KR101860912B1 - Monitoring Method For Quality Of Voice In Each Network Section, And LTE Quality Measurement System Performing The Same - Google Patents

Abstract

A method of monitoring voice quality per network section according to an embodiment of the present invention calculates a voice quality for a second section based on information obtained from an RTP (Real Time Protocol) packet transmitted from a second terminal to a first terminal ; Calculating a speech quality for the entire interval based on information obtained from a Real Time Transport Control Protocol (RTCP) generated by the first terminal that received the RTP packet; And calculating voice quality for a first interval by subtracting the voice quality for the second interval from the voice quality for the entire interval, wherein the entire interval is calculated from an LQMS (LTE Quality Measurement System ), And the second interval from the LQMS to the second terminal.

Description

A voice quality monitoring method for each network section, and a LQMS (LQMS) for performing the same, and an LTE Quality Measurement System Performing The Same

The present invention relates to a voice quality monitoring method for each network section and a LQMS for performing the same. More particularly, the present invention relates to a voice quality monitoring method for each network section capable of obtaining voice quality for a specific network section, .

Real-time multimedia services such as Voice over Long Term Evolution (VoLTE) deliver services using Real Time Protocol (RTP) to increase service processing speed. RTP provides a terminal-to-terminal network transmission function suitable for application services transmitted using multicast or unicast networks of real-time data such as audio, video and simulation data.

RTP is designed not to be implemented as a separate layer but to provide information required by a specific application service so that the processing of the protocol can be integrated into the processing of the application. Therefore, unlike existing protocols, RTP is a kind of customized protocol that changes or adds a header according to the needs of an application service, thereby making it a suitable protocol for an application service.

RTP does not perform resource reservation and accordingly can not guarantee quality of service (QoS) such as timely delivery and sequential delivery. Therefore, RTCP (Real Time Transport Control Protocol) extended by a control protocol is used .

The RTCP provides information on the current network environment and the received data quality to the calling party, and the calling party adapts automatically to the network environment according to the RTCP to adjust the transmission rate, thereby actively coping with the loss occurring in the network. Round Trip Time (RTT) can be calculated through transmission and reception of RTCP packets between terminals, and the network state can be estimated through this.

However, since the information about the data quality indicated by the RTCP is related to the network environment between the end points, it is not related to the specific network section. Therefore, when there is a quality problem, there is a limitation in determining which section of the entire network environment is causing the problem.

RFC3550 (RTP A Transport Protocol for Real-Time Applications) ITU-T-REC-G.107

The present invention provides a voice quality monitoring method for each network section that provides voice quality for a specific section so that a problem can be detected in which section of the entire network environment, and an LQMS for performing the voice quality monitoring method.

It is to be understood that both the foregoing general description and the following detailed description are exemplary and explanatory and are not restrictive of the invention, unless further departing from the spirit and scope of the invention as defined by the appended claims. It will be possible.

According to an aspect of the present invention, there is provided a method for monitoring voice quality per network section according to an embodiment of the present invention, the method comprising: receiving, by a second terminal, Calculating a speech quality for a second section; Calculating a speech quality for the entire interval based on information obtained from a Real Time Transport Control Protocol (RTCP) generated by the first terminal that received the RTP packet; And calculating voice quality for a first interval by subtracting the voice quality for the second interval from the voice quality for the entire interval, wherein the entire interval is calculated from an LQMS (LTE Quality Measurement System ), And the second interval from the LQMS to the second terminal.

According to an embodiment, the information obtained from the RTP packet includes at least one of a lost packet count, a burst packet loss count, a duplicate packet count, an initial packet a lost packet count, and a last packet lost count.

According to an embodiment, the information obtained from the RTP packet may include jitter information calculated using a capture time of the RTP packet and a time stamp of the RTP packet.

According to an embodiment, the information obtained from the RTP packet may include a DLSR (delay of the second terminal) of the second terminal in a result of subtracting the capture time of the RTCP generated by the first terminal from the capture time of the RTCP generated by the second terminal, lt; RTI ID = 0.0 > SR). < / RTI >

According to an embodiment, the information obtained from the RTCP may include packet loss information including at least one of an expected RTP packet count received and a lost packet count.

According to an embodiment, the information obtained from the RTCP may include jitter information obtained from the " interarrival jitter " of the RTCP.

According to the embodiment, the information obtained from the RTCP subtracts the DLSR of the second terminal from the result of subtracting the capture time of the RTCP generated by the first terminal from the capture time of the RTCP generated by the second terminal And delay information calculated by subtracting the DLSR of the first terminal from the result of subtracting the capture time of the RTCP generated by the second terminal from the capture time of the RTCP generated by the first terminal .

The method of monitoring voice quality per network section according to another embodiment of the present invention calculates a voice quality for a first section based on information obtained from an RTP (Real Time Protocol) packet transmitted from a first terminal to a second terminal ; Calculating a speech quality for the entire interval based on information obtained from a Real Time Transport Control Protocol (RTCP) generated by the second terminal that received the RTP packet; And calculating voice quality for a second section by subtracting the voice quality for the first section from the voice quality for the entire section, wherein the entire section is transmitted from the first terminal to an LQMS (LTE Quality Measurement System ), And the second interval from the LQMS to the second terminal.

According to the voice quality monitoring method for each network section according to an embodiment of the present invention and the LQMS for performing the same, the voice quality of a specific wireless network can be checked. Analysis and optimization of operation for improving service quality and additional investment.

The effects obtained by the present invention are not limited to the above-mentioned effects, and other effects not mentioned can be clearly understood by those skilled in the art from the following description will be.

