
The conventional technique described in Patent Document 3 accumulates all voice packet data during a call on the measurement target, and thus is difficult to apply to a communication terminal having a limited memory capacity such as a mobile phone. Further, since the accumulated voice packet data is transmitted to the voice evaluation system after the call is finished, a heavy load is imposed on the network.

  Further, in the prior art, quality information is transmitted at the end of a call or at the time of quality degradation. However, depending on the state of the network, it may be considered that the quality information is transmitted from the communication terminal. For example, if quality information is transmitted from a communication terminal when the network is in a congestion state, further congestion may be caused.

  In addition, in a terminal having a relatively low processing capability such as a mobile phone, the processing capability may be tight by transmitting quality information. However, if the processing capability is tight, the original processing such as voice signal processing may be performed. There is a risk that service quality will be degraded, for example, signal processing related to other services will be delayed and playback audio will be interrupted.

  The present invention has been made in view of such circumstances, and its purpose is to reduce the service quality when notifying the quality management device of the end-to-end service quality during a service session such as during an IP phone call. It is an object of the present invention to provide a communication terminal device and a computer program that can transmit quality information at a timing that can suppress as much as possible.

  In order to solve the above-described problems, a communication terminal apparatus according to the present invention includes a signal receiving unit that receives a signal from a communication partner via a communication network, and a quality information calculation that calculates quality information based on the received signal. Means, a quality information transmission buffer for storing the quality information, a signal reproduction means for reproducing a signal based on the received signal, and service quality fluctuation is within an allowable range even if there is a defect in the signal reproduction. A transmission timing calculation unit that sets a reproduction section as the transmission timing of the quality information, and a quality information transmission unit that transmits the quality information to the quality management apparatus at the transmission timing via the communication network. To do.

  The communication terminal device according to the present invention includes signal transmission means for transmitting a signal to the communication partner via the communication network, and the transmission timing calculation means has a problem in reproducing the transmission signal on the communication partner side. Even if the transmission interval in which the service quality variation is within the allowable range and the reproduction interval in which the service quality variation is within the allowable range even if there is a discrepancy in the signal reproduction, the transmission of the quality information is performed. It is characterized by timing.

  The communication terminal apparatus according to the present invention includes means for detecting a silence interval, and the silence interval is used as the transmission timing of the quality information.

  The communication terminal device according to the present invention includes means for detecting an image interval in which an image is stationary or an image interval in which there is little movement between consecutive images, and between image intervals in which an image is stationary or between consecutive images. It is characterized in that an image section with little motion is used as the transmission timing of the quality information.

  In the communication terminal device according to the present invention, the communication terminal device comprises means for determining whether the communication network is in a congestion state based on the quality information, and the transmission timing calculation means is determined to be in the congestion state In this case, the transmission of the quality information is suspended.

A computer program according to the present invention includes a quality information calculation function for calculating quality information based on a received signal received from a communication partner via a communication network, a function for storing the quality information in a quality information transmission buffer, and the reception A signal reproduction function for reproducing a signal based on a signal, and a transmission timing calculation function in which a reproduction section in which a variation in service quality is within an allowable range even if there is a defect in the signal reproduction is a transmission timing of the quality information The computer implements a quality information transmission function for transmitting quality information to a quality management device at the transmission timing via the communication network.
As a result, the communication terminal device described above can be realized using a computer.

  According to the present invention, when notifying the quality management apparatus of end-to-end service quality during a service session such as during an IP phone call, the quality information can be transmitted at a timing that can suppress degradation of service quality as much as possible. The effect is obtained.

Hereinafter, embodiments of the present invention will be described with reference to the drawings.
FIG. 1 is a block diagram showing a configuration of a communication service system according to an embodiment of the present invention. In FIG. 1, a communication terminal device (hereinafter referred to as a terminal) 1 sends and receives a signal packet 100 to and from a communication partner terminal 1 via an IP network 2. Examples of signals stored in the signal packet 100 include an audio signal and a video signal. Examples of the terminal 1 include an IP telephone terminal, an IP videophone terminal, and an IP videoconferencing terminal.

