KR102063976B1 - Variable buffering method for improving mouse-to-ear latency and voice quality on mission critical push to talk in wireless communication system and device therefor - Google Patents

Abstract

The present invention is to reduce the mouse-to-ear delay time by reducing the unnecessary buffering by performing a buffering for improving the voice quality in accordance with the MCPTT wireless communication environment or CPU loading environment in a wireless communication system supporting group communication A variable buffering method and apparatus are disclosed.
In the wireless communication system of the present invention for achieving this, the variable buffering method for improving the speech delivery time and the voice quality for the MCPTT service includes a transmission jitter observing unit installed in the PTT server, and how uniform the voice packets are continuously transmitted by the transmitting terminal. Continuously and in real time to monitor whether or not it is reached, while the de-jitter buffer management unit is installed in the app of the receiving terminal, by controlling the buffering in the OS unit, App unit, voice decoder stage according to the situation to reduce unnecessary buffering To be able.

Description

VARIABLE BUFFERING METHOD FOR IMPROVING MOUSE-TO-EAR LATENCY AND VOICE QUALITY ON MISSION CRITICAL PUSH TO TALK IN WIRELESS COMMUNICATION SYSTEM AND DEVICE THEREFOR }

The present invention relates to a variable buffering method and apparatus for improving speech delivery time and voice quality for an MCPTT service in a wireless communication system supporting group communication, and more particularly, in an MCPTT wireless communication system in a wireless communication system supporting group communication. The present invention relates to a variable buffering method and apparatus for reducing mouse-to-ear delay time by variably performing buffering for improving voice quality according to an environment or a CPU loading environment.

Recently, interest in special purpose mobile communication systems such as disaster safety nets and integrated wireless networks dedicated to railway networks is increasing, and most special purpose mobile communication systems are being built on LTE that is being used in commercial applications.

In such a special purpose dedicated mobile communication system, it is essential to provide a group communication service such as push-to-talk (PTT), and this service has a requirement to be provided in various environments such as a disaster situation.

3GPP has standardized the work item named MCPTT (Mission Critical Push-To-Talk) in connection with this PTT service, and utilizes direct communication between terminals as well as within mobile system coverage. It also defines services outside of one coverage and discusses technologies for them.

The most important performance indicator in MCPTT is latency, especially required by MCPTT to be less than 300ms for talk-to-ear latency.

Meanwhile, Mean Opinion Score (MOS), which is a performance index corresponding to sound quality in all voice services, is one of the important performance indicators. It is prescribed.

In a mobile communication system, a delay of several tens of microseconds to hundreds of microseconds may be additionally generated at a wireless access terminal according to a change in a wireless signal environment, and thus, a voice packet may arrive late or unbalanced in some cases.

In addition, depending on the amount of work being performed in the OS of the receiving terminal, the delay time of receiving processing for the voice packet may occur unbalanced.

In addition, the receiving packet may be unbalanced due to the CPU processing of the receiving terminal or the change of the wireless environment, and even in this situation, the receiving terminal may use a de-jitter buffer to satisfy a certain voice quality level. Voice packets received through the system are buffered and played back, but additional delay time occurs in proportion to the length of the buffering, which is not desirable in a disaster environment where mouse-to-ear delay time is important. It is necessary to optimize the buffering length to satisfy the to-ear delay time at the same time.

1 shows an overall configuration diagram of a general group communication system, in which two or more terminals 10 (transmitting terminal 10A and receiving terminal 10B) and a base station 22 / core network 24 performing group communication are shown. LTE network 20, and a PTT server 30 that provides a group communication service.

The terminal 10 transmits and receives voices, and at a specific moment, only one terminal in the group, that is, the transmitting terminal 10A, transmits a speech, and the receiving terminals 10B, which are terminals in the other group, receive the speech at a specific moment. In this case, before one terminal 10 in the group speaks, the PTT server 30 requests a speaking right and obtains it, and then speaks.

The PTT server 30 forms a terminal group, mediates the right to speak requested by the terminals in the group, and receives the speech of the transmitting terminal 10A and delivers the speech to the receiving terminals 10B.

LTE network 20 composed of the base station 22 / core network 24 between the transmitting terminal (10A) and the receiving terminal (10B) is a control message or voice packet related to the speech right arbitration through wireless communication with the terminals (10) Send and receive, and serves to provide wired and wireless connectivity between the terminal 10 and the PTT server 30.

FIG. 2 is a diagram illustrating a voice information transmission path in a group communication system, and converts speech received from the microphone 11 of the transmitting terminal 10A into compressed voice packet information through a voice encoder 12. This is transmitted to the modem 14 terminal for wireless transmission via the transmission terminal side OS / APP unit 13, the transmission is made to the LTE network (20).

When a voice packet comes into the PTT server 30 through the LTE network 20, the PTT server 30 copies and transmits the same voice packet to the receiving terminals 10B in the group.

When the receiving terminal 10B enters the modem 15 terminal through wireless communication, the receiving terminal 10B transmits the voice packet to the OS / App unit 16 on the receiving terminal side, and through the voice decoder 17. Decoded and finally reproduced by the speaker 18.

