WO2011090185A1 - 音声品質計測装置、音声品質計測方法およびプログラム - Google Patents
音声品質計測装置、音声品質計測方法およびプログラム Download PDFInfo
- Publication number
- WO2011090185A1 WO2011090185A1 PCT/JP2011/051162 JP2011051162W WO2011090185A1 WO 2011090185 A1 WO2011090185 A1 WO 2011090185A1 JP 2011051162 W JP2011051162 W JP 2011051162W WO 2011090185 A1 WO2011090185 A1 WO 2011090185A1
- Authority
- WO
- WIPO (PCT)
- Prior art keywords
- packet
- frame
- voice quality
- value
- network
- Prior art date
Links
Images
Classifications
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04W—WIRELESS COMMUNICATION NETWORKS
- H04W24/00—Supervisory, monitoring or testing arrangements
- H04W24/10—Scheduling measurement reports ; Arrangements for measurement reports
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L43/00—Arrangements for monitoring or testing data switching networks
- H04L43/08—Monitoring or testing based on specific metrics, e.g. QoS, energy consumption or environmental parameters
- H04L43/0823—Errors, e.g. transmission errors
- H04L43/0829—Packet loss
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/60—Network streaming of media packets
- H04L65/65—Network streaming protocols, e.g. real-time transport protocol [RTP] or real-time control protocol [RTCP]
-
- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/80—Responding to QoS
Definitions
- the present invention relates to a voice quality measuring device, voice quality measuring method and program for measuring voice quality, and in particular, flows in an IP section of a network such as a mobile CSIP (Continuous Service Improvement Program) network and a mobile EPC (Evolved Packet Core) network.
- the present invention relates to a voice quality measuring device, voice quality measuring method and program for measuring voice quality by collecting and analyzing packets.
- an apparatus for collecting voice packets flowing through a network such as a mobile network or an IP network to analyze voice quality. For example, by analyzing a header such as a UDP (User Datagram Protocol) header, an RTP (Real-time Transport Protocol) header, or an RTCP (Real-time Transport Control Protocol), the packet loss rate, round trip delay, or Direction delay, jitter, etc. are detected, and if at least one of them exceeds a predetermined threshold value, deterioration of voice quality is detected, and an analysis result or detection of deterioration is notified to a superordinate monitoring device.
- the voice quality analysis device for monitoring voice quality in voice communication service has been put to practical use by outputting or displaying on the monitoring terminal the monitoring device as it is or after processing or editing these.
- Patent Document 1 describes a listening quality evaluation apparatus that evaluates the listening quality of a voice-based IP packet media service provided via a packet communication network on the terminal side.
- Patent Document 2 describes a network voice quality control target value calculating device capable of managing a packet communication network in consideration of the influence of burst loss of packets on user experience quality for an application.
- Patent Document 3 a network voice quality management that calculates a network voice quality management target value consisting of performance information that can be actually measured from a packet communication network according to a voice subjective quality target value set for a voice application A target value calculation device is described.
- the above voice quality analysis device for example, only the RTP header of the RTP packet can be analyzed, so there is a problem that it is only possible to measure the occurrence of packet loss and the packet loss rate.
- voice quality is measured by sampling the IuUP (Iu User Plane) protocol frame or RFC (Request for Comments) 3267 payload format (Non-Patent Document 1) flowing in the mobile core network.
- IuUP Iu User Plane
- RFC Request for Comments
- 3267 payload format Non-Patent Document 1
- An object of the present invention is to provide a voice quality measuring device, a voice quality measuring method, and a program that solve the problems.
- the voice quality measuring device is A packet receiving unit for collecting packets conforming to a predetermined protocol from the network; An analysis unit that calculates a frame discard rate or a packet loss rate based on the information stored in the packet; And a quality measurement unit configured to estimate voice quality based on the frame loss rate or the packet loss rate.
- the voice quality measuring device is A packet receiving unit for collecting packets compliant with IuUP (Iu User Plane) protocol from the network; An IuUP analysis unit that calculates a frame discard rate based on at least one of a frame number, an FQC (Frame Quality Classifier), a header CRC (Cyclic Redundancy Check), and a payload CRC stored in the packet; And a quality measurement unit that estimates voice quality based on the frame discard rate.
- IuUP Iu User Plane
- FQC Framework Quality Classifier
- header CRC Cyclic Redundancy Check
- the voice quality measuring device is A packet receiving unit for collecting packets compliant with RFC (Request For Comments) 3267 protocol from the network; An RFC 3267 analysis unit that calculates a packet loss rate based on Q bits stored in the payload portion of the packet; And a quality measurement unit that estimates voice quality based on the packet loss rate.
- RFC Request For Comments
- the voice quality measurement method is Collecting a packet conforming to a predetermined protocol from the network; Calculating a frame discard rate or a packet loss rate based on the information stored in the packet; Estimating voice quality based on the frame loss rate or the packet loss rate.
- the voice quality measurement method is Collecting a packet compliant with IuUP (Iu User Plane) protocol from the network; Calculating a frame discard rate based on at least one of a frame number stored in the packet, a frame quality classifier (FQC), a header CRC (Cyclic Redundancy Check), and a payload CRC. Estimating voice quality based on the frame discard rate.
- IuUP Iu User Plane
- FQC frame quality classifier
- header CRC Cyclic Redundancy Check
- the program according to the sixth aspect of the present invention is A process of collecting packets conforming to a predetermined protocol from the network; Calculating a frame discard rate or a packet loss rate based on the information stored in the packet; And causing the computer to execute a process of estimating voice quality based on the frame loss rate or the packet loss rate.