1 is a configuration diagram of an LTE network according to the present invention.
FIG. 2 is a view for explaining traffic flows on an LTE network for facilitating understanding of the present invention. FIG.
FIG. 3 is a diagram illustrating an example of checking the reception quality of an RTP packet for each network section.
FIG. 4 is a diagram illustrating an embodiment for checking the transmission quality of an RTP packet for each network section.
5 is a diagram for explaining a method of calculating delay information for each network section.

Hereinafter, at least one embodiment related to the present invention will be described in detail with reference to the drawings. The suffix "module" and " part "for the components used in the following description are given or mixed in consideration of ease of specification, and do not have their own meaning or role.

1 is a configuration diagram of an LTE network according to the present invention.

1, the LTE network 1 includes a UE (User Equipment) 10, an Evolved Node B (eNB) 20, an S-GW (Serving Gateway) 30, a P- A data network gateway 40, an MME (Mobility Management Entity) 50, an HSS (Home Subscriber Server) 60, and an LQMS (LTE Quality Measurement System) 100.

The UE 10 is an LTE user terminal and is connected to the eNB 20 via the LTE-Uu interface 15. Here, the LTE Uu interface 15 defines a control plane for transmitting and receiving a control message as a wireless interface and a user plane for providing user data.

The eNB 20 provides a radio interface to the UE 10 and provides radio resource management functions such as radio bearer control, radio admission control, dynamic radio resource allocation, load balancing and inter-cell interference control do.

The S-GW 30 is an end point of an evolved universal terrestrial radio access network (E-UTRAN) and an evolved packet core (EPC), and an anchor point at handover between the eNB 20 and the 3GPP system do. The E-UTRAN is composed of at least one eNB 20 and the EPC is composed of the S-GW 30, the P-GW 40, the MME 50 and the HSS 60.

The P-GW 40 connects the UE 10 to an external PDN (Packet Data Network) 110 and performs a packet filtering function. In addition, the P-GW 40 assigns an IP address to the UE 10 and operates as a mobility anchoring point when performing handover between the 3GPP system and the non-3GPP system. In particular, the P-GW 40 receives the Policy and Charging Control (PCC) rules from the Policy and Charging Rule Function (PCRF), applies it to the corresponding service flow, Billing function. The PCRF provides policy control decisions and billing control functions as entities that perform policy and billing control.

The HSS 60 is a central database for storing a subscriber profile, and provides the MME 50 with user authentication information and a user profile.

The LTE network 1 provides a voice over LTE (VoLTE) service, and the data is transmitted in the form of a packet according to a Real Time Transport Protocol (RTP), that is, an RTP packet. RTP is a transport layer protocol for transmitting and receiving audio and video in real time, and is specified with RFC 1889dp RTCP (RTP Control Protocol). RTP is usually used as the upper communication protocol of UDP. The transmitting end can discard a packet with a large delay by taking a reproduction synchronization based on a timestamp. In addition, the receiving end checks the transmission delay and the bandwidth, and notifies the upper layer application of the transmitting end using the RTCP packet, so that the quality of service control can be realized by adjusting the coding speed and the like. RTCP is a protocol for controlling such RTP, and includes information on the current network environment and the received data quality.

The LQMS 100 includes a whole section, a first section, and a second section based on the RTP packet and the RTCP packet transmitted between the P-GW 40 and the PDN 110, Each service quality can be measured or calculated. For this, the LQMS 100 may include a quality calculator for measuring or calculating the quality of service of each interval.

PDN 110 refers to a network that is connected to a counterpart terminal of the UE 10 (e.g., 400 in FIG. 3). The entire interval refers to the entire network interval from the UE 10 to the counterpart UE. The first interval is a network interval from the UE 10 to the LQMS 100, and the second interval is an LQMS 100 to the counterpart terminal.

The LQMS 100 may use tapping to obtain an RTP packet and an RTCP packet transmitted between the P-GW 40 and the PDN 110. The tapping may be performed without causing loss or delay of data And means a technology capable of capturing a packet transmitted from a transmission line. That is, no loss or delay of the RTP packet or the RTCP packet occurs in the measurement and calculation of the quality of service for each network section by the LQMS 100.

Hereinafter, the interface between the elements constituting the LTE network 1 will be briefly described.

The LTE-Uu 15 provides a control plane and a user plane with a radio interface between the UE 10 and the eNB 20. [

The S1-U 25 is an interface between the eNB 20 and the S-GW 30, and provides a user plane. At this time, GTP (GPRS Tunneling Protocol) tunneling (GTP-U) for each bearer is provided.

X2 26 is an interface between the two eNBs 20, providing a control plane and a user plane. The X2-AP protocol is used in the control plane and GTP tunneling per bearer for data forwarding in the X2 handover in the user plane.

S5 35 is an interface between the S-GW 30 and the P-GW 40, providing a control plane and a user plane. At this time, the user plane provides bearer-specific GTP tunneling (GTP-U), and the control plane provides GTP tunnel management (GTP-C).

SGi 45 defines a user plane and a control plane as an interface between the P-GW 40 and the PDN 110. In the user plane, IETF based IP packet forwarding protocol is used, and in the control plane, protocols such as DHCP and RADIUS / Diameter are used.

S11 55 defines a control plane as an interface between the MME 50 and the S-GW 30, and GTP tunneling for each bearer is provided.

The S6a 65 is provided with a control plane as an interface between the HSS 60 and the MME 50, a Diameter protocol is used, and is used to exchange UE subscription information and authentication information.

The S1-MME 75 is an interface between the eNB 20 and the MME 50, and a control plane is defined, and the S1-AP protocol is used.

FIG. 2 is a view for explaining traffic flows on an LTE network for facilitating understanding of the present invention. FIG.

