• Title/Summary/Keyword: Packet Loss

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Analysis and Modeling of Traffic at Ntopia Subscriber Network of Korea Telecom (KT의 Ntopia가입자 망 트래픽 분석 및 모델링)

  • 주성돈;이채우
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.41 no.5
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    • pp.37-45
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    • 2004
  • As Internet technologies are mature, many new applications that are different characteristics are emerging. Recently we see wide use of P2P(Peer to Peer) applications of which traffic shows different statistical characteristics compared with traditional application such as web(HTTP) and FTP(File Transfer Protocol). In this paper, we measured subscriber network of KT(Korea Telecom) to analyze P2P traffic characteristics. We show flow characteristics of measured traffic. We also estimate Hurst parameter of P2P traffic and compare self-similarity with web traffic. Analysis results indicate that P2P traffic is much bustier than web traffic and makes both upstream traffic and downstream traffic be symmetric. To predict parameters related QoS such as packet loss and delays we model P2P traffic using two self-similar traffic models and predict both loss probability and mm delay then compare their accuracies. With simulation we show that the self-similar traffic models we derive predict the performance of P2P traffic accurately and thus when we design a network or evaluate its performance, we can use the P2P traffic model as reference input traffic.

An adaptive resynchronization technique for stream cipher system in HDLC protocol (HDLC 프로토콜에서 운용되는 동기식 스트림 암호 통신에 적합한 적응 난수열 재동기 기법)

  • 윤장홍;황찬식
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.22 no.9
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    • pp.1916-1932
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    • 1997
  • The synchronous stream cipher which require absoulte clock synchronization has the problem of synchronization loss by cycle slip. Synchronization loss makes the state which sender and receiver can't communicate with each other and it may break the receiving system. To lessen the risk, we usually use a continuous resynchronization method which achieve resynchronization at fixed timesteps by inserting synchronization pattern and session key. While we can get resynchronization effectively by continuous resynchroniation, there are some problems. In this paper, we proposed an adaptive resynchronization algorithm for cipher system using HDLC protocol. It is able to solve the problem of the continuous resynchronization. The proposed adaptive algorithm make resynchronization only in the case that the resynchronization is occurred by analyzing the address field of HDLC. It measures the receiving rate of theaddress field in the decision duration. Because it make resynchronization only when the receiving rate is greateer than the threshold value, it is able to solve the problems of continuous resynchronization method. When the proposed adaptive algorithm is applied to the synchronous stream cipher system in packet netork, it has addvance the result in R_e and D_e.

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Exploitation of Auxiliary Motion Vector in Video Coding for Robust Transmission over Internet (화상통신에서의 오류전파 제어를 위한 보조모션벡터 코딩 기법)

  • Lee, Joo-Kyong;Choi, Tae-Uk;Chung, Ki-Dong
    • The KIPS Transactions:PartB
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    • v.9B no.5
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    • pp.571-578
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    • 2002
  • In this paper, we propose a video sequence coding scheme called AMV (Auxiliary Motion Vector) to minimize error propagation caused by transmission errors over the Internet. Unlike the conventional coding schemes the AMY coder, for a macroblock in a frame, selects two best matching blocks among several preceding frames. The best matching block, called a primary block, is used for motion compensation of the destination macroblock. The other block, called an auxiliary block, replaces the primary block in case of its loss at the decoder. When a primary block is corrupted or lost during transmission, the decoder can efficiently and simply suppress error propagation to the subsequent frames by replacing the block with an auxiliary block. This scheme has an advantage of reducing both the number and the impact of error propagations. We implemented the proposed coder by modifying H.263 standard coding and evaluated the performance of our proposed scheme in the simulation. The simulation results show that AMV coder is more efficient than the H.263 baseline coder at the high packet loss rate.