- voice quality can be measured based on a packet conforming to a predetermined protocol (for example, IuUP protocol or RFC3267 protocol) collected from a mobile network Can be
- FIG. 1 is a block diagram showing the configuration of the voice quality measurement device 110 according to the present embodiment.
- the voice quality measuring apparatus 110 includes a packet receiving unit 111, an IuUP analyzing unit 114, and a quality measuring unit 115.
- the packet reception unit 111 receives an RTP packet containing an AMR (Adaptive Multi-Rate) IuUP protocol frame collected in the IP section of a mobile CSIP (Circuit Switched over IP) network, and outputs this to the IuUP analysis unit 114.
- AMR Adaptive Multi-Rate
- IuUP protocol frame is defined, for example, in 3rd Generation Partnership Project (3GPP) TS 25.415 standard (Non-Patent Document 2) and TS 26.102 standard (Non-Patent Document 3).
- the IuUP analysis unit 114 analyzes the sound quality deterioration due to the occurrence of bit errors, IuUP frame discarding, and the like. Specifically, the IuUP analysis unit 114 analyzes the IuUP protocol frame stored in the payload portion of the RTP packet, and performs the following processing.
- the IuUP analysis unit 114 refers to the value of the FQC (Frame Quality Classifier) field stored in the header portion of the IuUP protocol frame, and in the observation period (T), the value of the FQC field is other than 0 (Good) (for example, The number of frames, which is 1 and 2), is counted, the frame discard rate_1 is calculated based on the following equation, and the calculated frame discard rate_1 is output to the quality measurement unit 115.
- FQC Full Quality Classifier
- Frame discard rate _1 (N / M) * 100 (1)
- N in the equation (1) represents the number of frames in which the value of the FQC field is other than 0 in the observation period T.
- M in equation (1) represents the total number of RTP packets in the observation period T (ie, the total number of IuUP frames in the observation period T).
- the IuUP analysis unit 114 recalculates the value of the header CRC (Cyclic Redundancy Check) according to the method described in the 3GPP TS 25.415 standard (Non-Patent Document 2), and matches the value of the header CRC stored in the IuUP protocol frame. It is determined whether to do. If they do not match, the frame is discarded due to CRC NG, so the IuUP analysis unit 114 calculates a frame discard rate _ 2 and outputs it to the quality measurement unit 115.
- CRC Cyclic Redundancy Check
- the IuUP analysis unit 114 recalculates the value of the payload CRC according to the method described in the 3GPP TS 25.415 standard (Non-Patent Document 2), and determines whether or not the value of the payload CRC matches the value of the payload CRC stored in the IuUP protocol frame. judge. If they do not match, the frame is discarded due to CRC NG, so the IuUP analysis unit 114 calculates the frame discard rate_3 and outputs it to the quality measurement unit 115.
- the quality measurement unit 115 inputs the frame discard rate_1, the frame discard rate_2 and the frame discard rate_3 output from the IuUP analysis unit 114 within a predetermined observation period T, and the total frame discarding is performed according to the following equation Calculate the rate.
- MOS_e MOS value
- the quality measuring unit 115 estimates the MOS value using the relationship that the number of frame error concealment processing increases and the MOS value decreases as the frame discard rate increases.
- an objective evaluation value PESQ value may be used instead of the subjective evaluation MOS value. That is, a relational expression between the total frame loss rate and the PESQ value is created in advance offline based on experiments, and the quality measuring unit 115 estimates the PESQ value from the total frame loss rate to estimate the PESQ value. May be output.
- PESQ the details of PESQ are described in ITU-T Recommendation P.3. 862 (Non-Patent Document 4).
- the voice quality is calculated with respect to the total frame discard rate obtained by adding all three frame discard rates.
- voice quality may be calculated for each of the three frame discard rates, and the calculated voice quality may be separately output. At this time, it is possible to separate voice quality due to an error in the wireless access section and voice quality regarding the core network section.
- FIG. 2 shows, as an example, a configuration when quality measurement is performed by connecting the voice quality measurement device 110 according to the present embodiment to the IP section of the mobile CSIP network in communication between portable terminals by the mobile CSIP network.
- the mobile terminal 170 and the mobile terminal 171 perform voice communication (voice call) via the wireless access network 190, the mobile core network 180 and the wireless access network 191.
- the mobile core network 180 is a CSIP (Circuit Switched over IP) network, as an example. That is, the voice signal of the circuit switching is converted into an RTP packet by the voice communication device 150 and the voice communication device 151, and is sent to the mobile core network 180.
- CSIP Circuit Switched over IP
- the terminal 170 converts the input voice into a bit stream compressed and encoded by a voice compression coding method installed in the terminal, and outputs the bit stream.
- a bit rate of 12.2 kbps of AMR (Adaptive Multi-Rate) speech codec is used as a voice compression encoding method.
- AMR Adaptive Multi-Rate
- Details of the AMR are defined, for example, in the 3GPP TS 26.090 standard (Non-Patent Document 5).
- the AMR bit stream is stored in an IuUP (Iu User Plane) protocol frame when transmitted from the wireless access network 190 to the mobile core network 180 via the wireless access network 190.
- the IuUP protocol frame arrives at the mobile core network 180 and is input to the voice communication device 150.
- voice communication between the portable terminal 170 and the portable terminal 171 is, for example, communication by TrFO (Transcoder Free Operation) bypassing the audio codec. Therefore, the voice communication device 150 stores the IuUP protocol frame in the payload portion of the RTP packet, and then sends the RTP packet to the voice communication device 151 on the opposite terminal side using the RTP / UDP / IP protocol.