Referring to FIG. 2, an Internet traffic flow in the user plane of the LTE network reference model is shown. Reference numeral 200a denotes a traffic flow from the UE 10 to the Internet, hereinafter referred to as "uplink traffic flow", 200b denotes a traffic flow from the Internet to the UE 10, hereinafter referred to as " Business card.

IP packets are transmitted over the GTP tunnel on the S1-U (25) and S5 (35) interfaces, respectively. Here, the GTP tunnel is set for each EPS bearer through control signaling when the UE 10 initially connects to the LTE network.

Since a plurality of EPS bearers are set on one of the S1-U 25 and the S5 (35) interfaces, a tunnel endpoint identifier (TEID) is allocated upwardly and downwardly in each GTP tunnel establishment to distinguish them. When a GTP tunnel is set in the S1-U (25) interface, a TEID (UL S1-TEID) having an end point is allocated to the S-GW 20 upward and a TEID (DL S1-TEID) having an end point is allocated to the eNB 20 -TEID). Similarly, when a GTP tunnel is established in the S5 (35) interface, a TEID (UL S5-TEID) having an end point in the P-GW 40 and a TEID (DL S5-TEID) having an end point in the S- S5-TEID).

The eNB 20, the S-GW 30 and the P-GW 40 transmit a GTP tunnel header to the GTP packet header when the user IP packet is transmitted through the GTP tunnel on the S1-U 25 and the S5 The TEID that is allocated at the time of generation is inserted and transmitted. The S-GW 30 must have mapping information between the UL S1-TEID and the UL S5-TEID in order to terminate the S1-GTP tunnel in the upward direction and to transmit the user IP packet to the S5-GTP tunnel. Likewise, in the downlink, mapping information between DL S5-TEID and DL S1-TEID must be maintained.

Hereinafter, referring to FIG. 2, a procedure for each entity to process uplink and downlink Internet traffic flows will be described in detail.

First, in the uplink, the UE 10 transmits internally generated user IP packets to the eNB 20 via the LTE-Uu 15 interface. The eNB 20 adds the IP address of the S-GW 30 to the destination IP address, the source IP address of the eNB 20, the S1 GTP header having the UL S1-TEID set to the TEID, To the S-GW 30 via the GTP tunnel.

When receiving the user IP packet through the S1 GTP tunnel, the S-GW 30 sets the IP address of the P-GW 40 as the destination address, the IP address of the S-GW 30 as the originating address, and the UL S5-TEID as the TEID Configure the configured S5 GTP header. Then, the S5 GTP header is added to the user IP packet and transmitted to the P-GW 40 through the S5 GTP tunnel.

Subsequently, the P-GW 40 controls the S5 GTP header to extract user IP packets, and then transmits them to the Internet through IP routing.

Referring to the downlink traffic flow, the P-GW 40 receives a TCP packet directed to the UE 10 via the Internet. The P-GW 40 adds the S5 GTP header including the IP address of the S-GW 30 as the destination address, the IP address of the P-GW 40 as the source address and the DL S5-TEID as the TEID to the TCP packet, Packet, and transmits the packet to the S-GW 30 through the S5-GTP tunnel.

The S-GW 30 transmits the IP address of the eNB 20 as the destination address, the IP address of the S-GW 30 as the originating address, and the DL S1-TEID as the S1 GTP And transmits the header to the eNB 20 through the S1 GTP tunnel.

The eNB 20 transmits the user IP packet from which the S1 GTP header has been removed to the UE 10 via a data radio bearer (DRB), which is a bearer on the radio link.

It should be noted that the GTP tunnel of FIG. 2 is a user plane GTP tunnel for delivering user IP packets, hereinafter referred to as a "GTP-U tunnel ".

FIG. 3 is a diagram illustrating an example of checking the reception quality of an RTP packet for each network section.

3, the UE 200 is a monitored terminal (or a first terminal) corresponding to the UE 10 of FIG. 1 and the eNB / SPGW 300 is an eNB 20, an S-GW (30) and P-GW (40). UE 400 is also a major terminal (or second terminal) connected to PDN 110 of FIG.

In the following description, it is assumed that the RTP packet for the VoLTE service between the UE 200 and the UE 400 is voice. However, the scope of the present invention is not limited to this and the present invention can be applied to other multimedia packets such as video.

The UE 400 transmits an RTP packet Rx-RTP to the UE 200 and the RTP packet Rx-RTP corresponds to a reception tone received by the UE 200. [ In this specification, the name of Rx is added to the packet received by the UE 200, and the name of Tx is added to the packet transmitted by the UE 200. [

The LQMS 100 can calculate the voice quality for the second section using the RTP packet (Rx-RTP) transmitted through the second section.

The LQMS 100 may reflect the three pieces of information obtained from the RTP packet (Rx-RTP) when calculating the voice quality for the second section.

The three pieces of information correspond to packet loss information, jitter information, and delay information.

Wherein the packet loss information includes at least one of a lost packet count, a burst packet loss count, a duplicate packet count, an initial packet lost count and a last packet missing count last packet lost count).

The LQMS 100 can calculate the missing packet count, which is the number of missing packets using the sequence number of the RTP packet (Rx-RTP). The transmitting end (UE 400) of the RTP packet (Rx-RTP) assigns a unique sequence number to each RTP packet. The sequence number is incremented by one from an initial value (randomly determined) Lt; / RTI > Since a large number of RTP packets are successively transmitted at the time of transmission of the RTP packet (Rx-RTP), the LQMS 100 stores the sequence number of the received RTP packet and determines the number of missing RTP packets among consecutive sequence numbers Can be calculated.

