JP6389070B2 - Communication terminal, voice communication control method, and computer program - Google Patents

Description

  The present invention relates to a technique for controlling voice communication by a wireless terminal.

  In recent years, a technique for performing voice communication over an IP network by VoIP (Voice Over Internet Protocol) has been put into practical use. For example, Patent Document 1 proposes a technique for encoding audio data transmitted and received between terminals performing audio communication using different encoding methods between a transmission source terminal and a transmission destination terminal.

JP 2012-209638 A

  However, when voice communication by VoIP is realized by, for example, a wireless terminal, there is a problem that communication at the initial stage of a call becomes unstable. This is because the base station that accommodates the wireless terminal allocates a communication band to the wireless terminal based on the generated traffic.

  In view of the above circumstances, an object of the present invention is to provide a technology that enables a communication terminal to perform voice communication more stably.

  One aspect of the present invention is a communication terminal that performs voice communication using VoIP via a communication band allocation device that allocates a communication band corresponding to a communication amount to a communication terminal, and before starting voice communication of the first packet. It is a communication terminal provided with the audio | voice communication control part which transmits to.

  One aspect of the present invention is the communication terminal described above, wherein the voice communication control unit continuously transmits the first packet.

  One aspect of the present invention is the communication terminal described above, wherein the voice communication control unit transmits the first packet so that a communication amount sufficient to allocate a communication band necessary for voice communication is generated.

  One aspect of the present invention is the communication terminal described above, wherein the network status at the time before starting voice communication is determined based on the reception status of the first response packet received for each first packet. A network monitor unit for estimation is further provided.

  One aspect of the present invention is the communication terminal described above, wherein the voice communication control unit transmits a second packet during voice communication, and the network monitor unit receives each second packet. Based on the reception status of the second response packet, the status of the network during voice communication is estimated.

  One aspect of the present invention is the communication terminal described above, further including a display unit that displays an operation screen for a user to operate the communication terminal, wherein the network monitor unit is configured to start voice communication. Information indicating the network status at the time is displayed on the display unit.

  One aspect of the present invention is the communication terminal described above, wherein the network monitor unit displays information representing a network state during voice communication on the display unit.

  One aspect of the present invention is the communication terminal, wherein the voice communication control unit associates time, location, and information indicating a network status at the time of the location with time and location information. To determine the amount of transmission of the first packet.

  One aspect of the present invention is the communication terminal described above, wherein the network monitor unit is one or both of a state of the network before starting voice communication and a state of the network during voice communication. Is stored as the time and location information.

  One aspect of the present invention is a voice communication control method performed by a communication terminal that performs voice communication based on VoIP via a communication band allocation device that allocates a communication band corresponding to a communication amount to a communication terminal. An audio communication control method including an audio communication control step for transmitting before starting audio communication.

  One aspect of the present invention provides a voice communication control method in which a communication terminal that performs voice communication using VoIP via a communication band allocation device that allocates a communication band corresponding to a communication amount to a communication terminal performs voice communication on a first packet. This is a computer program for causing a computer to execute a voice communication control step to be transmitted before starting.

  According to the present invention, the communication terminal can perform voice communication more stably.

It is a figure which shows the structural example of the audio | voice communication system which performs audio | voice communication by VoIP. It is a system configuration figure showing the system configuration of the voice communication system of an embodiment. It is a functional block diagram which shows the function structure of the VoIP terminal 1 of embodiment. It is a figure which shows the specific example of the call quality table. It is a figure which shows the operation example of the audio | voice communication system of embodiment. It is a figure which shows the specific example of a dial screen. It is a figure which shows the specific example of a screen during a call. It is a sequence diagram which shows the flow of the 1st process in the audio | voice communication system of embodiment. It is a sequence diagram which shows the flow of the 2nd process in the audio | voice communication system of embodiment.

FIG. 1 is a diagram illustrating a configuration example of a voice communication system that performs voice communication using VoIP.
In the figure, a VoIP terminal 1-1, a VoIP terminal 1-2, a VoIP terminal 1-3, and a VoIP telephone 2 are terminals that perform voice communication by VoIP. The VoIP terminal 1-1, the VoIP terminal 1-2, and the VoIP terminal 1-3 are wireless terminals that perform voice communication by wireless communication such as a mobile phone or a smartphone. The VoIP telephone 2 is a fixed telephone having a voice communication function based on VoIP. The base station 3-1, the base station 3-2, and the base station 3-3 are base stations that accommodate the wireless terminals. The base station 3-1 accommodates the VoIP terminal 1-1. The base station 3-2 accommodates the VoIP terminal 1-2. The base station 3-3 accommodates the VoIP terminal 1-3.