  The terminal 1 transmits quality information 200 to the quality management server 3 via the IP network 2. The quality management server 3 performs service quality management and control based on the quality information 200.

  FIG. 2 is a block diagram illustrating a configuration of the terminal 1 according to the present embodiment. The terminal 1 shown in FIG. 2 handles audio signals or video signals. Hereinafter, when the audio signal and the video signal are not particularly distinguished, they are collectively referred to as a media signal.

  In FIG. 2, the terminal 1 receives a signal packet sent from the communication partner via the IP network 2 and stores the received signal packet in the jitter buffer 11. The jitter buffer 11 is a buffer for correcting a packet sequence error and interval variation that occur during transmission.

  For example, in an IP telephone system using VoIP (Voice over IP) technology, a transmitting terminal encodes and packetizes a voice signal every short time (for example, 20 milliseconds) to a communication partner terminal (receiving terminal). Send. Also, in a video delivery system such as a videophone, a transmission terminal encodes and packetizes a video signal frame by frame using an MPEG (Moving Picture Experts Group) method, and transmits the packet to a communication partner terminal (receiving terminal). At this time, RTP (Real-Time Transmission Protocol) is generally used as a transmission protocol. In packet transmission by RTP, packets may arrive at a receiving terminal in an order and interval different from the transmission order and transmission interval of packets from the transmitting terminal. For this reason, the jitter buffer 11 refers to the “sequence number” described in the RTP header, and corrects the order and interval of the received packets so as to coincide with the transmission order and transmission interval of the packets from the transmitting terminal. The packet that has passed through the jitter buffer 11 is sent to the decoding unit 12 and the quality information calculation unit 14, respectively.

  The decoding unit 12 decodes the encoded data included in the packet received from the jitter buffer 11. The decoded media signal (reproduction signal) is sent to the signal reproduction unit 13 and the quality information calculation unit 14, respectively. Furthermore, the decoding unit 12 calculates information (referred to as a quality influence parameter) indicating how much the decoded media signal affects the service quality. The quality influence parameter is sent to the transmission timing calculation unit 16.

  In the case of a speech signal, information representing how important the speech segment after decoding is in terms of speech quality is calculated. For example, it can be considered that a silent section in which the speaker's voice is hardly included is not very important for voice quality. As a result, the decoding unit 12 calculates information (for example, the normalized power value of the audio signal) for determining the silent section of the audio signal, and sends the information to the transmission timing calculation unit 16.

  In the case of a video signal, information representing how important the decoded frame is for video quality is calculated. In the MPEG system, which is a general encoding system for video signals, three frames (I picture, P picture, B picture) having different frames are included. An I picture can be reproduced independently, and a P picture can be reproduced using a difference from the previous I picture or P picture. The B picture is reproduced using a difference from the preceding and following I pictures or P pictures. Therefore, when the video signal is reproduced, the I picture is referred to by both the P picture and the B picture, so that it can be said that the influence on the video quality is larger than that of the P picture and the B picture. On the other hand, it is known that a B picture has less influence on video quality than an I picture and a P picture. Accordingly, the decoding unit 12 acquires the picture type (I picture, P picture, B picture) described in the MPEG picture header as information indicating how important the decoded frame is for video quality. The picture type is sent to the transmission timing calculation unit 16. For information other than the picture type, information indicating how important the decoded frame is in terms of video quality includes, for example, motion difference information from the previous frame.

  The signal reproduction unit 13 reproduces the media signal (reproduction signal) received from the decoding unit 12. The signal reproduction unit 13 has a reproduction buffer. The signal reproduction unit 13 stores the media signal received from the decoding unit 12 in the reproduction buffer and adjusts the reproduction speed.

  FIG. 3 shows a conceptual diagram of the signal reproduction unit 13. In FIG. 3, audio playback is taken as an example. The signal reproduction unit 13 has a reproduction buffer 31 for storing an audio signal, temporarily stores the audio signal received from the decoding unit 12 in the reproduction buffer 31, takes out the audio signal from the reproduction buffer 31 in accordance with the audio reproduction speed, Sound is played back by the speaker 32.