The LQMS 100 can calculate the continuous missing packet count, which is the number of continuously missing packets among the missing RTP packets, using the sequence number of the stored RTP packets.

Also, the LQMS 100 can calculate the redundant packet count, which is the number of packets corresponding to the redundant sequence number, using the sequence number of the stored RTP packet.

The LQMS 100 subtracts the sequence number of the first captured RTP packet (Rx-RTP) from the " highest sequence number received " of the RTCP for the RTP packet transmitted before the RTP packet (Rx- The first packet missing coefficient can be calculated.

That is, "highest sequence number received" means the highest sequence number among the sequence numbers of the RTP packets transmitted from the transmitting side 400, and the sequence number of the first captured packet in the "highest sequence number received" The number of missing packets having a sequence number lower than that of the RTP packet having the lowest sequence number among the captured packets can be calculated.

The LQMS 100 subtracts the sequence number of the last captured packet of the RTP packet (Rx-RTP) from the " highest sequence number received " of the RTCP of the transmitting side with respect to the RTP packet (Rx- The coefficient can be calculated.

That is, if the sequence number of the last packet captured in the " highest sequence number received " of the current RTCP is subtracted, the number of missing packets having a sequence number higher than that of the RTP packet having the highest sequence number among the captured packets have.

The jitter information is a deviation of the inter-arrival time and is calculated using the capture time of the RTP packet (Rx-RTP) and the time stamp of the RTP packet (Rx-RTP) .

The LQMS 100 may store the time of capture each time it receives an RTP packet (Rx-RTP), which is referred to as the capture time of the RTP packet (Rx-RTP). Each time the UE 400 transmits an RTP packet (Rx-RTP), the UE 400 records the transmission time in an RTP packet (Rx-RTP) and transmits the RTP packet (Rx-RTP).

For example, if the time stamp and capture time of the first RTP packet are t and t ', respectively, and the time stamp and capture time of the second RTP packet are t + a and t' + a ', respectively, t '+ a') - t '= a' and (t + a) -t = a becomes the transmission interval. Even if the transmission interval is constant due to the instability of the network communication, the inter-affiliation time may not be constant and fluctuate, and the jitter information is indicative of such a change (variation of the inter-affiliation time).

The delay information indicates a time delay required for transmission and reception of an RTP (or RTCP) packet in a second interval. A method for calculating delay information for each interval (a first interval, a second interval, or an entire interval) Will be described later.

The LQMS 100 may calculate the voice quality for the second section using the packet loss information, the jitter information, and the delay information.

The voice quality can be represented by an R-factor, which is a value obtained by quantifying the voice quality of the corresponding network. For example, the R-factor may have a positive value between 0 and 100, and the higher the R-factor, the better the quality.

The R-factor can be calculated by the following equation (1). Equation 1 is based on the e-model R-factor calculation method of the ITU-T-REC-G.107 standard, but the scope of the present invention is not limited thereto.

[Equation 1]

R Factor = Ro - Is - Id - Ie + A

Here, Ro denotes a basic noise-to-noise signal, and is a signal-to-noise ratio, which means noises due to a physical device between ends or noise due to ambient background sounds other than actual sounds.

Is is the sum of impulse or voice signal transmission, such as loud or small sound, quantization distortion, etc., which deteriorates the voice recording / reproduction quality. Id is a delay related to the voice signal, such as reception or noise of the sender, Impairments are delayed after voice signal transmission.

Ie is a value determined according to the percentage of each codec or packet loss (Effects of Equipment), and A is an advantage factor according to factors such as the type of user terminal.

The LQMS 100 can model parameters of each of Ro, Is, Id, Ie and A by referring to a reference table stored in advance based on the collected characteristics of the RTP packet and the RTCP. The reference table may be a table in which collected characteristics of RTP packets and RTCPs and specific values of respective parameters are mapped to each other.

In particular, the jitter information, the delay information, and the RTP packet size may be used as inputs in the calculation of Id. Also, the packet loss information can be used as an input in the calculation of Ie.

That is, Ie is a value reflecting the packet loss information for the second section, and Id is a value reflecting the jitter information for the second section and the delay information.

Accordingly, the LQMS 100 determines R (Rx-RTP) for the second section based on the packet loss information, the jitter information, and the delay information, which are information generated in the process of receiving the RTP packet -Factor (Rx-MOS_NW) can be calculated.

The UE 200 receives the RTP packet (Rx-RTP) and generates RTCP (Tx-RTCP), which is information on the quality of the RTP packet (Rx-RTP)

The LQMS 100 can calculate the R-factor (Rx-MOS_All) for the entire section based on the RTCP (Tx-RTCP), which is information on the quality of the RTP packet (Rx-RTP). This is because the RTCP (Tx-RTCP) is information on the quality of the RTP packet (Rx-RTP) received from the UE 400 to the UE 200.

The LQMS 100 may reflect three pieces of information obtained from the RTCP (Tx-RTCP) when calculating the voice quality for the entire section.

The three pieces of information correspond to packet loss information, jitter information, and delay information.

The packet loss information may include an expected RTP packet count, a lost packet count, and the like.

The LQMS 100 subtracts the highest sequence number received of the RTCP from the "highest sequence number received" of the currently captured RTCP (Tx-RTCP) for the RTP packet transmitted before the RTP packet (Rx-RTP) It is possible to calculate the expected received packet coefficient which is the total packet number of the RTP packet (Rx-RTP) corresponding to the (Tx-RTCP).

The LQMS 100 subtracts the " cumulative number of packet lost " of the RTCP for the RTP packet transmitted before the RTP packet (Rx-RTP) from the " cumulative number of packet lost " of the currently captured RTCP , The missing packet count which is the number of packets missing in the RTP packet (Rx-RTP) corresponding to the RTCP (Tx-RTCP) can be calculated.