The carrier IP network 4 is a carrier IP (Internet Protocol) network that provides communication services to users of the VoIP terminal 1-1, the VoIP terminal 1-2, and the VoIP telephone 2. A base station 3-1, a base station 3-2 and a VoIP telephone set 2 are connected to the provider IP network 4. Further, the provider IP network 4 is connected to a WAN (Wide Area Network) and can be connected to the IP network of another provider via the WAN.
The separate provider IP network 5 is an IP network of a provider that provides a communication service to the user of the VoIP terminal 1-3. A base station 3-3 is connected to the separate operator IP network 5.
The WAN 6 is a network that connects the provider IP network 4 and the separate provider IP network 5. The WAN 6 is, for example, a WAN such as the Internet, a dedicated line, and a wide area Ethernet (registered trademark).

FIG. 1 shows an example of the configuration of a voice communication system using VoIP, and the configuration of the voice communication system using VoIP may be different from that shown in FIG. For example, the number of IP networks of operators connected by the WAN 6 may be different from that in the figure. Similarly, the number of VoIP terminals 1 connected to each base station 3 may be different from that shown in FIG. Similarly, the numbers of base stations 3 and VoIP telephones 2 connected to the IP networks of the respective carriers may be different from those in the figure.
In addition, since the voice communication system using VoIP includes a conversion device that converts between packet switching communication and circuit switching communication, each VoIP terminal 1 and VoIP telephone 2 are connected to a fixed telephone line. It is also possible to perform voice communication with another telephone.

FIG. 2 is a system configuration diagram illustrating a system configuration of the voice communication system according to the embodiment.
In this embodiment, in order to simplify the description, it is assumed that a terminal that performs voice communication using VoIP uses a communication service of the same operator. In this case, the configuration of the voice communication system of the embodiment is the configuration shown in FIG. The voice communication system of the embodiment includes a VoIP terminal 1-1, a VoIP terminal 1-2, a base station 3-1, a base station 3-2, a provider IP network 4, and a response server 7. The VoIP terminal 1-1, the VoIP terminal 1-2, the base station 3-1, the base station 3-2, and the provider IP network 4 are the same as those in FIG. In the following description, VoIP terminal 1-1 and VoIP terminal 1-2 are described as VoIP terminal 1 unless otherwise specified. Similarly, the base station 3-1 and the base station 3-2 are referred to as a base station 3.
Note that the VoIP terminal 1 is not limited to a wireless terminal as in the example of the figure as long as it has a voice communication function based on VoIP.

The base station 3 (communication band allocation device) allocates a communication band for each VoIP terminal 1 accommodated by itself. Generally, the base station 3 uses only the minimum bandwidth necessary for communication control with its own device for terminals that do not perform user communication among the accommodated terminals in order to effectively use the radio bandwidth. assign. User communication is communication other than communication for the above-described communication control. For example, the user communication includes VoIP voice communication, communication for transmitting / receiving user data, communication for accessing a Web server, and the like. When the base station 3 detects that the user communication has occurred, the base station 3 allocates a communication band corresponding to the communication amount generated by the user communication to the terminal that has performed the user communication. However, there is a time lag from when the base station 3 detects the occurrence of user communication until the communication band is allocated. For this reason, the VoIP terminal 1 may not be allocated a sufficient communication band immediately after starting voice communication, and the call quality may deteriorate.
Note that the communication line to which the VoIP terminal 1 is connected is not limited by the type of line such as wireless and wired as long as it is a communication line that allocates a communication band according to the traffic. For example, the base station 3 may be a wired network switch that allocates a communication band by the above method. The response server 7 receives the first packet (first packet) and the second packet (second packet) transmitted from the VoIP terminal 1. The response server 7 generates a response packet for each received packet and transmits the response packet to the transmission source VoIP terminal 1. Details of the first packet and the second packet will be described later.

FIG. 3 is a functional block diagram illustrating a functional configuration of the VoIP terminal 1 according to the embodiment.
The VoIP terminal 1 includes a central processing unit (CPU), a memory, an auxiliary storage device, and the like connected by a bus, and executes a voice communication program. The VoIP terminal 1 functions as a device including a storage unit 11, a communication unit 12, a display unit 13, an input unit 14, a voice input / output unit 15, a network monitor unit 16, and a voice communication control unit 17 by executing a voice communication program.
Note that all or part of the functions of the VoIP terminal 1 may be realized by using hardware such as an application specific integrated circuit (ASIC), a programmable logic device (PLD), and a field programmable gate array (FPGA). The voice communication program may be recorded on a computer-readable recording medium. The computer-readable recording medium is, for example, a portable medium such as a flexible disk, a magneto-optical disk, a ROM, a CD-ROM, or a storage device such as a hard disk built in the computer system. The voice communication program may be transmitted via a telecommunication line.

  The storage unit 11 is configured using a storage device such as a magnetic hard disk device or a semiconductor storage device. The storage unit 11 stores in advance a VoIP application for realizing voice communication by VoIP. The storage unit 11 stores a call quality table 111 (time / location information) in advance.

FIG. 4 is a diagram showing a specific example of the call quality table 111.
The call quality table 111 is a table that stores information related to the quality of voice communication. The call quality table 111 has a call quality record for each position information. The call quality record has each value of position information, time information, call OK rate, and call NG rate. The position information is information representing the position of the own device. The position information is information such as latitude and longitude. The time information is information representing time. The call OK rate represents the probability that the quality of voice communication performed at the time indicated by the time information is good at the position indicated by the position information. The call NG rate represents the probability that the quality of voice communication performed at the time indicated by the time information is not good at the position indicated by the position information.

