• 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.

• Smart Media Journal
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• v.2 no.4
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• pp.51-59
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• 2013

With increasing transmission speed of backbone networks, it is getting to provide enough available bandwidth. However, the bandwidth is not effectively utilized in volumetric data transfer. This mainly comes from the transmission protocol, TCP, which is used for most applications. TCP is inherently difficult to adapt the available bandwidth because of it's own characteristic of transfer mechanism. UDT is a prominent application level data transfer protocol which is targeting high speed network. In this paper, we propose UDT parallel transfer technologies which is adaptive to network status and then evaluate their performance in two points of view. Firstly, we measure data transfer rate of UDT with rate congestion control methods, and compare them with basic UDT. Secondly, we apply parallel transfer technologies adapted to network status, and measure their performance. Experimental results showed that UDT rate congestion control method outperforms UDT with 106% improvement in RTT 100ms section set with jitter 30ms. In addition, performance of parallel transfer with rate congestion control method showed 107% improvement than that of parallel transfer in RTT 400ms section set with jitter 20ms.

• The KIPS Transactions:PartC
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• v.11C no.1
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• pp.89-98
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• 2004

AQM (Active Queue Management) techniques such as RED (Random Early Detection) which be proposed to solve the congestion of internet perform congestion control effectively for TCP data. However, in the situation where TCP and UDP share the bottleneck link, they can not solve the problems of the unfairness and long queueing delay. In this paper, we proposed an simple queue management algorithm, called PSRED (Protocol Sensitive RED), that improves fairness and decreases queueing delay. PSRED algorithm improves fairness and decreases average queue length by distinguishes each type of flow in using protocol field of packets and applies different drop functions to them respectively.

• KIPS Transactions on Computer and Communication Systems
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• v.2 no.3
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• pp.117-124
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• 2013

In this paper we propose a TFRC(TCP Friendly Rate Control) which guarantees a minimum rate in order to improve the efficiency of the previous TFRC which cannot distinguish congestion losses and wireless losses and decreases throughput both in wired and wireless networks. This TFRC technique is able to guarantee a minimum rate for video by restricting a loss event rate with packet loss probability about existing TFRC and constraining a rate reduction from the feedback timeout. When we experimented both the existing TFRC and the new one with TCP in the same network, we found that the latter is better than the former. Consequently, it shows that the proposed TFRC can improve video streaming quality using a guaranteed minimum transmission rate.

• Journal of the Institute of Electronics and Information Engineers
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• v.52 no.3
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• pp.25-31
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• 2015

In this paper, we propose PEP based on cross-layer and network coding in digital video broadcasting-return channel via satellite (DVB-RCS) networks. In particular, we propose not only a scheme of cross-layer information exchange for interaction between TCP and the resource allocation (RA) scheme in the link layer but also a tuning algorithm for the TCP contention window (CWND) by using information on the RA in the link layer and the redundancy rate for network-coded packets. The simulation results show that TCP CWND can be adjusted by RA information in the proposed protocol. Furthermore, through the transmission of network-coded packets and the proposed CWND tuning algorithm, TCP throughput is enhanced in lossy environment due to user mobility without the unnecessary resource overhead.

• Proceedings of the Korea Information Processing Society Conference
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• 2006.05a
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• pp.1097-1100
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• 2006

최근 네트워크 기술이 발전하면서 사용자들의 요구가 다양화되고 대용량의 멀티미디어 데이터에 대한 필요성이 증가하고 있다. 실시간성과 동시성이 중요시 되는 멀티미디어 데이터의 전송에는 UDP(User Datagram Protocol)가 사용되고 있는데, UDP는 현재의 인터넷 기반인 TCP (Transmission Control Protocol )와 경쟁관계에 있어 네트워크의 혼잡을 초래하는 경우가 많았다. 또한 TCP 트랙픽은 네트워크의 혼잡제어를 수행하지만 수신자의 관점을 고려하지 않은 점이 있어서 스트리밍 전송에는 적합하지 않은 단점을 가지고 있다. 본 논문에서는 UDP(User Datagram Protocol) 트래픽의 네트워크 형평성 문제를 해결하고 수신자의 관점을 고려하기 위해 네트워크에 우선순위를 적용하여 RTP/RTCP를 기반으로 수신자의 버퍼를 고려한 혼잡제어 알고리즘을 제안하였다. 시뮬레이션 결과를 통해 우선 순위가 있는 네트워크가 우선 순위가 없는 네트워크에 비해 패킷 손실률과 버퍼 점유도 면에서 스트리밍 전송에 적합한 환경을 제공하고 있음을 확인할 수 있었다.

