• Title/Summary/Keyword: network congestion

Search Result 908, Processing Time 0.023 seconds

Network Routing by Traffic Prediction on Time Series Models (시계열 모형의 트래픽 예측에 기반한 네트워크 라우팅)

  • Jung, Sang-Joon;Chung, Youn-Ky;Kim, Chong-Gun
    • Journal of KIISE:Information Networking
    • /
    • v.32 no.4
    • /
    • pp.433-442
    • /
    • 2005
  • An increase In traffic has a large Influence on the performance of a total network. Therefore, traffic management has become an important issue of network management. In this paper, we propose a new routing algorithm that attempts to analyze network conditions using time series prediction models and to propose predictive optimal routing decisions. Traffic congestion is assumed when the predicting result is bigger than the permitted bandwidth. By collecting traffic in real network, the predictable model is obtained when it minimizes statistical errors. In order to predict network traffic based on time series models, we assume that models satisfy a stationary assumption. The stationary assumption can be evaluated by using ACF(Auto Correlation Function) and PACF(Partial Auto Correlation Function). We can obtain the result of these two functions when it satisfies the stationary assumption. We modify routing oaths by predicting traffic in order to avoid traffic congestion through experiments. As a result, Predicting traffic and balancing load by modifying paths allows us to avoid path congestion and increase network performance.

A Network Adaptive SVC Streaming Protocol for Improving Video Quality (비디오 품질 향상을 위한 네트워크 적응적인 SVC 스트리밍 프로토콜)

  • Kim, Jong-Hyun;Koo, Ja-Hon;Chung, Kwang-Sue
    • Journal of KIISE:Information Networking
    • /
    • v.37 no.5
    • /
    • pp.363-373
    • /
    • 2010
  • The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

Delay-based Rate Control for Multimedia Streaming in the Internet (인터넷에서 멀티미디어 스트리밍을 위한 지연 시간 기반 전송률 제어)

  • Song Yong-Hon;Kim Nam-Yun;Lee Bong-Gyou
    • The Journal of Korean Institute of Communications and Information Sciences
    • /
    • v.31 no.9B
    • /
    • pp.829-837
    • /
    • 2006
  • Due to the internet network congestion, packets may be dropped or delayed at routers. This phenomenon degrades the quality of streaming applications that require high QoS requirements. The proposed algorithm in this paper, called DBRC(Delay-Based Rate Control), tries to cause router queue occupancy to reach a steady state or equilibrium by throttling the transmission rate of the multimedia traffics when network delays tend to increase and also probing for more bandwidth when network delays tend to decrease. Simulation results show that the proposed algorithm provides smooth transmission rate, nearly constant delay and low packet loss rates, compared with TFRC(TCP Friendly Rate Control) that is one of dominant multimedia congestion control algorithms.

First- and Second-best Pricing in Stable Dynamic Models (안정동력학 모형에서 최선 통행료 및 차선 통행료)

  • Park, Koo-Hyun
    • Journal of the Korean Operations Research and Management Science Society
    • /
    • v.34 no.4
    • /
    • pp.123-138
    • /
    • 2009
  • This study examined the first- and second-best pricing by stable dynamics in congested transportation networks. Stable dynamics, suggested by Nesterov and de Palma (2003), is a new model which describes and provides a stable state of congestion in urban transportation networks. The first-best pricing in user equilibrium models introduces user-equilibrium in the system-equilibrium by tolling the difference between the marginal social cost and the marginal private cost on each link. Nevertheless, the second-best pricing, which levies the toll on some, but not all, links, is relevant from the practical point of view. In comparison with the user equilibrium model, the stable dynamic model provides a solution equivalent to system-equilibrium if it is focused on link flows. Therefore the toll interval on each link, which keeps up the system-equilibrium, is more meaningful than the first-best pricing. In addition, the second-best pricing in stable dynamic models is the same as the first-best pricing since the toll interval is separately given by each link. As an effect of congestion pricing in stable dynamic models, we can remove the inefficiency of the network with inefficient Braess links by levying a toll on the Braess link. We present a numerical example applied to the network with 6 nodes and 9 links, including 2 Braess links.

TCP Performance Enhancement over the Wireless Networks by Using CPC and ZWSC (CPC와 ZWSC를 이용한 무선 망에서의 TCP 성능 향상 방안)

  • Lee, Myung-Sub;Park, Young-Min;Chang, Joo-Seok;Park, Chang-Hyeon
    • IEMEK Journal of Embedded Systems and Applications
    • /
    • v.1 no.1
    • /
    • pp.24-30
    • /
    • 2006
  • With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In these days, many papers describe about wireless-TCP which has suggested how to avoid congestion control when packet loss over the wireless network. In this paper, an enhancement scheme is proposed by modifying SNOOP scheme. To enhance the original SNOOP scheme, CPC(Consecutive Packet Control) and ZWSC(Zero Window Size Control) are added. The invocation of congestion control mechanism is now minimized by knowing the cause of packet loss. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compares to SNOOP by knowing the cause of packet loss at the base station.

  • PDF

OTP: An Overlay Transport Protocol for End-to-end Congestion and Flow Control in Overlay Networks

  • Kim, Kyung-Hoe;Kim, Pyoung-Yun;Youm, Sung-Kwan;Seok, Seung-Joon;Kang, Chul-Hee
    • Journal of IKEEE
    • /
    • v.11 no.4
    • /
    • pp.331-339
    • /
    • 2007
  • The problem of architecting a reliable transport system across an overlay network using split TCP connections as the transport primitive is mainly considered. The considered overlay network uses the application-level switch in each intermediate host. We first argue that natural designs based on store-and-forward principles that are maintained by split TCP connections of hop-by-hop approaches. These approaches in overlay networks do not concern end-to-end TCP semantics. Then, a new transport protocol-Overlay Transport Protocol (OTP)-that manages the end-to-end connection and is responsible for the congestion/flow control between source host and destination host is proposed. The proposed network model for the congestion and flow control mechanisms uses a new window size-Ownd-and a new timer in the source host and destination host. We validate our analytical findings and evaluate the performance of our OTP using a prototype implementation via simulation.

