• Title/Summary/Keyword: Traffic burst

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VLSI Design of Interface between MAC and PHY Layers for Adaptive Burst Profiling in BWA System (BWA 시스템에서 적응형 버스트 프로파일링을 위한 MAC과 PHY 계층 간 인터페이스의 VLSI 설계)

  • Song Moon Kyou;Kong Min Han
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.42 no.1
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    • pp.39-47
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    • 2005
  • The range of hardware implementation increases in communication systems as high-speed processing is required for high data rate. In the broadband wireless access (BWA) system based on IEEE standard 802.16 the functions of higher part in the MAC layer to Provide data needed for generating MAC PDU are implemented in software, and the tasks from formatting MAC PDUs by using those data to transmitting the messages in a modem are implemented in hardware. In this paper, the interface hardware for efficient message exchange between MAC and PHY layers in the BWA system is designed. The hardware performs the following functions including those of the transmission convergence(TC) sublayer; (1) formatting TC PDU(Protocol data unit) from/to MAC PDU, (2) Reed-solomon(RS) encoding/decoding, and (3) resolving DL MAP and UL MAP, so that it controls transmission slot and uplink and downlink traffic according to the modulation scheme of burst profile. Also, it provides various control signal for PHY modem. In addition, the truncated binary exponential backoff (TBEB) algorithm is implemented in a subscriber station to avoid collision on contention-based transmission of messages. The VLSI architecture performing all these functions is implemented and verified in VHDL.

A Study on the Bandwidth Allocation Control of Virtual Paths in ATM Networks for Multimedia Service (멀티미디어 서비스를 위한 ATM망에서 가상경로의 대역폭 할당 제어에 관한 연구)

  • Jang, Yung-Chul;Lee, Jung-Jei;Oh, Moo-Song
    • The Transactions of the Korea Information Processing Society
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    • v.4 no.6
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    • pp.1433-1442
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    • 1997
  • Algorithm design is needed to optimized bandwidth which satisfy quality of service(QoS) requirements of vary traffic classes for Multimedia service in ATM networks. The diverse flow characteristics e.g., burstiness, bit rate and burst length, have to guarantee the different quality of service(QoS) requirements in Asynchronous Transfer Mode(ATM). The QoS parameter may be measured in terms of cell loss probability and maximum cell dely. In this paper, we consider the ATM networks which the virtual path(vip) concept is implemented by applying the Markov Modulated Deterministic Process method. We develop an efficient algorithm to computer the minimum capacity required to satisfy all the QoS requirements when multiple classes of on-off source are multiplexed on single VP. Using above the result, we propose a simple algorithm to determine the VP combination to achieve the near optimum of total capacity required for satisfying the individual QoS requirements. Numerical results are also presented to demonstrate the performance of the algorithm, when compared to the optimal total capacity required.

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A Study on State Dependent RED and Dynamic Scheduling Scheme for Real-time Internet Service (실시간 인터넷 서비스를 위한 상태 의존 RED 및 동적 스케줄링 기법에 관한 연구)

  • 유인태;홍인기;서덕영
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.28 no.9B
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    • pp.823-833
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    • 2003
  • To satisfy the requirements of the real-time Internet services, queue management and scheduling schemes should be enhanced to accommodate the delay and jitter characteristic of them. Although the existing queue management schemes can address the congestion problems of TCP flows, they have some problems in supporting real-time services. That is, they show performance degradation when burst traffics are continuously going into the system after the queue is occupied at a predefined threshold level. In addition, under the congestion state, they show large jitter, which is not a desirable phenomenon for real-time transmissions. To resolve these problems, we propose a SDRED (State Dependent Random Early Detection) and dynamic scheduling scheme that can improve delay and jitter performances by adjusting RED parameters such as ma $x_{th}$ and $w_{q}$ according to the queue status. The SDRED is designed to adapt to the current traffic situation by adjusting the max,$_{th}$ and $w_{q}$ to four different levels. From the simulation results, we show that the SDRED decreases packet delays in a queue and has more stable jitter characteristics than the existing RED, BLUE, ARED and DSRED schemes.mes.mes.

Priority Polling and Dual Token Bucket based HCCA Algorithm for IEEE 802.11e WLAN (IEEE 802.11e WLAN을 위한 우선순위 폴링 및 이중 토큰 버킷 기반의 HCCA 알고리즘)

  • Lee, Dong-Yul;Lee, Chae-Woo
    • Journal of the Institute of Electronics Engineers of Korea CI
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    • v.46 no.4
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    • pp.31-41
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    • 2009
  • IEEE 802.11e proposed by IEEE 802.11 working group to guarantee QoS has contention based EDCA and contention free based HCCA. HCCA, a centralized polling based mechanism of 802.11e, needs a scheduling algorithm to allocate the network resource efficiently. The existing standard scheduler, however, is inefficient to support for QoS guarantee for real-time service having VBR traffic. To improve these limit, in this paper, we First, we propose priority polling algorithm which additionally considers the size of MSI and TXOP based on EDD algorithm to increase number of QSTAs. We also propose a dual token bucket TXOP allocation algorithm to reduce congestion caused by stations which enters network with considerable delay variance. TSPEC parameters, Maximum Burst Size (MBS) and Peak Data Rate (PR), are applied to design depth and token generation rate of two buckets. Simulation results show that the proposed algorithm has much higher performance compared with reference and SETT-EDD scheduler in terms of throughput and delay.

