• The Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.5B
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• pp.455-462
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• 2009

In this paper, we propose a buffer management scheme for decreasing the packet loss due to buffer overflow and improving the packet fairness between nodes in IEEE 802.11 based multi-hop mesh networks. In the proposed scheme, each mesh router that is an intermediate node receives fairly packet sent from neighboring mesh routers and mobile nodes, and it improves the reception ratio of multi-hop traffic of neighboring mesh routers. Therefore, the proposed scheme can reduce transmission delay and energy consumption. In order to improving the packet loss and the packet fairness, the proposed scheme uses the modified RTS/CTS under the IEEE 802.11 MAC protocol and reduces the packet loss by recognizing the packet size to send to the destination in advance. By using the simulation, we evaluated the proposed scheme in terms of the packet loss ratio and the number of received packet in each mesh router, and compare it to a traditional scheme.

• Journal of applied mathematics & informatics
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• v.17 no.1_2_3
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• pp.547-557
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• 2005

In this paper, we investigate the packet delay distribution of a dynamic bandwidth allocation(DBA) scheme in an Ethernet passive optical network(EPON). We focus on the gated-exhaustive vacation system. We assume that input packets arrive at an optical network unit(ONU) according to general interarrival distribution. We use a discrete time queueing model in order to find the packet delay distribution of the gated-exhaustive system with the primary transmission queue and the secondary input queue. We give some numerical examples to investigate the mean packet delays of the proposed queueing model to analyze the DBA scheme in an EPON.

• Proceedings of the IEEK Conference
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• 2000.11a
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• pp.213-216
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• 2000

In this paper, we propose a new scheme which improves the IP address lookup time. The new scheme is composed of two core technologies, named the prefix alignment and the prefix distance ordering. Now, as the Internet is being used commonly by improving the data transmission capacity, the need for enlarging the bandwidth of the Internet is on the rise. IP address lookup performance problem is an important obstacle in the router executing high speed packet forwarding. This results from the fact that the prefixes routing table is composed of and the traffic being processed in unit time are largely on the increase. The proposed lookup scheme is divided into two parts in technology, the one is the algorithm forming a routing database(routing table), the other is the lookup procedure in the actual packet processing.

• Journal of the Korean Operations Research and Management Science Society
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• v.19 no.3
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• pp.1-14
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• 1994

In this paper, an analytical model of the message delay in the DQDB (Distributed Queue Dual Bus) network is investigated. The DQDB network has been adopted as a subnetwork for the IEEE 802 MAN (Metropolitan Area Network) standard. The DQDB network consists of two high speed undirectional buses and a series of stations attached to both of the buses. Massages arriving at each station consists of severla packets according to its size. This system is approximated into " $B^{[x]}$/G/1 with exceptional first service queueing " by defining the concept of service time on a packet. The service time for a packet is defined as the time from the instant the packet arrives at the transmission buffer until the time the packet is fully transmitted. By using the BASTA property and the average work in the system, the mean message delay time is obtained.age work in the system, the mean message delay time is obtained.d.

• Korean Management Science Review
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• v.12 no.2
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• pp.89-103
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• 1995

Traditionally, the scarce and expensive resources in communication networks has been the bandwidth of transmission facilities; accordingly, the techniques used for networking and switching have been chosen so as to optimize the efficient use of that resource. One of them is the packet switching for data communications, which is much different from the circuit switching. It allows multiple users to share the data network facilities and bandwidth, rather than providing specific amounts of dedicated bandwidth to each users. One main challenge therefore has been to design and build cost-effective packet switching networks. In this paper, we develop a mathematical programming medel and heuristic algorithm for designing the packet switching networks with hop-count constraints. To develop an efficient heuristic, we apply the Lagrangian relaxation method. Some valid inequalities, which have been known as the tight constraints, are considered to improve the bound for Lagrangian sub-problem. The computational experiments show that the performance of the proposed heuristic is very satisfactory in both the speed and the quality of design solutions generated.

• Journal of information and communication convergence engineering
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• v.8 no.5
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• pp.513-518
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• 2010

This paper proposes a medium access control protocol for integrated voice and data services in slotted CDMA systems. The proposed protocol, which is named as RCSSMA (Reservation Code and Status Sensing Multiple Access), adopts a code reservation and status sensing schemes. RCSSMA protocol gives higher access priority to the voice traffic than data traffic for reducing the packet dropping probability. The voice terminal reserves an available spreading code to transmit voice packets during a talkspurt, whereas the data terminal transmits a packet over one of the available spreading codes that are not reserved by the voice terminals. In this protocol, the voice packets never contend with the data packets. Packet dropping probability and average data packet transmission delay are analyzed using a Markov chain model.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.6 no.4
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• pp.1006-1025
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• 2012

Improving the quality of service in IP networks is a major challenge for real-time voice communications. In particular, packet arrival-delay variation, so-called "jitter," is one of the main factors that degrade the quality of voice in mobile devices with the voice-over Internet protocol (VoIP). To resolve this issue, a receiver-based enhanced timing recovery algorithm combined with active jitter estimation is proposed. The proposed algorithm copes with the effect of transmission jitter by expanding or compressing each packet according to the predicted network delay and variations. Additionally, the active network jitter estimation incorporates rapid detection of delay spikes and reacts to changes in network conditions. Extensive simulations have shown that the proposed algorithm delivers high voice quality by pursuing an optimal trade-off between average buffering delay and packet loss rate.

• Journal of information and communication convergence engineering
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• v.9 no.1
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• pp.38-43
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• 2011

This paper proposes a centralized backoff control scheme for guaranteeing fair packet transmissions in S-ALOHA CDMA system. In S-ALOHA CDMA system, simultaneously transmitted packets act as multiple access interferences, so that unsuccessful packet transmissions are caused entirely by multiple access interferences. Therefore, in order to maximize the system performance, the levels of MAI should be kept at an optimal level. In the proposed scheme, the base station broadcasts the packet transmission probability, which is calculated based on the level of MAI. Mobile stations attempt to transmit their packet with the received probability. Simulation results show that the proposed scheme can guarantee a good fairness among all the mobile stations.

• ETRI Journal
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• v.35 no.4
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• pp.726-729
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• 2013

For secure multimedia communications, existing encryption techniques use an online session key for the key exchange, for which key size is limited to less than 10 digits to accommodate the latency condition caused by user devices only being able to handle low computational loads. This condition results in poor security of recorded encrypted data. In this letter, we propose a packet key scheme that encrypts real-time packets using a different key per packet for multimedia applications. Therefore, a key of a relatively small size can provide after-transmission confidentiality to data of a real-time session.

• Proceedings of the Korea Information Processing Society Conference
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• 2015.10a
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• pp.409-412
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• 2015

사용자들을 인증하기 위한 대표적인 프로토콜 중에 RADIUS(Remote Authentication Dial-In User Service)가 있다. 기존 RADIUS 서버는 클라이언트가 전송하는 request packet의 크기가 일정크기 이상이면, silently discard하는 프로세스를 수행한다. 하지만, 보안관련 속성이 계속해서 추가될 수 있는 XML방식의 request packet은 Larger packet으로 분류가 되면서 자칫, 패킷 크기로 인해서 인증실패가 될 수 있는 문제가 있다. 본 논문에서는 RADIUS 프로토콜이 전송방식으로 TCP 프로토콜을 사용한다는 가정 아래, 4096옥텟 이상의 패킷(Larger Packets)을 처리하기 위한 방안을 제시하고자 한다.