• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.7B
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• pp.477-487
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• 2008

Congestion control of the TCP reduces transmission rate when it detects packet loss because packet loss origines from congestion in the wired network. In the wireless network, packet loss comes from channel errors. Wired TCP degrades performance when there are wireless losses because it does not classify type of loss. These day, there are many researches which classify type of loss between congestion loss and wireless loss for wired-wireless hybrid network. For wireless TCP, many of existing algorithms are based on the estimated bandwidth or variations of packet arrival time. In this paper, we propose a new TCP scheme to distinguish the wireless packet losses from the congestion packet losses using MIB of the IEEE 802.11 MAC. We perform excessive simulations using the NS-2 network simulator and analyze the simulation results to compare the performance of the proposed algorithm to other well-known algorithms. From simulation results, we know that proposed algorithm improves performance about 12% and 32% compared with Spike algorithm and mBiaz algorithm, respectively.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.7
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• pp.1599-1609
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• 2014

TCP does not ensure the bandwidth and delay bound required for multimedia streaming services in broadband wireless network environments. In this paper, we propose a new congestion control scheme for efficient multimedia transmission, called COLO TCP (Concave Increase Slow Start Logarithmic Increase Congestion Avoidance TCP). The COLO TCP prevents the burst packet loss by applying the concave increase algorithm in slow start phase. In the congestion avoidance phase, COLO TCP uses the logarithmic increase algorithm that quickly recovers congestion window after packet loss. To highly utilize network bandwidth and reduce packet loss ratio, COLO TCP uses additive increase algorithm and adaptive decrease algorithm. Through simulation results, we prove that our COLO TCP is more robust for random loss. It is also possible for efficient multimedia transmission.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.112-114
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• 2002

인터넷은 무선 이동통신 환경에서 높은 지연 시간과 낮은 전송률로 인해 대용량의데이터를 전송하기 위한 기존의 TCP/IP, HTTP등의 인터넷 프로토콜은 적합하지 않다. 이러한 문제점을 해결하기 위해서 WAP이 등장하였으며, 성능 개선을 위해 여러 단체에서 연구 수행 중 이다. 본 논문에서는 Prefetching Scheme을 기존 WAP에 적용하여 그에 따른 성능변화를 시뮬레이션을 통해 분석한다.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.4
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• pp.369-377
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• 2008

Today, Ethernet technology is rapidly developing to have a bandwidth of 10Gbps beyond 1Gbps. In such high-speed networks, the existing method that host CPU processes TCP/IP in the operating system causes numerous overheads. As a result of the overheads, user applications cannot get the enough computing power from the host CPU. To solve this problem, the TCP/IP Offload Engine(TOE) technology was emerged. TOE is a specialized NIC which processes the TCP/IP instead of the host CPU. In this paper, we implemented a high-performance, lightweight TCP/IP(HL-TCP) for the TOE and applied it to an embedded system. The HL-TCP supports existing fundamental TCP/IP functions; flow control, congestion control, retransmission, delayed ACK, processing out-of-order packets. And it was implemented to utilize Ethernet MAC's hardware features such as TCP segmentation offload(TSO), checksum offload(CSO) and interrupt coalescing. Also we eliminated the copy overhead from the host memory to the NIC memory when sending data and we implemented an efficient DMA mechanism for the TCP retransmission. The TOE using the HL-TCP has the CPU utilization of less than 6% and the bandwidth of 453Mbps.

• The Journal of Korean Association of Computer Education
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• v.7 no.1
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• pp.37-44
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• 2004

Differentiated services network (DiffServ) aims to provide the same service to a group of connections that have similar Quality of Service requirements. One of the essential function to realize DiffServ is the traffic conditioning mechanism. The paper proposes the enhanced traffic conditioning mechanism which can assure the reserved rates of TCP and UDP flows and support fair distribution of excess bandwidth. The simulation results show that the new mechanism is rather insensitive of the effect of UDP against TCP throughput, and performs better both in terms of throughput assurance and fair distribution of excess bandwidth in case of well-provisioned and over-provisioned network environment.

• Journal of KIISE:Information Networking
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• v.28 no.1
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• pp.117-125
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• 2001

본 논문은 인터넷상에서 실시간 멀티미디어 서브를 제공하기 위한 분산 멀티미디어 시스템의 설계 및 구현 방안을 제안한다. 제안하는 시스템은 데이터와 제어 채널의 분리를 통하여 다양한 멀티미디어의 효율적인 전송과 기반으로 구성되며 멀티미디어의 효율적인 실시간 전송을 지원한다. 제어 채널은 TCP와 RTCP 패킷 포맷을 기반으로 구성되며 서버와 클라이언트간의 정보교환을 지원함으로써 서비스 환경에 대한 지속적인 모니터링 및 효과적인관리를 지원한다. 특히 제안하는 서비스 품질관리 모델은 전송률에 기반하며 전송률을 협상된 서비스 품질이 보장되는 최대치와 최소치 사이에서 유지함으로써 안정된 서비스 품질을 보장한다. 즉 서비스 환경의 동적인 변화에 대응 하여 적절한 전송률과 누락된 패킷에 대한 재전송 등을 결정한다. 성능분석 결과, 제안하는 모델은 환경변화에 대한 적응성 측면에서 우수성을 나타냈으며 서비스 품질보장이 중요한 서비스에 적합함을 알 수 있었다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.149-158
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• 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.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.12 no.8
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• pp.1384-1390
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• 2008

The TCP protocol is originally designed for wired network, however it performs very poor in wireless network due to different nature of wireless network from wired networks. In terms of TCP performance improvement in wireless Ad-hoc network, many researches show that small congestion window size of TCP connection can improve TCP performance. We propose a new TCP algorithm to improve TCP performance in wireless Ad-hoc network. The basic idea of our approach is that TCP receiver estimates the optimum window size and then sets congestion window limit of TCP sender to an optimum value by using the advertised window field in TCP ACK packet. From extensive computer simulation, the proposed algorithm shows superior performance than traditional TCP protocols in terms of packet delivery ratio and packet loss.

• 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.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.38B no.9
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• pp.700-706
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• 2013

In this paper, we propose a method for applying random linear network coding in satellite communication to improve reliability. In the proposed protocol, network-coded redundancy (NC-R) packets are transmitted in the PEP (Performance Enhancement Proxy). Therefore, if data packets is lost by wireless channel error, they can be recovered by NC-R packets. We also develop the TCP performance model of the proposed protocol and evaluate the performance of the proposed protocol. In the simulation results, It is shown that the proposed protocol can improve the TCP throughput as compared with that of the conventional TCP because the NC-R packets is sent by the sender-side PEP and the receiver-side PEP use these packets to recover the lost packets, resulting in reducing the packet loss in TCP.