- TrFO Transcoder Free Operation
- the voice communication device 151 receives the RTP packet, extracts the IuUP protocol frame stored in the RTP payload portion, and outputs it to the wireless access network 191.
- the wireless access network 191 In the radio access network 191, the 12.2 kbps AMR bit stream stored in the IuUP protocol frame is taken out and sent out to the portable terminal 171.
- the portable terminal 171 receives the 12.2 kbps AMR bit stream, decodes the bit stream, and reproduces voice.
- the voice quality measurement device 110 is in the upward direction (eg, the direction from the voice communication device 150 to the voice communication device 151) and the downward direction (eg, the direction from the voice communication device 151 to the voice communication device 150) of the IP section of the mobile core network 180.
- the IuUP protocol frame in the RTP packet storing K channel (K ⁇ 1) IuUP protocol frames is collected from both directions of the above, and based on the configuration shown in FIG. Analyze the frame and measure voice quality.
- FIG. 3 shows an example of the configuration in the case of applying the voice quality measurement device 110 of this embodiment to an LTE / EPC network.
- LTE stands for Long Term Evolution
- EPC stands for Evolved Packet Core.
- the details of the EPC are defined in the 3GPP TS 23.401 standard (Non-Patent Document 6).
- Non-Patent Document 6 Non-Patent Document 6
- the components given the same reference numerals perform the same operation.
- the mobile terminal 270 for LTE stores the AMR stream in the payload format of RFC 3267, further stores it in the payload of RTP, transmits UDP / IP transport on the LTE bearer, and transmits the LTE radio access network Receive at 220.
- the LTE radio access network 220 converts the RFC3267 payload format into an IuUP protocol frame, stores the IuUP protocol frame in the RTP payload, and sends it to the SP / GW 250 of the mobile EPC network.
- SP / GW is a generic name of S-GW and P-GW
- S-GW indicates Serving GW (Gateway)
- P-GW indicates Packet Data Network GW.
- the IuUP protocol frame is transferred while being stored in the payload of the RTP / UDP / IP packet.
- the opposite side SP / GW 251 receives the IuUP protocol frame and transfers it to the LTE radio access network 221.
- the LTE radio access network 221 converts the IuUP protocol frame stored in the RTP payload into the RFC3267 payload format, stores the RFC3267 payload format in the RTP payload, and places RTP / UDP / IP on the LTE bearer. It is sent to the other party's mobile phone 271.
- the mobile phone 271 receives the RTP, extracts the AMR stream stored in the RFC3267 payload format in the RTP payload, and decodes it by the AMR decoder to reproduce voice.
- the voice quality monitoring device 110 is an IuUP protocol frame in upstream and downstream RTP packets of K channels (K ⁇ 1) exchanged in the IP section between the SP / GW 250 and the SP / GW 251.
- the voice quality is measured based on the configuration shown in FIG.
- FIG. 4 is a block diagram showing the configuration of the speech analysis device 120 according to the present embodiment.
- the voice quality analysis device 120 includes a packet reception unit 113, an RFC 3267 analysis unit 116, an RTP header analysis unit 112, and a quality measurement unit 117.
- the packet reception unit 113 receives the RTP packet storing the AMRRFC 3267 payload, which is collected in the IP section of the mobile CSIP network, and outputs the RTP packet to the RTP header analysis unit 112 and the RFC 3267 analysis unit 116.
- the RTP header analysis unit 112 performs the following (packet loss) analysis. That is, the RTP header analysis unit 112 checks continuity for the sequence numbers stored in the RTP header for a predetermined observation period T (for example, several seconds), and the sequence numbers are missed (that is, continuity is determined). In the case of “lack”, the packet loss is determined, and the packet loss rate L_1 in the observation period T is calculated and output to the quality measurement unit 117.
- the RFC 3267 analysis unit 116 refers to the Q-bit value of the RFC 3267 payload format header stored in the RTP payload. When this numerical value is 0 (Damaged), it is determined that the bit stream of AMR stored in the RFC3267 payload is likely to contain an error, and the number of packets in which the Q bit is 0 in the observation period T is After counting, the error packet rate L_2 is calculated according to the following equation, and is output to the quality measurement unit 117.
- P in the equation (3) represents the number of packets in which the Q bit is 0 in the observation period T.
- M in the equation (3) represents the total number of RTP packets in the observation period T, similarly to M in the equation (1).
- the quality measurement unit 117 receives L_1 from the RTP header analysis unit 113 and L_2 from the RFC 3267 analysis unit 116, and calculates a total packet loss rate L_T according to the following equation for each observation period T.
- L_T L_1 + L_2 (4)
- the quality measuring unit 117 estimates the MOS value (MOS_e) from the total packet loss rate using this relational expression, and outputs the MOS_e value.
- the quality measuring unit 117 uses the MOS as the packet loss rate increases, as in the case of using the relationship that the MOS value decreases as the frame discard rate increases.
- the MOS value is estimated using the relationship that the value decreases.
- an objective evaluation value PESQ value may be used instead of the subjective evaluation MOS value. That is, a relational expression between the total packet loss rate and the PESQ value is created in advance offline based on experiments, and the quality measuring unit 117 estimates the PESQ value from the total packet loss rate to estimate the PESQ value. May be output.
- the voice quality is calculated for the total packet discard rate L_T obtained by adding each packet discard rate.
- voice quality may be calculated for each of the packet discard rates of L_1 and L_2, and the calculated voice quality may be separately output. As a matter of fact, it is possible to separate the voice quality for the core network zone and the voice quality due to an error in the radio access zone.
- the voice quality is measured based on both the packet discard rate by the Q bit of RFC3267 and the packet discard rate by the RTP header sequence number.