The " cumulative number of packet lost " means the accumulated number of missing packets. A late received packet is not regarded as a missing packet, and may be negative when there are duplicated packets. The subtraction of "cumulative number of packet lost" can be made by taking two's complement for the case that "cumulative number of packet lost" becomes negative.

The jitter information means a deviation of the inter-arrival time and can be obtained from the " interarrival jitter " of the currently captured RTCP (Tx-RTCP).

The interarrival jitter is a deviation of the inter-active time calculated by the UE 200 using the capture time of the RTP packet (Rx-RTP) and the time stamp of the RTP packet (Rx-RTP).

The delay information indicates a time delay required for transmission and reception of an RTP (or RTCP) packet in the entire interval. A method of calculating delay information for each interval (the first interval, the second interval, or the entire interval) Will be described later.

The LQMS 100 may calculate the voice quality for the entire interval using the packet loss information, the jitter information, and the delay information.

The voice quality may be represented by an R-factor as well as the voice quality for the second section. That is, the R-factor can be calculated by Equation (1), and the LQMS 100 can model parameters of each of Ro, Is, Id, Ie and A by referring to a reference table stored in advance based on the RTCP .

In particular, the jitter information and the delay information can be used as inputs in the calculation of Id. Also, the packet loss information can be used as an input in the calculation of Ie.

That is, Ie is a value reflecting the packet loss information for the entire section, and Id is a value reflecting the jitter information for the entire section and the delay information.

Therefore, the LQMS 100 calculates an R-factor for the entire interval based on the packet loss information, the jitter information, and the delay information, which are information generated in the process of receiving the RTP packet (Rx-RTP) (Rx-MOS_All) can be calculated.

The LQMS 100 subtracts the R-factor (Rx-MOS_NW) representing the received speech quality for the second section from the R-factor (Rx-MOS_All) representing the received speech quality for the entire section ) And R-factor (Rx-MOS_Air) indicating the received speech quality for the first section. That is, the LQMS 100 can acquire only the received voice quality for the first section of the entire section.

The first period includes a period between the UE 200 and the eNB / SPGW 300 (between the UE 10 and the eNB 20 in FIG. 1), and between the eNB / SPGW 300 and the LQMS 100 (In the interval between the eNB 20 and the point where the LQMS 100 is tapped in FIG. 1). It can be said that the quality of the received voice for the first section is determined by the quality of the received voice of the wireless network since almost no quality deterioration occurs in the wired network.

Therefore, the LQMS 100 can determine whether the UE 200 and the UE 200 are connected to each other based on R-factor (Rx-MOS_Air) 20) can be known. That is, the LQMS 100 can collect data on the received voice quality of the wireless networks of a plurality of base stations, and provide the collected data to the network manager. Or the LQMS 100 may provide the network manager with information about the base station having the received voice quality that does not meet a predetermined reference value among a plurality of base stations. Accordingly, the network manager can identify the base station whose reception voice quality has deteriorated to such a degree that a quality check is required, and can promptly perform quality check or facility investment, thereby improving user satisfaction with network quality.

In addition, the LQMS 100 compares the R-factor (Rx-MOS_NW) with the R-factor (Rx-MOS_Air) to determine which network interval has a decisive influence on the received voice quality of the R- It can be judged whether it is a section.

FIG. 4 is a diagram illustrating an embodiment for checking the transmission quality of an RTP packet for each network section.

Referring to FIG. 4, the UE 200 is a monitoring target terminal corresponding to the UE 10 in FIG. 1, and the eNB / SPGW 300 includes the eNB 20, the S-GW 30, GW 40 in Fig. The UE 400 is also a major terminal connected to the PDN 110 of FIG.

In the following description, it is assumed that the RTP packet for the VoLTE service between the UE 200 and the UE 400 is voice. However, the scope of the present invention is not limited to this and the present invention can be applied to other multimedia packets such as video.

The UE 200 transmits an RTP packet (Tx-RTP) to the UE 400 and the RTP packet (Tx-RTP) corresponds to a reception tone received by the UE 400.

The LQMS 100 can calculate the voice quality for the first interval using the RTP packet (Tx-RTP) transmitted through the first interval.

The LQMS 100 may reflect the three pieces of information obtained from the RTP packet (Tx-RTP) when calculating the voice quality for the first section.

The three pieces of information correspond to packet loss information, jitter information, and delay information.

The packet loss information may include a missing packet count, a consecutive missing packet count, a duplicate packet count, a first packet missing count, and a last packet missing count.

The LQMS 100 can calculate the missing packet count which is the number of missing packets using the sequence number of the RTP packet (Tx-RTP). The transmitting terminal (UE 200) of the RTP packet (Tx-RTP) assigns a unique sequence number to each RTP packet, and the sequence number is incremented by one from an initial value (randomly determined) Lt; / RTI > Since a large number of RTP packets are successively transmitted in the transmission of the RTP packet (Tx-RTP), the LQMS 100 stores the sequence number of the received RTP packet and determines the number of missing RTP packets among the consecutive sequence numbers Can be calculated.

The LQMS 100 can calculate the continuous missing packet count, which is the number of continuously missing packets among the missing RTP packets, using the sequence number of the stored RTP packets.

Also, the LQMS 100 can calculate the redundant packet count, which is the number of packets corresponding to the redundant sequence number, using the sequence number of the stored RTP packet.

The LQMS 100 subtracts the sequence number of the first captured packet of the RTP packet (Tx-RTP) from the " highest sequence number received " of the RTCP for the RTP packet transmitted before the RTP packet (Tx- The first packet missing coefficient can be calculated.

