Voice quality information collection device, voice quality information collection method, voice quality information collection system, and program

Abstract

This speech quality information gathering device (10) comprises: an SIP server log gathering unit (111) for gathering SIP server logs from an SIP server (2), each of the SIP server logs including an incoming telephone number, identification information pertaining to an operator making an incoming call, and a line number; an MG log gathering unit (112) for gathering MG logs from an MG (3), each of the MG logs including a TerminationID allocated according to a prescribed logic, and a delay time calculated using an RTCP report; and a data generation unit (12) for converting the line number included in each SIP server log to a TerminationID, integrating the converted SIP server logs and MG logs using the TerminationIDs, and generating log integration information.

Description

Technical Field

[0001] The present invention relates to a voice quality information collection device, a voice quality information collection method, a voice quality information collection system, and a program for collecting voice quality information in an IP telephone network.

Background Art

[0002] When so-called PSTN migration (IP conversion of fixed telephones), which is the migration of the PSTN (Public Switched Telephone Network) to an IP network, is completed, metal IP telephones are started, and direct IP connection is started between different company networks without going through the PSTN.

[0003] In a conventional public IP telephone network, in order to ensure a specified voice quality, test calls are continuously made to each area using a test terminal, and the voice quality is measured (see Non-Patent Document 1). The quality measurement method at this time is carried out in accordance with the guidelines of the TTC (Telecommunications Technology Committee) (see Non-Patent Document 2).

Prior Art Documents

Non-Patent Documents

[0004]

Non-Patent Document 1

Non-Patent Document 2

Summary of the Invention

Problems to be Solved by the Invention

[0005] However, when measuring the quality of an IP telephone network according to the prior art described in Non-Patent Document 1, as shown in FIG. 6, it is necessary to provide a test terminal 8 for measurement and a measuring instrument 9 at the originating side and the terminating side. Therefore, equipment costs and operation costs were incurred during quality measurement. In addition, since measurement is performed after providing the test terminal 8 for measurement and the measuring instrument 9, when a user reports a decrease in voice quality or the like, it was not possible to obtain the voice quality information about the call actually being used by the user.

[0006] Furthermore, in this quality measurement method, since test terminals are prepared on both the originating side and the terminating side, as shown in FIG. 7, when telephone services are provided by connecting IP networks to each other through IP interconnection without passing through the PSTN, even when trying to measure the quality of the telephone service connected to another company's network, it is not possible to install a test terminal on the other company's network side, and quality measurement could not be performed. As a result, when a voice quality problem occurred in a telephone call with another company's network, it was difficult to determine whether the cause of the failure was due to the own network or the other company's network.

[0007] In view of such points, the present invention has been made, and an object of the present invention is to reduce equipment costs and operation costs related to voice quality measurement of an IP telephone network, and to obtain voice quality information for each call being used by a user.

Means for Solving the Problems

[0008] To solve the above problems, a voice quality information collection device according to the present invention is a voice quality information collection device that collects voice quality information of a telephone service in which the IP telephone network of the originating carrier and the IP telephone network of the terminating carrier are IP interconnected. The originating terminal of a telephone call is connected to an MG (Media Gateway) that converts a metallic line and an IP line via a metallic accommodation device. The voice quality information collection device collects a SIP server log, which is a log including at least a called telephone number of a telephone call, identification information of the terminating carrier, and a line number for uniquely identifying the telephone call, from a SIP (Session Initiation Protocol) server belonging to the area where the originating terminal is accommodated. An MG log collection unit collects an MG log, which is a log including at least a start time and an end time of the telephone call, a TerminationID allocated for each line using a predetermined logic set in the MG, and a delay time of the telephone call calculated by the MG using an RTCP (Real Time Control Protocol) report, from the MG belonging to the area. A data generation unit generates converted SIP server log indicating the SIP server log in which the line number included in the SIP server log is converted into the TerminationID using the predetermined logic, and generates log integration information including the start time and the end time, the called telephone number, the identification information of the terminating carrier, and the delay time for the telephone call by associating the TerminationID included in the generated converted SIP server log with the TerminationID included in the MG log. It is characterized by comprising these components.

