• Journal of the Korean Institute of Telematics and Electronics S
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• v.36S no.7
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• pp.25-33
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• 1999

This paper proposes a handover algorithm for the mobile host in order to reduce the number of unnecessary retransmissions during handover. In our proposal, the home agent buffers user data after receiving the registration request message form the foreign agent and then transmits them again after a predefined time. We also analyze the packet disordering probability between two user data under an assumption that the home agent generates all user data. One of two user data is the one which has been generated just before the home agent receives the registration request message and the other is generated just after that. Simulation results show that our proposal provides efficient handover for a mobile host by reducing the packet disordering probalility.

• Proceedings of the Korea Society for Industrial Systems Conference
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• 1999.12a
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• pp.121-129
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• 1999

When continuous media are transmitted over the communication networks, asynchrony which can not maintain temporal relationships among packets may occur due to a random transit delay. There exist two types of synchronization schemes ; for guaranteed or non-guaranteed resource networks. The former which applies a resource reservation technique maintains delay characteristics, however, the latter supply a best-effort service. In this paper, I propose a intra-media synchronization scheme to transmit continuous media on general networks not guaranteeing a bounded delay tome. The scheme controls transmission times of the packets by estimating next delay time with the delay distribution. So, the arriving packets may be maintained within a limited delay boundary, and playout will be performed after buffering to smoothen small delay variations. The continually increasing delay due to network overload causes buffer underflow at the receiver. To solve it, the transmitter is required to speed up instantaneously. Too much increase of transmission-rate may cause network congestion. At that time, the transmitter drops the current packet when informed excessive delay from the receiver.

• The KIPS Transactions:PartC
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• v.9C no.6
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• pp.875-882
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• 2002

The active packets unlike traditional IP packets should be executed at each active node along their path. To execute the active program at each active node, the active packets for an active program should go through the same active nodes and all packets are delivered without any loss. This paper presents the new active engine for every active packet which execute an active program to be routed through the same intermediate active nodes and to be delivered reliably. Proposed active engine requires fewer changes to existing IP protocols and guarantees the reliable delivery of the active packets. Moreover, even if the sender does not have any information about the intermediate nodes every active packet is routed through the same intermediate active nodes. The simulation results show that proposed active engine achieves an efficient transmission with high data delivery and low communication overhead compared with the other existing transmission protocols.

• The Journal of the Korea Contents Association
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• v.12 no.4
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• pp.69-75
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• 2012

In this paper, to provide various services of QoS, and moreover applying traffic ratio to CAC(Call Admission Control) algorithm tested how long average data rate and the average packet delay time. When CAC algorithm is not applied, traffic mixture ratio is 1:1:4:4, the FTP Service 0.4, web services 0.6, streaming service 0.7, the packet delay requirements are not satisfied. On the other hand CAC Algorithm is applied, all the service of packet delay are satisfied with arrival rate. Therefore, we can make sure that applying of CAC of traffic control WWW, FTP, Video, VoIP can guarantee the various services of QoS.

• The KIPS Transactions:PartC
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• v.18C no.6
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• pp.433-440
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• 2011

In this paper, we propose a new scheduling method that can guarantee reliable jitter by minimizing the queue length variation in the streaming service provisioning such as IPTV and VoD. The amount of traffic to be delivered within a certain time is very fluid because MPEG-4 and H.264 encoders use VBR(Variable Bit Rate) for delivering video streaming traffic. This VBR characteristic increases the end-to-end propagation delay variation when existing scheduling methods are used for delivering video frames. Therefore, we propose the new scheduling method that can minimize change rate of queue length by adaptively controling service rate taking into account the size of bulk incoming packets and arrival rate for bulk streaming traffic. Video frames can be more reliably transmitted through the minimization of the queue length variation using the proposed method. We use the queueing model and also carry out OPNET simulation to validate the proposed method.

• Journal of the Korea Society of Computer and Information
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• v.15 no.12
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• pp.75-84
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• 2010

As broadcasting in Mobile Ad hoc NETwork (MANET) is the process that a node sends a packet to all other nodes in the network. it is used for routing protocols such as Ad hoc On demand Distance Vector (AODV) to disseminate control information for establishing the routes. In this paper, we propose Probabilistic Broadcasting mechanism based on Distance Ratio between sender and receive node in MANETs. The proposed approach is based on the combination of probability and distance based approach. A mobile node receiving broadcast packets determines the probability of rebroadcasting considering distance ratio from sender. The distance ratio of a node is calculated by the distance from sender and the length of radio field strength. As a node with high distance ratio is located far away from sender, rebroadcast probability is set to high value. On contrary, the low rebroadcast probability is set for a node with low distance ratio which is close to sender. So it reduces packets transmission caused by the early die-out of rebroadcast packets. Compared with the simple flooding and fixed probabilistic flooding by simulation, our approach shows better performances results. Proposed algorithm can reduce the rebroadcast packet delivery more than 30% without scanting reachability, where as it shows up to 96% reachability compared with flooding.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.9 no.1
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• pp.45-57
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• 2005

