The Influencing Factors of Network Packet Loss’s Long-Range Dependence has Impacts on the Packet Loss Rate

Jin Wang 보안공학연구지원센터 2015 International Journal of Multimedia and Ubiquitous Vol.10 No.11

In order to better establish no-reference video quality assessment model considering the network packet loss and further gain a better QoE evaluation to meet the needs of the user’s, so we build NS2+ MyEvalvid simulation platform to study the scale characteristic of the network packet loss, scale characteristic of packet loss through the influence of packet loss rate to influence QoE. The experimental results show that, packet loss processes has long-range dependence, the number of superimposed source N, shape parameter, Hurst parameter, the output link speed have impacts on long-range dependence. We came to the conclusion that when superimposed source N is more, shape parameter is smaller, Hurst parameter is bigger, the output link speed is smaller, packet loss’s long range dependence is larger, packet loss rate is high.

TCP 송수신자간의 큐사용률 추정을 이용한 송신자 기반의 패킷손실 구별기법

박미영(Mi-Young Park),정상화(Sang-Hwa Chung),이윤성(Yun-Sung Lee) 한국컴퓨터정보학회 2011 韓國컴퓨터情報學會論文誌 Vol.16 No.1

멀티홉 무선 네트워크에서 TCP가 동작하는 경우, 유선 네트워크와 무선 네트워크의 서로 다른 특성으로 인하여 TCP의 심각한 성능저하가 초래된다. 이것은 TCP가 무선오류로 인해 발생되는 패킷손실을 혼잡으로 인해 발생한 패킷손실로 간주하여 데이터 전송률을 불필요하게 감소시키기 때문이다. 이러한 성능저하를 피하기 위해서 혼잡손실과 무선손실을 구별하는 많은 기법들이 연구되어 왔으나, 이들 기법들은 무선손실에 대한 탐지정확도가 기대만큼 높지 않거나, 무선손실에 대한 탐지정확도가 높으면 혼잡손실에 대한 정확도가 낮아지는 경향을 보인다. 본 논문은 혼잡손실에 대한 탐지정확도의 희생을 최소화하면서, 무선손실에 대한 탐지정확도를 높이는 송신자 기반의 패킷손실 구별기법을 제안한다. 본 기법은 네트워크 혼잡과 상호 관련성이 높은TCP 송 수신자간의 큐 사용률을 추정하고, 추정된 큐 사용률과 특정 임계값을 비교하여 혼잡손실과 무선손실을 구별한다. 네트워크 시뮬레이터인 QualNet을 이용한 실험에서는 기존 기법과 제시된 기법간의 혼잡손실에 대한 탐지정확도와 무선손실에 대한 탐지정확도를 구분하여 비교평가하고, 홉 수 증가에 따른 성능향상을 비교평가 한다. 실험 결과는 멀티홉 무선 네트워크상에서 본 기법이 가장 높은 탐지정확도를 가질 뿐만 아니라 TCP의 성능을 가장 높게 향상시킴을 보인다. When TCP operates in multi-hop wireless networks, it suffers from severe performance degradation due to the different characteristics of wireless networks and wired networks. This is because TCP reacts to wireless packet losses by unnecessarily decreasing its sending rate assuming the losses as congestion losses. Although several loss differentiation algorithms (LDAs) have been proposed to avoid such performance degradation, their detection accuracies are not high as much as we expect. In addition the schemes have a tendency to sacrifice the detection accuracy of congestion losses while they improve the detection accuracy of wireless losses. In this paper, we suggest a new sender-based loss differentiation scheme which enhances the detection accuracy of wireless losses while minimizing the sacrifice of the detection accuracy of congestion losses. Our scheme estimates the rate of queue usage which is highly correlated with the congestion in the network path between a TCP sender and a receiver, and it distinguishes congestion losses from wireless losses by comparing the estimated queue usage with a certain threshold. In the extensive experiments based on a network simulator, QualNet, we measure and compare each detection accuracy of wireless losses and congestion losses, and evaluate the performance enhancement in each scheme. The results show that our scheme has the highest accuracy among the LDAs and it improves the most highly TCP performance in multi-hop wireless networks.

인터넷 상에서의 H.264 비디오 전송을 위한 패킷 손실 복원에 관한 연구

하호진(Hojin Ha),임창훈(Changhoon Yim),김영용(Young Yong Kim) 한국통신학회 2007 韓國通信學會論文誌 Vol.32 No.10C