- voice quality may be measured from the packet discard rate based only on Q bits of RFC3267. At this time, although the amount of processing can be reduced, errors in the wireless access section are not reflected in the voice quality.
- FIG. 5 is a diagram showing the IP section of the mobile CSIP network when the CSIP network is connected to the fixed network, and voice communication is performed between the mobile terminal connected to the CSIP network and the fixed terminal connected to the fixed network.
- the configuration when measuring the quality by connecting the voice quality measuring device 120 according to the embodiment is shown as an example.
- the components given the same reference numerals perform the same operation.
- the portable terminal 170 converts the input voice into a bit stream compression-encoded by an audio compression coding method installed in the terminal, and outputs the bit stream.
- a bit rate of 12.2 kbps of AMR (Adaptive Multi-Rate) speech codec is used as a voice compression encoding method.
- the AMR bit stream is stored in the IuUP protocol frame when it is transmitted from the wireless access network 190 to the mobile core network 180 via the wireless access network 190.
- the IuUP protocol frame arrives at the mobile core network 180 and is input to the voice communication device 160.
- the voice communication apparatus 160 extracts header information and a bit stream relating to 12.2 kbps AMR from the IuUP protocol frame, and stores them in the payload portion of the RTP packet.
- the opposite terminal is not a portable terminal but a terminal of a fixed network
- the voice communication device 160 constructs a payload format according to RFC3267, transfers it from the IuUP frame to a payload format including Q bits of RFC3267, and stores a 12.2 kbps AMR bit stream in the payload portion of RFC3267, RTP
- the RTP packet is sent to the gateway device 165 using the / UDP / IP protocol.
- the gateway device 165 receives the RTP packet, checks the RFC3267 payload format, and then checks the 12.2 kbps AMR stream stored in the payload portion of RFC3267 G. Codec conversion to G.711 stream, and G. 711 streams are output to the Public Switched Telephone Networks (PSTN) 200 in Synchronous Transfer Mode (STM).
- PSTN Public Switched Telephone Networks
- STM Synchronous Transfer Mode
- the telephone set 210 is connected to the PSTN network 200 and receives an audio signal.
- the voice communication in the direction from the telephone set 210 to the portable terminal 170 is different from the voice communication described above only in the direction, and thus the description thereof is omitted.
- the voice quality monitoring device 120 extracts RFC3267 payload format packets stored in upstream and downstream RTP packets for K channels (K ⁇ 1) exchanged in the IP section between the voice processing device 160 and the gateway device 165. Then, based on the configuration shown in FIG. 4, the voice quality is measured for the RFC3267 payload format.
- FIG. 6 shows an example of the configuration in the case of applying the voice quality measurement device 120 of this embodiment to an LTE / EPC network.
- components given the same reference numerals perform the same operation.
- the mobile terminal 270 for LTE stores the AMR stream in the payload format of RFC 3267, further stores it in the payload of RTP, transmits UDP / IP transport on the LTE bearer, and transmits the LTE radio access network Received by the SP / GW 250 of the mobile EPC network via 290.
- SP / GW is a generic name of S-GW and P-GW, S-GW indicates Serving GW, and P-GW indicates Packet Data Network GW.
- RTP / UDP / IP packets are transferred while storing the RFC3267 payload format.
- the SP / GW 251 receives this and transfers it to the LTE radio access network 291.
- the LTE radio access network 291 transmits RTP / UDP / IP on the LTE bearer as it is in RFC 3267 and sends it to the mobile phone 271 on the other side.
- the mobile phone 271 receives this, takes out the AMR stream stored in the RFC3267 payload format, and decodes it by the AMR decoder to reproduce speech.
- Voice quality monitoring apparatus 120 is an RFC 3267 stored in upstream and downstream RTP packets for K channels (KK1) exchanged in the IP section between SP / GW 250 and SP / GW 251.
- the payload format packet is collected, and the voice quality is measured based on the configuration shown in FIG. 4 for the information stored in the RFC3267 payload format.
- the IuUP protocol frame flowing in the mobile CSIP or mobile LTE / EPC core network is analyzed, and at least one of the frame number, FQC, header CRC and payload CRC stored in this is analyzed.
- the frame discard rate By calculating the frame discard rate based on one piece of information, it is possible to estimate and output a subjective MOS value or a PESQ value. According to such a voice quality measuring device, the amount of processing is extremely small, the reference signal is unnecessary, and the convenience is high.
- the voice quality measurement device of the above embodiment the RFC3267 payload format protocol flowing in the mobile CSIP or mobile LTE / EPC core network is analyzed, and based on only Q bits or both Q bits and sequence numbers.
- the packet discard rate By calculating the packet discard rate, the subjective MOS value or the PESQ value can be estimated and output. According to such a voice quality measuring device, the amount of processing is extremely small, the reference signal is unnecessary, and the convenience is high.
- the voice quality for the radio access network and the voice quality for transfer in the core network are separated by separately outputting the voice quality for each of the packet discard rate and the frame discard rate. be able to.
- the present invention also includes the inventions described below.
- a packet receiving unit for collecting a packet compliant with IuUP (Iu User Plane) protocol from a network
- An IuUP analysis unit that calculates a frame discard rate based on at least one of a frame number, an FQC (Frame Quality Classifier), a header CRC (Cyclic Redundancy Check), and a payload CRC stored in the packet;
- a quality measurement unit configured to estimate voice quality based on the frame discard rate.
- the IuUP analysis unit is characterized in that the frame discarding rate is calculated based on a ratio in which FQC stored in each of the plurality of packets collected during a predetermined period is a predetermined value.
