• Proceedings of the Korean Information Science Society Conference
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• 2003.10c
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• pp.151-153
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• 2003

이전의 계층화된 비디오 멀티캐스트에서는 각 계층의 크기를 그 하위 계층의 총합과 같도록 하여 TCP와 유사한 혼잡제어를 했었다. 하지만, 혼잡 시 비디오 데이터의 대역폭을 1/2로 줄이는 것은 사용자에게 인지되는 비디오 품질에 있어 큰 변화를 보이게 되므로 바람직하지 않고, 멀티캐스트를 사용함으로써 네트워크 전체에 트래픽을 줄이는 세션에게 아무런 이득도 주지 못하였다. 따라서 제안하는 방법에서는 기존 계층을 동일한 크기의 계층으로 나누고, 혼잡의 정도에 따라 계층의 감소 간격을 적절히 함으로써, 혼잡 해소 시에는 남아 있는 계층으로 데이터를 전달할 수 있게 하였다. 이러한 결과로 혼잡 시 비디오 품질이 급격히 떨어지는 것을 방지할 수 있고, 멀티캐스트를 사용하는 세션에게 이득을 줄 수 있다.

• IEMEK Journal of Embedded Systems and Applications
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• v.5 no.1
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• pp.7-17
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• 2010

ITS(Intelligent Transport System) as things are used for Broadcast service using TDMB/TPEG/NAVI rather than personal seamless service. It is attaching weight to Traffic information gathering, Charging, Settlement service. This research is applied to improve DCCP(Datagram Congestion Control Protocol) which has function as protecting data and preserving message boundary. The improving method is like that we solve data trust in UDP because Connection and Transmission overhead in UDP is less than in TCP. We fix the data loss which is generated from unordered delivery section of IP base wireless service by using DCCP protocol. We guarantee of connection with OBE(On-Board Equipment) and reliance about transmission of data by complement to mapping table and multi-hoping. Finally, We evaluate the performance about transmission of IP based data. We constructed a test-bed near research center for this test.

• Proceedings of the Korea Information Processing Society Conference
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• 2010.11a
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• pp.1019-1022
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• 2010

네트워크 기술의 발전으로 이용할 수 있는 대역폭이 증가하고 있다. 그에 따라 증가한 대역폭을 효율적으로 사용하기 위한 전송 기술이 요구되고 있다. TCP Vegas는 RTT(Round Trip Time)를 이용해 혼잡을 미리 예측하여 윈도우 크기를 조절하는 혼잡 제어 알고리즘을 사용한다. UDT는 높은 대역폭과 큰 RTT 환경에서 대용량 데이터를 전송하기 위해 제공된 응용 기반의 전송 프로토콜이다. 본 논문에서는 UDT에 혼잡예측 알고리즘을 적용한 새로운 UDT의 혼잡제어 알고리즘을 제안한다. 혼잡예측을 통해 혼잡한 구간, 혼잡하지 않은 구간을 나누어 혼잡윈도우를 갱신한다. 혼잡하지 않은 구간에서 혼잡윈도우를 증가시키고 혼잡한 구간에서 혼잡윈도우를 감소시킴으로써 기존의 UDT보다 성능이 개선되었음을 확인 할 수 있다.

• Journal of KIISE:Information Networking
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• v.35 no.2
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• pp.99-111
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• 2008

In recent years, the needs for WLANs(Wireless Local Area Networks) technology which can access to Internet anywhere have been dramatically increased particularly in SOHO(Small Office Home Office) and Hot Spot. However, unlike wired networks, there are some unique characteristics of wireless networks. These characteristics include the burst packet losses due to unreliable wireless channel. Note that burst packet losses, which occur when the distance between the wireless station and the AP(Access Point) increase or when obstacles move temporarily between the station and AP, are very frequent in 802.11 networks. Conversely, due to burst packet losses, the performance of 802.11 networks are not always as sufficient as the current application require, particularly when they use TCP at the transport layer. The high packet loss rate over wireless links can trigger unnecessary execution of TCP congestion control algorithm, resulting in performance degradation. In order to overcome the limitations of WLANs environment, MAC-layer LDA(Loss Differentiation Algorithm)has been proposed. MAC-layer LDA prevents TCP's timeout by increasing CRD(Consecutive Retry Duration) higher than burst packet loss duration. However, in the wireless channel with high packet loss rate, MAC-layer LDA does not work well because of two reason: (a) If the CRD is lower than burst packet loss duration due to the limited increase of retry limit, end-to-end performance is degraded. (b) energy of mobile device and bandwidth utilization in the wireless link are wasted unnecessarily by Reducing the drainage speed of the network buffer due to the increase of CRD. In this paper, we propose a new retransmission module based on Cross-layer approach, called BLD(Burst Loss Detection) module, to solve the limitation of previous link layer retransmission schemes. BLD module's algorithm is retransmission mechanism at IEEE 802.11 networks and performs retransmission based on the interaction between retransmission mechanisms of the MAC layer and TCP. From the simulation by using ns-2(Network Simulator), we could see more improved TCP throughput and energy efficiency with the proposed scheme than previous mechanisms.