본 논문은 인터넷에서 실시간 비디오 전송을 위하여 패킷 손실을 복원하기 위한 알고리즘을 제안한다. 인터와 인트라 프레임에 존재하는 시간축과 공간축의 의존성을 분석함으로써, 패킷 손실로 인한 에러 은닉과 에러 전파에 의한 비디오 화질의 왜곡을 최소화하도록 순방향 에러 정정 코드 (forward error correction, FEC)를 비디오 패킷에 할당한다. 최적의 FEC 패킷을 할당하기 위해서, 우선 패킷 손실로 인한 비디오 화질 저하의 크기를 패킷 왜곡 모델로 정형화한다. 그리고 주어진 채널환경과 패킷 왜곡 모델을 이용하여 적은 계산으로 패킷 정정 율에 비례하는 FEC 패킷 할당 알고리즘을 제안한다. 실험 결과에서, 제안된 알고리즘은 패킷 손실 네트워크 환경에서 많은 비디오 화질 향상을 가져왔으며, 패킷 손실 율의 증가에도 상대적으로 적은 화질 감소를 얻을 수 있었다. This paper presents an efficient packet loss resilient scheme for real-time video transmission over the Internet. By analyzing the temporal and spatial dependencies in inter- and intra-frames, we assign forward error correction codes (FEC) across video packets for minimizing the effect of error concealment and error propagation from packet loss. To achieve optimal allocation of FEC codes, we formulate the effect of packet loss on video quality degradation as packet distortion model. Then we propose an unequal FEC assignment scheme with low complexity based on packet correction rate, which uses the packet distortion model and includes channel status information. Simulation results show that the proposed FEC assignment scheme gives substantial improvement for the received video quality in packet lossy networks. Furthermore the proposed scheme achieves relatively smaller degradation of video quality with higher packet loss rates.

Adaptive Speech Streaming Based on Packet Loss Prediction Using Support Vector Machine for Software- Based Multipoint Control Unit over IP Networks

강진아,한미경,장종현,김홍국 한국전자통신연구원 2016 ETRI Journal Vol.38 No.6

An adaptive speech streaming method to improve the perceived speech quality of a software-based multipoint control unit (SW-based MCU) over IP networks is proposed. First, the proposed method predicts whether the speech packet to be transmitted is lost. To this end, the proposed method learns the pattern of packet losses in the IP network, and then predicts the loss of the packet to be transmitted over that IP network. The proposed method classifies the speech signal into different classes of silence, unvoiced, speech onset, or voiced frame. Based on the results of packet loss prediction and speech classification, the proposed method determines the proper amount and bitrate of redundant speech data (RSD) that are sent with primary speech data (PSD) in order to assist the speech decoder to restore the speech signals of lost packets. Specifically, when a packet is predicted to be lost, the amount and bitrate of the RSD must be increased through a reduction in the bitrate of the PSD. The effectiveness of the proposed method for learning the packet loss pattern and assigning a different speech coding rate is then demonstrated using a support vector machine and adaptive multirate-narrowband, respectively. The results show that as compared with conventional methods that restore lost speech signals, the proposed method remarkably improves the perceived speech quality of an SW-based MCU under various packet loss conditions in an IP network.

패킷 손실에 대한 스케일러블 비디오(SVC) 적응기법 및 성능분석

장의덕(Euy-doc Jang),김재곤(Jae-Gon Kim),Truong Cong Thang,강정원(Jung Won Kang) 한국방송·미디어공학회 2009 방송공학회논문지 Vol.14 No.6

SVC (Scalable Video Coding) is a new video coding standard to provide convergence media service in heterogeneous environments with different networks and diverse terminals through spatial-temporal-quality combined flexible scalabilities. This paper presents the performance analysis on packet loss in the delivery of SVC over IP networks and an efficient adaptation method to packet loss caused by buffer overflow. In particular, SVC with MGS (Medium Grained Scalability) as well as spatial and temporal scalabilities is addressed in the consideration of packet-based adaptation since finer adaptation is possible with a sufficient numbers of quality layers in MGS. The effect on spatio-temporal quality due to the packet loss of SVC with MGS is evaluated. In order to minimize quality degradation resulted by packet loss, the proposed adaptation of MGS based SVC first sets adaptation unit of AU (Access Unit) or GOP corresponding to allowed delay and then selectively discards packets in order of importance in terms of layer dependency. In the experiment, the effects of packet loss on quantitative qualities are analyzed and the effectiveness of the proposed adaptation to packet loss is shown.

가중치 기반 공정 큐잉에서 패킷 손실 차별화 및 보장

김태준(TaeJoon Kim),서봉수(Bongsue Suh) 한국정보기술학회 2013 한국정보기술학회논문지 Vol.11 No.10

WFQ (Weighted Fair Queuing) provides not only fairness among traffic flows in using bandwidth but also guarantees the Quality of Service (QoS) that individual flow requires, which is why it has been applied to the resource reservation protocol (RSVP)-capable router. The RSVP allocates an enough resource to satisfy both the rate and end-to-end delay requirements of the flow in a condition of no packet loss, and the WFQ scheduler guarantees those QoS requirements with the allocated resource. In practice, however, most QoS-guaranteed services allow a degree of packet loss, especially from 0.1% to 3% for Voice over IP, which depends on the codec used and their QoS requirements. Thus, WFQ needs to be revised not only to provide packet loss differentiation but also to guarantee the required packet loss. This paper enhances the WFQ to provide a packet loss differentiation and guarantee, and then investigates its performance. The performance evaluation showed that the proposed WFQ could increase the utilization of bandwidth by 8 ~ 12%.