- the voice quality measurement device according to appendix 1.
- the IuUP analysis unit recalculates the value of header CRC for each of a plurality of packets collected in a predetermined period, stores the value of header CRC recalculated for each packet, and the packet
- the voice quality measuring device characterized in that a frame discard rate is calculated based on a rate at which the value of the header CRC matches.
- the IuUP analysis unit recalculates the value of the payload CRC for each of a plurality of packets collected in a predetermined period, and stores the value of the payload CRC recalculated for each packet and the packet.
- the voice quality measuring device according to any one of appendices 1 to 3, characterized in that a frame discarding rate is calculated based on a rate at which the value of the payload CRC matches.
- the quality measuring unit is characterized by estimating voice quality based on a frame discarding rate obtained by adding together a plurality of frame discarding rates calculated by the IuUP analyzing unit according to mutually different methods. And the voice quality measuring device according to any one of appendices 2 to 4.
- a packet receiving unit that extracts a packet conforming to the RFC (Request For Comments) 3267 protocol from the network, An RFC 3267 analysis unit that calculates a packet loss rate based on Q bits stored in the payload portion of the packet; And a quality measurement unit configured to estimate speech quality based on the packet loss rate.
- RFC Request For Comments
- the RFC 3267 analysis unit is characterized in that the packet loss rate is calculated based on a ratio in which Q bits stored in each of the plurality of packets collected during a predetermined period have a predetermined value.
- the voice quality measuring device according to claim 6.
- the quality measurement unit according to any one of supplementary notes 1 to 7, characterized in that speech quality is estimated as a MOS (Mean Opinion Score) value or a PESQ (Perceptual Evaluation of Speech Quality) value.
- MOS Mobile Opinion Score
- PESQ Personal Evaluation of Speech Quality
- the quality measuring unit uses the relational expression of the frame loss rate or the packet loss rate and the MOS value or the PESQ value, which is created in advance by an experiment using a large amount of data offline.
- the voice quality measuring device according to appendix 8, wherein a MOS value or a PSEQ value is estimated from a discard rate or the packet loss rate.
- CSIP Continuous Service Improvement Program
- EPC Evolved Packet Core
- a step of collecting a packet compliant with IuUP (Iu User Plane) protocol from the network by the computer Calculating a frame discard rate based on at least one of a frame number stored in the packet, a frame quality classifier (FQC), a header CRC (Cyclic Redundancy Check), and a payload CRC. And e. Estimating the speech quality based on the frame discard rate.
- IuUP Iu User Plane
- FQC frame quality classifier
- header CRC Cyclic Redundancy Check
- the method includes a step of calculating a frame discarding rate based on a ratio in which the FQC stored in each of the plurality of packets collected during a predetermined period is a predetermined value.
- the voice quality measurement method according to appendix 11.
- the computer recalculates the value of header CRC for each of a plurality of packets collected in a predetermined period, and the value of header CRC recalculated for each packet and the header stored in the packet.
- the computer recalculates the value of payload CRC for each of a plurality of packets collected in a predetermined period, and the value of payload CRC recalculated for each packet and the payload stored in the packet 15.
- the voice quality measuring method according to any one of appendices 11 to 13, comprising the step of calculating a frame discard rate based on the rate at which the CRC value matches.
- the method includes a step of estimating speech quality based on a frame discarding rate obtained by adding together a plurality of frame discarding rates calculated by different methods from each other by the computer.
