• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.21-24
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• 2000

End-to-end congestion control mechanism have been critical to the robustness and stability of the Internet. Most of today's Internet traffic is TCP, and we expect this to remain so in the future. TCP/IP is the intermediate transport layer candidate for today's applications. TCP uses an adaptive window-based flow control. The congestion avoidance and control algorithms deployed by TCP aims at using the available network bandwidth. This paper compares different congestion control policies, and proposes the new design mechanism for future public networks

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4B
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• pp.255-269
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• 2006

The differentiated services(DiffServ) architecture provides packet level service differentiation through the simple and predefined Per-Hop Behaviors(PHBs). The Assured Forwarding(AF) PHB proposed as the assured services uses the RED-in/out(RIO) approach to ensusre the expected capacity specified by the service profile. However, the AF PHB fails to give good QoS and fairness to the TCP flows. This is because OUT(out- of-profile) packet droppings at the RIO buffer are unfair and sporadic during only network congestion while the TCP's congestion control algorithm works with a different round trip time(RTT). In this paper, we propose an Adaptive Regulating Drop(ARD) marker, as a novel dropping strategy at the ingressive edge router, to improve TCP fairness in assured services without a decrease in the link utilization. To drop packets pertinently, the ARD marker adaptively changes a Temporary Permitted Rate(TPR) for aggregate TCP flows. To reduce the excessive use of greedy TCP flows by notifying droppings of their IN packets constantly to them without a decrease in the link utilization, according to the TPR, the ARD marker performs random early fair remarking and dropping of their excessive IN packets at the aggregate flow level. Thus, the throughput of a TCP flow no more depends on only the sporadic and unfair OUT packet droppings at the RIO buffer in the core router. Then, the ARD marker regulates the packet transmission rate of each TCP flow to the contract rate by increasing TCP fairness, without a decrease in the link utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.5B
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• pp.427-434
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• 2003

This paper suggests an improved TCP congestion algorithm, which is more robust to lossy wireless environment than other algorithms such as TCP-Reno. The suggested algorithm decides on the size of a congestion window depending on both PER (Packet Error Rate) and its state, which is one of fast recovery state and slow start state. Some simulations are given to validate the suggested algorithm and the algorithm is compared with other TCP congestion algorithm from the point of view of performance measures such as a congestion window and throughput. The suggested algorithm has better throughput than other algorithm over wireless links with high PER and similar throughput to others over wireless links with low BER.

• Journal of Satellite, Information and Communications
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• v.8 no.1
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• pp.1-7
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• 2013

Currently satellite communication systems usually use the ACM(Adaptive Coding and Modulation) to extend the link availability and to increase the bandwidth efficiency. However, when ACM system is used for satellite communications, we should carefully consider TCP congestion control to avoid network congestions. Because MODCODs in ACM are changed to make a packet more robust according to satellite wireless link conditions, bandwidth of satellite forward link is also changed. Whereas TCP has a severe problem to control the congestion window for the changed bandwidth, then packet overflow can be experienced at MAC or PHY interface buffers. This is a reason that TCP in transport layer does not recognize a change of bandwidth capability form MAC or PHY layer. To overcome this problem, we propose the adaptive congestion control scheme of TCP for supporting ACM in Satellite PEP (Performance Enhancing Proxy) systems. Simulation results by using ns-2 show that our proposed scheme can be efficiently adapted to the changed bandwidth and TCP congestion window size, and can be useful to improve TCP performance.

• Journal of KIISE:Information Networking
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• v.27 no.3
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• pp.247-255
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• 2000

최근 이동통신 서비스의 보급이 날로 증가됨에 따라 무선 접속 인터넷 서비스이 사용에 대한 요구가 급증하고 있다. 그러나 인터넷에서 사용되는 TCP 프로토콜은 에러 발생율이 낮은 유선망을 고려하여 설계되었기 때문에 망에서 발생되는 패킷 손실을 망내의 폭주로 인한 것으로 가정하고 폭죽제어 알고리즘을 동작시켜 윈도우 크기를 줄인다. 그러나 무선통신망과 같이 에러 발생율이 높은 환경에서는 패킷 손실이 주로 엘 발생에 기인하는데, 이 경우 기존의 TCP 프로토콜을 사용하면 폭주제어 알고리즘이 동작되어 TCP의 성능을 저하시키는 문제점이 발생된다. 따라서 본 논문에서는 유무선 복합망에서 TCP 프로토콜의 성능을 개선하기 위한 Split ACKs 기법을 제안하였다. 이 기법은 기지국에서 무선링크의 패킷 손실 이후에 수신된 ACK 패킷을 여러 개로 쪼개서 TCP 송신측으로 전달한다. 따라서 여러 개의 ACK 패킷을 수신한 TCP 송신측은 폭죽제어 알고리즘이 동작되어 감소시킨 윈도우 크기를 빠르게 복귀시켜 주기 때문에, 저하된 TCP 프로토콜의 성능을 신속히 향상시킬 수 있다. 아울러 제안된 기법은 기존 TCP 프로토콜을 그대로 사용할 수 있으며, TCP의 End-to-end Semantics가 유지되는 장점이 있다. 시뮬레이션을 통한 성능분석 결과 이 기법은 기존의 TCP 프로토콜에 비해 약 20%의 성능향상을 보였다.