That is, "highest sequence number received" means the highest sequence number among the sequence numbers of the RTP packets transmitted from the transmitting side 200, and the sequence number of the first captured packet in the "highest sequence number received" The number of missing packets having a sequence number lower than that of the RTP packet having the lowest sequence number among the captured packets can be calculated.

The LQMS 100 subtracts the sequence number of the last captured packet of the RTP packet (Tx-RTP) from the " highest sequence number received " of the RTCP of the transmitting side with respect to the RTP packet (Tx- The coefficient can be calculated.

That is, if the sequence number of the last packet captured in the " highest sequence number received " of the current RTCP is subtracted, the number of missing packets having a sequence number higher than that of the RTP packet having the highest sequence number among the captured packets have.

The jitter information indicates a deviation of the inter-ARIB time, and can be calculated using the capture time of the RTP packet (Tx-RTP) and the time stamp of the RTP packet (Tx-RTP).

The LQMS 100 can store the time of capture each time it receives an RTP packet (Tx-RTP), which is referred to as the capture time of the RTP packet (Tx-RTP). Each time the UE 200 transmits an RTP packet (Tx-RTP), the UE 200 records the transmission time in an RTP packet (Tx-RTP) and transmits the RTP packet (Tx-RTP).

For example, if the time stamp and capture time of the first RTP packet are t and t ', respectively, and the time stamp and capture time of the second RTP packet are t + a and t' + a ', respectively, t '+ a') - t '= a' and (t + a) -t = a becomes the transmission interval. Even if the transmission interval is constant due to the instability of the network communication, the inter-affiliation time may not be constant and fluctuate, and the jitter information is indicative of such a change (variation of the inter-affiliation time).

The delay information means a time delay required for transmission and reception of an RTP (or RTCP) packet in a first interval, and a method for calculating delay information for each interval (a first interval, a second interval, or an entire interval) Will be described later.

The LQMS 100 may calculate the voice quality for the first period using the packet loss information, the jitter information, and the delay information.

The voice quality can be represented by an R-factor, which is a value obtained by quantifying the voice quality of the corresponding network. For example, the R-factor may have a positive value between 0 and 100, and the higher the R-factor, the better the quality.

The R-factor can be calculated by Equation 1 described in FIG. Equation 1 is based on the e-model R-factor calculation method of the ITU-T-REC-G.107 standard, but the scope of the present invention is not limited thereto. In Equation (1), the meanings of the parameters Ro, Is, Id, Ie and A are the same as those described in Fig.

The LQMS 100 can model parameters of each of Ro, Is, Id, Ie and A by referring to a reference table stored in advance based on the collected characteristics of the RTP packet and the RTCP.

In particular, the jitter information, the delay information, and the RTP packet size may be used as inputs in the calculation of Id. Also, the packet loss information can be used as an input in the calculation of Ie.

That is, Ie is a value reflecting the packet loss information for the first section, Id is a value reflecting the jitter information for the first section and the delay information.

Therefore, the LQMS 100 calculates R (Rx) for the first interval based on the packet loss information, the jitter information, and the delay information, which are information generated in the process of receiving the RTP packet (Tx-RTP) -Factor (Tx-MOS_Air) can be calculated.

The UE 400 receives the RTP packet (Tx-RTP) and can generate RTCP (Rx-RTCP), which is information on the quality of the RTP packet (Tx-RTP), and transmit it to the UE 200.

The LQMS 100 can calculate the R-factor (Tx-MOS_All) for the entire section on the basis of the RTCP (Rx-RTCP), which is information on the quality of the RTP packet (Tx-RTP). This is because the RTCP (Rx-RTCP) is information on the quality of the RTP packet (Tx-RTP) received from the UE 200 to the UE 400.

The LQMS 100 may reflect three pieces of information obtained from the RTCP (Rx-RTCP) when calculating the speech quality for the entire section.

The three pieces of information correspond to packet loss information, jitter information, and delay information.

The packet loss information may include an expected received packet count, a missing packet count, and the like.

The LQMS 100 subtracts the highest sequence number received of the RTCP from the "highest sequence number received" of the currently captured RTCP (Rx-RTCP) for the RTP packet transmitted before the RTP packet (Tx-RTP) It is possible to calculate the expected reception packet coefficient which is the total packet number of the RTP packet (Tx-RTP) corresponding to the Rx-RTCP.

The LQMS 100 subtracts the " cumulative number of packet lost " of the RTCP for the RTP packet transmitted before the RTP packet (Tx-RTP) from the " cumulative number of packet lost " of the currently captured RTCP , The missing packet count which is the number of packets missing in the RTP packet (Tx-RTP) corresponding to the RTCP (Rx-RTCP) can be calculated.

The " cumulative number of packet lost " means the accumulated number of missing packets. A late received packet is not regarded as a missing packet, and may be negative when there are duplicated packets. The subtraction of "cumulative number of packet lost" can be made by taking two's complement for the case that "cumulative number of packet lost" becomes negative.

The jitter information means a deviation of the inter-ARIB time and can be obtained from the " interarrival jitter " of the currently captured RTCP (Rx-RTCP).

The interarrival jitter is a deviation of the inter-ARIB time from the UE 400 using the capture time of the RTP packet (Tx-RTP) and the time stamp of the RTP packet (Tx-RTP).

The delay information indicates a time delay required for transmission and reception of an RTP (or RTCP) packet in the entire interval. A method of calculating delay information for each interval (the first interval, the second interval, or the entire interval) Will be described later.