Returning to the description of FIG.
The communication unit 12 is configured using a wireless communication interface such as Wi-Fi (Wireless Fidelity) (registered trademark). The communication unit 12 controls wireless communication with the base station 3.
The display unit 13 is configured using a display device such as a touch panel or a liquid crystal display. The display unit 13 displays an operation screen of the VoIP application in accordance with instructions from the network monitor unit 16 and the voice communication control unit 17. There are two operation screens: a dial screen and a call screen. Details of the dial screen and the call screen will be described later.

The input unit 14 is configured using an input device such as a touch panel or a keyboard. The input unit 14 receives an input of a user operation. The input unit 14 outputs user input information to the voice communication control unit 17.
The voice input / output unit 15 is configured using a voice input / output device such as a microphone or a speaker. The voice input / output unit 15 converts voice uttered by the user into a voice signal. The voice input / output unit 15 outputs the acquired voice signal to the voice communication control unit 17. The voice input / output unit 15 acquires a voice signal output from the voice communication control unit 17. The voice input / output unit 15 converts the acquired voice signal into voice and outputs the voice.

  The network monitor unit 16 acquires a response packet transmitted from the response server 7. The network monitor unit 16 estimates the network status based on the response packet reception status. The response packet reception status refers to, for example, the time from transmission of a packet to reception of the response packet (hereinafter referred to as “response time”) and the number of transmission packets for which the response packet has not been transmitted (hereinafter referred to as “packet loss”). And so on). The network monitoring unit 16 calculates network delay, communication fluctuation, and the like based on the response packet reception status. The network monitor unit 16 estimates the network status based on information such as the calculated network delay and communication fluctuation. The network monitor unit 16 displays the estimated network status on the display unit 13.

  The voice communication control unit 17 controls voice communication of the VoIP terminal 1. The voice communication control unit 17 is a functional unit that is generated when an operation for executing a VoIP application is input by a user. When an operation for executing the VoIP application is input, the VoIP application is read from the storage unit 11 by the CPU of the VoIP terminal 1 and developed on the memory.

  The voice communication control executed by the voice communication control unit 17 is roughly classified into a first process and a second process. The first process is a preprocess that is performed before the VoIP terminal 1 starts a voice communication by transmitting a call signal to the VoIP terminal 1 (hereinafter referred to as “opposite terminal”) as a call destination. In the first process, the VoIP terminal 1 transmits the first packet to the response server 7. This first packet is relayed to the response server 7 by the base station 3. The base station 3 that has received the first packet allocates a communication band corresponding to the traffic generated by the received first packet to the VoIP terminal 1. For this reason, the voice communication control unit 17 continuously transmits the first packets so that a communication amount sufficient to allocate a communication band necessary for voice communication using VoIP is generated. By transmitting the first packet, the VoIP terminal 1 can secure a communication band necessary for voice communication before starting voice communication.

In voice communication by VoIP, it is said that sufficient call quality can be obtained if a communication band of about 20 kbps is secured. Accordingly, the voice communication control unit 17 of the present embodiment may continuously transmit the first packet in order to secure a communication band of about 20 kbps, for example. For example, in this case, the VoIP terminal 1 may transmit 40-byte packets at 40 msec intervals.
Further, the voice communication control unit 17 may continuously transmit the first packet until the voice communication is started so that the communication band assigned to the base station 3 is maintained until the voice communication is started. . The voice communication control unit 17 starts transmission of the first packet at a time that is at least a time that is required for the base station 3 to allocate a communication band necessary for voice communication from the timing of starting voice communication. It may be configured as follows. Further, the voice communication control unit 17 may be configured to start transmission of the first packet at least by the timing at which the communication band is allocated by the base station 3 before the start of voice communication.

  Note that, by receiving a response packet (first response packet) to the first packet transmitted for securing the communication band, the network monitor unit 16 can estimate the state of the network before starting voice communication. it can. That is, by transmitting the first packet, the VoIP terminal 1 can secure a communication band before the start of voice communication and estimate a network state before the start of voice communication.

  The second process is a process for controlling voice communication from the start of the call to the end of the call. In the second process, the VoIP terminal 1 first receives the user's dial operation and outgoing call input. The voice communication control unit 17 transmits a transmission signal for starting voice communication to the opposite terminal with the opposite terminal indicated by the input information. When the opposite terminal receives the outgoing signal, voice communication between the originating VoIP terminal 1 and the opposite terminal is started. The voice communication control unit 17 generates voice data by performing processing such as encoding and compression on the voice signal output from the voice input / output unit 15. The voice communication control unit 17 divides the generated voice data into packets that are transmission units to the network. The voice communication control unit 17 transmits a packet obtained by dividing the voice data (hereinafter referred to as “voice packet”) to the opposite terminal.