Effect of the Invention

[0009] According to the present invention, it is possible to reduce the equipment cost and operation cost related to the voice quality measurement of the IP telephone network, and to obtain voice quality information in units of calls being used by users.

Brief Description of the Drawings

[0010]

Figure 1

Figure 2

Figure 3

Figure 4

Figure 5

Figure 6

Figure 7

Embodiments for Carrying Out the Invention

[0011] Next, embodiments for carrying out the present invention (hereinafter referred to as "this embodiment") will be described. FIG. 1 is a diagram showing the overall configuration of a voice quality information collection system 1 including a voice quality information collection device 10 according to this embodiment.

[0012] As shown in FIG. 1, in this embodiment, it is assumed that PSTN migration (IP conversion of fixed telephones) has been completed, metal IP telephones have been started, and direct IP connection has been started between the IP telephone networks of other companies (hereinafter sometimes referred to as "other company networks 600") without going through the PSTN. When migrating from the PSTN to the IP network, in the case of a conventional landline phone on the PSTN line, as shown by terminal 5a, after utilizing the conventional subscriber switch as the metal housing device 4, the metal line is used as it is. The landline phone (terminal 5a) is connected to the company's (Company A) IP phone network 100 via an MG (Media Gateway) 3 that converts the metal line and the IP line. Within the IP phone network 100, the SIP (Session Initiation Protocol) server 2 performs call control such as incoming and outgoing calls and call control, and provides communication services to IP phones (terminals 5b, 5c) and metal IP phones (terminal 5d) within the company's IP phone network 100, as well as providing communication services to IP phones (terminal 5e) and metal IP phones (not shown) on other company networks 600.

[0013] Note that the area where landline phones (terminals 5) such as metal IP phones and IP phones are installed has multiple areas (in FIG. 1, area X and area Y) set. And within that area, one or more SIP servers 2 and one or more MGs 3 are arranged. Also, the MG 3 is communicatively connected to one or more metal housing devices 4, and the metal housing device 4 is connected to a plurality of metal IP phones by metal lines.

[0014] The voice quality information collection device 10 according to this embodiment is communicatively connected to the SIP server 2 and the MG 3 within the company's IP phone network 100. The voice quality information collection device 10 collects information on voice quality within the IP phone network including other company networks 600 by collecting information on delay measurement using RTCP (Real Time Control Protocol) handled by the MG 3 within the IP phone network 100.

[0015] RTCP is a protocol that performs transmission and reception control when real-time transmitting voice data etc. using RTP (Real-time Transport Protocol) etc., and notifies a report that reports the quality of the data. The voice quality information collection device 10 according to this embodiment uses information on the RTT (Round Trip Time) delay time (hereinafter sometimes referred to as "RTT delay information" or "delay time") that can be created using the RTCP report received by MG3, which is the end point of RTCP, to collect voice quality information within the IP telephone network including the other company network 600.

[0016] <Voice quality information collection device> The voice quality information collection device 10 according to this embodiment combines the call unit RTT delay information included in the logs collected from MG3 and the called phone number and called carrier information included in the logs collected from the SIP server, and collects the delay information of voice data in units of areas within its own network and units of other carriers. This voice quality information collection device 10 is configured by a computer including a control unit, an input / output unit, and a storage unit (all not shown in the figure).

[0017] The input / output unit inputs and outputs information regarding each SIP server 2 within its own network and information between each MG3 and the like. This input / output unit is composed of a communication interface for transmitting and receiving information via a communication line and an input / output interface for inputting and outputting information between an input device such as a keyboard (not shown) and an output device such as a monitor.

[0018] The storage unit is composed of a hard disk, a flash memory, a RAM (Random Access Memory), etc. In this storage unit, a program for causing each function of the control unit to execute and information necessary for the processing of the control unit are temporarily stored.

[0019] The control unit manages all the processes executed by the voice quality information collection device 10, and as shown in FIG. 1, includes a log collection unit 11, a data generation unit 12, and a quality information generation unit 13.