The LQMS 100 may calculate the voice quality for the entire interval using the packet loss information, the jitter information, and the delay information.

The voice quality may be represented by an R-factor as well as the voice quality for the second section. That is, the R-factor can be calculated by Equation (1), and the LQMS 100 can model parameters of each of Ro, Is, Id, Ie and A by referring to a reference table stored in advance based on the RTCP .

In particular, the jitter information and the delay information can be used as inputs in the calculation of Id. Also, the packet loss information can be used as an input in the calculation of Ie.

That is, Ie is a value reflecting the packet loss information for the entire section, and Id is a value reflecting the jitter information for the entire section and the delay information.

Therefore, the LQMS 100 calculates R-Factor (R-Factor) for the entire interval based on the packet loss information, the jitter information, and the delay information, which are information generated in the process of receiving the RTP packet (Tx- (Tx-MOS_All) can be calculated.

The LQMS 100 subtracts the R-factor (Tx-MOS_Air) representing the transmission speech quality for the first section from the R-factor (Tx-MOS_All) representing the transmission speech quality for the entire section ) And R-factor (Tx-MOS_NW) indicating the transmission speech quality for the second section. That is, the LQMS 100 can obtain only the transmission speech quality for the second section of the entire section.

Here, the second interval is a network interval between the LQMS 100 and the UE 400 as a large terminal.

Therefore, the LQMS 100 recognizes the received voice quality of the network section between the LQMS 100 and the UE 400 based on the R-factor (Tx-MOS_NW) indicating the received voice quality for the second period . That is, the LQMS 100 can collect data on the transmission voice quality of the wireless network of the base station to which the UE 400 is connected, and provide the collected data to the network manager. Or the LQMS 100 may provide the network manager with information about base stations having a transmission voice quality that does not meet preset reference values among a plurality of base stations. Accordingly, the network manager can identify the base station whose reception voice quality has deteriorated to such a degree that a quality check is required, and can promptly perform quality check or facility investment, thereby enhancing user satisfaction with network quality.

The LQMS 100 also compares the R-factor (Tx-MOS_Air) with the R-factor (Tx-MOS_NW) It can be judged whether it is a section.

5 is a diagram for explaining a method of calculating delay information for each network section.

Referring to FIG. 5, a method of calculating delay information, which is a time delay required for transmission and reception of an RTP (or RTCP) packet in a first interval, will be described. The UE 200 or the UE 400 can simultaneously transmit the RTP packet and the RTCP packet to the counterpart side, and in the following description, it is assumed that the time delay required for transmission and reception of the RTP packet and the RTCP packet is the same .

The time delay required for transmission and reception of RTP (or RTCP) packets in the first interval can be obtained by a first round trip time (RTT). RTT means packet round-trip time.

The UE 400 transmits Rx-RTCP to the UE 200 through the LQMS 100 and the eNB / SPGW 300, and the LQMS 100 can record the capture time of the Rx-RTCP.

Thereafter, the UE 200 can receive Rx-RTCP and calculate DLSR (delay since last SR) and transmit it by including it in Tx-RTCP. The DLSR indicates the time from the reception of the last RTP packet to the transmission of the Tx-RTCP. That is, the DLSR indicates a delay time taken until the UE 200 receives the last RTP packet and transmits the Tx-RTCP.

When receiving the Tx-RTCP transmitted from the UE 200, the LQMS 100 can record the capture time of the Tx-RTCP.

The LQMS 100 can obtain the first RTT by subtracting the capture time of the Rx-RTCP from the capture time of the Tx-RTCP and subtracting the DLSR from the capture time of the Rx-RTCP. 5, the first RTT indicates a time (1) from the capture time of the Rx-RTCP of the LQMS 100 to the time when the reception of the last RTP packet from the UE 200 is completed and the time when the UE 200 receives the Tx- (2) of the LQMS 100 to the capture time of the Tx-RTCP.

Therefore, the first RTT represents the delay information for the first interval since it is the time delay required for transmission and reception of the RTP (or RTCP) packet in the first interval.

Now, a method for calculating delay information, which is a time delay required for transmission and reception of an RTP (or RTCP) packet in a second interval, will be described.

The time delay required for transmission and reception of RTP (or RTCP) packets in the second interval can be obtained by a second round trip time (RTT).

As described above, the UE 200 transmits the Tx-RTCP to the UE 400 through the eNB / SPGW 300 and the LQMS 100, and the LQMS 100 can record the capture time of the Tx-RTCP .

Then, the UE 400 can receive the Tx-RTCP, calculate the DLSR, and transmit it by including it in the Rx-RTCP. The DLSR indicates the time from the reception of the last RTP packet to the transmission of Rx-RTCP. That is, the DLSR indicates a delay time required for the UE 400 to receive the last RTP packet and transmit the Rx-RTCP.

When the LQMS 100 receives the Rx-RTCP transmitted from the UE 400, the LQMS 100 can record the capture time of the Rx-RTCP.

The LQMS 100 can obtain the second RTT by subtracting the capture time of Tx-RTCP from the capture time of Rx-RTCP and subtracting the DLSR from the capture time of Tx-RTCP. 5, the second RTT indicates the time (4) from the capture time of the Tx-RTCP of the LQMS 100 to the time when the reception of the last RTP packet from the UE 400 is completed and the time (⑤) from the sending time to the capture time of the Rx-RTCP of the LQMS (100).

Accordingly, the second RTT represents the delay information for the second interval since it is a time delay required for transmission and reception of the RTP (or RTCP) packet in the second interval.

The LQMS 100 may calculate delay information for the entire interval by summing the first RTT and the second RTT.

