• Proceedings of the IEEK Conference
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• 2000.11c
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• pp.165-168
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• 2000

Since the IP multicast was proposed, there have been many research works on reliable multicast protocols［1］. In multicast communication, many reliable multicast schemes were studied in order to overcome packet losses in the network. However the fact that packets are lost in the underlying networks but the solutions are sought in the end hosts makes the search for solutions difficult. If routers can identify packets before dropped from congestion, the routers can initiate recovery process. recovery scheme by routers is proposed. This scheme is much faster than by sender-initiated or receiver-initiated recovery and latency is smaller.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.4 no.4
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• pp.329-333
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• 2003

In this paper we analyze the performance of the low latency pre-registration handoff method of mobile IP by computer simulation. Packet losses and delays are evaluated in terms of system utilization. Foreign agents that participate in the handoff process can be modeled as queues representing input and output ports. Therefore, we propose an analytical model of pre-registration handoff by using open queueing network model. Simulation results are shown for validating analytical estimates.

• International Journal of Control, Automation, and Systems
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• v.6 no.4
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• pp.596-606
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• 2008

In this paper, we present a co-design methodology of dynamic optimal network-bandwidth allocation (ONBA) and adaptive control for networked control systems (NCSs) to optimize overall control performance and reduce total network-bandwidth usage. The proposed dynamic co-design strategy integrates adaptive feedback control with real-time scheduling. As part of this co-design methodology, a "closed-loop" ONBA algorithm for NCSs with communication constraints is presented. Network-bandwidth is dynamically assigned to each control loop according to the quality of performance (QoP) information of each control loop. As another part of the co-design methodology, a network quality of service (QoS)-adaptive control design approach is also presented. The idea is based on calculating new control values with reference to the network QoS parameters such as time delays and packet losses measured online. Simulation results show that this co-design approach significantly improves overall control performance and utilizes less bandwidth compared to static strategies.

• ETRI Journal
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• v.27 no.6
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• pp.802-805
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• 2005

A dynamic bandwidth allocation (DBA) scheme, an inter -optical network unit (ONU) bandwidth scheduling, is presented to provide quality of service (QoS) to different classes of packets in Ethernet passive optical networks (EPONs). This scheme, referred to as TADBA, is based on efficient threshold reporting from, and adaptive polling order rearranging of, ONUs. It has been shown that the network resources are efficiently allocated among the three traffic classes by guaranteeing the requested QoS, adaptively rearranging the polling orders, and avoiding nearly all fragmentation losses. Simulation results using an OPNET network simulator show that TADBA performs well in comparison to the available allocation scheme for the given parameters, such as packet delay and channel utilization.

• Journal of Communications and Networks
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• v.5 no.1
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• pp.65-72
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• 2003

Internet roundtrip delay/time (RTT) prediction plays an important role in detecting packet losses in reliable transport protocols for traditional web applications and determining proper transmission rates in many rate-based TCP-friendly protocols for Internet-based real-time applications. The widely adopted autoregressive and moving average (ARMA) model with fixed-parameters is shown to be insufficient for all scenarios due to its intrinsic limitation that it filters out all high-frequency components of RTT dynamics. In this paper, we introduce a novel parameter-varying RTT model for Internet roundtrip time prediction based on the information theory and the maximum entropy principle (MEP). Since the coefficients of the proposed RTT model are updated dynamically, the model is adaptive and it tracks RTT dynamics rapidly. The results of our experiments show that the MEP algorithm works better than the ARMA method in both RTT prediction and RTO estimation.

• Journal of Communications and Networks
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• v.9 no.3
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• pp.236-246
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• 2007

This paper investigates the performance of the transmission control protocol (TCP) transport protocol over IEEE 802.11 infrastructure based wireless networks. A wireless link is generally characterized by high transmission errors, random interference and a varying latency. The erratic packet losses usually lead to a curbing of the flow of segments on the TCP connection and thus limit TCP's performance. This paper examines the impact of the lossy nature of IEEE 802.11 wireless networks on the TCP performance and proposes a scheme to improve the performance of TCP over wireless links. A negative acknowledgment scheme, selective negative acknowledgment (SNACK), is applied on TCP over wireless networks and a series of ns-2 simulations are performed to compare its performance against that of other TCP schemes. The simulation results confirm that SNACK and its proposed enhancement SNACK-S, which incorporates a bandwidth estimation model at the sender, outperform conventional TCP implementations in 802.11 wireless networks.

• Journal of KIISE:Information Networking
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• v.30 no.2
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• pp.233-243
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• 2003

With the original Transmission Control Protocol(TCP) design, which is particularly targeted at the wired networks, a packet loss is assumed to be caused by the network congestion. In the wireless environment where the chances to lose packets due to transmission bit errors are not negligible, though, this assumption may result in unnecessary TCP performance degradation. In this paper, we propose three schemes that improve the ability to conceal the packet losses in the wireless network while limiting the degree of violating TCP end-to-end semantics to a temporary incidents. If there happens a packet loss at the wireless link and there is a chance that the loss is noticed by the sending TCP, the proposed schemes send an indirect acknowledgement. Each of the proposed schemes uses different criteria to decide whether there is a chance that the packet loss occurred in the wireless part is noticed by the sender. In order to limit the buffer overhead in the base, the indirect acknowledgements are issued only when the length of buffer is less than a certain threshold. We use simulation to compare the overhead and the performance of the proposed schemes, and to show that the proposed schemes improve the TCP performance compared to Snoop with a limited amount of buffer at the base station.

• Journal of Broadcast Engineering
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• v.10 no.4 s.29
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• pp.548-556
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• 2005

In wireless network environments, wireless channels are characterized by time-varying fading and interference conditions, which may lead to burst packet corruptions and delay variation. This can cause severe quality degradation of streaming media. To guarantee successful transmission of media over the hostile wireless networks, where channel conditions are highly fluctuating, a flexible and network-adaptive transport method is required. Thus, we propose a network-adaptive transport error control consisting of packet-level interleaved FEC and delay-constrained ARQ, which acts as an application-level transport method of streaming media to alleviate burst packet losses while adapting to the changing channel condition in wireless networks. The performances of the proposed network-adaptive transport error control, general error control schemes, and hybrid schemes are evaluated by a developed simulator at the transport-level and video quality of streaming media. Simulation results show that the proposed mechanism provides the best overall performance among compared other schemes in terms of the transport-level performance of error control and the performance of video quality for streaming media.

• Journal of the Institute of Electronics and Information Engineers
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• v.49 no.12
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• pp.73-80
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• 2012

In this article, we propose a P2P-based mobility management protocol for global seamless handover in heterogeneous wireless networks. Unlike previous mobility management protocols such as IETF MIPv4/6 and its variants, the proposed protocol can support global seamless handover without changing the existing network infrastructure. The idea of the proposed protocol is that the location management function for mobility management is separately supported from packet forwarding function, and bidirectional IP tunnels for packet transmission are dynamically constructed between two end-to-end mobile hosts. In addition, early handover techniques have been developed to avoid large handover delays and packet losses using the IEEE 802.21 Media Independent Handover functions. The architecture and signaling procedure of the proposed protocol have been designed in detail, and the mathematical analysis and simulation have been done for performance evaluation. The performance results show that the proposed protocol outperforms the existing MIPv6 and HMIPv6 in terms of handover latency and packet loss.

• 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.