[0020] The log collection unit 11 collects logs related to telephone calls from the SIP server 2 and MG3 within its own network. This log collection unit 11 is configured to include a SIP server log collection unit 111 and an MG log collection unit 112.

[0021] The SIP server log collection unit 111 collects logs related to telephone calls from the SIP server 2 (in FIG. 1, for example, the SIP server 2a in area X) belonging to the area to be measured for voice quality. Hereinafter, the logs collected from the SIP server 2 are referred to as "SIP server logs 20".

[0022] As shown in FIG. 2, this SIP server log 20 includes, as information on each telephone call, start time, end time, incoming phone number (described as "incoming number" in FIG. 2. The same applies hereinafter), identification information of the incoming carrier (described as "incoming carrier" in FIG. 2. The same applies hereinafter), and line information (line number: CIC).

[0023] The start time and the end time are the times when the communication connection of the telephone call starts and ends. The incoming phone number is the phone number of the terminal 5 on the receiving side. The identification information of the incoming carrier (incoming carrier) is information obtained from the IP address for identifying the carrier of the IP telephone network that houses the terminal 5 on the receiving side. The line information (line number: CIC) is a line number (CIC: Circuit Identification Code), which is information for uniquely identifying a telephone call.

[0024] The voice quality information collection device 10 may send a request to obtain the SIP server log 20 to the SIP server 2 belonging to the area to be measured for voice quality, and collect it from the SIP server 2. Alternatively, the SIP server 2 may be set to send the SIP server log 20 at predetermined intervals set in advance, and the voice quality information collection device 10 may be set to receive it.

[0025] Returning to FIG. 1, the MG log collection unit 112 collects logs related to telephone calls from MG3s (e.g., MG3a and MG3b in FIG. 1) belonging to the same area as the SIP server 2 that is the measurement target of voice quality. Hereinafter, the logs collected from MG3 are referred to as "MG logs 30". In FIG. 2, MG log 30a collected from MG3a in area X and MG log 30b collected from MG3b in the same area X are shown. As shown in FIG. 2, this MG log 30 (30a, 30b) includes, as information for each telephone call, start time, end time, line information (TerminationID), and delay time.

[0026] The start time and end time are the times when the communication connection of the telephone call starts and ends. The line information (TerminationID) is an identification number assigned by MG3 for each line. The delay time is information on the RTT delay time (RTT delay information) calculated by each MG3 for each call from the RTCP report.

[0027] The calculation of the RTT delay time by MG3 will be described. MG3 calculates the RTT delay time based on the report obtained by RTCP, which is the control protocol of RTP. Specifically, MG3 transmits RTCP_SR (Sender Report) to the receiving-side MG3 or IP phone (terminal 5), and stores the transmission time. On the receiver side, the time (T) from receiving RTCP_SR until sending RTCT_RR (Receiver Report) is measured, and the measured time (T) is described in RTCT_RR and responded to the transmitting-side MG3. The transmitting-side MG3 calculates the difference between the time when RTCP_SR was transmitted and the time when RTCT_RR was received, and then subtracts the time (T) described in RTCT_RR to calculate the RTT delay time.

[0028] The voice quality information collection device 10 may send a request to obtain the MG log 30 to the MG3 belonging to the same area as the SIP server 2 that is the measurement target of the voice quality, and collect it from each MG3. Alternatively, the MG log 30 may be set to be transmitted from each MG3 at predetermined intervals set in advance, and the voice quality information collection device 10 may receive it.

[0029] Returning to FIG. 1, the data generation unit 12 associates the SIP server log 20 and the MG log 30 with line information, and generates call unit log integration information 40 (see FIG. 2). This data generation unit 12 includes a SIP server log conversion unit 121, an MG log aggregation unit 122, and a log integration unit 123.

[0030] The SIP server log conversion unit 121 converts the line information (line number: CIC) included in the SIP server log 20 into the TerminationID, which is the format of the line information described in the MG log 30. Note that the SIP server log after converting the line information into the TerminationID is referred to as the "converted SIP server log 21".