- the voice quality measurement method according to any one of the above.
- the computer is characterized in that the packet loss rate is calculated based on a ratio in which the Q bits stored in each of the plurality of packets collected in a predetermined period are a predetermined value.
- Voice quality measurement method described in.
- IuUP Iu User Plane
- FQC frame quality classifier
- header CRC Cyclic Redundancy Check
Landscapes
- Engineering & Computer Science (AREA)
- Multimedia (AREA)
- Computer Networks & Wireless Communication (AREA)
- Signal Processing (AREA)
- Environmental & Geological Engineering (AREA)
- Data Exchanges In Wide-Area Networks (AREA)
- Maintenance And Management Of Digital Transmission (AREA)
- Mobile Radio Communication Systems (AREA)
- Telephonic Communication Services (AREA)
Abstract
Description
本発明は、日本国特許出願:特願2010-013098号(2010年1月25日出願)の優先権主張に基づくものであり、同出願の全記載内容は引用をもって本書に組み込み記載されているものとする。
所定のプロトコルに準拠したパケットをネットワークから採取するパケット受信部と、
前記パケットに格納された情報に基づいて、フレーム廃棄率またはパケットロス率を算出する解析部と、
前記フレーム廃棄率または前記パケットロス率に基づいて音声品質を推定する品質計測部と、を備えている。
IuUP(Iu User Plane)プロトコルに準拠したパケットをネットワークから採取するパケット受信部と、
前記パケットに格納されたフレーム番号、FQC(Frame Quality Classifier)、ヘッダCRC(Cyclic Redundancy Check)及びペイロードCRCのうちの少なくともいずれか1つに基づいて、フレーム廃棄率を算出するIuUP解析部と、
前記フレーム廃棄率に基づいて音声品質を推定する品質計測部と、を有する。
RFC(Request For Comments)3267プロトコルに準拠したパケットをネットワークから採取するパケット受信部と、
前記パケットのペイロード部分に格納されたQビットに基づいて、パケットロス率を算出するRFC3267解析部と、
前記パケットロス率に基づいて音声品質を推定する品質計測部と、を有する。
所定のプロトコルに準拠したパケットをネットワークから採取する工程と、
前記パケットに格納された情報基づいて、フレーム廃棄率またはパケットロス率を算出する工程と、
前記フレーム廃棄率または前記パケットロス率に基づいて音声品質を推定する工程と、を含む。
IuUP(Iu User Plane)プロトコルに準拠したパケットをネットワークから採取する工程と、
前記パケットに格納されたフレーム番号、FQC(Frame Quality Classifier)、ヘッダCRC(Cyclic Redundancy Check)及びペイロードCRCのうちの少なくともいずれか1つに基づいて、フレーム廃棄率を算出する工程と、
前記フレーム廃棄率に基づいて音声品質を推定する工程と、を含む。
所定のプロトコルに準拠したパケットをネットワークから採取する処理と、
前記パケットに格納された情報基づいて、フレーム廃棄率またはパケットロス率を算出する工程と、
前記フレーム廃棄率または前記パケットロス率に基づいて音声品質を推定する処理と、をコンピュータに実行させる。
第1の実施形態に係る音声品質計測装置について、図面を参照して説明する。
第2の実施形態に係る音声品質計測装置について、図面を参照して説明する。
前記パケットに格納されたフレーム番号、FQC(Frame Quality Classifier)、ヘッダCRC(Cyclic Redundancy Check)及びペイロードCRCのうちの少なくともいずれか1つに基づいて、フレーム廃棄率を算出するIuUP解析部と、
前記フレーム廃棄率に基づいて音声品質を推定する品質計測部と、を備えていることを特徴とする音声品質計測装置。
前記パケットのペイロード部分に格納されたQビットに基づいて、パケットロス率を算出するRFC3267解析部と、
前記パケットロス率に基づいて音声品質を推定する品質計測部と、を備えていることを特徴とする音声品質計測装置。
前記パケットに格納されたフレーム番号、FQC(Frame Quality Classifier)、ヘッダCRC(Cyclic Redundancy Check)及びペイロードCRCのうちの少なくともいずれか1つに基づいて、フレーム廃棄率を算出する工程と、
前記フレーム廃棄率に基づいて音声品質を推定する工程と、を含むことを特徴とする音声品質計測方法。
前記パケットのペイロード部分に格納されたQビットに基づいて、パケットロス率を算出する工程と、
前記パケットロス率に基づいて音声品質を推定する工程と、を含むことを特徴とする音声品質計測方法。
前記パケットに格納されたフレーム番号、FQC(Frame Quality Classifier)、ヘッダCRC(Cyclic Redundancy Check)及びペイロードCRCのうちの少なくともいずれか1つに基づいて、フレーム廃棄率を算出する処理と、
前記フレーム廃棄率に基づいて音声品質を推定する処理と、をコンピュータに実行させることを特徴とするプログラム。
前記パケットのペイロード部分に格納されたQビットに基づいて、パケットロス率を算出する処理と、
前記パケットロス率に基づいて音声品質を推定する処理と、をコンピュータに実行させることを特徴とするプログラム。
111、113 パケット受信部
112 RTPヘッダ解析部
114 IuUP解析部
115、117 品質計測部
116 RFC3267解析部
150、151、160 音声通信装置
165 ゲートウェイ装置
170、171、270、271 携帯端末
180 モバイルコアネットワーク
190、191 無線アクセス網
200 PSTN網
210 電話機
220、221、290、291 LTE無線アクセス網
250、251 S-P/GW
280 EPCネットワーク
Claims (17)
- 所定のプロトコルに準拠したパケットをネットワークから採取するパケット受信部と、
前記パケットに格納された情報に基づいて、フレーム廃棄率またはパケットロス率を算出する解析部と、
前記フレーム廃棄率または前記パケットロス率に基づいて音声品質を推定する品質計測部と、を備えていることを特徴とする音声品質計測装置。 - 前記パケット受信部は、IuUP(Iu User Plane)プロトコルに準拠したパケットをネットワークから採取し、
前記解析部は、前記パケットに格納されたフレーム番号、FQC(Frame Quality Classifier)、ヘッダCRC(Cyclic Redundancy Check)及びペイロードCRCのうちの少なくともいずれか1つに基づいて、フレーム廃棄率を算出することを特徴とする、請求項1に記載の音声品質計測装置。 - 前記解析部は、所定の期間に採取された複数のパケットのそれぞれに格納されたFQCが所定の値である割合に基づいて、フレーム廃棄率を算出することを特徴とする、請求項2に記載の音声品質計測装置。
- 前記解析部は、所定の期間に採取された複数のパケットのそれぞれについてヘッダCRCの値を再計算し、各パケットに対して再計算されたヘッダCRCの値と該パケットに格納されたヘッダCRCの値とが一致する割合に基づいて、フレーム廃棄率を算出することを特徴とする、請求項2又は3に記載の音声品質計測装置。
- 前記解析部は、所定の期間に採取された複数のパケットのそれぞれについてペイロードCRCの値を再計算し、各パケットに対して再計算されたペイロードCRCの値と該パケットに格納されたペイロードCRCの値とが一致する割合に基づいて、フレーム廃棄率を算出することを特徴とする、請求項2乃至4のいずれか1項に記載の音声品質計測装置。
- 前記品質計測部は、前記解析部によって互いに異なる方法で算出された複数のフレーム廃棄率を足し合わせて得られたフレーム廃棄率に基づいて音声品質を推定することを特徴とする、請求項3乃至5のいずれか1項に記載の音声品質計測装置。
- 前記パケット受信部は、RFC(Request For Comments)3267プロトコルに準拠したパケットをネットワークから採取し、
前記解析部は、前記パケットのペイロード部分に格納されたQビットに基づいて、パケットロス率を算出することを特徴とする、請求項1に記載の音声品質計測装置。 - 前記解析部は、所定の期間に採取された複数のパケットのそれぞれに格納されたQビットが所定の値である割合に基づいて、パケットロス率を算出することを特徴とする、請求項7に記載の音声品質計測装置。