In general, the receiving terminal 10B performs an operation of performing a certain level of buffering using the de-jitter buffer without reproducing the voice packet received from the transmitting terminal 10A, which is used for non-uniform reception of the voice packet. To prevent deterioration of voice quality.

Due to the wired / wireless communication process between the transmitting terminal 10A and the receiving terminal 10B and an unbalanced processing time in the OS / App unit, the voice packet may be imbalanced, and the characteristics of the voice reproducing that must be continuously reproduced such as streaming This imbalance can directly affect voice quality.

3 is a diagram illustrating a voice quality degradation state due to unbalanced reception of a packet in a conventional group communication system, and RTP (Real Time Protocol) is a protocol for supporting real-time multimedia data. For example, the RTP data transmitted from the transmitting terminal 10A is sequentially transmitted from the RTP data (seq: 161) to the RTP data (seq: 333) in sequence order, for example, between the transmitting terminal 10A and the LTE network 20. Some voice packets (eg, RTP data (seq: 250, 251, 252)) arrive late due to a weak electric field, or a weak electric field is generated between the LTE network 20 and the receiving terminal 10B. For example, when the RTP data (seq: 330, 331, 332) arrives late, when the voice portion of the late arriving packet has already passed in time while the voice is being played, the late arriving packet The voice portion (T11, T12 section of the voice waveform shown on the right side of FIG. 3) is processed by silence, and in this case, the voice quality is seriously degraded.

Therefore, the receiving terminal 10B buffers the packet in the corresponding de-jitter buffers 19A and 19B at the OS / App unit 16 or the voice decoder 17 stage, and plays back the voice after a certain level of buffering is performed. As shown in FIG. 4, even when some packets are unbalanced, such as RTP data (seq: 250, 251, 252) or RTP data (seq: 330, 331, 332), the delayed regeneration is performed. Since the probability of timely performing packet decoding and reproducing in the normal voice reproducing process increases, the packets that arrive late by the weak electric field are normally regenerated as in the T21 and T22 sections of the voice waveforms shown on the right side of FIG. 4. Voice quality deterioration does not occur.

However, due to the nature of disaster environment, Mouth-to-ear needs to be within 300ms. However, if the existing buffering technique is used as it is, the basic delay due to buffering occurs, so it is difficult to satisfy the criteria of 300ms. .

Especially in the case of a good wireless signal environment or CPU loading condition, the unnecessary buffering may increase the mouth-to-ear delay time, and accordingly, the necessity of varying the buffering for improving the voice quality is needed. have.

Republic of Korea Patent Publication No. 10-2014-0103121 (published Aug. 25, 2014) Republic of Korea Patent Publication No. 10-2008-0094843 (published 24 October 2008)

The present invention has been invented in view of the above circumstances, and it is possible to reduce mouse-to-ear delay time by variably performing buffering for improving voice quality according to MCPTT wireless communication environment or CPU loading environment. It is a technical problem to provide a variable buffering method and apparatus for improving speech delivery time and voice quality for an MCPTT service in a wireless communication system.

In addition, the present invention provides a transmission jitter observing unit in the PTT server, and continuously and in real time monitor how uniformly the voice packets transmitted continuously by the transmitting terminal is reached, while the de-jitter buffer management unit in the App portion of the receiving terminal Variable buffering method for improving speech delivery time and voice quality for the MCPTT service in a wireless communication system to reduce unnecessary buffering by managing the buffering at the OS unit, the App unit, and the voice decoder stage according to the situation. And it is a technical problem to provide an apparatus.

를 결정하고, 결정된

로 수신단말에 디-지터 버퍼 길이를 가변으로 조정하도록 명령하는 것을 특징으로 한다.In the wireless communication system of the present invention for solving this technical problem, in the variable buffering method for improving speech delivery time and voice quality for the MCPTT service, a plurality of terminals including a transmitting terminal and a receiving terminal through a single PTT server A variable buffering method for improving speech delivery time and voice quality for an MCPTT service in a wireless communication system performing group communication, comprising: a transmission jitter observing unit configured to observe transmission jitter of a transmission terminal in a PTT server, and continuously from a transmission terminal If it is determined whether the arrival time interval of the received voice packet is uniform, and it is determined that the arrival time interval is not uniform, then the minimum de-jitter buffer length that the receiving terminal should set is based on the distribution of the measured voice packet arrival time interval.

Determined, determined

It is characterized by instructing the receiving terminal to variably adjust the de-jitter buffer length.

The transmission jitter observing unit of the PTT server measures the statistical distribution of the arrival time intervals of the received voice packets when measuring whether the arrival time intervals of the voice packets received from the transmitting terminal are uniform.

The statistical distribution of arrival time intervals of the voice packet is characterized in that the average and variance of the arrival time interval of the voice packet.

The de-jitter buffer length commanded by the transmitting jitter observer of the PTT server to be set to the receiving terminal is a result of obtaining a de-jitter buffer length through which the MOS index can obtain a predetermined score or more from the receiving terminal through previous experiments. It is characterized by calculating through the mapping from.