[0031] The SIP server log conversion unit 121 uses the same logic as the predetermined logic (TerminationID generation logic) used for generating the TerminationID in the MG3 to convert the line number (CIC) into the TerminationID and generate the converted SIP server log 21 (see FIG. 2). This TerminationID generation logic can be arbitrarily set by the operators of each IP telephone network. The SIP server log conversion unit 121 generates the TerminationID as information that can uniquely identify a line using the same predetermined logic as the MG3, for example, by using the voice channel information of the call path used in the line number (CIC).

[0032] Returning to FIG. 1, when there are a plurality of MG3s in the area to be measured for voice quality, the MG log aggregation unit 122 aggregates a plurality of MG logs 30 collected from each MG3 (in FIG. 2, MG logs 30a and 30b). Note that the MG log after aggregating the plurality of MG logs 30 is referred to as the "aggregated MG log 31". When there is one MG3, the MG log 30 directly becomes the aggregated MG log 31.

[0033] Specifically, for a plurality of MG logs 30, the MG log aggregation unit 122 integrates and sorts the logs in time series based on the call start time, and generates an aggregated MG log 31 (see FIG. 2). The reason why the MG log aggregation unit 122 sorts the MG logs 30 in time series is that when using voice channel information for generating the TerminationID, due to the limitation of the number of voice channels, the same TerminationID may be generated after a certain period of time. By using time information, even when there are phone calls with the same TerminationID, the call start times are different, so that the phone calls can be uniquely identified.

[0034] Returning to FIG. 1, the log integration unit 123 generates log integration information 40 by integrating the converted SIP server log 21 in which the line information is converted into a TerminationID by the SIP server log conversion unit 121 and the aggregated MG log 31 in which a plurality of MG logs 30 are aggregated by the MG log aggregation unit 122, based on the TerminationID.

[0035] Specifically, the log integration unit 123 uses the TerminationID, which is information common to the converted SIP server log 21 and the aggregated MG log 31, to integrate the two logs and generate log integration information 40 (see FIG. 2). As shown in FIG. 2, the log integration information 40 includes information on the start time, end time, incoming phone number (incoming number), incoming carrier, and delay time as information on each phone call. By doing so, the voice quality information collection device 10 can acquire voice quality information in units of calls that the user is actually using.

[0036] Returning to FIG. 1, the quality information generation unit 13 uses the log integration information 40 generated by the log integration unit 123 to group telephone calls in units of arrival areas and arrival carriers from the information on the incoming telephone number and the incoming carrier indicated by the log integration information 40, aggregate the information on the delay time, and generate quality information 50 (see FIG. 3). Here, the quality information generation unit 13 obtains information on the area defined by the 0ABJ number starting with the outside area code by searching for any number of digits from 1 digit to all digits for the incoming telephone number for a forward match. Also, the quality information generation unit 13 identifies the incoming carrier based on the identification information of the incoming carrier obtained from the IP address.

[0037] FIG. 3 is a diagram for explaining the generation process of the quality information 50 executed by the quality information generation unit 13 of the voice quality information collection device 10 according to the present embodiment. As indicated by the symbol α in FIG. 3, the quality information generation unit 13 identifies "Area X", which is an area such as the 23 wards of Tokyo, based on the incoming telephone number "03". Also, based on the incoming carrier "a.jp", it identifies "Company A", which is its own company. Then, the delay time of the incoming carrier in this arrival area is set to "5 ms". Also, as indicated by the symbol β in FIG. 3, based on the incoming telephone number "0422", it identifies "Area Y", which is an area such as Musashino City, Tokyo. Also, based on the incoming carrier "a.jp", it identifies "Company A", which is its own company. Then, as the delay time of the incoming carrier in this arrival area, for example, the two delay information "10 ms" and "12 ms" are averaged to "11 ms". Also, as indicated by the symbol γ in FIG. 3, based on the incoming telephone number "06", it identifies "Area Z", which is an area of Osaka City, Osaka Prefecture. Also, based on the incoming carrier "b.jp", it identifies "Company B", which is another company. Then, the delay time of the incoming carrier in this arrival area is set to "40 ms".