A Study of Performance Analysis on Effective Multiple Buffering and Packetizing Method of Multimedia Data for User-Demand Oriented RTSP Based Transmissions Between the PoC Box and a Terminal (PoC Box 단말의 RTSP 운용을 위한 사용자 요구 중심의 효율적인 다중 수신 버퍼링 기법 및 패킷화 방법에 대한 성능 분석에 관한 연구)

  • Bang, Ji-Woong;Kim, Dae-Won
    • Journal of Korea Multimedia Society
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    • v.14 no.1
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    • pp.54-75
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    • 2011
  • PoC(Push-to-talk Over Cellular) is an integrated technology of group voice calls, video calls and internet based multimedia services. If a PoC user can not participate in the PoC session for various reasons such as an emergency situation, lack of battery capacity, then the user can use the PoC Box which has a similar functionality to the MM Box in the MMS(Multimedia Messaging Service). The RTSP(Real-Time Streaming Protocol) method is recommended to be used when there is a transmission session between the PoC box and a terminal. Since the existing VOD service uses a wired network, the packet size of RTSP-based VOD service is huge, however, the PoC service has wireless communication environments which have general characteristics to be used in RTSP method. Packet loss in a wired communication environments is relatively less than that in wireless communication environment, therefore, a buffering latency occurs in PoC service due to a play-out delay which means an asynchronous play of audio & video contents. Those problems make a user to be difficult to find the information they want when the media contents are played-out. In this paper, the following techniques and methods were proposed and their performance and superiority were verified through testing: cross-over dual reception buffering technique, advance partition multi-reception buffering technique, and on-demand multi-reception buffering technique, which are designed for effective picking up of information in media content being transmitted in short amount of time using RTSP when a user searches for media, as well as for reduction in playback delay; and same-priority packetization transmission method and priority-based packetization transmission method, which are media data packetization methods for transmission. From the simulation of functional evaluation, we could find that the proposed multiple receiving buffering and packetizing methods are superior, with respect to the media retrieval inclination, to the existing single receiving buffering method by 6-9 points from the viewpoint of effectiveness and excellence. Among them, especially, on-demand multiple receiving buffering technology with same-priority packetization transmission method is able to manage the media search inclination promptly to the requests of users by showing superiority of 3-24 points above compared to other combination methods. In addition, users could find the information they want much quickly since large amount of informations are received in a focused media retrieval period within a short time.

Consolidation of Metro Networks and Access Networks by using Long-reach WDM-PON (장거리 전송 파장분할 다중방식 수동형 광가입자망을 이용한 메트로망과 가입자망 통합 방안)

  • Lee Sang-Mook;Mun Sil-Gu;Kim Min-Hwan;Lee Chang-Hee
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.43 no.5 s.347
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    • pp.59-67
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    • 2006
  • We demonstrate bidirectional long-reach 35-channel dense wavelength division multiplexing-passive optical network(DWDM-PON) based on wavelength-locked Fabry-Perot laser diodes (F-P LDs). The mode control of F-P LD enhances output power at decreased the required injection power. We show packet-loss-free transmission in all 70 channels at 125 Mb/s per channel line rate through 70 km of single mode fiber without optical amplifier The DWDM-PON can consolidate a metro network into an access network by bypassing the central offices within its reach. The proposed DWDM-PON can accommodate about 80 subscribers with an EDFA-based broadband light source. Further expansion up to 100 subscribers is possible with a semiconductor-based BLS.

Cognitive Impairment and Decreased Quality of Life in Elderly Patients with Subsyndromal Depression (노인 아증후군적 우울증 환자의 인지기능 및 삶의 질 저하)