를 결정하고, 결정된

로 디-지터 버퍼 길이를 가변으로 조절하는 것을 특징으로 한다.The de-jitter manager of the receiving terminal is a de-jitter buffer length that the receiving terminal should set based on the received signal sensitivity of the voice packet received from the PTT server or the load environment of the OS.

Determined, determined

Low-jitter buffer is characterized in that the variable length.

The de-jitter buffer manager of the receiving terminal calculates a Reference Signal Received Power (RSRP) or Signal to Interference and Noise Ratio (SINR), which is an indicator of received signal sensitivity of the receiving terminal, and de-jitter buffer based on the RSRP and SINR. It is characterized by adjusting the length.

The load environment of the OS unit may calculate a load (%) of the CPU of the receiving terminal and determine the de-jitter buffer length based on the load (%) of the CPU.

The receiving terminal is characterized in that to calculate the length of the de-jitter buffer that the MOS index can obtain a predetermined score or more from the result obtained through the previous experiment through mapping.

와 디-지터 관리부에 의해 설정되는

에 기반하여 디-지터 버퍼 길이를 결정하는 것을 특징으로 한다.The receiving terminal is received from a PTT server

Set by the di-jitter manager

Determining the de-jitter buffer length based on the.

The receiving terminal may adjust the length of the de-jitter buffer by using the determined de-jitter buffer length and play the voice packet received from the transmitting terminal.

를 설정하여 수신단말이 디-지터 버퍼 길이를 설정하도록 명령하는 송신 지터 관측부; 상기 수신단말의 App부에 구비되어, PTT 서버로부터 수신하는 음성 패킷의 수신 신호 감도나 OS의 부하량 환경을 기반으로 수신단말이 설정해야 할 디-지터 버퍼 길이인

를 결정하고, 결정된

로 디-지터 버퍼 길이를 가변으로 조절하도록 된 디-지터 관리부;를 포함하여 이루어진 것을 특징으로 한다.In addition, in the wireless communication system of the present invention for solving the above technical problem, a variable buffering device for improving speech delivery time and voice quality for the MCPTT service, LTE network consisting of a transmitting terminal, a receiving terminal, a base station / core network, and A variable buffering device for improving speech delivery time and voice quality for an MCPTT service in a wireless communication system including a PTT server providing a group communication service between a transmitting terminal and a receiving terminal, wherein the PTT server is provided with a voice from a transmitting terminal. When a packet is received, the time interval between arrivals of voice packets is measured, and the minimum de-jitter buffer length that the receiving terminal should set based on the statistical distribution of time intervals between voice packets.

A transmission jitter observing unit for instructing the receiving terminal to set the de-jitter buffer length by setting a? It is provided in the App unit of the receiving terminal, the de-jitter buffer length to be set by the receiving terminal based on the received signal sensitivity of the voice packet received from the PTT server or the load environment of the OS

Determined, determined

And a de-jitter manager configured to variably adjust the low-jitter buffer length.

The statistical distribution of arrival time intervals of the voice packet is characterized in that the average and variance of the arrival time interval of the voice packet.

The de-jitter buffer length commanded by the transmitting jitter observer of the PTT server to be set to the receiving terminal is a result of obtaining a de-jitter buffer length through which the MOS index can obtain a predetermined score or more from the receiving terminal through previous experiments. It is characterized by calculating through the mapping from.

The receiving terminal is characterized in that to calculate the length of the de-jitter buffer that the MOS index can obtain a predetermined score or more from the result obtained through the previous experiment through mapping.

와 디-지터 관리부에 의해 설정되는

에 기반하여 디-지터 버퍼 길이를 결정하는 것을 특징으로 한다.The receiving terminal is received from a PTT server

Set by the di-jitter manager

Determining the de-jitter buffer length based on the.

According to the present invention, the de-jitter buffer can be variably adjusted according to various wireless communication situations of the transmitting and receiving terminals, and thus, MOS corresponding to Mouth-to-Ear performance and voice quality corresponding to voice delay time according to the instant situation. You can optimize performance at the same time.

In addition, when a voice packet arrives poorly at the PTT server due to poor reception signal sensitivity of the transmitting terminal, the de-jitter buffer may be lengthened to prevent sudden degradation of voice quality.

In addition, when the voice signal arrives to the server satisfactorily due to the improved reception signal sensitivity of the transmitting terminal, the mouth-to-ear delay time can be minimized by shortening the de-jitter buffer length.

On the other hand, when a voice packet arrives poorly due to poor reception signal sensitivity or the like of the receiving terminal, the de-jitter buffer length can be lengthened to prevent sudden degradation of voice quality.

When the voice packet arrives satisfactorily due to the improved reception signal sensitivity of the receiving terminal, the mouth-to-ear delay time can be minimized by shortening the buffer length.

When the voice packet is unevenly processed due to the increased load of the OS of the receiving terminal, the de-jitter buffer length is increased to prevent sudden deterioration of voice quality, while the voice packet is reduced due to the reduced load of the OS of the receiving terminal. For faster processing, shorter de-jitter buffer lengths can minimize mouth-to-ear latency.

As described above, if the voice quality is improved by minimizing Mouth-to-Ear delay time by variably adjusting the de-jitter buffer, the group communication quality of the disaster terminal user may be improved, thereby contributing to increasing the success rate of disaster task performance. .

