• The Journal of Korean Institute of Communications and Information Sciences
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• v.21 no.1
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• pp.91-102
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• 1996

The multicast routing algorithm is necessary to transmit multimedia traffic efficiently in ATM (asynchronous transfer mode) networks. In this paper, we propose the multicast routing algorithm which is based on VP/VC characteristic. The proposed algorithm is based on VP tree concept and using cost function which is based on VP/VC switching. The cost funication is composed of link cost, delay and weighting factor on delay and the weighting factor is calculated by delay sensitivity of the traffic. The proposed algorithm can choose delay bounded path which satisfies delay constraint, moreover it can choose optimal path among VPs which has the same link cost and satisfying delay constraint. With controlling weighting factor, proposed algorithm can set-up efficient path. When the weighting factor sets to be between 0.8 and 1, experimental results show that the perforance of proposed scheme is approximated to that of cost optimal algorithm and strongly delay optimized algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37A no.10
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• pp.876-884
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• 2012

This paper introduces two multi-hop delay factors which can be caused by conventional TDMA scheduling; queueing delay and delay by random link scheduling, and proposes a new sequential scheduling scheme to resolve these two factors. We also simulate the TDMA network with the proposed link scheduling scheme and compare it with conventional(random) link scheduling scheme in terms of end-to-end packet transmission delay. From the simulation results, the more the average hop distance increases, the more the difference of the delay performance of both scheduling schemes increases. When the average number of hops is 2.66, 4.1, 4.75, and 6.3, the proposed sequential scheduling scheme reduces the average end-to-end delay by about 22%, 36%, 48%, and 55% respectively when compared to the random scheduling scheme.

• Journal of KIISE:Information Networking
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• v.31 no.4
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• pp.393-404
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• 2004

Recently, mSCTP (Mobile SCTP) has been proposed as a transport layer approach for supporting mobility. mSCTP is based on the ‘multi-homing’ feature of Stream Control Transmission Protocol(SCTP), and utilize the functions to dynamically add or delete IP addresses of end points to or from the existing connectionin order to support mobility. In this paper, we propose a mechanism to determine when to add or delete an W address, utilizing the link layer radio signal strength information in order to enhance the performance of mSCTP We also propose a mechanism for a mobile node to initiate the change of data delivery path based on link layer radio signal strength information. In addition, if it takes long time to acquire new data path, we propose an approach for reducing handover latency. The simulation results show that the performance of proposed transport layer mobility support mechanism is competitive compared to the traditional network layer mobility supporting approach. Especially, when the moving speed of mobile node is fast, it shows better performance than the traditional network layer approaches.

• The KIPS Transactions:PartC
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• v.17C no.4
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• pp.379-384
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• 2010

Mobile nodes in high mobility ad hoc networks might come together in specific areas. In non-uniformly distributed networks, traffic load can be concentrated to intermediate nodes between dense clusters, and networks performance can be degraded. In this paper, we proposed a cluster based routing protocol that heavy traffic nodes adaptively react according to traffic load. The simulation result shows that the proposed protocol reduce packet loss and end-to-end delay.

• Journal of KIISE:Computer Systems and Theory
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• v.27 no.5
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• pp.518-528
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• 2000

Messages in distributed real-time systems are categorized into two groups: synchronous messages and asynchronous messages. Synchronous messages, such as sampled audio and image data,are generated periodically with delivery time constraints. Protocols should guarantee the end-to-enddeadlines for such messages. Asynchronous messages are non-periodic and may arrive in a randomway with no strict time constraints.In this paper, we propose TTP/C(timed token protocol with concession), an extension of TTPprotocol, to achieve higher timeliness guarantee for synchronous messages in distributed real-timesystems. In TTP/C, a node concedes the allocated bandwidth to other nodes with urgent synchronousmessages to be sent provided that the node has no urgent messages, TTP/C works very well evenif the synchronous messages are generated with some jittering by nodes. The simulation results showthe improved performance of TTP/C protocol for guaranteeing synchronous messages deadlinescomeared to the existing TTP protocols.

• Journal of the Korea Society of Computer and Information
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• v.12 no.3
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• pp.177-183
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• 2007

Active queue management (AQM) refers to a family of packet dropping mechanisms for router queues that has been proposed to support end-to-end congestion control mechanisms in the Internet. The proposed AQM algorithm by the IETF is Random Early Detection (RED). The RED algorithm allows network operators simultaneously to achieve high throughput and low average delay. However. the resulting average queue length is quite sensitive to the level of congestion. In this paper, we propose the Refined Adaptive RED(RARED), as a solution for reducing the sensitivity to parameters that affect RED performance. Based on simulations, we observe that the RARED scheme improves overall performance of the network. In particular, the RARED scheme reduces packet drop rate and improves goodput.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2011.10a
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• pp.633-636
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• 2011

The paper deals with the performance of contention-based medium access control (MAC) protocols for underwater sensor networks. We extensively analyze the number of received-packets and the end-to-end delay of ALOHA, CSMA, CSMA-RTS-CTS and CSMA-RTS-CTS-ACK protocols using a Qualnet underwater network simulator which accommodates diverse underwater acoustic channel environments. Using simulation results, we support an engineering table to determine an adequate contention-based MAC protocol for underwater sensor networks.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.47 no.5
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• pp.9-17
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• 2010