According to the voice quality monitoring method for each network section according to the embodiment of the present invention, the voice quality of a specific wireless network can be confirmed. By analyzing the voice quality degradation section through such information, operational optimization for improving service quality, Investment and other measures.

The method described above can be implemented as computer-readable code on a computer-readable recording medium. The computer-readable recording medium includes all kinds of recording media storing data that can be decoded by a computer system. For example, it may be a ROM (Read Only Memory), a RAM (Random Access Memory), a magnetic tape, a magnetic disk, a flash memory, an optical data storage device, or the like. In addition, the computer-readable recording medium may be distributed and executed in a computer system connected to a computer network, and may be stored and executed as a code readable in a distributed manner.

It will be apparent to those skilled in the art that various modifications and variations can be made in the present invention without departing from the spirit or scope of the invention as defined in the appended claims. It will be understood that various modifications and changes may be made.

Claims (16)

Calculating a voice quality for a second section based on information obtained from an RTP (Real Time Protocol) packet transmitted from the second terminal to the first terminal;
Calculating a speech quality for the entire interval based on information obtained from a Real Time Transport Control Protocol (RTCP) generated by the first terminal that received the RTP packet; And
And calculating a speech quality for the first section by subtracting the speech quality for the second section from the speech quality for the entire section,
Wherein the entire interval includes the first interval from the first terminal to the LQMS (LTE Quality Measurement System), and the second interval from the LQMS to the second terminal. The method according to claim 1,
Wherein the information obtained from the RTP packet includes:
A lost packet count, a burst packet loss count, a duplicate packet count, an initial packet lost count, and a last packet lost count. The packet loss information including at least one of the packet loss information and the packet loss information. The method according to claim 1,
Wherein the information obtained from the RTP packet includes:
Wherein the jitter information is calculated using a capture time of the RTP packet and a time stamp of the RTP packet. The method according to claim 1,
Wherein the information obtained from the RTP packet includes:
(DLSR) of the second terminal is subtracted from the capture time of the RTCP generated by the first terminal from the capture time of the RTCP generated by the second terminal, A method for monitoring voice quality per network section. The method according to claim 1,
The information obtained from the RTCP includes,
Packet loss information including at least one of an expected RTP packet count received and a lost packet count. The method according to claim 1,
The information obtained from the RTCP includes,
And jitter information obtained from the " interarrival jitter " of the RTCP. The method according to claim 1,
The information obtained from the RTCP includes,
The delay information calculated by subtracting the DLSR of the second terminal from the capture time of the RTCP generated by the first terminal from the capture time of the RTCP generated by the second terminal, And delay information calculated by subtracting the DLSR of the first terminal from a result of subtracting the capture time of the RTCP generated by the second terminal from the capture time of the RTCP. Calculating a speech quality for a first interval based on information obtained from an RTP (Real Time Protocol) packet transmitted from the first terminal to the second terminal;
Calculating a speech quality for the entire interval based on information obtained from a Real Time Transport Control Protocol (RTCP) generated by the second terminal that received the RTP packet; And
And calculating a speech quality for the second section by subtracting the speech quality for the first section from the speech quality for the entire section,
Wherein the entire interval includes the first interval from the first terminal to the LQMS (LTE Quality Measurement System), and the second interval from the LQMS to the second terminal. 9. The method of claim 8,
Wherein the information obtained from the RTP packet includes:
A lost packet count, a burst packet loss count, a duplicate packet count, an initial packet lost count, and a last packet lost count. The packet loss information including at least one of the packet loss information and the packet loss information. 9. The method of claim 8,
Wherein the information obtained from the RTP packet includes:
Wherein the jitter information is calculated using a capture time of the RTP packet and a time stamp of the RTP packet. 9. The method of claim 8,
Wherein the information obtained from the RTP packet includes:
Wherein the first terminal extracts the DLSR of the first terminal from the capture time of the RTCP generated by the second terminal from the capture time of the RTCP generated by the first terminal, Way. 9. The method of claim 8,
The information obtained from the RTCP includes,
Packet loss information including at least one of an expected RTP packet count received and a lost packet count. 9. The method of claim 8,
The information obtained from the RTCP includes,
And jitter information obtained from the " interarrival jitter " of the RTCP. 9. The method of claim 8,
The information obtained from the RTCP includes,
The delay information calculated by subtracting the DLSR of the second terminal from the capture time of the RTCP generated by the first terminal from the capture time of the RTCP generated by the second terminal, And delay information calculated by subtracting the DLSR of the first terminal from a result of subtracting the capture time of the RTCP generated by the second terminal from the capture time of the RTCP. In an LQMS (LTE Quality Measurement System) for monitoring voice quality per network section,
The second terminal calculates the voice quality for the second section based on the information obtained from the RTP (Real Time Protocol) packet transmitted to the first terminal, and transmits the RTCP The speech quality of the entire section is calculated based on the information obtained from the Real Time Transport Control Protocol, and the speech quality of the first section is subtracted from the speech quality of the entire section, And a quality calculation unit for calculating the quality of the image,
Wherein the entire interval comprises the first interval from the first terminal to the LQMS and the second interval from the LQMS to the second terminal. In an LQMS (LTE Quality Measurement System) for monitoring voice quality per network section,
The first terminal calculates a voice quality for a first interval based on information obtained from an RTP (Real Time Protocol) packet transmitted to the second terminal, and the RTCP The speech quality of the entire section is calculated based on the information obtained from the Real Time Transport Control Protocol and the speech quality of the first section is subtracted from the speech quality of the entire section, And a quality calculation unit for calculating the quality of the image,
Wherein the entire interval comprises the first interval from the first terminal to the LQMS and the second interval from the LQMS to the second terminal.

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