In addition, if the Mouth-to-Ear delay time is minimized according to the environment, disaster group communication users can promptly communicate, which can contribute to the successful execution of the disaster mission through smooth communication.

In addition, if the MOS index, which is the voice quality, is maximized according to the environment, disaster group communication users can communicate more clearly in the event of a disaster such as a fire, thereby contributing to the successful execution of the disaster mission through smooth communication.

1 is an overall configuration diagram of a general group communication system,
2 is a detailed configuration diagram of a conventional group communication system for explaining a voice information transmission path;
FIG. 3 is an exemplary diagram illustrating voice quality deterioration due to unbalanced reception of a packet in a conventional group communication system. FIG.
4 is an exemplary diagram illustrating overcoming voice quality deterioration due to unbalanced reception of packets through buffering in a conventional group communication system;
5 is a diagram illustrating a detailed configuration of a group communication system according to the present invention;
FIG. 6 is an exemplary diagram illustrating a procedure of detecting a PTT server and adjusting a de-jitter buffer length to minimize degradation of voice quality when a wireless signal environment becomes poor in a transmission period according to the present invention.
FIG. 7 is an exemplary view illustrating a procedure of detecting a PTT server and adjusting a de-jitter buffer length to minimize Mouth-to-Ear delay time when a wireless signal environment is recovered in a transmission interval according to the present invention.
8 is an exemplary view illustrating a procedure of adjusting a de-jitter buffer length by a receiving terminal when packet reception occurs unevenly due to deterioration of a wireless signal receiving environment at a modem terminal of a receiving terminal according to the present invention;
FIG. 9 is an exemplary view illustrating a procedure of adjusting a de-jitter buffer length by a receiving terminal when packet processing occurs unevenly due to deterioration of the CPU loading environment of the OS at the receiving terminal according to the present invention.

Specific structural or functional descriptions of the embodiments according to the inventive concept disclosed herein are provided only for the purpose of describing the embodiments according to the inventive concept. It may be embodied in various forms and is not limited to the embodiments described herein.

Embodiments according to the inventive concept may be variously modified and have various forms, so embodiments are illustrated in the drawings and described in detail herein. However, this is not intended to limit the embodiments in accordance with the concept of the invention to the specific forms disclosed, and includes all changes, equivalents, or substitutes included in the spirit and scope of the present invention.

Terms such as first or second may be used to describe various components, but the components should not be limited by the terms. The terms are used only for the purpose of distinguishing one component from another, for example without departing from the scope of the rights according to the inventive concept, and the first component may be called a second component and similarly the second component. The component may also be referred to as the first component.

When a component is said to be "connected" or "connected" to another component, it may be directly connected to or connected to that other component, but it may be understood that other components may exist in the middle. Should be. On the other hand, when a component is said to be "directly connected" or "directly connected" to another component, it should be understood that no other component exists in the middle. Other expressions describing the relationship between components, such as "between" and "immediately between" or "neighboring to" and "directly neighboring", should be interpreted as well.

The terminology used herein is for the purpose of describing particular example embodiments only and is not intended to be limiting of the invention. Singular expressions include plural expressions unless the context clearly indicates otherwise. In this specification, terms such as "comprise" or "have" are intended to indicate that there is a feature, number, step, action, component, part, or combination thereof described herein, but one or more other features. It is to be understood that the present invention does not exclude the possibility of the presence or the addition of numbers, steps, operations, components, parts, or combinations thereof.

Unless defined otherwise, all terms used herein, including technical or scientific terms, have the same meaning as commonly understood by one of ordinary skill in the art. Terms such as those defined in the commonly used dictionaries should be construed as having meanings consistent with the meanings in the context of the related art, and are not construed in ideal or excessively formal meanings unless expressly defined herein. Do not.

FIG. 5 is a diagram illustrating a detailed configuration of a group communication system in which the present invention is performed, and similarly to the conventional group communication system illustrated in FIGS. 1 and 2, two or more terminals 10 (transmitting terminal 10A). And a LTE network 20 composed of a receiving terminal 10B), a base station 22 / core network 24, and the like, and a PTT server 30 that provides group communication services. Components having the same reference numerals as the illustrated components perform the same operations, and thus detailed descriptions thereof will be omitted in order to avoid redundant description. However, in addition to the conventional group communication system, the transmission jitter observing unit 36 is provided in the PTT server 30 to perform the present invention, and the OS / App unit in the receiving terminal 10B includes the OS unit 162 and the App. It can be separated into the unit 164, and is further provided with a de-jitter buffer management unit 166.

The transmission jitter observing unit 36 provided in the PTT server 30 continuously monitors in real time how uniformly the voice packets continuously transmitted by the transmitting terminal 10A are reached. The packet copying unit 32 serves to copy the packets received from the transmitting terminal 10A to the receiving terminals in the group in the same manner, and the packet distribution unit 34 copies the copied packets to each receiving terminal ( 10B) to transmit.

