• Proceedings of the Korean Information Science Society Conference
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• 2007.06b
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• pp.412-417
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• 2007

멀티 포트 네트워크 인터페이스 카드는 지원 가능한 대역폭의 합이 포트의 수에 따라서 결정된다. 따라서 I/O 버스와의 대역폭 균형을 맞출 수 있는 장점이 있다. 또한 상대적으로 저렴한 스위치 가격으로 높은 대역폭을 지원해 줄 수 있다. 이러한 이유에서 최근 멀티 포트 네트워크 인터페이스 카드는 고 가용성 서버뿐만 아니라 고성능 서버에서도 사용되기 시작하고 있다. 본 논문에서는 이러한 멀티 포트 네트워크 인터페이스 카드가 지원할 수 있는 최대 대역폭을 분석한다. 특히 최근에 등장한 멀티 코어 프로세서 서버에서 TCP/IP 성능 측정을 수행하여 멀티 코어 자원을 최대한 활용하는지를 분석한다. 분석 결과 현재 리눅스가 제공하는 인터럽트 분산 정책 및 패킷 처리 기법으로는 멀티 포트 네트워크 인터페이스 카드의 높은 대역폭 특성을 최대한 활용하기에는 부족함을 밝힌다. 또한 각 포트 별로 들어오는 네트워크 흐름의 특성이 서로 다를 때에 시스템이 그에 신속히 적응하지 못함을 측정 결과를 통해서 보인다. 이러한 측정 및 분석 결과는 멀티 코어 시스템에서 멀티 포트 NIC을 최대한 활용하기 위한 리눅스의 향상 필요성을 시사하며 그를 위한 방안을 제시할 수 있다.

• Journal of the Korea Society for Simulation
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• v.18 no.3
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• pp.91-102
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• 2009

The average window size is most closely related to average throughput. In order to improve fairness, the proposed dropper tries to control the window size of each flow to equal level by intentional packet drop. Intentional packet drop is performed only to the flows that have been occupied bandwidth in a large amount. Because of intentional packet drop, this flow cut down its transmission rate to a half. Accordingly, somewhat capacity of core link comes into existence. And other flow can use this new capacity of this link. Hence other flows have more throughput than before. In this paper, we propose the TRC (Transmission Rate Control) Dropper improving the fairness between individual flows of aggregated sources on DiffServ network. It has the fairness improvement mechanism mentioned above paragraph.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.8B
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• pp.667-680
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• 2008

In this work, we present the results of our empirical study on 802.11 wireless LAN network traffic. We collect the packet trace from existing campus wireless LAN infra-structure. We analyzed four different data sets: aggregate traffic, upstream traffic, downstream traffic, tcp only packet trace from aggregate traffic. We analyze the time series aspect of underlying traffic (byte count process and packet count process), marginal distribution of time series, and packet size distribution. We found that in all four data sets there exist long-range dependent property in byte count and packet count process. Inter-arrival distribution is well fitted with Pareto distribution. Upstream traffic, i.e. from the user to Internet, exhibits significant difference in its packet size distribution from the rests. Average packet size of upstream traffic is 151.7 byte while average packet size of the rest of the data sets are all greater than 260 bytes. Packets with full data payloads constitutes 3% and 10% in upstream traffic and the downstream traffic, respectively. Despite the significant difference in packet size distribution, all four data sets have similar Hurst values. The Hurst alone does not properly explain the stochastic characteristics of the underlying traffic. We model the underlying traffic using fractional-ARIMA (FARIMA) and fractional Gaussian Noise (FGN). While the fractional Gaussian Noise based method is computationally more efficient, FARIMA exhibits superior performance in accurately modeling the underlying traffic.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.3B
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• pp.339-348
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• 2004

Packet classification is an important factor to support various services such as QoS guarantee and VPN for users in Internet. Packet classification is a searching process for best matching rule on rule tables by employing multi-field such as source address, protocol, and port number as well as destination address in If header. In this paper, we propose hardware based packet classification algorithm by employing tree bitmap of multi-bit trio. We divided prefixes of searching fields and rule into multi-bit stride, and perform a rule searching with multi-bit of fixed size. The proposed scheme can reduce the access times taking for rule search by employing indexing key in a fixed size of upper bits of rule prefixes. We also employ a marker prefixes in order to remove backtracking during searching a rule. In this paper, we generate two dimensional random rule set of source address and destination address using routing tables provided by IPMA Project, and compare its memory usages and performance.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.4
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• pp.1344-1367
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• 2014