[0038] The quality information generation unit 13 groups the telephone calls in the log integration information 40 by the arrival area unit and the arrival business operator unit, and aggregates the delay time of the telephone calls common to the arrival area and the arrival business operator based on a predetermined aggregation logic. For example, the quality information generation unit 13 can calculate information such as the average value, maximum value, minimum value, and variance of the delay time based on the delay time of the quality information 50, and generate the quality information 50 (see FIG. 3).

[0039] In addition, when the arrival-side terminal 5 is not a metal IP phone (landline phone) and the area cannot be specified because the phone number does not conform to the area defined by the 0ABJ number, the quality information generation unit 13 uses the information of the identifiable arrival business operator (for example, "Company B") to generate the delay time information for the entire operator without specifying the area.

[0040] In the quality information 50 shown in FIG. 3, the delay time of Company B is "40 ms", compared with the delay times of Company A being "5 ms" and "11 ms". Therefore, it can be seen that there is no problem with the voice quality of the IP phone network on the side of Company A, and there is a problem with the voice quality (delay) in the IP phone network on the side of Company B.

[0041] Thus, according to the voice quality information collection system 1 including the voice quality information collection device 10 according to this embodiment, when acquiring the quality information of the IP phone network, it is not necessary to provide a test terminal and a measuring instrument. Therefore, the equipment cost and operation cost related to voice quality measurement can be reduced. In addition, the voice quality information collection device 10 can acquire voice quality information in units of calls being used by the user. Furthermore, since the voice quality information collection device 10 can acquire information regarding the voice quality in a telephone call with another company's network, it is possible to compare the voice quality information of a telephone call under an IP connection with another company's network and the voice quality information of a telephone call within its own network. This facilitates the isolation when a voice quality trouble occurs and makes it possible to shorten the time for problem solving.

[0042] ≪Flow of processing≫ Next, the process flow executed by the voice quality information collection system 1 including the voice quality information collection device 10 according to this embodiment will be described. FIG. 4 is a sequence diagram showing the process flow executed by the voice quality information collection system 1 including the voice quality information collection device 10.

[0043] First, the SIP server 2 belonging to a certain area generates a SIP server log 20 (see FIG. 2) (step S1) and transmits it to the voice quality information collection device 10. Then, the log collection unit 11 (SIP server log collection unit 111) of the voice quality information collection device 10 collects the SIP server log 20 (step S2). This SIP server log 20 includes at least the incoming call number, the identification information of the incoming carrier, and the information of the line number (CIC) as the line information.

[0044] Also, one or more MG3s belonging to the same area as the area to which the SIP server 2 belongs generate an MG log 30 (see FIG. 2) (step S3). At this time, MG3 calculates the RTT delay time (delay time) using the RTCP report. This MG log 30 includes at least the start time, the end time, the delay time, and the information of the TerminationID assigned by MG3 as the line information. MG3 transmits the generated MG log 30 to the voice quality information collection device 10. Then, the log collection unit 11 (MG log collection unit 112) of the voice quality information collection device 10 collects the MG log 30 (step S4).

[0045] Regarding the process of collecting the SIP server log 20 in steps S1 and S2 and the process of collecting the MG log 30 in steps S3 and S4, the order does not matter.

[0046] The data generation unit 12 (SIP server log conversion unit 121) of the voice quality information collection device 10 converts the line information (line number: CIC) included in the SIP server log 20 into TerminationID, which is the format of the line information described in the MG log 30, based on a predetermined logic, and generates a converted SIP server log 21 (see Figure 2) (step S5).

[0047] Also, the MG log aggregation unit 122 of the voice quality information collection device 10 aggregates a plurality of MG logs 30 collected from each MG3 and generates an aggregated MG log 31 (see Figure 2) (step S6). The order of the processes in step S5 and step S6 does not matter.

[0048] Next, the data generation unit 12 (log integration unit 123) of the voice quality information collection device 10 integrates the converted SIP server log 21 and the aggregated MG log 31 by TerminationID and generates log integration information 40 (see Figure 2) (step S7).