The de-jitter buffer management unit 166 provided in the receiving terminal 10B is operated by the OS unit 162 and voice decoder 17 to prevent deterioration of voice quality due to uneven reception of voice packets. It serves to manage the buffering of the buffers 19A and 19B variably according to circumstances.

Hereinafter, the detailed and exemplary operation of the present invention will be described in detail with reference to FIGS. 5 to 9.

를 설정한다.The transmission jitter observing unit 36 of the PTT server 30 should measure the arrival time interval between the received voice packets when receiving the voice packets from the transmitting terminal 10A and set them by the receiving terminal 10B based on the distribution. Is the minimum de-jitter buffer length

Set.

값을 수신단말(10B)에게 전달한다.The transmission jitter observing unit 36 of the PTT server 30 may be

The value is transmitted to the receiving terminal 10B.

를 설정한다.The OS section 162 or the modem 15 side of the receiving terminal 10B estimates how unbalanced the received voice packet is according to the CPU loading or the received signal sensitivity of the OS section 162.

Set.

The de-jitter buffer manager 166 of the receiving terminal 10B determines the size of the de-jitter buffers 19A and 19B in real time and the size of the de-jitter buffers 19A and 19B as follows.

는 전체 디-지터 버퍼 크기를 나타낸다.here,

Represents the total de-jitter buffer size.

Detailed operations of the PTT server 30 according to the present invention are as follows.

에 대해 아래와 같이 지속적으로 계산한다.The transmission jitter observing unit 36 of the PTT server 30 side is an inter-arrival time between voice packets transmitted and received by the transmitting terminal 10A.

Calculate continuously for

은 n번째 음성 패킷의 도달 시간을 나타내고,

은 n-1번째 음성 패킷의 도달 시간을 나타낸다.here,

Represents the arrival time of the nth voice packet,

Represents the arrival time of the n-1 th voice packet.

을 계산하여 분포를 산출하고, 산출된

분포에 대해 매핑 테이블을 이용하여 아래와 같이

를 산출한다.As such, the arrival time interval between voice packets in real time

To calculate the distribution,

Using the mapping table for the distribution,

Calculate

는 음성 패킷 간 도달 시간 간격의 평균값이며,

는 음성 패킷 간 도달 시간 간격의 분산값을 나타낸다.here,

Is the average of the arrival time intervals between voice packets,

Denotes the variance of the arrival time interval between voice packets.

매핑 함수는 실험을 통해 특정 기준을 맞출 수 있는 디-지터 버퍼 길이를 산출하는 형태로 정의를 하는 것으로, 현 MCPTT 시스템 상에서 평균이 mean, 분산이 var인 정규분포로 음성 트래픽을 발생시킬 때 평균 평가점수(Mean Opinion Score; MOS)이 3.0 + α (α는 margin을 두기 위한 고정값) 이상이 되는 최소한의 디-지터 버퍼 길이를 실험을 통해 구한다.

Mapping function D that meet specific criteria by experiment - that the definitions in the form of calculating a de-jitter buffer length, when the average of the generated voice traffic in the normal distribution, mean, variance var average over current MCPTT system Experimentally find the minimum de-jitter buffer length where the Mean Opinion Score (MOS) is greater than 3.0 + α (α is a fixed value for margining).

정보를 주기적으로 계산하고 이를 수신단말(10B)에 주기적으로 전달하거나 변화량이 일정 수준 이상 존재할 때에 이벤트성으로 전달한다.PTT server 30 through the above formula

The information is periodically calculated and delivered to the receiving terminal 10B periodically or when the amount of change exists over a certain level.

Next, detailed operations of the receiving terminal 10B side according to the present invention will be described.

를 아래와 같이 정의한다.The de-jitter buffer manager 166

Define as

는 실험을 통해 특정 기준을 맞출 수 있는 디-지터 버퍼 길이를 산출하는 형태로 정의를 하는 것으로, 현 MCPTT 시스템 상에서 수신 신호 환경과 관련된 지표인 Reference Signal Received Power (RSRP) 및 Signal to Interference and Noise Ratio (SINR)이 각각 a, b일 때 Mean Opinion Score (MOS) 평균 평가점수가 3.0 + α (α는 margin을 두기 위한 고정값) 이상이 되는 최소한의 디-지터 버퍼 길이를 실험을 통해 구한다.here

Is defined in the form of calculating the de-jitter buffer length that can meet specific criteria through experiments. Reference Signal Received Power (RSRP) and Signal to Interference and Noise Ratio, which are indicators related to the reception signal environment on the current MCPTT system, are defined. Experimentally find the minimum de-jitter buffer length that mean Mean Opinion Score (MOS) score is more than 3.0 + α (α is fixed value for margin) when (SINR) is a and b, respectively.

는 실험을 통해 특정 기준을 맞출 수 있는 디-지터 버퍼 길이를 산출하는 형태로 정의를 하는 것으로, 현 MCPTT 시스템 상에서 CPU 로딩이 c%일 때 Mean Opinion Score (MOS) 평균 펑가점수가 3.0 + α (α는 margin을 두기 위한 고정값) 이상이 되는 최소한의 디-지터 버퍼 길이를 실험을 통해 구한다.Also here

Is defined in the form of calculating the de-jitter buffer length that can meet specific criteria through experiments. Mean CPU pin average mean PGA score is 3.0 + α (c%) when the CPU load is c% on the current MCPTT system. α is experimentally determined to be the minimum de-jitter buffer length that is greater than or equal to a fixed value for margining.

