• KIISE Transactions on Computing Practices
• /
• v.22 no.8
• /
• pp.381-386
• /
• 2016

Recently full-duplex communication in wireless networks is enabled by the advancement of self-interference cancellation technology. Full-duplex radio is a promising technology for next-generation wireless local area networks (WLAN) because it can simultaneously transmit and receive signals within the same frequency band. Since legacy medium access control (MAC) protocols are designed based on half-duplex communication, they are not suitable for full-duplex communication. In this paper, we discuss considerations of full-duplex communication and propose a novel full-duplex MAC protocol. We conducted a simulation to measure the throughput of our MAC protocol. Through the simulation results, we can verify that significant throughput gains of the proposed full-duplex MAC protocol, thus comparing the basic full-duplex MAC protocol.

• The KIPS Transactions:PartC
• /
• v.13C no.4 s.107
• /
• pp.441-446
• /
• 2006

In recent years, WDM networks have received much attention as the Internet backbone networks because of the explosive growth of the Internet IP-based traffic. The Optical Burst Switching (OBS) has been proposed as an effective optical switching technology in the WDM networks. The OBS has the advantages in 1) the high usage rate of the bandwidth, and 2) no necessity of optical buffer. However, the OBS has the burst-contention problem in the networks. The deflection routing is proposed as one of means to solve this problem. In this paper, we propose a new routing method to minimize burst loss in the deflection routing based networks. In addition, we propose a QoS control method using a new routing algorithm. Finally, we show the variety of the proposed methods by computer simulations.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2000.04a
• /
• pp.356-361
• /
• 2000

이벤트 서비스는 멀티미디어 통신, 전자상거래 등과 같은 대용량의 분산 응용 시스템에서 발생하는 비동기적 이벤트의 전송을 제어하기 위한 서비스이다. 하지만 기존의 이벤트 서비스는 분산 환경의 실시간 응용 시스템을 지원하기 위해 필요한 실시간 지원 기능들이 결여되어 있어 이를 지원할 수 있는 실시간 이벤트 서비스 처리 방안이 요구된다. 본 논문에서 제안하는 RTEF(Real-Time Event Filtering) 서비스는 기존의 ORB 구조에 실시간 이벤트 필터링, 실시간 모니터 및 QoS 저장소를 포함시킨 실시간 이벤트 서비스 미들웨어이다. 특히, RTEF는 이벤트 사용자의 실시간 요구사항을 분산 실시간 응용시스템에 반영시키기 위하여 이벤트 필터링 기능을 실시간 처리에 맞게 강화 시켜 기존의 시스템에서 지원하지 못한 사용자의 실시간 요구사항을 지원하는데 중점을 두었다. 분산 실시간 응용시스템에 RTEF를 적용하면 사용자에게 자신이 원하는 실시간 요구조건에 맞게 필터링된 이벤트 처리결과를 제공하게 되므로 궁극적으로 보다 높은 서비스(QoS)를 제공하게 된다. 또한 이 과정을 통해 불필요한 이벤트가 필터링 됨으로써 전체적인 네트웍 트래픽(traffic)을 감소시키는 효과를 가져온다.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.11 no.12
• /
• pp.2265-2270
• /
• 2007

This paper discusses a tester of functionality and call processing performance, based on the MEGACO/H.248 protocol that both IETF and ITU-T recommend as a media gateway control protocol, of both a media gateway controller and an access gateway which constitute a next generation convergence network. Effective methods, a functional architecture and implementation for such testification are provided. Especially included are not only a virtual emulation function of analog subscriber lines connecting to an access gateway, but also a tester emulated as a counter system of the protocol for the testifying a media gateway controller and an access gateway system.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
• /
• 2009.10a
• /
• pp.987-990
• /
• 2009

The MAC protocol of HiperLAN/2 is based on TDMA/TDD. The radio channels are assigned by AP(Access Point) that is centrally operated. Mobile terminal that data transmission is necessary to uplink requests radio resource to AP through RCH channel. The changing number of RCHs in each MAC frame is important because too many RCHs may result in a waste of radio resources and too few RCHs may result in many access collisions and prolong time that connect to AP. Therefore, number of RCH should be allocated properly according to traffic. From these viewpoint, this paper proposes an advanced scheme that dynamically changed the number of RCH which is based on the number of success and collision of RR message in previous MAC frame. To prove efficiency of proposed scheme, a lots of simulations are conducted and analyzed.