The buffer size of the reproduction buffer 31 is preset for each terminal. It can be estimated based on the buffer size of the reproduction buffer 31 how many seconds later the audio signal decoded by the decoding unit 12 is reproduced. As shown in FIG. 3, assuming that the total time of the audio signals stored in the reproduction buffer 31 is L milliseconds, the time until the audio signal 300 newly sent from the decoding unit 12 to the signal reproduction unit 13 is reproduced. Time t is given by:
t = L

  The signal reproduction unit 13 outputs a signal representing the total time of media signals (reproduction signals) accumulated in the reproduction buffer to the transmission timing calculation unit 16.

  The quality information calculation unit 14 calculates quality information for each predetermined quality information transmission period (for example, every 8 seconds) based on the packet received from the jitter buffer 11 or the media signal (reproduction signal) received from the decoding unit 12. The quality information represents service quality. Examples of the service quality include voice quality during a call in the IP telephone service, voice quality and video quality during a call in the videophone service, and the like. Examples of the quality information include parameters indicating the state of the network (packet loss rate, delay, jitter, etc.), quality parameters included in a report block defined in “RFC3611 RTCP XR”, and the like. Furthermore, when a wireless communication terminal (for example, a mobile phone) is used, it is desirable to acquire quality information in a lower layer such as wireless reception intensity and calculate quality information in the lower layer. The quality information calculation unit 14 sends the quality information to the transmission timing calculation unit 16 and the quality packet generation unit 15.

  The quality packet generation unit 15 shapes the quality information received from the quality information calculation unit 14 into a predetermined format, and generates a quality packet obtained by packetizing the shaped quality information. An example of the predetermined format is a description format using XML (eXtensible Markup Language) shown in FIG. The XML format can describe quality information without being trapped by the type of quality information. When sending quality information to the quality management server 3, for example, a publish method of SIP (Session Initiation Protocol) can be used. In the publish method of SIP, an XML format can be used. The quality packet generator 15 sends the quality packet to the quality packet transmitter 17.

  The transmission timing calculation unit 16 calculates the transmission timing of the quality packet. This transmission timing calculation method will be described later. The transmission timing calculation unit 16 instructs the quality packet transmission unit 17 to transmit the quality packet.

  The quality packet transmission unit 17 transmits the quality packet to the quality management server 3 according to the transmission timing instructed from the transmission timing calculation unit 16. The quality packet transmission unit 17 includes a transmission buffer for adjusting the quality packet transmission timing. The quality packet transmission unit 17 temporarily stores the quality packet received from the quality packet generation unit 15 in the transmission buffer, extracts the quality packet from the transmission buffer in accordance with the transmission timing instructed from the transmission timing calculation unit 16, and the quality management server 3 to send.

  The signal input unit 21 inputs a media signal to be sent to a communication partner. For example, an audio signal is input in the case of an IP phone, and an audio signal and a video signal are input in the case of a video phone.

  The encoding unit 22 encodes the media signal input by the signal input unit 21. The encoded media signal (transmission signal) is sent to the signal packet sending unit 23. Further, the encoding unit 22 calculates a quality influence parameter that represents how much the encoded media signal affects the service quality. This quality influence parameter is the same as the quality influence parameter calculated by the decoding unit 12. The quality influence parameter is sent to the transmission timing calculation unit 16.

  The signal packet transmission unit 23 packetizes the encoded media signal (transmission signal) received from the encoding unit 22 and transmits the signal packet to a communication partner terminal. Further, the signal packet transmission unit 23 outputs a signal representing the timing for transmitting the signal packet to the transmission timing calculation unit 16.

  Next, the quality information transmission timing calculation method according to the present embodiment will be described.

  Although quality information transmission is based on transmission to the quality management server 3 every preset quality information transmission cycle (for example, every 8 seconds), it is desirable to avoid quality information transmission leading to deterioration of service quality as much as possible. For example, in a terminal having a relatively low processing capability such as a mobile phone, the processing capability may be tight by transmitting quality information. There is a risk that service quality will be degraded, for example, signal processing related to other services will be delayed and playback audio will be interrupted. In order to avoid such a situation as much as possible, the transmission timing is calculated so that the quality information is transmitted at a timing at which the service quality fluctuation caused by the quality information transmission falls within an allowable range.