Further, the voice communication control unit 17 acquires a voice packet transmitted from the opposite terminal. The voice communication control unit 17 combines the acquired voice packets to acquire voice data. The voice communication control unit 17 performs processing such as decoding and decompression on the acquired voice data to generate a voice signal. The voice communication control unit 17 outputs the generated voice signal to the voice input / output unit 15. When the user inputs a call end operation, the voice communication control unit 17 transmits an end signal to the opposite terminal and ends the voice communication.
Further, the voice communication control unit 17 transmits the second packet to the response server 7 at a predetermined timing while performing voice communication with the opposite terminal. By transmitting the second packet during voice communication, the VoIP terminal 1 can estimate the state of the network during voice communication.

FIG. 5 is a diagram illustrating an operation example of the voice communication system according to the embodiment.
In the figure, the vertical axis represents time. In the figure, the VoIP terminal 1 transmits the first packet to the response server 7 at time t ₁₁ , time t ₂₁ , time t _31, and time t ₄₁ . The VoIP terminal 1 receives a response packet to the first packet at time t ₁₂ , time t ₂₂ , time t _32, and time t ₄₂ , respectively. RTT1, RTT2, RTT3 and RTT4, respectively, indicating the time _{t 11,} the time t _21, the first packet transmitted at time _{t 31} and time _{t 41 RTT (Round Trip Time)} . The network monitor unit 16 calculates a statistical value such as an average or a variance for these RTTs. The network monitor unit 16 estimates the state of the network before starting voice communication by comparing the calculated statistical value with a preset threshold value.

Further, in the figure, the VoIP terminal 1 transmits the second packet to the response server 7 at time t ₅₁ and time t ₆₁ . Then, VoIP terminal 1, a response packet to the second packet (second response packet), respectively, are received at time _{t 52} and time _{t 62.} RTT5 and RTT6 each represent RTT second packet transmitted at time _{t 51} and time _{t 61.} Similarly to the first packet, the network monitoring unit 16 calculates the statistical value such as the average or variance for the RTT of the second packet, thereby estimating the state of the network during voice communication.

  Note that the network monitoring unit 16 may calculate a statistical value based on all the RTTs in the estimation of the network status based on the RTT of the first packet (or the second packet), or may be based on a part of the RTTs. Statistical values may also be calculated. When calculating a statistical value based on some RTTs, for example, the network monitoring unit 16 calculates a statistical value based on the latest N RTTs among the RTTs of the first packet (or the second packet). Also good.

The network monitor unit 16 may estimate the network status by comparing the statistical value calculated as described above with a preset statistical value threshold. For example, the network monitor unit 16 may estimate that the network status is good when the calculated statistical value is smaller than the threshold value. Further, the network monitor unit 16 may estimate that the network status is bad when the calculated statistical value is larger than the threshold value.
Note that the network monitor unit 16 may estimate the state of the network by comparing the calculated statistical value with a plurality of threshold values. For example, the network monitor unit 16 may estimate that the network status is good or bad at a plurality of stages based on a plurality of threshold values.

FIG. 6 is a diagram showing a specific example of a dial screen.
Reference numeral 131 in the figure represents a dial screen. The dial screen 131 is a screen displayed before the VoIP terminal 1 starts voice communication. The dial screen 131 includes a communication status display area 1311, a destination display area 1312, a dial display area 1313, and a call start button 1314.
The communication status display area 1311 displays information representing the network status estimated in the first process. The network status displayed in the communication status display area 1311 may be displayed in the form of characters as shown in the figure, or may be displayed in another mode. For example, in the communication status display area 1311, a graphic having a color corresponding to the network status may be displayed, or a score indicating the network status may be displayed. Further, for example, the score indicating the network status may be displayed as a numerical value, or may be expressed as the number or size of the displayed graphic.
By displaying information representing the network status in the communication status display area 1311, the user can start voice communication after confirming that a communication band necessary for voice communication has been secured. For example, the network monitor unit 16 calculates the communication band based on the transmission amount of the first packet and the reception amount of the response packet. The communication band may be calculated based on the amount of voice packets received per unit time. When the calculated communication band exceeds a predetermined threshold, the network monitor unit 16 displays “good” in the communication status display area 1311 to notify that the communication band necessary for voice communication has been secured. The network monitor unit 16 may estimate the required RTT based on the target communication band, and may estimate the network status by comparing the estimated RTT with the actual RTT. For example, the network monitor unit 16 may estimate that a sufficient communication band is secured when the estimated RTT is larger than the actual RTT. For example, the network monitor unit 16 may estimate that a sufficient communication band is not secured when the estimated RTT is smaller than the actual RTT. The network monitor unit 16 may display the network status estimated in this way in the communication status display area 1311.

The destination display area 1312 is an area where the telephone number of the destination input by the user is displayed.
The dial display area 1313 is an area in which a dial for the user to input a telephone number of a destination is displayed.
The call start button 1314 is a button for inputting an operation for the user to make a call to the telephone number displayed in the destination display area 1312. When the call start button 1314 is pressed, the voice communication control unit 17 transmits a call signal to the opposite terminal indicated by the telephone number displayed in the callee display area 1312 and starts voice communication. When the voice communication control unit 17 starts voice communication, the voice communication control unit 17 displays a screen during a call on the display unit 13.