를 주기적으로 계산한다.The de-jitter management unit 166 present in the App unit 164 of the receiving terminal 10B periodically receives RSRP, SINR, and CPU loading information from the modem 15 and the OS unit 162, through the above equation.

Calculate periodically.

정보를 조합하여 최종적으로 디-지터 버퍼(19A)(19B)의 길이인

를 항상 계산하고, 이 값이 일정 수준 이상 변경되면 보이스 디코더(17) 및 APP부(16)에서 운영하는 디-지터 버퍼 길이를

로 조정한다.The de-jitter management unit 166 present in the App unit 164 of the receiving terminal 10B receives from the PTT server 30.

Information is combined to finally determine the length of the de-jitter buffers 19A and 19B.

Is always calculated, and if this value is changed by a certain level or more, the length of the de-jitter buffer operated by the voice decoder 17 and the APP unit 16 is determined.

Adjust to.

As described above, in the wireless communication system supporting the group communication, the mouse-to-ear delay time can be reduced by reducing the unnecessary buffering by variably performing the buffering for improving the voice quality according to the MCPTT wireless communication environment or the CPU loading environment.

FIG. 6 is an exemplary diagram illustrating a procedure of detecting the PTT server 30 and adjusting the de-jitter buffer length accordingly when the wireless signal environment becomes poor due to weak electric field in the transmission period.

6 is a diagram illustrating an example of a base station in a process in which RTP data, which is a voice packet periodically transmitted by the transmitting terminal 10A, is transmitted uplink from sequence RTP data (seq: 161) to sequence RTP data (seq: 333) in sequence number order, for example. For example, the transmitter 10A enters the weak field and transmits from the RTP data (seq: 250) to the weak field.

The voice packet transmitted after the transmitting terminal 10A enters the weak electric field area is received unevenly in the receiving terminal 10B, so that the interval of the voice packet is delayed in time due to increased jitter, thereby delaying some sections. This silence is processed and the MOS deteriorates.

On the other hand, the voice packet transmitted by the transmitting terminal 10A in the weak field area is unbalanced to the PTT server 30, the arrival time interval distribution between voice packets is different from before.

를 다시 계산하여 새로운 값인

를 산출하고, 이 값을 수신단말(10B)들에게 전달한다.The PTT server 30 continuously calculates the mean and variance values in real time through statistics on the time interval samples, and when the variance value is greater than before, the PTT server 30

To recalculate the new value,

Is calculated, and the value is transmitted to the receiving terminals 10B.

값을 통해 최종 버퍼 길이인

값을 다시 계산하고, 이에 따라 디-지터 버퍼 길이를 더 길게 재조정한다.Receiving terminals 10B are delivered from the server

Value is the final buffer length

Recalculate the value, and readjust the de-jitter buffer length longer accordingly.

As a result, the playback of the received voice packet is delayed due to an increase in the buffer length, so that even if the voice packet arrives disproportionately late, the playback can be performed on time, and the MOS index is increased as the silence section is resolved.

FIG. 7 is an exemplary view illustrating a procedure of adjusting the de-jitter buffer length accordingly to minimize the Mouth-to-Ear delay time when the poor wireless signal environment is recovered in the transmission section in contrast to the case of FIG. 6. .

When RTP data, which is a voice packet, is periodically transmitted uplink to the PTT server 30 in the transmitting terminal 10A, if the PTT server 30 is, for example, before the RTP data (seq: 161), the transmitting terminal 10A When the RTP data transmitted is determined to be poor in the wireless signal environment and is adjusted according to the de-jitter buffer length, the transmitting terminal 10A is close to the base station from the time when the RTP data (seq: 161) is received. If the transmitting terminal 10A enters the strong electric field for reasons such as losing, this causes the receiving terminal 10B to receive voice packets more uniformly, resulting in faster arrival of the voice packets than the playback time due to jitter reduction, and Mouth-to. -Ear delay may occur.

Meanwhile, as the voice packet reaches the PTT server 30 more uniformly than before, the distribution of arrival time intervals between voice packets is different.

를 다시 계산하여 새로운 값인

를 산출하고, 이 값을 수신단말(10B)들에게 전달한다.When the PTT server 30 continuously calculates the average score and the variance value through statistics on arrival time interval samples in real time and detects that the variance value is smaller than before, the PTT server 30

To recalculate the new value,

Is calculated, and the value is transmitted to the receiving terminals 10B.

을 통해 최종 디-지터 버퍼 길이인

값을 다시 계산하고, 계산 결과에 따라 디-지터 버퍼 길이를 더 짧게 재조정한다.Receiving terminals 10B are delivered from the PTT server 30

Through the final de-jitter buffer length

Recalculate the value and readjust the shorter de-jitter buffer length depending on the result of the calculation.

Accordingly, the mouth-to-ear delay time can be further reduced by applying less time buffering without degrading the MOS due to the de-jitter buffer length reduction for voice packets received after the readjustment of the de-jitter buffer length. .