  Specifically, in the case of a service that handles voice (for example, IP phone service), if the playback voice is interrupted in a voiced section that contains a relatively large amount of speaker's voice, it is likely that the voice quality is significantly lowered. . On the other hand, in a silent section in which the speaker's voice is hardly included, even if the playback voice is interrupted, it is difficult to perceive the interruption of the voice, and it is not so much felt that the voice quality has deteriorated. Based on such knowledge, even if there is a discrepancy in signal reproduction in the silent section, it is assumed that the service quality fluctuation is within an allowable range. Therefore, if quality information is transmitted in a silent section, service quality fluctuations can be kept within an allowable range, and degradation of service quality can be suppressed as much as possible.

  In the case of a service that handles video (for example, a videophone service), even if the playback image is interrupted in an image section where the image is still, it is difficult to perceive the interruption of the playback image, and the video quality is not much reduced. I can't feel it. Similarly, it is difficult to perceive an image break even in an image section where there is little movement between consecutive images, and it is not so much felt that the video quality has deteriorated. Based on such knowledge, even if there is a discrepancy in signal reproduction in an image section where images are stationary and an image section where there is little motion between consecutive images, the service quality fluctuation is assumed to be within an allowable range. . Therefore, if quality information is transmitted in an image section where images are stationary or in an image section where there is little movement between successive images, service quality fluctuations can be kept within an allowable range, and deterioration in service quality is minimized. Can do.

  In addition, transmitting quality information when the network is in a congested state places an additional load on the network, which can be a cause of service quality degradation. For this reason, when it is determined that the network is in a congestion state, it is preferable to maintain transmission of quality information.

FIG. 5 is a flowchart showing a flow of quality information transmission timing calculation processing according to the present embodiment. Note that FIG. 5 shows an example of a service that handles voice.
In FIG. 5, the transmission timing calculation unit 16 receives information used for quality information transmission timing calculation from each unit.

  From the decoding unit 12 and the encoding unit 22, the normalized power value of the audio signal is received as information for determining the silent section of the audio signal. The normalized power value received from the decoding unit 12 relates to the sound to be reproduced. The normalized power value received from the encoding unit 22 relates to the sound to be transmitted.

  Quality information is received from the quality information calculation unit 14. From the signal reproduction unit 13, a signal (reproduction timing information) representing the total time of the reproduction signals accumulated in the reproduction buffer is received. The reproduction timing information indicates the timing at which audio corresponding to the normalized power value received from the decoding unit 12 is reproduced.

  From the signal packet transmission unit 23, a signal (signal packet transmission timing information) indicating timing for transmitting a signal packet is received. The signal packet transmission timing information indicates the timing at which audio corresponding to the normalized power value received from the encoding unit 22 is transmitted.

  In step SP1, the transmission timing calculation unit 16 determines whether the network is in a congestion state based on the quality information. In the quality information, for example, when the packet loss rate is larger than a preset threshold value, it is determined that the network is in a congestion state. In addition to the packet loss rate, the network congestion state may be determined using round-trip delay time or jitter.

  As a result of step SP1, when it is determined that the network is in a congested state, transmission of quality information is suspended. Therefore, in this case, the quality packet transmission unit 17 does not transmit the quality packet. On the other hand, if it is determined that the network is not congested, the process proceeds to step SP2.

  In step SP2, the transmission timing calculation unit 16 calculates transmission timing candidates.

Here, the transmission timing candidate calculation process will be described.
First, based on the normalized power value received from the decoding unit 12, a silent section in the reproduced sound is determined. Similarly, based on the normalized power value received from the encoding unit 22, the silent section in the transmission voice is determined. For the determination of the silent section, a threshold for determining the silent section is provided in advance, and a section having a normalized power value less than the threshold is determined as the silent section. By obtaining the normalized power value in units of packets, it is possible to determine a silent section in units of packets. FIG. 6 is a graph for explaining the silent section determination processing. In the example of FIG. 6, the normalized power value is obtained every 20 milliseconds. When the normalized power value is greater than or equal to the threshold, it is determined as a voiced section, and when the normalized power value is less than the threshold, it is determined as a silent section.

