• Title/Summary/Keyword: TCP throughput

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The Performance Improvement using Rate Control in End-to-End Network Systems (종단간 네트워크 시스템에서 승인 압축 비율 제어를 이용한 TCP 성능 개선)

  • Kim, Gwang-Jun;Yoon, Chan-Ho;Kim, Chun-Suk
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.9 no.1
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    • pp.45-57
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    • 2005
  • In this paper, we extend the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. Even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congetion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps. By analyzing the periodic bursty behavior of the source IP queue, we derive estimated for the maximum queue size and arrive at a simple predictor for the degraded throughput, applicable for relatively general situations.

Modified RTT Estimation Scheme for Improving Throughput of Delay-based TCP in Wireless Networks (무선 환경에서 지연기반 TCP의 성능 향상을 위한 수정된 RTT 측정 기법)

  • Kang, Hyunsoo;Park, Jiwoo;Chung, Kwangsue
    • Journal of KIISE
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    • v.43 no.8
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    • pp.919-926
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    • 2016
  • In a wireless network, TCP causes the performance degradation because of mistaking packet loss, which is caused by characteristics of wireless link and throughput oscillation due to change of devices connected on a limited bandwidth. Delay based TCP is not affected by packet loss because it controls window size by using the RTT. Therefore, it can solve the problem of unnecessary degradation of the rate caused by misunderstanding reason of packet loss. In this paper, we propose an algorithm for improving the remaining problems by using delay based TCP. The proposed scheme can change throughput adaptively by adding the RTT, which rapidly reflects the network conditions to BaseRTT. It changes the weight of RTT and the increases and decreases window size based on the remaining amount of the buffer. The simulation indicated that proposed scheme can alleviate the throughput oscillation problem, as compared to the legacy TCP Vegas.

TCP Congestion Control of Transfer Rate-based in End-to-End Network Systems (종단간 네트워크 시스템에서 전송율 기반 TCP 혼잡제어)

  • Bae, Young-Geun;Yoon, Chan-Ho;Kim, Gwang-Jun
    • The Journal of the Korea institute of electronic communication sciences
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    • v.1 no.2
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    • pp.102-109
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    • 2006
  • In this paper, we improve the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. For example, even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congestion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps.

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Performance Evaluation of MPTCP over Shared Bottleneck Link (공유된 병목 링크를 경유하는 MPTCP의 성능 평가)

  • Nguyen, Van Dien;Ro, Soonghwan
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.40 no.1
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    • pp.70-77
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    • 2015
  • In this paper, we present experimental results evaluating the performance of the Multipath TCP over shared bottleneck path in series of benchmark tests. In summary, we find that the Multipath TCP's fairness as well as its competitive responds to the change of network conditions such as latency, loss rate${\cdots}$ MPTCP is extremely unfair and powerful with regular TCP in ideal network conditions but its throughput decreases clearly even less than regular TCP in worse network conditions with very high latency, higher packet loss rate.

Research on the enhancement of throughput for traffic in WLAN (초고속 무선 랜에서 트래픽 간의 처리율 향상을 위한 연구)

  • Song, Byunjin;Lee, Seonhee
    • Journal of Satellite, Information and Communications
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    • v.10 no.3
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    • pp.53-56
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    • 2015
  • In this paper, we want provide improved services with faster transmission, IEEE 802.11n was standardized. A-MPDU (Aggregation MAC Protocol Data UNIT) is a vital function of the IEEE 802.11n standard, which was proposed to improve transmission rate by reducing frame transmission overhead. In this paper, we show the problems of TCP retransmission with A-MPDU and propose a solution utilizing the property of TCP cumulative ACK. If the transmission of an MPDU subframe fails, A-MPDU mechanism allows selective re-transmission of failed MPDU subframe in the MAC layer. In TCP traffic transmission, however, a failed MPDU transmission causes TCP Duplicate ACK, which causes unnecessary TCP re-transmission. Furthermore, congestion control of TCP causes reduction in throughput. By supressing unnecessary duplicate ACKs the proposed mechanism reduces the overhead in transmitting redundant TCP ACKs, and transmitting only a HS-ACK with the highest sequence number. By using the RACK mechanism, through the simulation results, it was conrmed that the RACK mechanism increases up to 20% compared the conventional A-MPDU, at the same time, it tightly assures the fairness among TCP flows.

Prevention of Buffer Overflow in the Mobility Support Router for I-TCP (I-TCP를 위한 이동성 지원 라우터에서의 버퍼 오버플로우 방지)

  • 김창호;최학준;장주욱
    • Journal of KIISE:Information Networking
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    • v.31 no.1
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    • pp.20-26
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    • 2004
  • A congestion control algorithm to prevent buffer overflow in MSR(Mobility Support Router) for I-TCP is proposed. Due to high bit error rate and frequent hand-offs over wireless environment, the current congestion control scheme in TCP Reno over mixed(wired and wireless) network exhibits lower throughput than the throughput achieved over wired only network. I-TCP has been proposed to address this by splitting a TCP connection into two TCP connections over wired section and wireless section, respectively. However, buffer overflow in MSR may occur whenever there are excessive bit errors or frequent hand-offs. This may lead to the loss of packets acked by MSR(resident in buffer) to the sender, but not received by the receiver, breaking TCP end-to-end semantics. In this Paper, a new scheme is proposed to prevent the MSR buffer from overflow by introducing “flow control” between the sender and the MSR. Advertised window for the TCP connection between the sender and the MSR is tied to the remaining MSR buffer space, controlling the flow of packets to the MSR buffer before overflow occurs.