  • Ryu, Jae Sung;Kim, Moon Doo;Lee, Chang In;Park, Joon Hyuk
    • Korean Journal of Biological Psychiatry
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    • v.20 no.2
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    • pp.45-53
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    • 2013
  • Objectives Non-major depression with fewer symptoms than required for a Diagnostic and Statistical Manual of Mental Disorders-4th edition diagnosis of major depressive disorder (MDD) has consistently been found to be associated with functional impairment. In this study, we aim to estimate the cognitive impairment and the quality of life in elderly patients with subsyndromal depression (SSD) compared with non-depressive elderly (NDE). Methods The Korean version of Mini International Neuropsychiatric Interview was administered to 194 outpatients with depression and 108 normal controls. SSD is defined as having five or more current depressive symptoms with core depressive symptoms (depressive mood or loss of interest or pleasure) during more than half a day and more than seven days over two weeks. Depression was evaluated by the Korean form of Geriatric Depression Scale of a 15-item short version. Global cognition was assessed by Mini-Mental State Examination in the Korean version of CERAD assessment packet (MMSE-KC). Subjective cognitive impairment was assessed by the Subjective Memory Complaint Questionnaire. Quality of life was evaluated by the Korean Version of Short-Form 36-Item Health Survey. Results The mean score of the MMSE-KC in the SSD group was lower than that in the NDE group with adjustment for age, gender, and education [F = 4.270, p = 0.04, analysis of covariance (ANCOVA)]. If we defined those having Z-score of MMSE-KC < -1.5 as a high risk group of cognitive impairment, the odds ratio for the high risk group of cognitive impairment was 1.86 [95% confidence intervals (CI) 1.04-3.34] in SSD and 7.57 (95% CI 3.50-16.40) in MDD compared to NDE. The scores of physical component summary (F = 9.274, p = 0.003, ANCOVA) and mental component summary (F = 53.166, p < 0.001, ANCOVA) in the SSD group were lower than those in the NDE group with adjustment for age, gender, and education. Conclusions The subjects with SSD, as well as those with MDD, showed impairment of global cognition and also experienced low quality of life in both physical and mental aspects, compared to the NDE group.

Error Resilient Video Coding Techniques Using Multiple Description Scheme (다중 표현을 이용한 에러에 강인한 동영상 부호화 방법)

  • 김일구;조남익
    • Journal of Broadcast Engineering
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    • v.9 no.1
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    • pp.17-31
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    • 2004
  • This paper proposes an algorithm for the robust transmission of video in error Prone environment using multiple description codingby optimal split of DCT coefficients and rate-distortionoptimization framework. In MDC, a source signal is split Into several coded streams, which is called descriptions, and each description is transmitted to the decoder through different channel. Between descriptions, structured correlations are introduced at the encoder, and the decoder exploits this correlation to reconstruct the original signal even if some descriptions are missing. It has been shown that the MDC is more resilient than the singe description coding(SDC) against severe packet loss ratecondition. But the excessive redundancy in MDC, i.e., the correlation between the descriptions, degrades the RD performance under low PLR condition. To overcome this Problem of MDC, we propose a hybrid MDC method that controls the SDC/MDC switching according to channel condition. For example, the SDC is used for coding efficiency at low PLR condition and the MDC is used for the error resilience at high PLR condition. To control the SDC/MDC switching in the optimal way, RD optimization framework are used. Lagrange optimization technique minimizes the RD-based cost function, D+M, where R is the actually coded bit rate and D is the estimated distortion. The recursive optimal pet-pixel estimatetechnique is adopted to estimate accurate the decoder distortion. Experimental results show that the proposed optimal split of DCT coefficients and SD/MD switching algorithm is more effective than the conventional MU algorithms in low PLR conditions as well as In high PLR condition.

Efficient Transmission of Scalable Video Streams Using Dual-Channel Structure (듀얼 채널 구조를 이용한 Scalable 비디오(SVC)의 전송 성능 향상)

