• Journal of KIISE:Information Networking
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• v.37 no.5
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• pp.363-373
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• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.45 no.10
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• pp.29-39
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• 2008

Greedy routing which is easy to apply to geographic wireless sensor networks is frequently used. Greedy routing works well in dense networks whereas in sparse networks it may fail. When greedy routing fails, it needs a recovery algorithm to get out of the communication void. However, additional recovery algorithm causes problems that increase both the amount of packet transmission and energy consumption. Communication void is a condition where all neighbor nodes are further away from the destination than the node currently holding a packet and it therefore cannot forward a packet using greedy forwarding. Therefore we propose a VODUA(Virtually Ordered Distance Upgrade Algorithm) as a novel idea to improve and solve the problem of void. In VODUA, nodes exchange routing graphs that indicate information of connection among the nodes and if there exist a stuck node that cannot forward packets, it is terminated using Distance Cost(DC). In this study, we indicate that packets reach successfully their destination while avoiding void through upgrading of DC. We designed the VODUA algorithm to find valid routes through faster delivery and less energy consumption without requirement for an additional recovery algorithm. Moreover, by using VODUA, a network can be adapted rapidly to node's failure or topological change. This is because the algorithm utilizes information of single hop instead of topological information of entire network. Simulation results show that VODUA can deliver packets from source node to destination with shorter time and less hops than other pre-existing algorithms like GPSR and DUA.

• Journal of the Korean Institute of Intelligent Systems
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• v.13 no.6
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• pp.692-697
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• 2003

The researches on the Internet based tole-operation have received increased attention for the past few years. In this paper, we implement the Internet based tele-operating system. In order to transmit robustly the surroundings and control information of the robot, we make a data as a packet type. Also in order to transmit a very large image data, we use PEG compressive algorithm. The central problem in the Internet based tele-operation is the data transmission latency or data-loss. For this specific problem, we introduce an autonomous mobile robot with a 2-layer fuzzy controller. Also, we implement the color detection system and the robot can perceive the object. We verify the efficacy of the 2-layer fuzzy controller by applying it to a robot that is equipped with various input sensors. Because the 2-layer fuzzy controller can control robustly the robot with various inputs and outputs and the cost of control is low, we hope it will be applied to various sectors.

• Journal of KIISE:Information Networking
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• v.34 no.6
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• pp.435-445
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• 2007

Snoop protocol is one of the efficient schemes to compensate TCP packet loss and enhance TCP throughput in wired-cum-wireless networks. However, Snoop protocol has a problem: it cannot perform local retransmission efficiently under the bursty-error prone wireless link. To solve this problem, SACK-Aware-Snoop and SNACK mechanism have been proposed. These approaches improve the performance by using SACK option field between base station and mobile host. However in the wireless channel with high packet loss rate, SACK-Aware-Snoop and SNACK mechanism do not work well because of two reason: (a) end-to-end performance is degraded because duplicate ACKs themself can be lost in the presence of bursty error, (b) energy of mobile device and bandwidth utilization in the wireless link are wasted unnecessarily because of SACK option field in the wireless link. In this paper, we propose a new local retransmission scheme based on Cross-layer approach, called Cross-layer Snoop(C-Snoop) protocol, to solve the limitation of previous localized link layer schemes. C-Snoop protocol includes caching lost TCP data and performing local retransmission based on a few policies dealing with MAC-layer's timeout and local retransmission timeout. From the simulation result, we could see more improved TCP throughput and energy efficiency than previous mechanisms.

• Journal of KIISE
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• v.42 no.10
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• pp.1254-1267
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• 2015

Recently, excessive installations of APs have caused WLAN interference, and many techniques have been suggested to solve this problem. The AP aggregation technique serves to reduce active APs by moving station connections to a certain AP. Since this technique forcibly moves station connections, the transmission performance of some stations may deteriorate. The AP transmit power control technique may cause station disconnection or deterioration of transmission performance when power is reduced under a certain level. The combination of these two techniques can reduce interference through AP aggregation and narrow the range of interferences further through detailed power adjustment. However, simply combining these techniques may decrease the probability of power adjustment after aggregation and increase station disconnections upon power control. As a result, improvement in performance may be insignificant. Hence, this study suggests a scheme to combine the AP aggregation and the AP transmit power control techniques in OpenFlow-based WLAN to ameliorate the disadvantages of each technique and to reduce interferences efficiently by performing aggregation for the purpose of increasing the probability of adjusting transmission power. Simulations reveal that the average transmission delay of the suggested scheme is reduced by as much as 12.8% compared to the aggregation scheme and by as much as 18.1% compared to the power control scheme. The packet loss rate due to interference is reduced by as much as 24.9% compared to the aggregation scheme and by as much as 46.7% compared to the power control scheme. In addition, the aggregation scheme and the power control scheme decrease the throughput of several stations as a side effect, but our scheme increases the total data throughput without decreasing the throughput of each station.