• The KIPS Transactions:PartC
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• v.13C no.7 s.110
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• pp.889-896
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• 2006

ATM Forum has defined a guaranteed frame rate (GFR) service to serve Internet traffic efficiently. The GFR service provides virtual connections (VCs) for minimum cell rate (MCR) guarantees and allows them to fairly share the residual bandwidth. And ATM Forum has recommended a frame-based generic cell rate algorithm (F-GCRA) as a frame classifier, which determines whether an Am cell is eligible to use the guaranteed bandwidth in a frame level. An ATM switch accommodates cells in its buffer or drops them in a frame level according to current buffer occupancy. A FIFO shared buffer has so simple structure as to be feasibly implemented in switches, but has not been able to provide an MCR guarantee for each VC without buffer management based on per-VC accounting. In this paper, we enhance the F-GCRA frame classifier to guarantee an MCR of each VC without buffer management based on per-VC accounting. The enhanced frame classifier considers burstness of TCP traffic caused by congestion control algorithm so as to enable each VC to use its reserved bandwidth sufficiently. In addition, it is able to alleviate the unfairness problem in usage of the residual bandwidth. Simulation results show that the enhanced frame classifier satisfies quality of services (QoSs) of the GFR service for the TCP traffic.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.12
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• pp.34-43
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• 2007

Recently, in addition to the sharp increase of mobile nodes, various kinds of wireless technologies are available for mobile nodes. If IPv6 technology is applied to the network, multi-homing terminals which have several public IP addresses on one interface will be common. Accordingly, there are many research activities on mobility management for multi-interface, multi-homming nodes. In this paper we propose an extended fast handover mechanism for multi-interface MIPv6 environments that uses multi-interface FBU (MFBU) message instead of the existing FBU message. The MFBU message has the "tunnel destination" mobility option that points a specific tunnel destination other than NAR, and "T" flag that indicates the existence of tunnel destination option. The proposed mechanism can improve the TCP performance by mitigating packet reordering during FMIPv6 handover that can cause unnecessary congestion control due to 3 duplicate ACKs. In this paper, we implemented a multi-Interface MIPv6 simulator by extending a single-interface MIPv6 simulator in NS-2, and showed that the performance of TCP traffic is improved by using the proposed multi-interface fast MIPv6.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.4
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• pp.720-740
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• 2011

Wi-Fi-enabled portable devices have recently been introduced into the consumer electronics market. These devices download or upload content, from or to a host machine, such as a personal computer, a laptop, a home gateway, or a media server. This paper investigates the fairness among multiple Wi-Fi-enabled portable devices in a home network when they are simultaneously communicated with the host machine. First, we present that, a simple IEEE 802.11-based home network suffers from unfairness, and the fairness is exaggerated by the wireless link errors. This unfairness is due to the asymmetric response of the TCP to data-packet loss and to acknowledgment-packet loss, and the wireless link errors that occur in the proximity of any node; the errors affect other wireless devices through the interaction at the interface queue of the home gateway. We propose a QoS-provisioning framework in order to achieve per-device fairness and service differentiation. For this purpose, we introduce the medium access price, which denotes an aggregate value of network-wide traffic load, per-device link usage, and per-device link error rate. We implemented the proposed framework in the ns-2 simulator, and carried out a simulation study to evaluate its performance with respect to fairness, service differentiation, loss and delay. The simulation results indicate that the proposed method enforces the per-device fairness, regardless of the number of devices present and regardless of the level of wireless link errors; furthermore it achieves high link utilization with only a small amount of frame losses.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.15 no.1
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• pp.195-215
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• 2021

Multipath Transport Control Protocol (MPTCP) is a transport layer protocol that enables multiple TCP connections across various paths. Due to path heterogeneity, it incurs more energy in a multipath wireless network. Recent work presents a set of approaches described in the literature to support systems for energy consumption in terms of their performance, objectives and address issues based on their design goals. The existing solutions mainly focused on the primary system model but did not discourse the overall system performance. Therefore, this paper capitalized a novel stochastically multipath scheduling scheme for data and path capacity variations. The scheduling problem formulated over MPTCP as a stochastic optimization, whose objective is to maximize the average throughput, avoid network congestion, and makes the system more stable with greater energy efficiency. To design an online algorithm that solves the formulated problem over the time slots by considering its mindrift-plus penalty form. The proposed solution was examined under extensive simulations to evaluate the anticipated stochastic optimized MPTCP (so-MPTCP) outcome and compared it with the base MPTCP and the energy-efficient MPTCP (eMPTCP) protocols. Simulation results justify the proposed algorithm's credibility by achieving remarkable improvements, higher throughput, reduced energy costs, and lower-end to end delay.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.43 no.11 s.353
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• pp.149-158
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• 2006

We proposed a method for improving a performance of TCP downstream between a desktop PC as a fixed host and a PDA as a mobile host in a wired and wireless network based on IEEE 802.11x wireless LAN. With data transmission between these heterogeneous terminals a receiving time during downstream is slower than that during upstream by 20% at maximum. The reason is that their congestion window size will be oscillated due to a significantly lower packet processing rate at receiver compared to a packet sending rate at sender. Thus it will cause to increase the number of control packets to negotiate their window size. To mitigate these allergies, we proposed two distinct methods. First, by increasing a buffer size of a PDA at application layer an internal processing speed of a socket receive buffer of TCP becomes faster and then the window size is more stable. However, a file access time in a PDA is kept nearly constant as the buffer size increases. With the buffer size of 32,768bytes the receiving time is faster by 32% than with that of 512bytes. Second, a delay between packets to be transmitted at sender should be given. With an inter-packet delay of 5ms at sender a resulting receiving time is faster by 7% than without such a delay.