End-to-End Congestion Control of High-Speed Gigabit-Ethernet Networks based on Smith's Principle

  • Lee, Seung-Hyub;Cho, Kwang-Hyun
    • Proceedings of the IEEK Conference
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    • 2000.07a
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    • pp.101-104
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    • 2000
  • Nowadays, the issue of congestion control in high-speed communication networks becomes critical in view of the bandwidth-delay products for efficient data flow. In particular, the fact that the congestion is often accompanied by the data flow from the high-speed link to low-speed link is important with respect to the stability of closed-loop congestion control. The Virtual-Connection Network (VCN) in Gigabit Ethernet networks is a packet-switching based network capable of implementing cell- based connection, link-by-link flow-controlled connection, and single- or multi-destination virtual connections. VCN described herein differ from the virtual channel in ATM literature in that VCN have link-by-link flow control and can be of multi-destination. VCNs support both connection-oriented and connectionless data link layer traffic. Therefore, the worst collision scenario in Ethernet CSMA/CD with virtual collision brings about end-to-end delay. Gigabit Ethernet networks based on CSMA/CD results in non-deterministic behavior because its media access rules are based on random probability. Hence, it is difficult to obtain any sound mathematical formulation for congestion control without employing random processes or fluid-flow models. In this paper, an analytical method for the design of a congestion control scheme is proposed based on Smith's principle to overcome instability accompanied with the increase of end-to-end delays as well as to avoid cell losses. To this end, mathematical analysis is provided such that the proposed control scheme guarantees the performance improvement with respect to bandwidth and latency for selected network links with different propagation delays. In addition, guaranteed bandwidth is to be implemented by allowing individual stations to burst several frames at a time without intervening round-trip idle time.

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MPLS and Video Stream broadcast multicast transport optimization through convergence (MPLS와 멀티캐스트 융합을 통한 Video Stream 방송 전송 최적화)

  • Hwang, Seong-Kyu;Han, Seung-Jo
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.18 no.6
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    • pp.1330-1336
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    • 2014
  • QoS techniques and transmitted in real-time communication with the advancement of technology a variety of applications and services are available these days, mobile devices bogeuphwa LTE technology to the development of multimedia services with high quality can be realized. In order to satisfy this condition simply with a router with an increased bandwidth expansion by considering the increase in the routing table of the network scalability problems included. Burst traffic data to be distributed according to the environment is to be centered. To do this, the destination -based routing method to transmit the current paper -based (Source routing) routing settings are required. In this paper, published by the IETF, IP switching system based on standardized protocol Label Switching Multi-Protocol Label Switching (MPLS) network by using the existing Best Effect is difficult to guarantee QoS for multimedia transmission in MPLS network environment using optimized QoS guarantees to transmit the multicast.

A Handoff Mechanism to Avoid Congestion in Wireless Cells (무선 셀에서의 혼잡 발생을 피하는 핸드오프 방안)

  • 변해선;이미정
    • Journal of KIISE:Information Networking
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    • v.30 no.5
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    • pp.595-603
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    • 2003
  • To provide mobile nodes with continuous communication services, it is important to reduce the packet losses during handoffs. The handoffs of mobile nodes cause packet losses and decrease of TCP throughput on account of a variety of factors. One of those is the congestion in the new cell. Due to the congestion, not only the node moving into the cell but also the already existing nodes that were successfully communicating in the cell suffer the performance degradation. In this paper we propose a new handoff mechanism called‘packet freeze control’, which avoids the congestion caused by handoffs by regulating the influx of traffic burst into the new cell. Packet freeze control is applicable to a wireless network domain in which FAs(Foreign Agents) are connected hierarchically and constitute a logical tree. It gradually increases the number of packets transferred to the new cell by buffering packets in the FAs on the packet delivery path over the wireless network domain. The simulation results show that the proposed mechanism not only reduces the packet losses but also enhances the TCP throughput of other mobile nodes in the cell.

Performance Evaluation of Scaling based Dynamic Time Warping Algorithms for the Detection of Low-rate TCP Attacks (Low-rate TCP 공격 탐지를 위한 스케일링 기반 DTW 알고리즘의 성능 분석)

  • So, Won-Ho;Shim, Sang-Heon;Yoo, Kyoung-Min;Kim, Young-Chon
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.44 no.3 s.357
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    • pp.33-40
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    • 2007
  • In this paper, low-rate TCP attack as one of shrew attacks is considered and the scaling based dynamic time warping (S-DTW) algorithm is introduced. The low-rate TCP attack can not be detected by the detection method for the previous flooding DoS/DDoS (Denial of Service/Distirbuted Denial of Service) attacks due to its low average traffic rate. It, however, is a periodic short burst that exploits the homogeneity of the minimum retransmission timeout (RTO) of TCP flows and then some pattern matching mechanisms have been proposed to detect it among legitimate input flows. A DTW mechanism as one of detection approaches has proposed to detect attack input stream consisting of many legitimate or attack flows, and shown a depending method as well. This approach, however, has a problem that legitimate input stream may be caught as an attack one. In addition, it is difficult to decide a threshold for separation between the legitimate and the malicious. Thus, the causes of this problem are analyzed through simulation and the scaling by maximum auto-correlation value is executed before computing the DTW. We also discuss the results on applying various scaling approaches and using standard deviation of input streams monitored.