• Journal of Internet Computing and Services
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• v.3 no.3
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• pp.67-74
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• 2002

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rate to send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link, An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used for feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• 제어로봇시스템학회:학술대회논문집
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• 2004.08a
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• pp.474-479
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• 2004

This paper describes a feedback-based congestion control algorithm to improve TCP performance over wireless network. In this paper, we adjust the packet marking probability at the router for Max-Min fair sharing of the bandwidth and full utilization of the link. Using the successive ECN (Explicit Congestion Notification), the proposed algorithm regulates the window size to avoid the congestion and sees the packet loss only due to the wireless link error. Based on the asymptotic analysis, it is shown that the proposed algorithm guarantees the QoS of the wireless TCP. The effectiveness of the proposed algorithm is demonstrated by simulations.

• Journal of Korea Multimedia Society
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• v.19 no.5
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• pp.924-931
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• 2016

In this paper, Various network adaptive MAC scheduling technique is proposed to effectively overcome limits of narrow bandwidth and low transmission speed in underwater. UPFC(Underwater Packet Flow Control) is a technique to reduce both the number of transmission and transmission time using three types (Normal, Blocked, Parallel) of data transmission. In this technique, the load information, in which a transmission node have, is transmitted to destination node using marginal bit in reserved header. Then the transmitted information is referred to determine weighting factor. According to the weighting factor, scheduling is dynamically changed adaptively. The performance of UPFC is analyzed and flow control technique which can be applied to Cluster Based Network and Ad Hoc network as well.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.5 no.6
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• pp.1147-1165
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• 2011

We present a simple and cost effective rate control scheme for streaming video over a wireless channel by using the information of mobile devices' buffer level. To prevent buffer fullness and emptiness at receivers, the server should be able to adjust sending rate according to receivers' buffer status. We propose methods to adjust sending rate based on the buffer level and discrete derivative of the buffer occupancy. To be compatible with existing network protocols, we provide methods to adjust sending rate by changing the inter-packet delay (IPD) at the server side. At every round-trip time, adjustments of sending rate are made in order to achieve responsiveness to sudden changes of buffer availabilities. A series of simulations and the prototype system showed that the proposed methods did not cause buffer overflows and it can maintain smoother rate control and react to bandwidth changes promptly.

• Journal of the Korean Institute of Telematics and Electronics A
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• v.31A no.4
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• pp.21-29
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• 1994

Three protocols based on the slotted Aloha technique are proposed for very high-speed optical fiber local area networks using wavelength division multiplexing (WDM) passive star topology and the throughputs and delays are derived. For getting a high probability in successful transmission of control packets determining the transmission of a data packet, we adopt control mini slot groups in these protocols. The retransmission probability is also considered in analysis. Both throughput and delay of three protocols are compared and analyzed by varying the number of control solt groups, the retransmission probability the length of a data packet and the number of channels. The numerical analysis shows that the proposed protocols adopted the control slot groups give the increase of throughput and the decrease of delay.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.8
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• pp.3809-3822
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• 2017

Wireless Sensor Networks hold the limelight because of significant potential for distributed sensing of large geographical areas. The radio duty cycling mechanism that turns off the radio periodically is necessary for the energy conservation, but it deteriorates the network congestion when the traffic load is high, which increases the packet loss and the delay too. Although many papers for WSNs have tried to mitigate network congestion, none of them has mentioned the congestion problem caused by the radio duty cycling of MAC protocols. In this paper, we present a simple and efficient congestion control technique that operates on the radio duty cycling MAC protocol. It detects the congestion by checking the current queue size. If it detects the congestion, it extends the network capacity by adding supplementary wakeup times. Simulation results show that our proposed scheme highly reduces the packet loss and the delay.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.11 no.9
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• pp.3502-3509
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• 2010

Streaming service in the wireless mobile network environment has been a very challenging issue due to the dynamic uncertain nature of the channels. Overhead such as congestion, latency, and jitter lead to the problem of performance degradation of an adaptive multi-streaming service. This paper proposes a AMSS (Adaptive Multi-level Streaming Service) mechanism to reduce the performance degradation due to overhead such as variable network bandwidth, mobility and limited resources of the wireless mobile network. The proposed AMSS optimizes streaming services by: 1) use of fuzzy similarity metric, 2) minimization of packet loss due to buffer overflow and resource waste, and 3) minimization of packet loss due to congestion and delay. The simulation result shows that the proposed method has better performance in congestion control and packet loss ratio than the other existing methods of TCP-based method, UDP-based method and VBM-based method. The proposed method showed improvement of 10% in congestion control ratio and 8% in packet loss ratio compared with VBM-based method which is one of the best method.

• The Journal of the Korea Contents Association
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• v.7 no.1
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• pp.131-138
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• 2007

In wired networks, packet losses mostly occur due to congestion. TCP reacts to the congestion by decreasing its congestion window, thus to reduce network utilization. In wireless networks, however, losses may occur due to the high bit-error rate of the transmission medium or due to fading and mobility. Nevertheless, TCP still reacts to packet losses according to its congestion control scheme, thus to reduce the network utilization unnecessarily. This reduction of network utilization causes the performance of TCP to decrease. In this paper, we predict packet loss by using RSS(Received Signal Strengths) on the wireless and suggest adding RSS flag bit in ACK packet of MH. By using RSS flag bit in ACK, the FH(Fixed Host) decides whether it adopt congestion control scheme or not for the maximum throughput. The result of the simulation by NS-2 shows that the proposed mechanism significantly increases sending amount and receiving amount by 40% at maximum.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.2 no.1 s.2
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• pp.93-100
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• 2003

In this farer, An ADSRC(Advanced Dedicated Short Range Communication) packet communication system developed by ETRI is introduced. The ADSRC system has been developed to provide high-speed, short-range wireless racket communication in roadside environment for mobile office services. The requirements of the ADSRC system for mobile office services and the system design specification to meet them with regard to mobile of nce environment are discussed. The ADSRC packet communication systems consist of the MAC(Medium Access Control) Processor block the OFDM() modem block and the RF block. The MAC processor block handles medium access control. The OFDM modem transmits data packets at up to 24Mbps adaptively and recovers the data from RF block. The ADSRC packet communication system architecture is described.

• Journal of KIISE:Information Networking
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• v.32 no.3
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• pp.382-390
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• 2005

As today's networks evolve towards an If-based integrated network, the role of transmission control protocol(TCP) has been increasing as well. As a well-known issue, the performance of TCP is affected by its loss recovery mechanism that is comprised of two algorithms; fast retransmit and fast recovery. Although retransmission timeout(RTO) caused by multiple packet losses can be avoided by using selective acknowledgement(SACK) option, RTO cannot be avoided if a retransmitted packet is lost. Therefore, we propose a simple modification to make it possible for a TCP sender using SACK option to detect a lost retransmission. In order to evaluate the proposed algorithm, simulations have been performed for two scenarios where packet losses are random and correlated. Simulation results show that the proposed algorithm can improve TCP performance significantly.