As such, when a poor wireless signal environment is recovered, a buffer length adjustment process for reducing unnecessary de-jitter buffer length and minimizing Mouth-to-Ear delay time is performed through a control operation in the PTT server 30.

8 shows a procedure in which the receiving terminal 10B adjusts the de-jitter buffer length accordingly when packet reception occurs unevenly due to deterioration of a wireless signal receiving environment in the modem 15 terminal on the receiving terminal 10B side. It is an exemplary figure which shows.

As the wireless signal reception environment of the receiving terminal 10B deteriorates, a receiving packet error frequently occurs, and thus, a time interval between arrivals of voice packets delivered to the App unit 164 of the receiving terminal 10B becomes uneven. As a result, MOS degradation occurs in the receiving terminal 10B.

값을 계산한다.The modem 15 terminal of the receiving terminal 10B transmits the deteriorated RSRP and SINR information to the App unit 164, and the App unit 164 of the receiving terminal 10B transmits the new information.

Calculate the value.

계산값을 기반으로 최종 버퍼 길이인

값을 다시 계산하고, 이에 따라 디-지터 버퍼 길이를 더 길게 재조정한다.In addition, the de-jitter management unit 166 of the App unit 164 may be a new device.

Based on the calculation, the final buffer length is

Recalculate the value, and readjust the de-jitter buffer length longer accordingly.

As a result, the playback of the received voice packet is delayed due to an increase in the buffer length, so that even if the voice packet arrives disproportionately late, the playback can be performed on time, and the MOS index increases as the silence section is resolved.

9 illustrates a procedure in which the receiving terminal 10B adjusts the de-jitter buffer length accordingly when packet processing occurs unevenly due to deterioration of the CPU loading environment in the OS unit 162 on the receiving terminal 10B side. It is an exemplary figure to show.

When the receiving terminal 10B temporarily receives an increase in the load of the OS unit due to an internal factor in the process of receiving a voice packet from the PTT server 30, and the CPU loading increases (for example, after RTP data (seq: 250)) As a result, the processing of the voice packet in the App unit 164 of the receiving terminal 10B is delayed, resulting in uneven reproduction of the voice packet, which causes MOS degradation.

값을 계산한다.At this time, the OS unit 162 of the receiving terminal 10B transmits an increase in CPU loading to the App unit 164, and the App unit 164 of the receiving terminal 10B receives a new value.

Calculate the value.

계산값을 기반으로 최종 버퍼 길이인

값을 다시 계산하고, 이에 따라 디-지터 버퍼 길이를 더 길게 재조정한다.Afterwards, in the App unit 164,

Based on the calculation, the final buffer length is

Recalculate the value, and readjust the de-jitter buffer length longer accordingly.

As a result, the playback of the voice packets received in the future is delayed due to an increase in the buffer length, so that even if the voice packets are disproportionately delayed, the playback can be performed on time, and the MOS is improved as the silence section is eliminated.

Subsequently, when the receiving terminal 10B receives the voice packet more uniformly because the load of the OS unit 162 is reduced or the receiving signal environment is changed, the receiving terminal 10B has a de-jitter buffer length. Short adjustment to improve Mouth-to-Ear performance.

When the load of the OS unit 162 where the receiving terminal 10B is temporarily increased is reduced again or the receiving signal environment is good and the RSRP or SINR is increased, this causes the voice packet at the App unit 164 of the receiving terminal 10B. There is room for further improvements in Mouth-to-Ear performance with shorter processing of de-jitter buffer lengths.

값을 계산한다.In this case, the OS unit 162 or the modem 15 terminal of the receiving terminal 10B transmits CPU loading or RSRP / SINR information to the App unit 164, and the App unit 164 of the receiving terminal 10B transmits the information. Through new

Calculate the value.

계산 값을 기반으로 최종 버퍼 길이인

값을 다시 계산하고, 이에 따라 디-지터 버퍼 길이를 더 짧게 재조정한다.Thereafter, the de-jitter buffer management unit 166 of the App unit 164 provides a new device.

Based on the calculated value

Recalculate the value and readjust the de-jitter buffer length accordingly.

Accordingly, due to the decrease in the buffer length of the voice packet received by the receiving terminal 10B in the future, less time buffering may be applied without MOS degradation, thereby further reducing mouth-to-ear delay time.

10-terminal, 10A-transmitting terminal,
10B-receiving terminal, 11-microphone,
12-voice encoder, 13-OS / APP section of the transmitting terminal,
14, 15-modem, 16-OS / App section,
162-OS part, 164-App part,
166-de-jitter buffer manager, 17-voice decoder,
19A, 19B-de-jitter buffer, 20-LTE network,
22-base station, 24-core network,
30-PTT server, 32-packet copy,
34-packet distributor, 36-transmit jitter observer.