• Journal of KIISE:Information Networking
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• v.34 no.6
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• pp.466-474
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• 2007

Asymmetric networks that the downlink bandwidth is larger than the uplink bandwidth may cause the degradation of the TCP performance due to the uplink congestion. In order to solve this problem, this paper designs and implements the Dynamic Segment Size Control mechanism which offers a suitable segment size for current networks. The proposed mechanism does not require any changes in customer premises but suppress the number of ACKs using segment reassembly technique to avoid the uplink congestion. The gateway which adapted the Dynamic Segment Size Control mechanism, detects the uplink congestion condition and dynamically measures the bandwidth asymmetric ratio and the packet loss ratio. The gateway reassembles some of segments received from the server into a large segment and transmits it to the client. This reduces the number of corresponding ACKs. In this mechanism, the SACK option is used when occurs the bit error during the transmission. Based on the simulation in the GEO satellite network environment, we analyzed the performance of the Dynamic Segment Size Control mechanism.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.8A
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• pp.745-751
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• 2006

The transmitter requires knowledge of the channel status information in order to adopt the adaptive modulation and coding scheme(AMC) for OFDM system. But in the outdoor environment which the users have high mobility, the channel status information from the users is outdated, so that it induces the degradation of system throughput and packet error rate(PER) performance. To solve this problem, researches about applying channel prediction technique to the AMC scheme have been proceeded. Most channel prediction techniques assume that there is no channel variation in the predefined time duration, e.g., a slot. As a result, those techniques cannot compensate the degradation of PER performance resulting from the rapid variation of channel during the slot duration. This paper introduces a novel channel prediction technique for OFDM/FDD system to support adaptive modulation and coding scheme over rapidly time-varying multipath fading channel. The proposed channel prediction technique considers the time-varying nature of channel during the slot duration. Simulation results show that the AMC scheme of OFDM/FDD system utilizing the proposed channel prediction technique can guarantee the target PER of 1% without any loss of system throughput compared with the case supported by the conventional channel prediction under ITU-R Veh A 30km/h.

• 대한공업교육학회지
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• v.33 no.2
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• pp.287-299
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• 2008

HTTP is one of the most widely used protocols of the WWW. Currently it uses TCP as the transport layer protocol to provide reliability. The HTTP uses separate TCP connection for each file request and adds unnecessary head-of-line blocking overhead for the file retrieval. The web application is short sized and affected by the increased handover latency of TCP in wireless environment. SCTP has attractive features such as multi-streaming and multi-homing. SCTP's multi-streaming and multi-homing avoid head-of-line blocking problem of TCP and reduce handover latency of TCP in wired and wireless environment. Mean response time is the important measure in most web application. In this paper, we present the comparison of mean response time between HTTP over SCTP with that of HTTP over TCP in wired and wireless environments using NS-2 simulator. We measured mean response time for varying packet loss rate, bandwidth, RTT, and the number of web objects in wired environment and mean response time and packet loss rate for varying moving speed and region size in wireless environment. Our experimental result shows that SCTP reduces the mean response time of TCP based web traffic.

• Journal of KIISE:Information Networking
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• v.29 no.4
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• pp.375-385
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• 2002

The IETF has defined the Intserv model and the RSVP signaling protocol to improve QoS capability for a set of newly emerging services including voice and video streams that require high transmission bandwidth and low delay. However, since the current Intserv model requires each router to maintain the states of each service flow, the complexity and the overhead for processing packets in each rioter drastically increase as the size of the network increases, giving rise to the scalability problem. This motivates our work; namely, we investigate and devise new control schemes to enhance the scalability of the Intesev model. To do this, we basically resort to the SCORE network model, extend it to fairly well adapt to the three services presented in the Intserv model, and devise schemes of the QoS scheduling, the admission control, and the edge and core node architectures. We also carry out the computer simulation by using ns-2 simulator to examine the performance of the proposed scheme in respects of the bandwidth allocation capability, the packet delay, and the packet delay variation. The results show that the proposed scheme meets the QoS requirements of the respective three services of Intserv model, thus we conclude that the proposed scheme enhances the scalability, while keeping the efficiency of the current Intserv model.

• Journal of Internet Computing and Services
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• v.4 no.3
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• pp.47-55
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• 2003

Multicast protocols ore developed in order to support group communications efficiently, However. there still exist some issues to be solved for deploy multicast protocol in the public internet. This paper refers problem of existing Reliable Multicast Protocols and redesigns component function, and proposes enhanced reliable multicast transport protocol, This paper proposed a improved multicast transport scheme in NACK based reliable multicast. This scheme is much faster than by sender-initiated or receiver-initiated recovery and latency is smaller. Designed components are implemented in UNIX environment using C programming longuage. Then the protocol was evaluated performance through simulation. As the result, proposed protocol is better than existing protocols in both of transmission delay and packet loss. Especially, proposed protocol in this paper con be used in multicast services needed high reliability.