[0049] Subsequently, the quality information generation unit 13 of the voice quality information collection device 10 groups the telephone calls in units of the called area and the called carrier based on the called telephone number and the information of the called carrier indicated by the log integration information 40, and aggregates the information on the delay time based on a predetermined aggregation logic to generate quality information 50 (see Figure 3) (step S8). Here, the predetermined aggregation logic is set in advance, for example, as the average value, maximum value, minimum value, variance, etc. of the delay time.

[0050] In this way, the voice quality information collection system 1 can collect the voice quality information of the IP telephone network without providing a test terminal or a measuring instrument, so that the equipment cost and operation cost related to voice quality measurement can be reduced. Also, the voice quality information collection device 10 can acquire voice quality information in units of calls being used by the user. Furthermore, since the voice quality information collection device 10 can acquire the quality information between the terminal 5 belonging to another company's network, it becomes possible to easily isolate the problem when the voice quality trouble occurs.

[0051] <Hardware Configuration> The voice quality information collection device 10 according to the present embodiment is realized by a computer 900 configured as shown in FIG. 5, for example. FIG. 5 is a hardware configuration diagram showing an example of a computer 900 that realizes the functions of the voice quality information collection device 10 according to the present embodiment. The computer 900 includes a CPU (Central Processing Unit) 901, a ROM (Read Only Memory) 902, a RAM 903, an HDD (Hard Disk Drive) 904, an input / output I / F (Interface) 905, a communication I / F 906, and a media I / F 907.

[0052] The CPU 901 operates based on a program stored in the ROM 902 or the HDD 904 and performs control by the control unit. The ROM 902 stores a boot program executed by the CPU 901 when the computer 900 is started up, a program related to the hardware of the computer 900, and the like.

[0053] The CPU 901 controls an input device 910 such as a mouse and a keyboard, and an output device 911 such as a display and a printer via the input / output I / F 905. The CPU 901 acquires data from the input device 910 via the input / output I / F 905 and outputs the generated data to the output device 911. Note that a GPU (Graphics Processing Unit) or the like may be used together with the CPU 901 as a processor.

[0054] The HDD 904 stores programs executed by the CPU 901 and data used by the programs. The communication I / F 906 receives data from other devices via a communication network (for example, NW (Network) 920) and outputs it to the CPU 901, and transmits data generated by the CPU 901 to other devices via the communication network.

[0055] The media I / F 907 reads the program or data stored in the recording medium 912 and outputs it to the CPU 901 via the RAM 903. The CPU 901 loads the program related to the target process from the recording medium 912 onto the RAM 903 via the media I / F 907 and executes the loaded program. The recording medium 912 is an optical recording medium such as a DVD (Digital Versatile Disc) or a PD (Phase change rewritable Disk), a magneto-optical recording medium such as an MO (Magneto Optical disk), a magnetic recording medium, a semiconductor memory, or the like.

[0056] For example, when the computer 900 functions as the voice quality information collection device 10 of the present invention, the CPU 901 of the computer 900 realizes the functions of the voice quality information collection device 10 by executing the program loaded on the RAM 903. Also, the data in the RAM 903 is stored in the HDD 904. The CPU 901 reads and executes the program related to the target process from the recording medium 912. In addition, the CPU 901 may read the program related to the target process from another device via the communication network (NW920).