- 前記品質計測部は、音声品質をMOS(Mean Opinion Score)値又はPESQ(Perceptual Evaluation of Speech Quality)値として推定することを特徴とする、請求項1乃至8のいずれか1項に記載の音声品質計測装置。
- 前記品質計測部は、あらかじめオフラインで多量のデータを用いた実験によって作成しておいた、フレーム廃棄率又はパケットロス率とMOS値又はPESQ値との関係式を用いて、該フレーム廃棄率又は該パケットロス率からMOS値又はPSEQ値を推定することを特徴とする、請求項9に記載の音声品質計測装置。
- 前記ネットワークは、モバイルCSIP(Continuous Service Improvement Program)ネットワークット又はモバイルEPC(Evolved Packet Core)ネットワークであることを特徴とする、請求項1ないし10のいずれか1項に記載の音声品質計測装置。
- 所定のプロトコルに準拠したパケットをネットワークから採取する工程と、
前記パケットに格納された情報基づいて、フレーム廃棄率またはパケットロス率を算出する工程と、
前記フレーム廃棄率または前記パケットロス率に基づいて音声品質を推定する工程と、を含むことを特徴とする音声品質計測方法。 - IuUP(Iu User Plane)プロトコルに準拠したパケットをネットワークから採取する工程と、
前記パケットに格納されたフレーム番号、FQC(Frame Quality Classifier)、ヘッダCRC(Cyclic Redundancy Check)及びペイロードCRCのうちの少なくともいずれか1つに基づいて、フレーム廃棄率を算出する工程と、
前記フレーム廃棄率に基づいて音声品質を推定する工程と、を含むことを特徴とする、請求項12に記載の音声品質計測方法。 - RFC(Request For Comments)3267プロトコルに準拠したパケットをネットワークから採取する工程と、
前記パケットのペイロード部分に格納されたQビットに基づいて、パケットロス率を算出する工程と、
前記パケットロス率に基づいて音声品質を推定する工程と、を含むことを特徴とする、請求項12に記載の音声品質計測方法。 - 所定のプロトコルに準拠したパケットをネットワークから採取する処理と、
前記パケットに格納された情報基づいて、フレーム廃棄率またはパケットロス率を算出する工程と、
前記フレーム廃棄率または前記パケットロス率に基づいて音声品質を推定する処理と、をコンピュータに実行させることを特徴とするプログラム。 - IuUP(Iu User Plane)プロトコルに準拠したパケットをネットワークから採取する処理と、
前記パケットに格納されたフレーム番号、FQC(Frame Quality Classifier)、ヘッダCRC(Cyclic Redundancy Check)及びペイロードCRCのうちの少なくともいずれか1つに基づいて、フレーム廃棄率を算出する処理と、
前記フレーム廃棄率に基づいて音声品質を推定する処理と、をコンピュータに実行させることを特徴とする、請求項15に記載のプログラム。 - RFC(Request For Comments)3267プロトコルに準拠したパケットをネットワークから採取する処理と、
前記パケットのペイロード部分に格納されたQビットに基づいて、パケットロス率を算出する処理と、
前記パケットロス率に基づいて音声品質を推定する処理と、をコンピュータに実行させることを特徴とする、請求項15に記載のプログラム。
Priority Applications (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
JP2011550974A JPWO2011090185A1 (ja) | 2010-01-25 | 2011-01-24 | 音声品質計測装置、音声品質計測方法およびプログラム |
US13/575,211 US20120281589A1 (en) | 2010-01-25 | 2011-01-24 | Audio quality measurement apparatus, audio quality measurement method, and program |
Applications Claiming Priority (2)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
JP2010-013098 | 2010-01-25 | ||
JP2010013098 | 2010-01-25 |
Publications (1)
Publication Number | Publication Date |
---|---|
WO2011090185A1 true WO2011090185A1 (ja) | 2011-07-28 |
Family
ID=44306986
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
PCT/JP2011/051162 WO2011090185A1 (ja) | 2010-01-25 | 2011-01-24 | 音声品質計測装置、音声品質計測方法およびプログラム |
Country Status (3)
Country | Link |
---|---|
US (1) | US20120281589A1 (ja) |
JP (1) | JPWO2011090185A1 (ja) |
WO (1) | WO2011090185A1 (ja) |
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JP2014068321A (ja) * | 2012-09-27 | 2014-04-17 | Nec Access Technica Ltd | 転送装置、通信システム、転送方法およびプログラム |
Families Citing this family (7)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
CN105100508B (zh) | 2014-05-05 | 2018-03-09 | 华为技术有限公司 | 一种网络语音质量评估方法、装置和系统 |
US9325838B2 (en) | 2014-07-22 | 2016-04-26 | International Business Machines Corporation | Monitoring voice over internet protocol (VoIP) quality during an ongoing call |
CN106797380A (zh) * | 2014-07-30 | 2017-05-31 | 奥普图林克公司 | 语音优化实现装置 |
US10924607B2 (en) * | 2018-12-21 | 2021-02-16 | T-Mobile Usa, Inc. | Soft drop indicator based on UE triggers |
US11722544B2 (en) * | 2021-08-24 | 2023-08-08 | Motorola Mobility Llc | Electronic device that mitigates audio/video communication degradation of an image stream of a remote participant in a video communication session |
US11606406B1 (en) | 2021-08-24 | 2023-03-14 | Motorola Mobility Llc | Electronic device that mitigates audio/video communication degradation of an image stream of a local participant in a video communication session |
US11765215B2 (en) * | 2021-08-24 | 2023-09-19 | Motorola Mobility Llc | Electronic device that supports individualized dynamic playback of a live video communication session |
Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JP2005244609A (ja) * | 2004-02-26 | 2005-09-08 | Nippon Telegr & Teleph Corp <Ntt> | ネットワーク品質管理目標値算出方法および装置、並びに、ネットワーク品質監視方法および装置 |
WO2007045273A1 (en) * | 2005-10-17 | 2007-04-26 | Telefonaktiebolaget Lm Ericsson (Publ) | Method and apparatus for estimating speech quality |
JP2008205698A (ja) * | 2007-02-19 | 2008-09-04 | Nec Corp | インターワーキング装置 |