• Journal of Institute of Control, Robotics and Systems
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• v.18 no.8
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• pp.774-779
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• 2012

We design and implement a RTT (Round Trip Time) based TCP (Transmission Control Protocol) for USN (Ubiquitous Sensor Network). We adopt a basic update algorithm for window size from FAST TCP that uses the queuing delay at link as the congestion measure. The designed TCP estimates the queuing delay at link from the measured RTT in the network layer, and updates the window size based on the estimated queuing delay. The designed TCP allows to utilize the full capacity of USN links and avoids the waste of the given link capacity that is common without the flow control in the transport layer. The experiment results show that the window size of the sender converges within a small range of variations without any packet loss, and verify the stability and performance of the designed TCP.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.11 no.6
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• pp.81-86
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• 2011

In one of the most impacting schemes proposed to address the TCP throughput problem over network coding, the network coding layer sends an acknowledgement if an innovative linear combination is received, even when a new packet is not decoded. Although this scheme is very effective, its implementation requires a limit on the coding window size. This limitation causes low TCP throughput in the presence of packet reordering. We argue that a TCP variant detecting a packet loss relying only on timers is effective in dealing with the packet reordering problem in network coding environments as well. Also we propose a new network coding layer to support such a TCP variant. Simulation results for a 2-path environment show that our proposed scheme improves TCP throughput by 19%.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.11
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• pp.4068-4086
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• 2014

Due to the prevalence of powerful mobile terminals and the rapid advancements in wireless communication technologies, the wireless video streaming service has become increasingly more popular. Recent studies show that video streaming services via Transmission Control Protocol (TCP) are becoming more practical. TCP has more advantages than User Diagram Protocol (UDP), including firewall traversal, bandwidth fairness, and reliability. However, each video service shares an equal portion of the limited bandwidth because of the fair sharing characteristics inherent in TCP and this bandwidth fair sharing cannot always guarantee the video quality for each user. To solve this challenging problem, an Adaptive Multiple TCP (AM-TCP) scheme is proposed in this paper to guarantee the video quality for mobile devices in wireless networks. AM-TCP adaptively controls the number of TCP connections according to the video Rate Distortion (RD) characteristics of each stream and network status. The proposed scheme can minimize the total distortion of all participating video streams and maximize the service quality by guaranteeing the quality of each video streaming session. The simulation results show that the proposed scheme can significantly improve the quality of video streaming in wireless networks.

• Journal of Internet Computing and Services
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• v.8 no.1
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• pp.15-23
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• 2007

Wireless LAN (IEEE 802.11) uses traditional TCP for reliable data transmission, But it brings the unintentional packet loss which is not congestion loss caused by handoff, interference, and fading in wireless LAN. In wireless LAN, TCP experiences performance degradation because it consumes that the cause of packet loss is congestion, and it decrease the sending rate by activating congestion control algorithm. This paper analyzes that correlation of throughput and buffer size for wireless buffer tuning. We find MBT (Maximum Buffer Threshold) which does not increase the throughput through the analysis, For calculation of MBT, we experiment the throughput by using high volume music data which is creased by real-time performance of piano. The experiment results is shown that buffer tuing based on MBT shows 20.3%, 21.4%, and 45.4% throughput improvement under 5ms RTT, 10ms RTT, and 20ms RTT, respectively, comparing with the throughput of operation system default buffer size, In addition, we describe that The setting of TCP buffer size by exceeding MBT does not have an effect on the performance of TCP.

• The KIPS Transactions:PartC
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• v.11C no.2
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• pp.245-252
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• 2004

Stream Control Transmission Protocol(SCTP) is a new connection-oriented, reliable delivery transport protocol operating on top of an unreliable connectionless packet service such as IP. It inherits many of the functions developed for TCP, including flow control and packet loss recovery functions. In addition, it also supports transport layer multihoming and multistreaming In this paper, we study the impact of multi-homing on the performance of SCTP. We first compare performance of single-homed SCTP. multi-homed SCTP, TCP Reno and TCP SACK. We, then describe potential flaw in the current SCTP retransmission policy, when SCTP host is multihomed. Our Results show that SCTP performs better than TCP Reno and TCP SACK due to several changes from TCP in its congestion control mechanism. In particular. multi-homed SCTP shows the best result among the compared schemes. Through experimentation for multi-homed SCTP, we found that the current SCTP retransmission policy nay deteriorate the perfomance when the retransmission path it worse than the original path. Therefore, the condition of retransmission path is a very important factor In SCTP performance and a proper mechanism would be required to measure the condition of the retransmission path.