[0057] <Effect> Hereinafter, the effects of the voice quality information collection device 10 and the like according to the present invention will be described. The voice quality information collection device according to the present invention is a voice quality information collection device 10 that collects voice quality information of a telephone service in which an IP telephone network 100 of a originating carrier and an IP telephone network 600 of a terminating carrier are IP interconnected. The originating terminal of a telephone call is connected to an MG3 that converts a metallic line and an IP line via a metallic housing device 4. The voice quality information collection device 10 collects a SIP server log 20, which is a log including at least a called telephone number of the telephone call, identification information of the terminating carrier, and a line number for uniquely identifying the telephone call, from a SIP server 2 belonging to the area where the originating terminal 5 is housed. An MG log collection unit 112 that collects an MG log 30, which is a log including at least a start time and an end time of the telephone call, a TerminationID assigned for each line using a predetermined logic set in the MG3, and a delay time of the telephone call calculated by the MG3 using an RTCP report, from the MG3 belonging to the area. A data generation unit 12 that generates a converted SIP server log 21 indicating the SIP server log in which the line number included in the SIP server log 20 is converted into a TerminationID using a predetermined logic, and generates log integration information 40 for the telephone call including the start time and the end time, the called telephone number, the identification information of the terminating carrier, and the delay time by associating the TerminationID included in the generated converted SIP server log 21 with the TerminationID included in the MG log 30. It is characterized by comprising.

[0058] Thus, according to the voice quality information collection device 10, when acquiring the quality information of the IP telephone network, it is not necessary to provide a test terminal or a measuring instrument as in the prior art. Therefore, it is possible to reduce the equipment cost and operation cost related to voice quality measurement. In the prior art, since the line information was calculated by different logics in the logs of the SIP server 2 and the MG3, the logs could not be integrated. In contrast, according to the voice quality information collection device 10, for each telephone call, log integration information 40 integrating the SIP server log 20 and the MG log 30 can be generated. Thereby, the voice quality information collection device 10 can acquire voice quality information in units of calls being used by the user. Therefore, when a user or the like reports a deterioration in voice quality, the voice quality of the voice call actually used by the user can be confirmed.

[0059] The voice quality information collection device according to the present invention is characterized in that it further includes a quality information generation unit 13 that identifies the arrival area indicating the area to which the arrival-side terminal belongs from the incoming telephone number, identifies the arrival carrier from the identification information of the arrival carrier, groups the log integration information 40 in units of arrival areas and arrival carriers, and aggregates the delay information of the telephone call to generate quality information.

[0060] In this way, since the voice quality information collection device 10 can acquire information regarding the voice quality in a telephone call with another company's network, it is possible to compare the voice quality information of a telephone call under an IP connection with another company's network and the voice quality information of a telephone call within its own network. Thereby, it becomes easy to distinguish when a voice quality trouble occurs, and it is possible to realize shortening of the problem-solving time.

[0061] Note that the present invention is not limited to the embodiments described above, and many modifications are possible by those having ordinary knowledge in the art within the technical idea of the present invention.

Explanation of Reference Numerals

[0062] 1 Voice Quality Information Collection System 2 SIP Server 3 MG (Media Gateway) 4 Metal Containment Device 5 Terminal (Metal IP Phone, IP Phone) 10 Voice Quality Information Collection Device 11 Log Collection Unit 12 Data generation unit 13 Quality information generation unit 20 SIP server log 21 Transformed SIP server log 30 MG log 31 Aggregated MG log 40 Log integration information 50 Quality information 111 SIP server log collection unit 112 MG log collection unit 121 SIP server log conversion unit 122 MG log aggregation unit 123 Log integration unit

Claims

1. A voice quality information collecting device that collects voice quality information of a telephone service in which the IP telephone network of the originating carrier and the IP telephone network of the terminating carrier are IP interconnected, The originating terminal of the telephone call is connected via a metal accommodating device to an MG (Media Gateway) that converts between a metal line and an IP line, The voice quality information collecting device, A SIP server log collecting unit that collects a SIP server log, which is a log including at least a terminating telephone number of a telephone call, identification information of the terminating carrier, and a line number for uniquely identifying the telephone call, from a SIP (Session Initiation Protocol) server belonging to the area where the originating terminal is accommodated, An MG log collecting unit that collects an MG log, which is a log including at least a start time and an end time of the telephone call, a TerminationID assigned for each line using a predetermined logic set in the MG, and a delay time of the telephone call calculated by the MG using an RTCP (Real Time Control Protocol) report, from the MG belonging to the area, A data generation unit that generates converted SIP server logs indicating the SIP server logs in which the line numbers included in the SIP server logs are converted into the TerminationID using the predetermined logic, and generates log integration information including the start time and end time, the terminating telephone number, the identification information of the terminating carrier, and the delay time for the telephone call by associating the TerminationID included in the generated converted SIP server logs with the TerminationID included in the MG logs, A voice quality information collecting device, characterized by comprising the above.