In this paper, we extend the performance of bidirectional TCP connection over end-to-end network that uses transfer rate-based flow and congestion control. The sharing of a common buffer by TCP packets and acknowledgement has been known to result in an effect called ack compression, where acks of a connection arrive at the source bunched together, resulting in unfairness and degraded throughput. The degradation in throughput due to bidirectional traffic can be significant. Even in the simple case of symmetrical connections with adequate window size, the connection efficiency is improved about 20% for three levels of background traffic 2.5Mbps, 5.0Mbps and 7.5Mbps. Otherwise, the throughput of jitter is reduced about 50% because round trip delay time is smaller between source node and destination node. Also, we show that throughput curve is improved with connection rate algorithm which is proposed for TCP congetion avoidance as a function of aggressiveness threshold for three levels of background traffic 2.5Mbps, 5Mbps and 7.5Mbps. By analyzing the periodic bursty behavior of the source IP queue, we derive estimated for the maximum queue size and arrive at a simple predictor for the degraded throughput, applicable for relatively general situations.

• Proceedings of the Korea Information Processing Society Conference
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• 2002.11a
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• pp.45-48
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• 2002

무선 이동통신환경에서는 시간에 따라 네트워크에 의한 데이터 손실, 일관성 없는 패킷의 도착 간격 등과 같은 문제와 전송 속도의 저하가 발생함으로써 일정한 수준의 QoS 를 제공하기 어렵다. 이러한 문제점에 대처하기 위하여 다중 비트율 코딩, 트랜스코딩 등과 같은 기술이 제안되었다. 그러나 이러한 방식들은 서버의 과도한 연산량 요구나 저장 공간 낭비라는 단점을 가지고 있다. 본 논문에서는 이러한 문제점을 해결하기 위해 다중 비트율 코딩 방식과 스케일러블 코딩 방식을 결합하여 네트워크 QoS 변화에 유연하게 적응될 수 있는 방식을 제시한다. 기존의 심플 혹은 메인 프로파일이 네트워크의 상태가 일정하다는 것을 전제로 하는 반면, 스케일러블 프로파일은 기본 데이터와 부가 데이터를 분리하여 하나의 파일로 관리되므로 서버의 연산량을 줄이고, 저장 공간도 절약한다. 또한 같은 스트림에 대해서도 단말기의 능력에 따라서 부가 데이터를 복호화 할지를 결정할 수 있어 여러 사용자에게 같은 스트림 데이터를 전송할 수 있는 장점이 있다.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.9
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• pp.19-28
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• 2010

The GTS(Guaranteed Time Slot) of the IEEE 802.15.4 standard, which is the contention free access mechanism, is used for low-latency applications or applications requiring specific data bandwidth. But it has some problems such as delay of service due to FIFS(First In First Service) scheduling. In this paper, we proposes a weight based GTS allocation scheme for fair queuing in IEEE 802.15.4 LR-WPAN. The proposed scheme uses a weight that formed by how much more weight we give to the recent history than to the older history for a new GTS allocation. This scheme reduces service delay time and also guarantees transmission simultaneously within a limited time. The results of the performance analysis shows that our approach improves the performance as compared to the native explicit allocation mechanism defined in the IEEE 802.15.4 standard.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.49 no.4
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• pp.26-33
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• 2012

Please WBAN(Wireless Body Area Networks) is wireless communication of within 3 m radius from inside and outside of the body. WBAN is many uses IEEE 802.15.4 MAC protocol for WBAN MAC protocol requirements. IEEE 802.15.4 MAC protocol applies GTS(Guaranteed Time Slot) allocation scheme for guarantee packet delivery of application which requests QoS(Quality of Service). However, GTS allocation scheme of IEEE 802.15.4 MAC protocol uses FIFS(First In First Service) queueing. So it reduces data transmission reliability and has many problems. Therefore, IEEE 802.15.4 applies not WBAN MAC protocol for emergency data transmission. In this paper, we proposes a priority based MAC protocol for emergency data transmission in wireless body area networks. the proposal MAC protocol sets priority the data. So GTS reqeuest throughput of emergency data increasing and GTS allocation delay of emergency data decreasing.