Claims (15)

In a variable buffering method for improving speech delivery time and voice quality for an MCPTT service in a wireless communication system in which a plurality of terminals including a transmitting terminal and a receiving terminal perform group communication through a single PTT server,
The PTT server includes a transmission jitter observing unit for observing transmission jitter of the transmission terminal, and if it is determined whether the arrival time intervals of the voice packets continuously received from the transmission terminal are uniform, and then it is determined that they do not reach uniformly, Based on the distribution of voice packet arrival time intervals, the minimum de-jitter buffer length

Determined, determined

Command the receiving terminal to vary the de-jitter buffer length,
The de-jitter buffer length commanded by the transmitting jitter observer of the PTT server to be set to the receiving terminal is a result of obtaining a de-jitter buffer length through which the MOS index can obtain a predetermined score or more from the receiving terminal through previous experiments. Variable buffering method for improving speech delivery time and voice quality for MCPTT service in a wireless communication system, characterized in that it is calculated by mapping from.
The method of claim 1, wherein the transmission jitter observing unit of the PTT server measures the statistical distribution of the arrival time interval of the received voice packet when measuring whether the arrival time interval of the voice packet received from the transmitting terminal is uniform. Variable buffering method for improving speech delivery time and voice quality for MCPTT service in a wireless communication system.
The method of claim 2, wherein the statistical distribution of the arrival time interval of the voice packet is a mean and variance of the arrival time interval of the voice packet variable for improving speech delivery time and voice quality for the MCPTT service in a wireless communication system Buffering method.
delete The method of claim 1, wherein the de-jitter management unit of the receiving terminal is a length of the de-jitter buffer to be set by the receiving terminal based on the received signal sensitivity of the voice packet received from the PTT server or the load environment of the OS.

Determined, determined

A variable buffering method for improving speech delivery time and voice quality for a MCPTT service in a wireless communication system, characterized in that the low-jitter buffer length is controlled to be variable.
The receiver of claim 5, wherein the de-jitter buffer manager of the receiving terminal calculates a Reference Signal Received Power (RSRP) or Signal to Interference and Noise Ratio (SINR) that is a receiving signal sensitivity indicator of the receiving terminal, and calculates the RSRP and SINR. A variable buffering method for improving speech delivery time and voice quality for an MCPTT service in a wireless communication system, characterized in that it adjusts the length of the de-jitter buffer based on the basis.
The wireless communication system of claim 5, wherein the load environment of the OS calculates a load (%) of the CPU of the receiving terminal and determines the de-jitter buffer length based on the load (%) of the CPU. Variable buffering method for improving speech delivery time and voice quality for MCPTT service.
The MCPTT service in a wireless communication system according to claim 5, wherein the receiving terminal calculates a de-jitter buffer length for which a MOS index can obtain a predetermined score or more through mapping from a result obtained through a previous experiment. Variable buffering method for improving speech delivery time and voice quality.
The method of claim 5, wherein the receiving terminal is received from a PTT server

Set by the di-jitter manager

A variable buffering method for improving speech delivery time and voice quality for an MCPTT service in a wireless communication system, characterized in that the de-jitter buffer length is determined based on the M-JTT.
10. The MCPTT service of claim 9, wherein the receiving terminal adjusts the length of the de-jitter buffer by using the determined de-jitter buffer length to reproduce the voice packet received from the transmitting terminal. Variable buffering method for improving speech delivery time and speech quality.
Variable for improving speech delivery time and voice quality for MCPTT service in wireless communication system including LTE network consisting of transmitting terminal and receiving terminal, base station / core network, and PTT server providing group communication service between transmitting terminal and receiving terminal A buffering device,
It is provided in the PTT server, measures the arrival time interval between voice packets received when receiving a voice packet from the transmitting terminal, the minimum de-jitter buffer to be set by the receiving terminal based on the statistical distribution of the arrival time interval between voice packets Length

A transmission jitter observing unit for instructing the receiving terminal to set the de-jitter buffer length by setting a?
It is provided in the App unit of the receiving terminal, the de-jitter buffer length to be set by the receiving terminal based on the received signal sensitivity of the voice packet received from the PTT server or the load environment of the OS

Determined, determined

And a de-jitter manager configured to variably control the low-jitter buffer length.
The de-jitter buffer length commanded by the transmitting jitter observer of the PTT server to be set to the receiving terminal is a result of obtaining a de-jitter buffer length through which the MOS index can obtain a predetermined score or more from the receiving terminal through previous experiments. Variable buffering device for improving speech delivery time and voice quality for MCPTT service in a wireless communication system, characterized in that it is configured to calculate through mapping from
12. The method of claim 11, wherein the statistical distribution of the arrival time intervals of the voice packets is a mean and variance of the arrival time intervals of the voice packets. Buffering device.
12. The variable buffering apparatus of claim 11, wherein the speech transmission time and voice quality for the MCPTT service are improved in a wireless communication system.
12. The MCPTT service of claim 11, wherein the receiving terminal calculates a de-jitter buffer length for which a MOS index can obtain a predetermined score or more through mapping from a result obtained through a previous experiment. Variable buffering device to improve speech delivery time and voice quality.
The method of claim 11, wherein the receiving terminal is received from a PTT server

Set by the di-jitter manager

Variable buffering device for improving speech delivery time and voice quality for MCPTT service in a wireless communication system, characterized in that for determining the de-jitter buffer length based on.

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