Family Cites Families (4)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JP3616592B2 (ja) * | 2001-09-20 | 2005-02-02 | 日本電気株式会社 | Atm網の音声処理システム、及び、その音声処理方法 |
EP1440589A1 (en) * | 2001-10-19 | 2004-07-28 | Nokia Corporation | Multicast transmission to a radio access network |
US7729346B2 (en) * | 2004-09-18 | 2010-06-01 | Genband Inc. | UMTS call handling methods and apparatus |
US7792150B2 (en) * | 2005-08-19 | 2010-09-07 | Genband Us Llc | Methods, systems, and computer program products for supporting transcoder-free operation in media gateway |
-
2011
- 2011-01-24 WO PCT/JP2011/051162 patent/WO2011090185A1/ja active Application Filing
- 2011-01-24 US US13/575,211 patent/US20120281589A1/en not_active Abandoned
- 2011-01-24 JP JP2011550974A patent/JPWO2011090185A1/ja active Pending
Patent Citations (3)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JP2005244609A (ja) * | 2004-02-26 | 2005-09-08 | Nippon Telegr & Teleph Corp <Ntt> | ネットワーク品質管理目標値算出方法および装置、並びに、ネットワーク品質監視方法および装置 |
WO2007045273A1 (en) * | 2005-10-17 | 2007-04-26 | Telefonaktiebolaget Lm Ericsson (Publ) | Method and apparatus for estimating speech quality |
JP2008205698A (ja) * | 2007-02-19 | 2008-09-04 | Nec Corp | インターワーキング装置 |
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
JP2014068321A (ja) * | 2012-09-27 | 2014-04-17 | Nec Access Technica Ltd | 転送装置、通信システム、転送方法およびプログラム |
Also Published As
Publication number | Publication date |
---|---|
US20120281589A1 (en) | 2012-11-08 |
JPWO2011090185A1 (ja) | 2013-05-23 |
Similar Documents
Publication | Publication Date | Title |
---|---|---|
WO2011090185A1 (ja) | 音声品質計測装置、音声品質計測方法およびプログラム | |
JP5668687B2 (ja) | 音声品質解析装置、音声品質解析方法およびプログラム | |
EP1938496B1 (en) | Method and apparatus for estimating speech quality | |
Singh et al. | VoIP: State of art for global connectivity—A critical review | |
US7986634B2 (en) | Apparatus and method for measuring quality of sound encoded with a variable band multi-codec | |
Ortega et al. | Comparison between the real and theoretical values of the technical parameters of the VoIP codecs | |
US9973402B2 (en) | Transmission device, receiving device, and relay device | |
EP2137727B1 (en) | Method and device for transmitting data in a communication system | |
JP4217121B2 (ja) | Ipネットワークシステムにおける音声品質評価方法および音声品質調整装置 | |
EP2514143B1 (en) | Connection analysis in communication systems | |
Zhang et al. | Perceived QoS assessment for Voip networks | |
Bhebhe et al. | VoIP performance over HSPA with different VoIP clients | |
Hammer et al. | Corrupted speech data considered useful: Improving perceived speech quality of voip over error-prone channels | |
Toral-Cruz et al. | Traffic analysis for IP telephony | |
Fitzpatrick | An E-Model based adaptation algorithm for AMR voice calls | |
Meddahi et al. | " Packet-e-model": e-model for wireless VoIP quality evaluation | |
Gambhir | Objective measurement of speech quality in VoIP over wireless LAN during handoff | |
Jung et al. | Improving wireless VoIP quality by using adaptive packet coding | |
Matousek et al. | On-line monitoring of VoIP quality using IPFIX | |
KR100939128B1 (ko) | 영상 통화를 위한 단말 및 방법 | |
Orosz et al. | VoicePerf: A Quality Estimation Approach for No-reference IP Voice Traffic | |
Georgieva et al. | Investigation of Algorithms for VoIP Signaling | |
Wang et al. | VolP QoS performance evaluation in a commercial environment | |
JP2016139848A (ja) | 音声品質推定装置、方法及びプログラム | |
Barbaresi et al. | Performance Evaluation of Quality of VoIP service over UMTS-UTRAN R99 |
Legal Events
Date | Code | Title | Description |
---|---|---|---|
121 | Ep: the epo has been informed by wipo that ep was designated in this application |
Ref document number: 11734791 Country of ref document: EP Kind code of ref document: A1 |
|
WWE | Wipo information: entry into national phase |
Ref document number: 2011550974 Country of ref document: JP |
|
NENP | Non-entry into the national phase |
Ref country code: DE |
|
WWE | Wipo information: entry into national phase |
Ref document number: 13575211 Country of ref document: US |
|
122 | Ep: pct application non-entry in european phase |
Ref document number: 11734791 Country of ref document: EP Kind code of ref document: A1 |