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

• Journal of KIISE:Information Networking
• /
• v.28 no.4
• /
• pp.527-539
• /
• 2001

The wireless asynchronous transfer mode (ATM) system has the advantage of providing the broadband services with various quality-of-service requirements to the mobile terminal efficiently by utilizing the ATM technology developed for the wired ATM system. The available bit rate (ABR) service among various ATM services utilizes the available bandwidth remaining in the ATM link, which allows the efficient bandwidth usage. During the handoff of the mobile terminal, however, the queue length in the access point (AP) which resides in the boundary of the wired ATM network and the wireless ATM network may increase abruptly. In this paper, we propose a scheme which prevents the buffer-overflow and buffer-underflow in the AP during the handoff of the wireless ABR connection in the wireless ATM system using binary feedback rate-based ABR traffic control. This scheme controls the source's cell generation rate during both handoff period and some time interval after the completion of the handoff procedure. The simulation results show that the proposed scheme prevents the buffer-overflow and buffer-underflow. The proposed scheme can contribute to increasing the throughput of the wireless ABR service during handoff by preventing the buffer overflow and underflow during handoff period.

• Journal of the Institute of Electronics Engineers of Korea TC
• /
• v.43 no.10 s.352
• /
• pp.8-17
• /
• 2006

Recently, the high-speed Internet users increase rapidly and broadband networks have been widely deployed. However, the current TCP congestion control algorithm was designed for relatively narrowband network environments, and thus its performance is inefficient for traffic transport in broadband networks. To remedy this problem, the TCP having an enhanced congestion control algorithm is required for broadband networks. In this paper, we propose an improved TCP congestion control that can sufficiently utilize the large available bandwidth in broadband networks. The proposed algorithm predicts the available bandwidth by using ACK information and RTT variation, and prevents large packet losses by adjusting congestion window size appropriately. Also, it can rapidly utilize the large available bandwidth by enhancing the legacy TCP algorithm in congestion avoidance phase. In order to evaluate the performance of the proposed algorithm, we use the ns-2 simulator. The simulation results show that the proposed algorithm improves not only the utilization of the available bandwidth but also RTT fairness and the fairness between contending TCP flows better than the HSTCP in high bandwidth delay product network environment.

• The Journal of the Institute of Internet, Broadcasting and Communication
• /
• v.16 no.1
• /
• pp.139-145
• /
• 2016

Dumbbell is the most basic topology that can be used in almost all kind of network experiment within it or just by little expansion. While Transmission Control Protocol TCP is the basic protocol that is used for the connectivity among networks and stations. TCP major and basic goal is to provide path and services to different applications for communication. For that reason TCP has to transfer a lot of data through a communication medium that cause serious congestion problem. To calculate the congestion problem, different kind of pre-cure solutions are developer which are Loss Based Variant and Delay Based Variant. While LBV keep track of the data that is going to be passed through TCP protocol, if the data packets start dropping that means congestion occurrence which notify as a symptom, TCP CUBIC use LBV for notifying the loss. Similarly the DBV work with the acknowledgment procedure that is used in when data ACK get late with respect to its set data rate time, TCP COMPOUND/VAGAS are examples of DBV. Many algorithms have been purposed to control the congestion in different TCP variants but the loss of data packets did not completely controlled. In this paper, the congestion control algorithms are implemented and corresponding results are analyzed in Dumbbell topology, it is typically used to analyze the TCP traffic flows. Fairness of throughput is evaluated for different TCP variants using network simulator (NS-2).

• Journal of KIISE:Information Networking
• /
• v.30 no.1
• /
• pp.134-142
• /
• 2003

We present an ATM switch architectures based on the multistage interconnection network(MIN) for the efficient multicast traffic control. Many of these applications require multicast connections as well as point-to-point connections. Muiticast connection in which the same message is delivered from a source to arbitrary number of destinations is fundamental in the areas such as teleconferencing, VOD(video on demand), distributed data processing, etc. In designing the multicast ATM switches to support those services, we should consider the fairness(impartiality) and priority control, in addition to the overflow problem, cell processing with large number of copies, and the blocking problem. In particular, the fairness problem which is to distribute the incoming cells to input ports smoothly is occurred when a cell with the large copy number enters upper input port. In this case, the upper input port sends before the lower input port because of the calculating method of running sum, and therefore cell arrived into lower input port Is delayed to next cycle to be sent and transmission delay time becomes longer. In this paper, we propose the cell splitting and group splitting algorithm, and also the fairness scheme on the basis of the nonblocking characteristics for issuing appropriate copy number depending on the number of Input cell in demand. We evaluate the performance of the proposed schemes in terms of the throughput, cell loss rate and cell delay.