  • Yoo, Homin;Lee, Jaemyoun;Park, Juyoung;Han, Sanghwa;Kang, Kyungtae
    • KIPS Transactions on Computer and Communication Systems
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    • v.2 no.9
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    • pp.381-392
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    • 2013
  • During the last decade, the multitude of advances attained in terminal computers, along with the introduction of mobile hand-held devices, and the deployment of high speed networks have led to a recent surge of interest in Quality of Service (QoS) for video applications. The main difficulty is that mobile devices experience disparate channel conditions, which results in different rates and patterns of packet loss. One way of making more efficient use of network resources in video services over wireless channels with heterogeneous characteristics to heterogeneous types of mobile device is to use a scalable video coding (SVC). An SVC divides a video stream into a base layer and a single or multiple enhancement layers. We have to ensure that the base layer of the video stream is successfully received and decoded by the subscribers, because it provides the basis for the subsequent decoding of the enhancement layer(s). At the same time, a system should be designed so that the enhancement layer(s) can be successfully decoded by as many users as possible, so that the average QoS is as high as possible. To accommodate these characteristics, we propose an efficient transmission scheme which incorporates SVC-aware dual-channel repetition to improve the perceived quality of services. We repeat the base-layer data over two channels, with different characteristics, to exploit transmission diversity. On the other hand, those channels are utilized to increase the data rate of enhancement layer data. This arrangement reduces service disruption under poor channel conditions by protecting the data that is more important to video decoding. Simulations show that our scheme safeguards the important packets and improves perceived video quality at a mobile device.

An Interference Reduction Scheme Using AP Aggregation and Transmit Power Control on OpenFlow-based WLAN (OpenFlow가 적용된 무선랜 환경에서 AP 집단화 및 전송 파워 조절에 기반한 간섭 완화 기법)

  • Do, Mi-Rim;Chung, Sang-Hwa;Ahn, Chang-Woo
    • Journal of KIISE
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    • v.42 no.10
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    • pp.1254-1267
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    • 2015
  • Recently, excessive installations of APs have caused WLAN interference, and many techniques have been suggested to solve this problem. The AP aggregation technique serves to reduce active APs by moving station connections to a certain AP. Since this technique forcibly moves station connections, the transmission performance of some stations may deteriorate. The AP transmit power control technique may cause station disconnection or deterioration of transmission performance when power is reduced under a certain level. The combination of these two techniques can reduce interference through AP aggregation and narrow the range of interferences further through detailed power adjustment. However, simply combining these techniques may decrease the probability of power adjustment after aggregation and increase station disconnections upon power control. As a result, improvement in performance may be insignificant. Hence, this study suggests a scheme to combine the AP aggregation and the AP transmit power control techniques in OpenFlow-based WLAN to ameliorate the disadvantages of each technique and to reduce interferences efficiently by performing aggregation for the purpose of increasing the probability of adjusting transmission power. Simulations reveal that the average transmission delay of the suggested scheme is reduced by as much as 12.8% compared to the aggregation scheme and by as much as 18.1% compared to the power control scheme. The packet loss rate due to interference is reduced by as much as 24.9% compared to the aggregation scheme and by as much as 46.7% compared to the power control scheme. In addition, the aggregation scheme and the power control scheme decrease the throughput of several stations as a side effect, but our scheme increases the total data throughput without decreasing the throughput of each station.

Implementation of a Sensor Node with Convolutional Channel Coding Capability (컨벌루션 채널코딩 기능의 센서노드 구현)

  • Jin, Young Suk;Moon, Byung Hyun
    • Journal of Korea Society of Industrial Information Systems
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    • v.19 no.1
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    • pp.13-18
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    • 2014
  • Sensor nodes are used for monitoring and collecting the environmental data via wireless sensor network. The wireless sensor network with various sensor nodes draws attention as a key technology in ubiquitous computing. Sensor nodes has very small memory capacity and limited power resource. Thus, it is essential to have energy efficient strategy for the sensor nodes. Since the sensor nodes are operating on the same frequency bands with ISM frequency bands, the interference by the devices operating on the ISM band degrades the quality of communication integrity. In this paper, the convolutional code is proposed instead of ARQ for the error control for the sensor network. The proposed convolutional code was implemented and the BER performance is measured. For the fixed transmitting powers of -19.2 dBm and -25dBm, the BER with various communication distances are measured. The packet loss rate and the retransmission rate are calculated from the measured BER. It is shown that the porposed method obtained about 9~12% and 12-19% reduction in retransmission rate for -19.2 dBm and -25 dBm respectively.