Mobile traffic is increasing a masse because of the propagation of the Internet and the development of wireless mobile technology. Accordingly, the Network Local Mobility Management (NETLMM) working group [1] of the Internet Engineering Task Force (IETF) has standardized Proxy Mobile IPv6 (PMIPv6) [2] as a protocol for accomplishing the transmissibility of mobile terminals. PMIPv6 is a network-led IP-based mobility management protocol, which can control terminal mobility without depending on the type of access system or the capability of the terminal. By combining PMIPv6 and the mobility of Session Initiation Protocol (SIP), we can establish terminal mobility and session mobility through a more effective route. The mobility function can be improved and the overlap of function reduced as compared to that in the case of independent operation. PMIPv6 is appropriate for a non-real-time service using TCP, and SIP is appropriate for a real-time service using RTP/UDP. Thus, in the case of a terminal using both services, an effective mobility management is possible only by using PMIPv6 together with SIP. In order to manage mobility in this manner, researches on PMIPv6-SIP are in progress. In line with this trend, this paper suggests a new PMIPv6-SIP architecture where when a mobile terminal conducts a handover, a network-led handover while maintaining the session without the addition of a special function or middleware is possible along with effective performance evaluation through mathematical modeling by comparing the delay and the packet loss that occur during the handover to the Pure-SIP.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.21 no.8
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• pp.18-25
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• 2020

With the development of water internet technology, data communication between objects is expanding. Research related to data communication technology between vehicles that incorporates related technologies into vehicles has been actively conducted. For data communication between mobile terminals, data stability, reliability, and real-time performance must be guaranteed. The 5 GHz Wi-Fi band, which is advantageous in bandwidth, communications speed, and wireless saturation of the wireless network, was selected as the data communications network between vehicles. This study analyzes how to design and implement a 5 GHz Wi-Fi network in a vehicle network. Considering the characteristics of the mobile communication terminal device, a continuous variable communications structure is proposed to enable high-speed data switching. We simplify the access point access procedure to reduce the latency between wireless terminals. By limiting the Transmission Control Protocol Internet Protocol (TCP/IP)-based Dynamic Host Configuration Protocol (DHCP) server function and implementing it in a broadcast transmission protocol method, communication delay between terminal devices is improved. Compared to the general commercial Wi-Fi communication method, the connection operation and response speed have been improved by five seconds or more. Utilizing this method can be applied to various types of event data communication between vehicles. It can also be extended to wireless data-based intelligent road networks and systems for autonomous driving.

• Journal of Korea Multimedia Society
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• v.1 no.1
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• pp.98-108
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• 1998

In the multimedia tele-medical system the medical informations are stored and managed in multimedia database. Also, multimedia DBMS is essential in order to manage large scale medical informations, and the remote access function is necessary for the distributed processing at all around the hospital. For the multimedia tele-medical information that is composed of image/picture, data, video, and audio, a high-speed telecommunication network is necessary that can provide separated connections for each medical information type with different QoS. The commercial DBMSs are based on the TCP/IP socket API(Application Programming Interface) that does not provide multiple QoS. Also, each commercial DBMS has its own API that is incompatible with other DBMS. In this paper, we propose a multimedia DBMS agent for the remote access of the multimedia database in the tele-medical system. The proposed multimedia DBMS agent is based on the ATM API that can provide high-speed data transfer capability and multiple QoS connections. Also, the proposed multimedia DBMS agent is independent of the commercial DBMS. We explain the functional architecture of the multimedia DBMS agent, implementation technology on the ATM network environment, and the result of performance analysis.

• Journal of Navigation and Port Research
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• v.28 no.10 s.96
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• pp.827-832
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• 2004

Up to now a lot of the study on ship collision avoidance systems have proceeded actively. However the rate of ship collision accidents hasn't decreased yet. If there is collision risk in close quarters situation in spite of maneuvering ship for collision avoidance according to the system, only use of TCP A and DCP A as input factor for collision risk decision is not useful to avoiding collision action. For the recent 5 years by the analysis of first observation distance about approaching ship in domestic collision accidents, nearly $45\%$ of accidents is close first observation less than 2 miles. Therefore it is essential part for safety navigations to study for collision avoidance action in close encounter. In this paper, as a fundamental study of supporting collision avoidance maneuvering for navigators, we proposed ship collision avoidance support model in close quarters situation through analysis qf collision accidents to effectively get rid of the causes.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.704-711
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• 2003

Recently, the demands for real-time service and multimedia data are rapidly increasing. There are significant redundancies between header fields both within the same packet header and in consecutive packets belonging to the same packet stream. But there are many overheads in using the current UDP/IP protocol. Header compression is considered to enhance the transmission efficiency for small size of payload. By sending the static field information only once initially and by utilizing dependencies and predictability for other fields, the header size can be significantly reduced for most packets. This work describes an implementation for header compression of the headers of U/UDP protocols to reduce overhead on Ethernet network. Typical UDP/IP Header packets can be compressed down to 7 bytes and the header compression system is designed and implemented on the Linux environment. Using the designed Header compression system between a server and a client have the advantage of effective data throughput in network.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.1-8
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• 2005

In this paper, we study a call admission control algorithm for supporting multicast service under the BcN environment where broadcasting, communication and Internet are converging to be one. It is necessary to control service requests with a certain criteria in order to guarantee QoS because the system capacity is limited. As a possible solution, we divide one multicast service into 3 classes and set up a threshold per each class to control service request. Especially, for the purpose of system benefit, we define system pay-off rate 'GAIN' with the term 'Reward' and 'Penalty' according to admit and reject service request. And we confine the range of threshold which makes GAIN to be maximized. For the performance analysis, we model the system as M/M/m/m queueing system, investigate GAIN under various conditions and show the effectiveness of the proposed algorithm.