• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.613-627
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• 2004

In this paper, we propose an Aggregate Fairness Maker (ARM) required for an Edge router to improve fairness of throughput among the flows of Assured Service in DiffServ with different round trip time (RTT) and we propose a user flow Three Color Marker (uf-TCM) as a flow marker that marks packets from the flow as green, yellow, or red. A yellow packet is the packet that consumes loss token in uf-TCM as well as that is demoted green packet in AM due to disobey the aggregate traffic profile. The proposed AFH promotes yellow packet to green packet or demotes green packet to yellow packet through the fair method without per-flow management, and it improves the feirness of throughput among the flows as well as link utilization. A yellow packet and a red packet have the same drop precedence at Core Router in our scheme. So we can use the RIO buffer management scheme. We evaluated the performance of our proposed AFM and the REDP Marker that was proposed to improve fairness without per-flow management. Simulation results show that, compared with the REDP marker, proposed AFM can improve performance of throughput fairness among the flows with different RTT and link utilization under the over-provisioning, exact-provisioning, and under-provisioning network environments at Multiple DiffServ domains as well as at Single DiffServ domain.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.42 no.2 s.302
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• pp.1-10
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• 2005

The current implementation of TCP for the Internet is not efficient when used for Mobile Ad hoc Networks (MANENTs). This is because TCP assumes that all packet losses are caused by congestion, whereas transmission errors are a main reason for packet losses in wireless networks. To remedy this situation and increase performance, we propose an end-to-end method of using propagation delays and the differences between propagation delays to distinguish the causes for packet losses. The proposed method has two characteristics: Firstly, it is energy-efficient because this solution is only initiated when a packet loss is detected. Secondly, our approach considers only the one way propagation delay and is more accurate in determining causes for packet losses than existing methods which consider round trip time. Petri net models of the proposed TCP and of the standard TCP have been built and simulations have been performed on them. Our simulation results show that the proposed approach increases throughput and reduces propagation delay compared with standard TCP.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.4A
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• pp.377-386
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• 2008

In this paper, a new handover management scheme has been proposed to reduce handover latency and to support fast handover without packet losses, so that it may be applicable to the wireless mobile Internet system such as IPv6-based WiBro system. To minimize the handover latency in processing of movement detection, we propose the handover management scheme which simplifies the handover message exchanging procedure between mobile subscriber station (MSS) and network by integrating layer 2 and layer 3 handovers efficiently based on the layer2 information. To reduce the processing delay from new care-of-address (NCoA) configuration during handover, we propose that NCoA is created, distributed and managed by new access control router (NACR). In addition, in order to minimize the packet transmission delay and eliminate the packet losses, the proposed scheme employs a crossover router (CR) which is upper network located over PACR and NACR and employs the packet buffering for MSS. The simulation study shows that the proposed scheme achieves loss-free packet delivery and low latency in the environment of narrow overlapped cell area or high velocity of the MSS, comparing the performance with the conventional schemes.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.19 no.10
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• pp.2263-2268
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• 2015

This paper describes packet scheduling techniques that can be used to alleviate the impact of interference. The mechanism is consisted of interference estimation and master delay police. Proposed scheduling police is effective in reducing packet loss and delay. Another advantage worth mentioning, are the additional saving s in the transmitter power since packet are not transmitted when channel is bad. This paper gives that scheduling policy works only with data traffic since voice packets need to be sent at fixed intervals. However, if the delay variance is constant and the delay can be limited to a slot, it may be worthwhile to use DM packet for voice.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4C
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• pp.353-364
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• 2002

To prevent the performance degradation of TCP due to packet loss in the smooth handoff by the route optimization extension of Mobile If protocol, a buffering of packets at a base station is needed. A buffering of packets at a base station recovers the packets dropped during the handoff by forwarding the buffered packets at the old base station to the mobile user. But, when the mobile user moves to a new foreign network which is connected to a congested router, the buffered packets forwarded by the old base station are dropped and the link utilization performance degraded due to increased congestion by the forwarded packets. In this paper, when the mobile user moves to a new foreign network which is connected to a congested router, Ive propose a packet forwarding control scheme required far the old base station to improve TCP performance in mobile networks. The old base station forwards or discards the buffered packets during handoff by proposed packet forwarding control scheme based on congestion states of RED(Random Early Detection) at the congested router. Simulation results slow that link utilization performance can be improved by applying proposed packet forwarding control scheme.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.5
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• pp.21-31
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• 2002

As the MPEG streaming traffics are transmitted over the Internet, MPEG packet losses cause error propagation in both spatial and temporal domains which in turn leads to severe degradation in image quality. This image degradation is become worse by the bursty packet loss at the network. In this paper, we propose a new error spreading method to reduce the effects of bursty packet losses over the Internet. The proposed error spreading method spreads out the not utilize packet losses widely, and can be used for videocasting and video-broadcasting that do not utilize the ARQ method. The proposed method can not only reduce the consecutive losses of image blocks, but also improve the performance of error concealment. Experimental results show that the deterioration of MPEG video quality can be reduced by spreading out the MPEG packet losses.