The dial screen 131 may display other information not shown in the figure. For example, the current time or a picture of the owner of the opposite terminal as the destination may be displayed.
Further, for example, available communication means may be displayed on the dial screen 131 according to the network status. Usable communication means are, for example, voice communication using a telephone line, mail, chat, and the like. In this case, the VoIP terminal 1 stores in advance information in which each communication unit is associated with a network state estimated to be usable by each communication unit. The network monitoring unit 16 selects an available communication means according to the estimated network status. The network monitor unit 16 displays on the dial screen 131 a display for selecting a communication means that is estimated to be usable, and prompts the user to input a selection. The input unit 14 receives an input of user selection. The VoIP terminal 1 executes the communication means selected by the user. Thus, by displaying the communication means that can be used according to the network status, the user can select the communication means according to the network status.

FIG. 7 is a diagram illustrating a specific example of a screen during a call.
Reference numeral 132 in the figure represents a screen during a call. The on-call screen 132 is a screen displayed when the VoIP terminal 1 is performing voice communication. The in-call screen 132 includes a communication status display area 1321, a call destination display area 1322, and a call end button 1323.
The communication status display area 1321 displays information representing the network status estimated in the second process. In the communication status display area 1321, characters representing the network status may be displayed as shown in the figure, or may be displayed in another manner. For example, in the communication status display area 1321, a graphic having a color corresponding to the network status may be displayed, or a score indicating the network status may be displayed. Further, for example, the score indicating the network status may be displayed as a numerical value, or may be expressed as the number or size of the displayed graphic.

The called party display area 1322 is an area in which the telephone number of the opposite terminal during voice communication with the own apparatus is displayed.
The call end button 1323 is a button for the user to input an operation to end voice communication. When the call end button 1323 is pressed, the voice communication control unit 17 displays the dial screen 131 on the display unit 13.
Note that other information not shown in the figure may be displayed on the call screen 132. For example, information such as call time and call charge may be displayed.

FIG. 8 is a sequence diagram illustrating a flow of the first process in the voice communication system according to the embodiment.
First, the user operates the input unit 14 to input an operation for executing a VoIP application to the VoIP terminal 1. The VoIP terminal 1 accepts user input (step S101). The VoIP terminal 1 reads the VoIP application from the storage unit 11 and executes it (Step S102). The voice communication control unit 17 displays a dial screen on the display unit 13 (step S103). The voice communication control unit 17 transmits the first packet to the response server 7 (step S104).

  The base station 3 receives the first packet transmitted from the VoIP terminal 1 (step S105). The base station 3 allocates a communication band corresponding to the communication amount generated by the first packet to the VoIP terminal 1 (step S106). In step S104, the VoIP terminal 1 transmits the first packet so that a communication amount sufficient to allocate a communication band necessary for voice communication by VoIP is generated. Therefore, in step S106, the base station 3 allocates a communication band required for VoIP voice communication to the VoIP terminal 1. The base station 3 relays the received first packet to the response server 7 (step S107).

  The response server 7 receives the first packet relayed by the base station 3 (step S108). The response server 7 generates a response packet for each received first packet and transmits it to the VoIP terminal 1 (step S109).

  The base station 3 receives the response packet transmitted from the response server 7 (step S110). The base station 3 relays the received response packet to the VoIP terminal 1 (step S111).

The VoIP terminal 1 receives the response packet relayed by the base station 3 (step S112). The network monitor unit 16 of the VoIP terminal 1 estimates the network status based on the response packet reception status (step S113). The network monitor unit 16 displays the network status on the display unit 13 (step S114).
A plurality of first packets are transmitted as in the operation example of FIG. Therefore, the VoIP terminal 1 receives a plurality of response packets. Each time a response packet is received, the network monitoring unit 16 may appropriately estimate the network status based on the reception status of the response packets received so far. Further, the network monitor unit 16 may estimate the network status every time a predetermined number of response packets are received. Further, the network monitor unit 16 may estimate the network status based on the response packet received within a predetermined time.

FIG. 9 is a sequence diagram illustrating a flow of the second process in the voice communication system according to the embodiment.
First, the user inputs the telephone number of the opposite terminal as the call destination from the dial screen. The VoIP terminal 1 receives an input of a telephone number (step S201). Next, the user inputs a call operation from the dial screen to start a call. The VoIP terminal 1 receives an input of a call operation (step S202). The voice communication control unit 17 of the VoIP terminal 1 transmits a call signal to the opposite terminal using the input telephone number (step S203). When the opposite terminal receives the outgoing signal, voice communication between the VoIP terminal 1 and the opposite terminal is started. When voice communication is started, voice data is transmitted and received between the VoIP terminal 1 and the opposite terminal.