2. A quality information generation unit that identifies a terminating area indicating the area to which the terminating terminal belongs from the terminating telephone number, identifies the terminating carrier from the identification information of the terminating carrier, groups the log integration information in units of the terminating area and the terminating carrier, and aggregates the delay information of the telephone call to generate quality information The voice quality information collecting device according to claim 1, further characterized by comprising the above.

3. A voice quality information collecting method of a voice quality information collecting device that collects voice quality information of a telephone service in which the IP telephone network of the originating carrier and the IP telephone network of the terminating carrier are IP interconnected, The originating terminal of the phone call is connected to an MG that converts between a metallic line and an IP line via a metallic housing device. The voice quality information collection device: collects an SIP server log, which is a log containing at least the incoming phone number of the phone call, the identification information of the incoming carrier, and the line number for uniquely identifying the phone call, from an SIP server belonging to the area where the originating terminal is housed; collects an MG log, which is a log containing at least the start time and end time of the phone call, the TerminationID assigned to each line using a predetermined logic set in the MG, and the delay time of the phone call calculated by the MG using an RTCP report, from the MG belonging to the area; generates a converted SIP server log indicating the SIP server log in which the line number included in the SIP server log is converted to the TerminationID using the predetermined logic, and generates log integration information for the phone call, including the start time and end time, the incoming phone number, the identification information of the incoming carrier, and the delay time, by associating the TerminationID included in the generated converted SIP server log with the TerminationID included in the MG log; A voice quality information collection method characterized by performing the above steps.

4. The voice quality information collection device further: identifies the incoming area indicating the area to which the receiving terminal belongs from the incoming phone number, identifies the incoming carrier from the identification information of the incoming carrier, groups the log integration information in units of the incoming area and the incoming carrier, and aggregates the delay information of the phone call to generate quality information. The voice quality information collection method according to claim 3, characterized by performing the above steps.

5. A voice quality information collection system for collecting voice quality information of a phone service in which the IP telephone network of the originating carrier and the IP telephone network of the receiving carrier are IP interconnected, comprising a voice quality information collection device, an MG connected to a metallic housing device that houses the originating terminal and converts between a metallic line and an IP line, and an SIP server that controls the connection of phone calls in the IP telephone network. The SIP server belonging to the area where the originating terminal is accommodated generates a SIP server log which is a log including at least the incoming phone number of a phone call, the identification information of the incoming carrier, and the line number for uniquely identifying the phone call, and transmits it to the voice quality information collection device. The MG belonging to the area generates an MG log which is a log including at least the start time and end time of the phone call, the TerminationID assigned for each line using a predetermined logic set in the MG, and the delay time of the phone call calculated by the MG using an RTCP report, and transmits it to the voice quality information collection device. The voice quality information collection device is provided with a SIP server log collection unit, an MG log collection unit, and a data generation unit. The SIP server log collection unit collects the SIP server log. The MG log collection unit collects the MG log. The data generation unit generates a converted SIP server log indicating the SIP server log in which the line number included in the SIP server log is converted into the TerminationID using the predetermined logic, and associates the TerminationID included in the generated converted SIP server log with the TerminationID included in the MG log, thereby generating log integration information including the start time and end time, the incoming phone number, the identification information of the incoming carrier, and the delay time for the phone call. A voice quality information collection system characterized by the above.

6. The voice quality information collection device further includes a quality information generation unit. The quality information generation unit identifies the incoming area indicating the area to which the incoming terminal belongs from the incoming phone number, identifies the incoming carrier from the identification information of the incoming carrier, groups the log integration information in units of the incoming area and the incoming carrier, and aggregates the delay information of the phone call to generate quality information. The voice quality information collection system according to claim 5, characterized by the above.

7. A program for causing a computer to function as the voice quality information collection device according to claim 1 or claim 2.

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