• International Journal of Computer Science & Network Security
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• v.22 no.7
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• pp.91-102
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• 2022

With the evolution of wireless sensor network (WSN) technology, the routing policy has foremost importance in the Internet of Things (IoT). A systematic routing policy is one of the primary mechanics to make certain the precise and robust transmission of wireless sensor networks in an energy-efficient manner. In an IoT environment, WSN is utilized for controlling services concerning data like, data gathering, sensing and transmission. With the advantages of IoT potentialities, the traditional routing in a WSN are augmented with decision-making in an energy efficient manner to concur finer optimization. In this paper, we study how to combine IoT-based deep learning classifier with routing called, Kriging Regressive Deep Belief Neural Learning (KR-DBNL) to propose an efficient data packet routing to cope with scalability issues and therefore ensure robust data packet transmission. The KR-DBNL method includes four layers, namely input layer, two hidden layers and one output layer for performing data transmission between source and destination sensor node. Initially, the KR-DBNL method acquires the patient data from different location. Followed by which, the input layer transmits sensor nodes to first hidden layer where analysis of energy consumption, bandwidth consumption and light intensity are made using kriging regression function to perform classification. According to classified results, sensor nodes are classified into higher performance and lower performance sensor nodes. The higher performance sensor nodes are then transmitted to second hidden layer. Here high performance sensor nodes neighbouring sensor with higher signal strength and frequency are selected and sent to the output layer where the actual data packet transmission is performed. Experimental evaluation is carried out on factors such as energy consumption, packet delivery ratio, packet loss rate and end-to-end delay with respect to number of patient data packets and sensor nodes.

• Journal of the Korea Society of Computer and Information
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• v.16 no.1
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• pp.133-142
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• 2011

When TCP operates in multi-hop wireless networks, it suffers from severe performance degradation due to the different characteristics of wireless networks and wired networks. This is because TCP reacts to wireless packet losses by unnecessarily decreasing its sending rate assuming the losses as congestion losses. Although several loss differentiation algorithms (LDAs) have been proposed to avoid such performance degradation, their detection accuracies are not high as much as we expect. In addition the schemes have a tendency to sacrifice the detection accuracy of congestion losses while they improve the detection accuracy of wireless losses. In this paper, we suggest a new sender-based loss differentiation scheme which enhances the detection accuracy of wireless losses while minimizing the sacrifice of the detection accuracy of congestion losses. Our scheme estimates the rate of queue usage which is highly correlated with the congestion in the network path between a TCP sender and a receiver, and it distinguishes congestion losses from wireless losses by comparing the estimated queue usage with a certain threshold. In the extensive experiments based on a network simulator, QualNet, we measure and compare each detection accuracy of wireless losses and congestion losses, and evaluate the performance enhancement in each scheme. The results show that our scheme has the highest accuracy among the LDAs and it improves the most highly TCP performance in multi-hop wireless networks.

• Journal of Institute of Control, Robotics and Systems
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• v.13 no.11
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• pp.1099-1105
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• 2007

In this paper, modified two-port time-domain passivity approach is proposed for stable bilateral control of teleoperators under time-varying communication delay. We separate input and output energy at each port of a bilateral controller, and propose a sufficient condition for satisfying the passivity of the bilateral controller including time-delay. Output energy at the master port should be less than the transmitted input energy from the slave port with time-delay, and output energy at the slave port should be less than the transmitted input energy from the master port with time-delay. For satisfying above two conditions, two passivity controllers are attached at each port of the bilateral controller. A packet reflector with wireless internet connection is used to introduce serious time-varying communication delay of teleoperators. Average amount of time-delay was about 190(msec) for round trip, and varying between 175(msec) and 275(msec). Moreover some data packet was lost during the communication due to UDP data communication. Even under the serious time-varying delay and packet loss communication condition, the proposed approach can achieve stable teleoperation in free motion and hard contact as well.

• Journal of Korea Multimedia Society
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• v.9 no.7
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• pp.844-853
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• 2006

The IEEE 802.17 Resilient Packet Ring (RPR) for future Local, Metropolitan, and Wide Area Networks was recently standardized, but it still suffer from delay jitter deterioration and even some bandwidth loss under unbalanced overload. In order to overcome these drawbacks, this paper proposes a bandwidth redistribution based fairness control method, compatible with the legacy one, in which each congested node measures the amount of available bandwidth of its bottleneck link resulted from regulating upstream nodes' shares of the link bandwidth, calculates optimal fair rate with the number of uptream nodes requiring more bandwidth, and then redistributes the available bandwidth to the upstream nodes by advertising the rate. The performance evaluation results show that the proposed method fairly redistributes 95% of the bottleneck link bandwidth with even only two redistributions.