Optimum TCP/IP Packet Size for Maximizing ATM Layer Throughput in Wireless ATM LAN

  • Lee, Ha-Cheol
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.31 no.11B
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    • pp.953-959
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    • 2006
  • This paper provides optimum TCP/IP packet size that maximizes the throughput efficiency of ATM layer as a function of TCP/IP packet length for several values of channel BER over wireless ATM LAN links applying data link error control schemes to reduce error problems encountered in using wireless links. For TCP/IP delay-insensitive traffc requiring reliable delivery, it is necessary to adopt data link layer ARQ protocol. So ARQ error control schemes considered in this paper include GBN ARQ, SR ARQ and type-I Hybrid ARQ, which ARQ is needed, but FEC can be used to reduce the number of retransmissions. Especially adaptive type-I Hybrid ARQ scheme is necessary for a variable channel condition to make the physical layer as SONET-like as possible.

An improved performance of TCP traffic connection congestion control in wireless networks (무선네트워크에서 TCP 트래픽 연결 혼잡제어에 관한 성능 개선)

  • Ra Sang-dong;Na Ha-sun;Park Dong-suk
    • Journal of the Korea Institute of Information and Communication Engineering
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    • v.10 no.2
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    • pp.264-270
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    • 2006
  • In this paper we verified that the window based TCP performance of throughput can be improved by the traffic connection efficiency. and have studied the performance of traffic congestion control that is controlling transmission rate. In wireless network, the bidirectional node is run by estimating the usage rate of link of error control idle and the throughput is shown by transmitting segments. The throughput rate shows almost no delay due to the bidirectional traffic connection efficiency up to the allowable point as increasing the transport rate by the critical value, depending on the size of end-to-end node queue of the increase of transport rate. This paper reports the performance improvement as the number of feedback connection traffic congestion control increases because of the increase of the number of asynchronous transport TCP connections.

A Study on Local Retransmission Timeout of AT-Snoop Protocol (AT-Snoop 프로토콜의 지역 재전송 시간에 관한 연구)

  • Cho Yong bum;Cho Sung joon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.30 no.4B
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    • pp.218-225
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    • 2005
  • Although Snoop protocol can enhance TCP throughput efficiently in a wired-cum-wireless environment, it has a problem in performing local packet retransmissions under a burst error-prone wireless link. AT-Snoop protocol is proposed to cope with this Snoop protocol's problem by adopting adaptive timer. In this paper, TCP throughputs of AT-Snoop protocol have been analyzed with varying wireless link conditions and the ways of setting parameters of AT-Snoop protocol for higher TCP throughput are found out through computer simulations. From the simulation results, AT-Snoop protocol's two parameters, local retransmission threshold value and local retransmission timeout value, are closely related with the fading changing rate. To get higher TCP throughput, local retransmission threshold value and local retransmission timeout value should be set to a little bit larger values than average WSRTT(Wireless Smoothed Round Trip Time) and mean bad period of the wireless link, respectively.

Analytical Modeling of TCP Dynamics in Infrastructure-Based IEEE 802.11 WLANs

  • Yu, Jeong-Gyun;Choi, Sung-Hyun;Qiao, Daji
    • Journal of Communications and Networks
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    • v.11 no.5
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    • pp.518-528
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    • 2009
  • IEEE 802.11 wireless local area network (WLAN) has become the prevailing solution for wireless Internet access while transport control protocol (TCP) is the dominant transport-layer protocol in the Internet. It is known that, in an infrastructure-based WLAN with multiple stations carrying long-lived TCP flows, the number of TCP stations that are actively contending to access the wireless channel remains very small. Hence, the aggregate TCP throughput is basically independent of the total number of TCP stations. This phenomenon is due to the closed-loop nature of TCP flow control and the bottleneck downlink (i.e., access point-to-station) transmissions in infrastructure-based WLANs. In this paper, we develop a comprehensive analytical model to study TCP dynamics in infrastructure-based 802.11 WLANs. We calculate the average number of active TCP stations and the aggregate TCP throughput using our model for given total number of TCP stations and the maximum TCP receive window size. We find out that the default minimum contention window sizes specified in the standards (i.e., 31 and 15 for 802.11b and 802.11a, respectively) are not optimal in terms of TCP throughput maximization. Via ns-2 simulation, we verify the correctness of our analytical model and study the effects of some of the simplifying assumptions employed in the model. Simulation results show that our model is reasonably accurate, particularly when the wireline delay is small and/or the packet loss rate is low.