  The voice communication control unit 17 transmits the second packet to the response server 7 during voice communication (step S204). The response server 7 generates a response packet for each received second packet and transmits it to the VoIP terminal 1 (step S205). The network monitor unit 16 of the VoIP terminal 1 estimates the network status based on the response packet reception status (step S206). The network monitor unit 16 displays the estimated network status on the call screen (step S207). The VoIP terminal 1 transmits the second packet at a predetermined timing until the voice communication is completed, and repeatedly executes the above-described network state estimation.

  When the conversation with the call partner is finished, the user inputs an operation for ending the call from the dial screen. The VoIP terminal 1 receives an input of a call end operation (step S208). The voice communication control unit 17 of the VoIP terminal 1 transmits an end signal for ending the voice communication to the opposite terminal (step S209). When the opposite terminal receives the end signal, the VoIP terminal 1 ends the voice communication with the opposite terminal (step S210).

  The VoIP terminal 1 of the embodiment configured as described above transmits the first packet to the response server 7 before starting the voice communication. By transmitting the first packet, the base station 3 assigns a communication band necessary for the VoIP terminal 1 to perform VoIP communication. Therefore, communication at the initial stage of the call is more stable, and the VoIP terminal 1 can perform voice communication more stably.

  The VoIP terminal 1 estimates the network status based on the reception status of the response packet transmitted from the response server 7. Therefore, the user can determine whether or not to make a call in consideration of the network status. The VoIP terminal 1 can perform voice communication more stably when the user selects a timing when the network condition is good and makes a call.

<Modification>
The VoIP terminal 1 may change the transmission amount of the first packet by changing the size of the first packet or the transmission interval while continuously transmitting the first packet.
For example, when the RTT is large or when a sufficient communication band is not secured, the VoIP terminal 1 may increase the size of the first packet. In this case, the VoIP terminal 1 may shorten the transmission interval of the first packet. The VoIP terminal 1 may increase the transmission amount of the first packet by changing the size or transmission interval of the first packet as described above. Further, the VoIP terminal 1 may reduce the size of the first packet when the RTT is small or when a sufficient communication band is secured. In this case, the VoIP terminal 1 may increase the transmission interval of the first packet. The VoIP terminal 1 may reduce the transmission amount of the first packet by changing the size or transmission interval of the first packet as described above. Thus, by changing the size of the first packet, the VoIP terminal 1 can secure a communication band according to the network status. Note that the VoIP terminal 1 may increase the transmission amount of the first packet so as to further secure the communication band even when the network condition is good.
In addition, the transmission amount of the first packet for allocating a communication band necessary for voice communication varies depending on the type of carrier or network. In addition, there is a correlation between the RTT and the transmission amount of the first packet for allocating a communication band necessary for voice communication. Based on these facts, the VoIP terminal 1 may be configured as follows.
Information representing the relationship between the transmission amount of the first packet for allocating a communication band necessary for voice communication and the type of carrier or network is stored in advance. The VoIP terminal 1 determines the transmission amount of the first packet for allocating a communication band necessary for voice communication based on the type of carrier and network to which the VoIP terminal 1 is connected and the RTT. The VoIP terminal 1 may determine the size or transmission interval of the first packet according to the determined transmission amount of the first packet.

  RTT correlates with communication fluctuations (network conditions). Further, the relationship between the RTT and the network status varies depending on the position of the VoIP terminal 1. Therefore, the network status may be estimated based on the position of the VoIP terminal 1 and the RTT. In that case, the VoIP terminal 1 may store in advance information representing the relationship between the location, the RTT, and the network status, or may acquire the information from another device. In addition, the VoIP terminal 1 may store the network state estimated from the transmission of the second packet in association with the position and the RTT.

  In addition, the VoIP terminal 1 obtains the RTT necessary for keeping the call quality within the allowable range at the position of the own device based on the information indicating the relationship between the position of the VoIP terminal 1, the RTT, and the network status. Then, the size or transmission interval of the first packet may be changed based on the acquired RTT.

  The voice communication control unit 17 of the VoIP terminal 1 may determine the transmission amount of the first packet based on information held in the call quality table 111, radio wave intensity, and the like. For example, when the call quality table 111 is registered as in the example of FIG. 4, the VoIP terminal 1 has a voice communication call OK rate at the time “14:00” at the location indicated by the position information “A”. Judge that it is “0%”. In this case, the voice communication control unit 17 may be configured to determine that the state of the network is in a state where voice communication cannot be performed, and not to transmit the first packet. Similarly, when the radio field intensity is equal to or lower than a predetermined threshold, the voice communication control unit 17 determines that voice communication cannot be performed with sufficient call quality and does not transmit the first packet. May be. Also, for example, the VoIP terminal 1 determines that the rate of voice communication call OK performed at the time “10:00” at the location indicated by the position information “A” is “100%”. In this case, the voice communication control unit 17 may be configured to transmit the first packet with a smaller transmission amount. In the transmission of the first packet, the above-described control is performed, so that the VoIP terminal 1 can effectively use the resources of the system related to voice communication. For example, it is possible to suppress the network from being congested due to the transmission of the first packet. In addition, for example, the VoIP terminal 1 can suppress battery consumption of its own device by transmitting an appropriate amount of the first packet.