• 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.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.403-405
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• 2002

지금까지 TCP 혼잡 제어를 위한 여러 종류의 메커니즘들이 Connection을 적응성 있게 제어하기 위하여 사용되어 왔지만, TCP 혼잡 제어 메커니즘들은 성능상의 여러 문제점을 가지고 있는게 사실이다. 이에 IETF에서는 AQM(Active Queue Management) 메커니즘으로 RED 알고리즘을 권고했다. 그러나 이 또한 상이한 네트워크에서는 파래메터 설정에 따른 문제점이 있어, 네트워크 상황에 적절하게 대응하지 못하는 단점이 있다. 이러한 RED알고리즘의 문제점을 극복하고, 효율성을 개선하기 위해서 SRED, BLUE, FRED, DRED 등 다양한 AQM 메커니즘들이 제시되고 있다. 본 논문에서는 네트워크 트래픽 상황에 따라 적응성을 갖고 Threshold의 변경에 사용되는 패킷 손실율을 구하는데 있어 트래픽을 고려한가중치를 줌으로써 트래픽 상황을 반영하도록 했고, Threshold 설정에 있어 적응성 있는 단계를 통하여 큐 안정성을 개선하도록 하였다. 제안한 알고리즘의 성능 분석은 NS 시뮬레이터를 사용하였고, 제안한 Load Adaptive DRED 알고리즘과 DRED 알고리즘의 버퍼 관리 기법의 성능 비교 분석을 통하여 큐 안정성의 개선된 성능을 확인하였다.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11b
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• pp.1249-1252
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• 2002

클러스터와 같은 네트워크 컴퓨팅 환경에서는 신속하고 신뢰성이 보장되는 데이터 전송이 요구된다. 신뢰성 보장을 위해서 일반적으로 사용되는 전송 프로토콜은 TCP 이다. 그러나 클러스터의 하부 네트워크로서 많이 사용되는 Myrinet 은 cut-through 스위칭 방식을 기반으로 하기 때문에 네트워크 혼잡(congestion)이 발생하지 않는다. 따라서 TCP 의 혼잡 제어(congestion control) 등과 같은 루틴들은 Myrinet 상에서 불필요한 오버헤드를 발생시킨다. 본 논문은 Myrinet 네트워크에서 흐름 제어(flow control)만으로도 신뢰성을 보장할 수 있음을 보이고 TCP 보다 오버헤드가 적은 UDP에 흐름 제어를 구현한 RUM(Reliable UDP on Myrinet)을 제안한다. 성능 측정 결과, RUM 은 신뢰성을 보장함과 동시에 TCP 보다 최대 34% 더 높은 처리량(throughput)을 보이며, UDP 와 비슷한 낮은 단방향 지연시간(one-way latency)을 보장함을 알 수 있다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4C
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• pp.353-364
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• 2002

To prevent the performance degradation of TCP due to packet loss in the smooth handoff by the route optimization extension of Mobile If protocol, a buffering of packets at a base station is needed. A buffering of packets at a base station recovers the packets dropped during the handoff by forwarding the buffered packets at the old base station to the mobile user. But, when the mobile user moves to a new foreign network which is connected to a congested router, the buffered packets forwarded by the old base station are dropped and the link utilization performance degraded due to increased congestion by the forwarded packets. In this paper, when the mobile user moves to a new foreign network which is connected to a congested router, Ive propose a packet forwarding control scheme required far the old base station to improve TCP performance in mobile networks. The old base station forwards or discards the buffered packets during handoff by proposed packet forwarding control scheme based on congestion states of RED(Random Early Detection) at the congested router. Simulation results slow that link utilization performance can be improved by applying proposed packet forwarding control scheme.