  Next, based on the reproduction timing information received from the signal reproduction unit 13, the timing for reproducing the silent section in the reproduced audio is calculated. Similarly, based on the signal packet transmission timing information received from the signal packet transmission unit 23, the timing at which the silent section in the transmission voice is transmitted is calculated. The timing at which the silent section is reproduced is set as a timing candidate for transmitting quality information (transmission timing candidate). More preferably, a section in which the timing at which the silent section is reproduced matches the timing at which the silent section is transmitted is set as a transmission timing candidate.

In general, it is known that the ratio of silent sections in conversational speech is about 60%. Therefore, as illustrated in FIG. 7, there are many silent sections. In the example of FIG. 7, the timing at which the silent part in the received voice is reproduced is Ri, the timing at which the silent part in the transmitted voice is transmitted is Si, and the timing at which both coincide is “R ₂ = S ₃ ” and “ R ₈ = S ₇ ”. If this timing is set as a transmission timing candidate, it is not necessary to place a burden on both the audio playback and audio transmission processes.

Returning to FIG.
In step SP3, the transmission timing calculation unit 16 calculates the transmission timing from the transmission timing candidates. In this transmission timing calculation process, a timing larger than the quality information transmission cycle and closest to the quality information transmission cycle is calculated among a plurality of transmission timing candidates. When a plurality of transmission timing candidates are calculated as illustrated in FIG. 8, the transmission timing candidate G is larger than the quality information transmission cycle and closest to the quality information transmission cycle is “transmission timing candidate G”. Is the transmission timing. The transmission timing calculation unit 16 notifies the quality packet transmission unit 17 of the transmission timing of the calculation result.

  The quality packet transmission unit 17 transmits the quality packet according to the transmission timing notified from the transmission timing calculation unit 16.

  In FIG. 5, the case of a service that handles audio is given as an example, but the flow of processing is the same for a service that handles video. In the case of a video signal, an image section where an image is stationary and an image section where there is little motion between consecutive images are set as transmission timing candidates. Specifically, for example, a B picture section is set as a transmission timing candidate based on an MPEG picture type. Alternatively, based on motion difference information with respect to the previous frame, an image section where the image is stationary and an image section where there is little motion between successive images are set as transmission timing candidates.

  As described above, according to this embodiment, even if there is a discrepancy in signal reproduction, the reproduction section in which the service quality fluctuation is within the allowable range is set as the transmission timing of the quality information. Specifically, in the audio signal, the silent period is set as the transmission timing of the quality information. In the video signal, an image interval in which an image is stationary or an image interval in which there is little movement between consecutive images is set as the transmission timing of quality information. Thereby, the effect that the fall of the service quality which arises by quality information transmission can be suppressed as much as possible is acquired.

  If it is determined that the communication network is in a congested state, transmission of quality information is suspended. As a result, it is possible to avoid applying a further load to the communication network in a congested state and to prevent deterioration of service quality.

  Note that the terminal 1 according to the present embodiment may be realized by dedicated hardware, or is configured by a computer system such as a personal computer, and implements the functions of each unit of the terminal 1 shown in FIG. The function may be realized by executing a program to do so.

Further, the program for realizing each step shown in FIG. 5 is recorded on a computer-readable recording medium, and the program recorded on the recording medium is read by the computer system and executed, thereby executing the quality information transmission timing. A calculation process may be performed. Here, the “computer system” may include an OS and hardware such as peripheral devices.
“Computer-readable recording medium” refers to a flexible disk, a magneto-optical disk, a ROM, a writable nonvolatile memory such as a flash memory, a portable medium such as a DVD (Digital Versatile Disk), and a built-in computer system. A storage device such as a hard disk.