  Furthermore, the VoIP terminal 1 may be configured to display the above-mentioned call OK rate and call NG rate on the display unit 13. By displaying these pieces of information on the display unit 13, the user can more appropriately select whether or not to perform voice communication.

  The call quality table 111 may be configured not only to store information prepared in advance but also to store information generated by the VoIP terminal 1 itself. In this case, the VoIP terminal 1 acquires information on the quality of voice communication based on the network state estimated in the first process and the second process, and the location and time at which the network state is estimated. The VoIP terminal 1 may be configured to generate a call quality record based on the acquired information and register it in the call quality table 111.

  Note that the use of information held in the call quality table 111 is not limited to the above-described use for determining the transmission amount of the first packet. Information held in the call quality table 111 may be used for any process as long as it contributes to the improvement of the quality of voice communication. For example, the VoIP terminal 1 may weight the estimated network status using information stored in the call quality table 111. By this weighting, the VoIP terminal 1 can more accurately estimate the network status.

  Further, the information held in the call quality table 111 may be information transmitted from another device. For example, the VoIP terminal 1 transmits the estimated network status to the response server 7. The response server 7 compares the information transmitted from each VoIP terminal 1 to optimize the relationship among the position information, time information, call OK rate, and call NG rate. The response server 7 feeds back the optimized information to each VoIP terminal 1 so that each VoIP terminal 1 can estimate the network status with higher accuracy. When the response server 7 feeds back the above information to the VoIP terminal 1, the response server 7 may transmit the information embedded in the header portion or payload of the response packet. Information is fed back by a response packet, thereby suppressing an increase in network traffic.

For example, when the response server 7 receives the first packet or the second packet, the response server 7 selects information on the call OK rate or the call NG rate corresponding to the VoIP terminal 1 that is the transmission source of the packet from the optimized information. The response server 7 transmits the selected information to the VoIP terminal 1 that is the transmission source of the packet.
For example, the response server 7 may hold the number of VoIP terminals 1 located in a predetermined range as the terminal number information. In this case, the VoIP terminal 1 transmits the first packet or the second packet together with the position information of the own device. When the response server 7 receives the first packet or the second packet, the response server 7 transmits the number-of-terminals information in the range where the VoIP terminal 1 is located to the VoIP terminal 1 that is the transmission source of the packet. By transmitting the terminal number information in the range where the VoIP terminal 1 is located to the VoIP terminal 1, the user of the VoIP terminal 1 can grasp how many VoIP terminals 1 are communicating within the same range. It becomes possible.

The network monitor unit 16 may estimate the network status based on the above terminal number information. For example, the network monitor unit 16 may adjust the number of RTTs used for network state estimation according to the acquired terminal number information. For example, in a place and time zone where the number of terminals is large and a change in the network status is large, the network monitoring unit 16 increases the number of RTTs used for estimating the network status. As a result, a change in the network status is easily reflected in the estimation result of the network status.
The voice communication control unit 17 may control the transmission amount of the first packet based on the terminal number information. For example, in a place and time zone where the number of terminals is large and the network condition is not good, the voice communication control unit 17 reduces the size of the first packet or lengthens the transmission interval of the first packet. As a result, the transmission amount of the first packet transmitted per unit time is reduced, and further deterioration of the network situation due to the transmission of the first packet is suppressed.

  The VoIP terminal 1 may display the network status for each location on the display unit 13. For example, the VoIP terminal 1 stores map information in advance, and displays a map near the position of the own device on the display unit 13. When the response server 7 receives the first packet or the second packet, the response server 7 transmits the status of the network around the place where the VoIP terminal 1 is located to the VoIP terminal 1 that is the transmission source of the packet. The VoIP terminal 1 colors the map displayed on the display unit 13 with a color according to the network status. . Thus, by displaying the map representing the network status, the user of the VoIP terminal 1 can determine to which location the network status will be favorable.

The VoIP terminal 1 may be configured to transmit a transmission signal after transmitting the first packet when an operation for starting voice communication is input. By configuring the VoIP terminal 1 in this way, a user can start a call after a communication band necessary for voice communication is secured, regardless of the timing of inputting an operation for starting voice communication. It becomes.
In addition, when the VoIP terminal 1 is an opposite terminal, the VoIP terminal 1 may be configured to start voice communication after transmitting the first packet when receiving a transmission signal. By configuring the VoIP terminal 1 in this way, the user can start a call after a communication band necessary for voice communication is ensured regardless of the timing of inputting an off-hook operation.
In addition, when the own device is the opposite terminal, the VoIP terminal 1 may be configured to transmit the first packet after receiving the outgoing signal and before notifying the user of the incoming call. By configuring the VoIP terminal 1 in this way, the user can start a call after a communication band necessary for voice communication is secured.