In this paper we propose an improved hierarchical routing protocol for wireless hybrid mesh network. The proposed method efficiently manages network topology and reduces overhead traffic for setting routing path by considering link stability. The simulation results show that the proposed method outperforms the HOLSR (hierarchical optimized link state routing) method in aggregate goodput, packet delivery ratio, and end-to-end delay.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.9
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• pp.1145-1157
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• 1999

본 논문에서는 RFC 2022에서 제안한 MARS 모델을 기반으로 하여 단일 대규모 클러스터를 가지는 ATM 망에서 다중의 멀티캐스팅 서버(MCS)를 이용한 멀티캐스팅 방안을 제안하고 그 성능을 평가하였다. 클러스터 내의 한 ATM 호스트가 특정 IP 멀티캐스트 그룹에 가입할 경우 ATM 호스트의 위치와 이미 존재하는 멀티캐스팅 서버들 사이의 전송 지연을 고려하여 가능한 한 종단간 전송 지연을 최소화하는 멀티캐스팅 서버를 선택하는 방안을 기술하였다. 이 방안은 최단거리 경로 알고리즘(shortest path algorithm)에 기반하여 최적의 MCS를 선정하고 송수신자 사이의 최소 지연을 가지는 멀티캐스트 트리를 구성한다. 다양한 망 위상에서 MCS의 분포 패턴을 다르게 할 경우에도 이 방안은 멀티캐스트 트리의 평균 전달 지연을 줄이는 것을 시뮬레이션을 통하여 확인하였다.Abstract In this paper, we proposed a scheme to support multiple MCSs over a single and large cluster in ATM networks, evaluated its performance by simulation. When an ATM host requests joining into a specific multicast group, the MARS designate a proper MCS among the multiple MCSs for the group member to minimize the average path delay between the sender and the group members. This scheme constructs a multicast tree through 2-phase partial multicast tree construction based upon the shortest path algorithm.We reduced the average path delay in multicast tree using our scheme under various cluster topologies and MCS distribution scenarios.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.975-983
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• 1999

최근 네트워크의 속도가 빨라지고 멀티미디어 데이터를 다루기 위한 기술들이 개발됨에 따라 많은 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 그러나 이들 응용프로그램들은 수신자에게 전송되는 영상.음성의 품질이 낮기 때문에 기대만큼 빠르게 확산되지 못하고 있다. 영상.음성의 품질이 낮은 이유는 현재 인터넷이 실시간 응용프로그램이 요구하는 만큼 빠르고 신뢰성 있게 데이터를 전송할 수 없기 때문이다. 현재 인터넷의 내부구조를 바꾸지 않고 품질을 높이기 위해 많은 연구들이 진행되고 있는데 그 중 하나는 동적으로 변화하는 인터넷의 상태에 맞게 멀티캐스트 트래픽의 전송율을 조절하는 종단간의 흐름제어이다. 본 논문은 기존의 흐름제어 기법인 IVS와 RLM의 성능을 개선시키기 위한 두 가지 흐름제어 기법을 소개한다. IVS는 송신자가 주기적으로 측정된 네트워크 상태에 따라 전송율을 일정하게 조절한다. 송신자가 하나의 데이타 스트림을 생성하는 IVS와는 달리 RLM에서는 송신자가 계층적 코딩에 의하여 생성된 여러개의 데이타 스트림을 전송하고 각 수신자는 자신의 네트워크 상태에 맞게 데이타 스트림을 선택하는 기법이다. 그러나 IVS는 송신자가 전송율을 일정하게 증가시키고, RLM은 각자의 네트워크 상태를 고려하지 않고 임의의 시간에 하나 이상의 데이타 스트림을 받기 때문에 성능을 저하시킬 수 있다. 본 논문에서는 TCP-like IVS와 Adaptive RLM이라는 두 가지 새로운 기법을 소개한다. TCP-like IVS는 송신자가 전송율을 동적으로 결정하고, Adaptive RLM은 하나 이상의 데이타 스트림을 받기 위해 적당한 시간을 선택할 수 있다. 본 논문에서는 시뮬레이션을 통해 여러 가지 네트워크 구조에서 두 가지 방식이 기존의 방식에 비하여 더욱 높은 대역폭 이용율과 10~20% 정도 적은 패킷손실율을 이룬다는 것을 보여준다.Abstract Nowadays, many multimedia applications for the Internet are introduced as the network gets faster and many techniques manipulating multimedia data are developed. These multimedia applications, however, do not spread widely and are not fast as expected at their introduction time due to the poor quality of image and voice delivered at receivers. The poor quality is mainly attributed to that the current Internet can not carry data as fast and reliably as the real-time applications require. To improve the quality without modifying the internal structure of the current Internet, many researches are conducted. One of them is an end-to-end flow control of multicast traffic adapting the sending rate to the dynamically varying Internet state. This paper proposes two flow-control techniques which can improve the performance of the two conventional techniques; IVS and RLM. IVS statically adjusts the sending rate based on the network state periodically estimated. Differently from IVS in which a sender produces one single data stream, in RLM a sender transmits several data streams generated by the layered coding scheme and each receiver selects some data streams based on its own network state. The more data streams a receiver receives, the better quality of image or voice the receiver can produce. The two techniques, however, can degrade the performance since IVS increases its sending rate statically and RLM accepts one more data stream at arbitrary time regardless of the network state respectively. We introduce two new techniques called TCP-like IVS and Adaptive RLM; TCP-like IVS can determine the sending rate dynamically and Adaptive RLM can select the right time to add one more data stream. Our simulation experiments show that two techniques can achieve better utilization and less packet loss by 10-20% over various network topologies.