  • PDF

Improve ARED Algorithm in TCP/IP Network (TCP/IP 네트워크에서 ARED 알고리즘의 성능 개선)

  • Nam, Jae-Hyun
    • Journal of the Korea Society of Computer and Information
    • /
    • v.12 no.3
    • /
    • pp.177-183
    • /
    • 2007
  • Active queue management (AQM) refers to a family of packet dropping mechanisms for router queues that has been proposed to support end-to-end congestion control mechanisms in the Internet. The proposed AQM algorithm by the IETF is Random Early Detection (RED). The RED algorithm allows network operators simultaneously to achieve high throughput and low average delay. However. the resulting average queue length is quite sensitive to the level of congestion. In this paper, we propose the Refined Adaptive RED(RARED), as a solution for reducing the sensitivity to parameters that affect RED performance. Based on simulations, we observe that the RARED scheme improves overall performance of the network. In particular, the RARED scheme reduces packet drop rate and improves goodput.

  • PDF

Optimization of Long-term Generator Maintenance Scheduling considering Network Congestion and Equivalent Operating Hours (송전제약과 등가운전시간을 고려한 장기 예방정비계획 최적화에 관한 연구)

  • Shin, Hansol;Kim, Hyoungtae;Lee, Sungwoo;Kim, Wook
    • The Transactions of The Korean Institute of Electrical Engineers
    • /
    • v.66 no.2
    • /
    • pp.305-314
    • /
    • 2017
  • Most of the existing researches on systemwide optimization of generator maintenance scheduling do not consider the equivalent operating hours(EOHs) mainly due to the difficulties of calculating the EOHs of the CCGTs in the large scale system. In order to estimate the EOHs not only the operating hours but also the number of start-up/shutdown during the planning period should be estimated, which requires the mathematical model to incorporate the economic dispatch model and unit commitment model. The model is inherently modelled as a large scale mixed-integer nonlinear programming problem and the computation time increases exponentially and intractable as the system size grows. To make the problem tractable, this paper proposes an EOH calculation based on demand grouping by K-means clustering algorithm. Network congestion is also considered in order to improve the accuracy of EOH calculation. This proposed method is applied to the actual Korean electricity market and compared to other existing methods.

A Congestion Control Algorithm for the fairness Improvement of TCP Vegas (TCP Vegas의 공정성 향상을 위한 혼잡 제어 알고리즘)

  • 오민철;송병훈;정광수
    • Journal of KIISE:Information Networking
    • /
    • v.31 no.3
    • /
    • pp.269-279
    • /
    • 2004
  • The most important factor influencing the robustness of the Internet Is the end-to-end TCP congestion control. However, the congestion control scheme of TCP Reno, the most popular TCP version on the Internet, employs passive congestion indication. It makes worse the network congestion. Recently, Brakmo and Peterson have proposed a new version of TCP, which is named TCP Vegas, with a fundamentally different congestion control scheme from that of the Reno. Many studies indicate that the Vegas is able to achieve better throughput and higher stability than the Reno. But there are two unfairness problems in Vegas. These problems hinder the spread of the Vegas in current Internet. In this paper, in order to solve these unfairness problems, we propose a new congestion control algorithm called TCP PowerVegas. The existing Vegas depends mainly only on the rtt(round trip time), but the proposed PowerVegas use the new congestion control scheme combined the Information on the rtt with the information on the packet loss. Therefore the PowerVegas performs the congestion control more competitively than the Vegas. Thus, the PowerVegas is able to solve effectively these unfairness problems which the Vegas has experienced. To evaluate the proposed approach, we compare the performance among PowerVegas, Reno and Vegas under same network environment. Using simulation, the PowerVegas is able to achieve better throughput and higher stability than the Reno and is shown to achieve much better fairness than the existing Vegas.

Improving TCP Performance by Limiting Congestion Window in Fixed Bandwidth Networks (고정대역 네트워크에서 혼잡윈도우 제한에 의한 TCP 성능개선)

  • Park, Tae-Joon;Lee, Jae-Yong;Kim, Byung-Chul
    • Journal of the Institute of Electronics Engineers of Korea TC
    • /
    • v.42 no.12
    • /
    • pp.149-158
    • /
    • 2005
  • This paper proposes a congestion avoidance algorithm which provides stable throughput and transmission rate regardless of buffer size by limiting the TCP congestion window in fixed bandwidth networks. Additive Increase, Multiplicative Decrease (AIMD) is the most commonly used congestion control algorithm. But, the AIMD-based TCP congestion control method causes unnecessary packet losses and retransmissions from the congestion window increment for available bandwidth verification when used in fixed bandwidth networks. In addition, the saw tooth variation of TCP throughput is inappropriate to be adopted for the applications that require low bandwidth variation. We present an algorithm in which congestion window can be limited under appropriate circumstances to avoid congestion losses while still addressing fairness issues. The maximum congestion window is determined from delay information to avoid queueing at the bottleneck node, hence stabilizes the throughput and the transmission rate of the connection without buffer and window control process. Simulations have performed to verify compatibility, steady state throughput, steady state packet loss count, and the variance of congestion window. The proposed algorithm can be easily adopted to the sender and is easy to deploy avoiding changes in network routers and user programs. The proposed algorithm can be applied to enhance the performance of the high-speed access network which is one of the fixed bandwidth networks.