The voice communication control unit 17 may be configured as a functional unit having the function of the network monitor unit 16. By integrating this function, the VoIP terminal 1 may be configured as an apparatus that does not include the network monitor unit 16.
Further, the VoIP terminal 1 may transmit the first packet to a device other than the response server 7. For example, the VoIP terminal 1 may transmit the first packet to a device such as a DNS server or a WEB server.
The VoIP terminal 1 may be configured to transmit a first packet by designating a fictitious destination. The VoIP terminal 1 may be configured to transmit a first packet that does not designate a specific destination. For example, the VoIP terminal 1 may transmit the first packet by broadcast. With this configuration, the VoIP terminal 1 can secure a communication band without requiring the response server 7.

  You may make it implement | achieve the VoIP terminal 1 in embodiment mentioned above with a computer. In that case, a program for realizing this function may be recorded on a computer-readable recording medium, and the program recorded on this recording medium may be read into a computer system and executed. Here, the “computer system” includes an OS and hardware such as peripheral devices. The “computer-readable recording medium” refers to a storage device such as a flexible medium, a magneto-optical disk, a portable medium such as a ROM and a CD-ROM, and a hard disk incorporated in a computer system. Furthermore, the “computer-readable recording medium” dynamically holds a program for a short time like a communication line when transmitting a program via a network such as the Internet or a communication line such as a telephone line. In this case, a volatile memory inside a computer system serving as a server or a client in that case may be included and a program held for a certain period of time. Further, the program may be a program for realizing a part of the above-described functions, and may be a program capable of realizing the functions described above in combination with a program already recorded in a computer system. You may implement | achieve using programmable logic devices, such as FPGA (Field Programmable Gate Array).

  The embodiment of the present invention has been described in detail with reference to the drawings. However, the specific configuration is not limited to this embodiment, and includes designs and the like that do not depart from the gist of the present invention.

DESCRIPTION OF SYMBOLS 1, 1-1 to 1-3 ... VoIP terminal, 11 ... Memory | storage part, 111 ... Call quality table, 12 ... Communication part, 13 ... Display part, 131 ... Dial screen, 1311 ... Communication status display area, 1312 ... Call destination Display area, 1313 ... Dial display area, 1314 ... Call start button, 132 ... In-call screen, 1321 ... Communication status display area, 1322 ... Call destination display area, 1323 ... Call end button, 14 ... Input section, 15 ... Voice input Output unit 16 Network monitor unit 17 Voice communication control unit 2 VoIP telephone 3, 3-1 to 3-3 base station 4 operator IP network 5 other provider IP network 6 ... WAN (Wide Area Network), 7 ... Response server

Claims (11)

A communication terminal that performs voice communication by VoIP via a communication band allocation device that allocates a communication band according to a communication amount to a communication terminal,
A voice communication control unit that transmits the first packet before starting voice communication ;
The voice communication control unit transmits the first packet so that a communication amount sufficient to allocate a communication band necessary for voice communication is generated.
Communication terminal. VoI via a communication band allocation device that allocates a communication band corresponding to the communication volume to a communication terminal
A communication terminal for performing voice communication by P,
A voice communication control unit that transmits the first packet before starting voice communication;
The voice communication control unit determines an amount to transmit the first packet based on time and location information in which a time, a location, and information indicating a network state at the time of the location are associated with each other. To
Communication terminal. The voice communication control unit continuously transmits the first packet;
The communication terminal according to claim 1 or 2. A network monitoring unit for estimating a network state at a time point before starting voice communication based on a reception state of the first response packet received for each of the first packets;
The communication terminal as described in any one of Claim 1 to 3 . The voice communication control unit transmits a second packet during voice communication,
The network monitoring unit estimates a network status during voice communication based on a reception status of a second response packet received for each second packet;
The communication terminal according to claim 4 . A display unit for displaying an operation screen for a user to operate the communication terminal;
The network monitor unit causes the display unit to display information representing a network status at a time before starting voice communication.
The communication terminal according to claim 4 or 5 . The network monitor unit causes the display unit to display information indicating a network status during voice communication.
The communication terminal according to claim 6 . The network monitor unit accumulates one or both of the status of the network at the time before starting voice communication and the status of the network during voice communication as the time and location information.
The communication terminal according to any one of claims 4 to 7 . A voice communication control method performed by a communication terminal that performs voice communication by VoIP via a communication band allocation device that allocates a communication band according to a communication amount to a communication terminal,
The first packet have a voice communication control step of transmitting before starting the voice communication,
In the voice communication control step, the first packet is transmitted so that a communication amount sufficient to allocate a communication band necessary for voice communication is generated.
Voice communication control method. A voice communication control method performed by a communication terminal that performs voice communication by VoIP via a communication band allocation device that allocates a communication band according to a communication amount to a communication terminal,
A voice communication control step of transmitting the first packet before starting voice communication;
In the voice communication control step, the amount of the first packet to be transmitted is determined based on the time and location information in which the time, the location, and information indicating the network status at the time of the location are associated with each other. To
Voice communication control method.   The computer program for functioning a computer as a communication terminal as described in any one of Claim 1 to 8.

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