신뢰성 있는 단방향 데이터 전송 시스템 설계

김동욱(Dongwook Kim),민병길(Byunggil Min) 한국정보보호학회 2016 정보보호학회논문지 Vol.26 No.6

단방향 전송 기술에서 해결해야 할 이슈들 중에 한 가지는 TCP 기반의 데이터 전송에서 발생하는 패킷 손실을 줄이는 것이다. 잘 알려진 에러 수정 기법들을 활용해서 패킷 손실을 줄일 수 있다. 하지만, 기존의 여러 기법들을 활용한다고 하더라도, 링크 에러와 버퍼 오버플로우에 의한 패킷 손실은 여전히 발생할 수 있다. 본 논문에서는 신뢰성 있는 단방향 데이터 전송 시스템(RED, REliable Data diode)을 제안한다. RED는 기존의 단방향 전송 기술과 마찬가지로 TCP 기반의 데이터 전송을 지원하기 위해 TCP 프록시 기법을 활용한다. RED 송신시스템은 TCP 패킷의 지연 전송을 활용하여 버퍼 오버플로우에 의한 패킷 손실을 줄일수 있다. 또한, RED 송신시스템은 패킷의 중요도 및 여유 자원을 고려하여, RED 수신시스템에게 다수의 동일한 패킷을 복제 및 전송함으로써, 단방향 전송 링크에서의 링크 에러에 의한 패킷 손실을 줄일 수 있다. One of the issues, which is dealed with in undirectional data transmission technology, is reducing the packet loss in TCP based data transfer. We can decrease the packet loss by using several well known error correction approaches. Although we utilize those previous approaches, the packet loss by both link errror and buffer overflow could be occurred. In this paper, we propose the RED(REliable Data diode). RED also uses the TCP proxy approach for supporting the TCP based data transfer which is similar with the existing unidirectional data transmission technologies. The RED transmission system could alleviate the packet loss caused by buffer overflow by exploiting the delaying transmission of TCP packets. Furthermore, in order to reduce the packett loss caused by link error in the unidirectional transmission link, the RED transmission system transmits one or more duplicated packets to the RED reception system by considering both the remaining resources and packet importance.

Performance Analysis of Cognitive Radio Networks Adopting Packet Discarding or Buffering Policy

Min Zhang,Bin Li 보안공학연구지원센터 2015 International Journal of Grid and Distributed Comp Vol.8 No.5

In a cognitive radio network (CRN), a preempted secondary user (SU) is placed in a call level queue to wait for accessing another free channel. If the availability of channels is transparent to SUs, packets will be generated during their waiting time and the performance of the CRN will be influenced by which way to handle these packets. This paper mainly analyzes the performance of a CRN adopting discarding or buffering policy, Here, an analytical model is developed to derive the formulas for both call level performance measures (i.e., call blocking probability) and packet level performance measures (i.e., packet delay, packet loss ratio and throughput) for SUs. Numerical results show that theoretical models are consistent with simulation results. The major observations include (i) the performances of a CRN degrade as the primary user (PU) or SU call arrival rate increases. (ii) Under discarding policy, the call blocking probability and packet delay are smaller than adopting buffering policy, while the packet delay and throughput are greater. (iii) Adopting different policies cause a smaller effect on call blocking probability and throughput than on packet loss ratio and packet delay.

패킷 손실에 의해 잘못된 토큰 버킷 파라메타를 정정하는 비디오 정보 파일을 가진 비디오 스트리밍 수신기

이현노(Hyun-No Lee),김동회(Dong-Hoi Kim) 한국디지털콘텐츠학회 2016 한국디지털콘텐츠학회논문지 Vol.17 No.3

Video streaming traffics which are arrived in receiver have irregular traffic patterns by many problems over the network path. Particularly, if these received traffics enter into replay buffer without any operation, the overflow and underflow effects are made according to the buffer status. There was an existing scheme which automatically set up token bucket parameters using the video information file under the assumption of the lossless packet on network. The existing scheme has a problem which can set up the wrong token bucket parameters by the lost packets on the practical networks with video packet loss. Therefore, this paper proposes a new scheme which reset up video file information to correct the wrong token bucket parameters in case of packet loss in practical networks with packet loss. Through the simulation, it was found that the proposed scheme would have better performance than the existing scheme in terms of overflow generation and packet loss.

음성 특성을 이용한 G.711 패킷 손실 은닉 알고리즘의 성능개선

한승호,김진술,이현우,류원,한민수,Han Seung-Ho,Kim Jin-Sul,Lee Hyun-Woo,Ryu Won,Hahn Min-Soo 대한음성학회 2006 말소리 Vol.57 No.-

Because a packet loss brings about degradation of speech quality, VoIP speech coders have PLC (Packet Loss Concealment) mechanism. G.711, which is a mandatory VoIP speech coder, also has the PLC algorithm based on pitch period replication. However, it is not robust to burst losses. Thus, we propose two methods to improve the performance of the original PLC algorithm in G.711. One adaptively utilizes voiced/unvoiced information of adjacent good frames regarding to the current lost frame. The other is based on adaptive gain control according to energy variation across the frames. We evaluate the performance of the proposed PLC algorithm by measuring a PESQ value under different random and burst packet loss simulating conditions. It is shown from the experiments that the performance of the proposed PLC algorithm outperforms that of PLC employed in ITU-T Recommendation G.711.