Further, the “computer-readable recording medium” means a volatile memory (for example, DRAM (Dynamic Random Access Memory)), etc., which hold programs for a certain period of time.
The program may be transmitted from a computer system storing the program in a storage device or the like to another computer system via a transmission medium or by a transmission wave in the transmission medium. Here, the “transmission medium” for transmitting the program refers to a medium having a function of transmitting information, such as a network (communication network) such as the Internet or a communication line (communication line) such as a telephone line.
The program may be for realizing a part of the functions described above. Furthermore, what can implement | achieve the function mentioned above in combination with the program already recorded on the computer system, and what is called a difference file (difference program) may be sufficient.

As mentioned above, although embodiment of this invention was explained in full detail with reference to drawings, the specific structure is not restricted to this embodiment, The design change etc. of the range which does not deviate from the summary of this invention are included.
For example, the communication terminal device according to the present invention includes a reception-only terminal. For example, it may be a receiving terminal for a service that distributes audio and video at a concert venue in real time.

It is a block diagram which shows the structure of the communication service system which concerns on one Embodiment of this invention. It is a block diagram which shows the structure of the communication terminal device 1 which concerns on one Embodiment of this invention. It is a conceptual diagram of the signal reproduction | regeneration part 13 shown in FIG. It is an example of the quality information which concerns on one Embodiment of this invention. It is a flowchart which shows the flow of the quality information transmission timing calculation process which concerns on one Embodiment of this invention. It is a graph for demonstrating the silence area determination process which concerns on one Embodiment of this invention. It is an example of the transmission timing candidate which concerns on one Embodiment of this invention. It is an example of the transmission timing candidate which concerns on one Embodiment of this invention.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 ... Communication terminal device, 2 ... IP network (packet communication network), 3 ... Quality management server (quality management apparatus), 11 ... Jitter buffer, 12 ... Decoding part, 13 ... Signal reproduction | regeneration part, 14 ... Quality information calculation part, DESCRIPTION OF SYMBOLS 15 ... Quality packet generation part, 16 ... Transmission timing calculation part, 17 ... Quality packet transmission part, 21 ... Signal input part, 22 ... Encoding part, 23 ... Signal packet transmission part, 31 ... Reproduction buffer

Claims (6)

Signal receiving means for receiving a signal from a communication partner via a communication network;
Quality information calculating means for calculating quality information based on the received signal;
A quality information transmission buffer for storing the quality information;
Signal reproducing means for reproducing a signal based on the received signal;
A transmission timing calculation means for setting a reproduction section in which a service quality variation is within an allowable range even if there is a flaw in the signal reproduction as a transmission timing of the quality information;
Quality information transmitting means for transmitting quality information to the quality management device at the transmission timing via the communication network;
A communication terminal device comprising: Comprising signal transmission means for transmitting a signal to a communication partner via the communication network;
The transmission timing calculation means includes a transmission section in which a variation in service quality is within an allowable range even if there is a reproduction failure of the transmission signal on the communication partner side, and a service quality even if there is a failure on the signal reproduction. The communication terminal apparatus according to claim 1, wherein a section in which a reproduction section whose variation is within an allowable range coincides with the transmission timing of the quality information. Means for detecting silent sections;
The communication terminal apparatus according to claim 1 or 2, wherein a silent period is set as a transmission timing of the quality information. Means for detecting an image interval in which the image is stationary or an image interval in which there is little movement between successive images,
The communication terminal device according to claim 1 or 2, wherein an image section in which an image is stationary or an image section in which movement between successive images is small is set as the transmission timing of the quality information. Means for determining whether the communication network is congested based on the quality information;
The communication terminal apparatus according to claim 1, wherein the transmission timing calculation unit suspends transmission of the quality information when it is determined that the communication network is in a congested state. A quality information calculation function for calculating quality information based on a received signal received from a communication partner via a communication network;
A function of storing the quality information in a quality information transmission buffer;
A signal reproduction function for reproducing a signal based on the received signal;
A transmission timing calculation function in which a reproduction section in which service quality variation is within an allowable range even if there is a defect in the signal reproduction is a transmission timing of the quality information;
A quality information transmission function for transmitting quality information to the quality management device at the transmission timing via the communication network;
A computer program for causing a computer to realize the above.

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