• The Transactions of The Korean Institute of Electrical Engineers
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• v.57 no.11
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• pp.2124-2126
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• 2008

In the wireless communication systems, the demand of multimedia services is also increased. Unlike typical data packets, realtime service such as VoIP packets have delay bound and low loss rate requirement. In this paper we propose a new scheduling algorithm that be able to allocate resources to the different kinds of services such as VoIP and data packet. The proposed algorithm considers both time delay and channel condition toe determine the priority. Simulation results show that the proposed algorithm works more efficiently than the conventional algorithms.

• Proceedings of the IEEK Conference
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• 2004.06a
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• pp.171-174
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• 2004

Packet classification is an essential architectural component in implementing the quality-of-service (QoS) in today's Internet which provides a best-effort service to ail of its applications. Multiple header fields of incoming packets are compared against a set of rules in packet classification, the highest priority rule among matched rules is selected, and the packet is treated according to the action of the rule. In this Paper, we proposed a new packet classification scheme based on parallel multiple hashing on tuple spaces. Simulation results using real classifiers show that the proposed scheme provides very good performance on the required number of memory accesses and the memory size compared with previous works.

• ETRI Journal
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• v.28 no.1
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• pp.31-44
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• 2006

In a packet switching network, congestion is unavoidable and affects the quality of real-time traffic with such problems as delay and packet loss. Packet fair queuing (PFQ) algorithms are well-known solutions for quality-of-service (QoS) guarantee by packet scheduling. Our approach is different from previous algorithms in that it uses hardware time achieved by sampling a counter triggered by a periodic clock signal. This clock signal can be provided to all the modules of a routing system to get synchronization. In this architecture, a variant of the PFQ algorithm, called digitized delay queuing (DDQ), can be distributed on many line interface modules. We derive the delay bounds in a single processor system and in a distributed architecture. The definition of traffic contribution improves the simplicity of the mathematical models. The effect of different time between modules in a distributed architecture is the key idea for understanding the delay behavior of a routing system. The number of bins required for the DDQ algorithm is also derived to make the system configuration clear. The analytical models developed in this paper form the basis of improvement and application to a combined input and output queuing (CIOQ) router architecture for a higher speed QoS network.

• The KIPS Transactions:PartC
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• v.9C no.1
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• pp.21-30
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• 2002

Nowadays the fair queuing field is studied actively for solving the unfair queuing problem which degrades the service performance of various applications on network. Above all DRR is a scheme that has lower work complexity and can be implemented easily. It guarantees the fair service by serving each queue every round Service Quantum(SQ) that includes the service deficit of the previous round. But it increases the delay by the numbers of unnecessary resetting of SQ or round circulation for the service that have the packet size over maximum several kbyte. In this paper, We propose the method that sets SQ dynamically to minimize this problem and the method that minimizes the work complexity on processing of packet. The proposed scheme minimizes the delay on various applications in enterprise environment by setting SQ dynamically and processing the packet simply.

• Journal of Broadcast Engineering
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• v.12 no.2
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• pp.80-92
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• 2007

IPTV Services that is representative union service of broadcasting and telecommunication need guarantee of QoS, efficiency of multicasting, and hish bandwidth on the network. Because typical TDM based metro transport network was designed by transporting fixed voice traffic with stable and recovering method, it has a defect of bottleneck and a waste of bandwidth for acceptance of data traffic with burst feature and then all of data are treated equally at the transport network because it cannot classify between advanced high end service and best effort low end service. for completely resolving this kind of problem about increasing burst traffic and QoS issues, firstly we need to new design for transport network. This paper presents transformation method from TDM based metro transport network to packet based transport network and advantage and effectiveness of packet based transport network and also indicates technical factor and characters about method of packet transport system. As a result of research, the Packet Transport System, which is a transmission network for packet delivery, take in not only a specific character of legacy TDM but QoS, Multicast and high bandwidth, then, it is able to keep an effective bandwidth and a stabilized performance of packet transmissions. Additionally, if a fault be occurred on an optical link, the system is able to guarantee a differential QoS by an each service class using an algorithm to make certain of a traffic existence and contain a protective mechanism.

• Proceedings of the IEEK Conference
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• 1999.11a
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• pp.121-124
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• 1999

In a packet-switched wireless cellular network, a packet destined to a mobile station is queued at a base station and then broadcast over the base station's cell. When an active mobile station leaves a cell, there re-main packets which are destined to the mobile and not yet delivered to it at the cell's base station. For application which are sensitive to packet losses, such back- logged packets must be forwarded to the new base station. Otherwise, an end-to-end retransmission may be required. However, an increase in packet delay is incurred by employing the packet forward scheme, since a packet may be forwarded many times before it is delivered to the destined mobile station. For an enhanced quality-of-service level, it is preferred to reduce tile packet delay time. In this paper, we develop an analytical approximation method for deriving mean packet delay times. Using the approximation and simulation methods, we investigate the effect of network parameters on the packet delay time.

• Information and Communications Magazine
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• v.24 no.12
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• pp.22-30
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• 2007

본고에서는 현재 3GPP(3rd Generation Partner Project)에서 표준화 작업이 진행중인 EPS(Evolved Packet Service) 시스템에서, Seamless radio mobility 기술에 대해서 알아본다. 본고는 EPS 표준 규격과 관련 기고를 참고하여 작성하였다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.4B
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• pp.255-269
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• 2006

The differentiated services(DiffServ) architecture provides packet level service differentiation through the simple and predefined Per-Hop Behaviors(PHBs). The Assured Forwarding(AF) PHB proposed as the assured services uses the RED-in/out(RIO) approach to ensusre the expected capacity specified by the service profile. However, the AF PHB fails to give good QoS and fairness to the TCP flows. This is because OUT(out- of-profile) packet droppings at the RIO buffer are unfair and sporadic during only network congestion while the TCP's congestion control algorithm works with a different round trip time(RTT). In this paper, we propose an Adaptive Regulating Drop(ARD) marker, as a novel dropping strategy at the ingressive edge router, to improve TCP fairness in assured services without a decrease in the link utilization. To drop packets pertinently, the ARD marker adaptively changes a Temporary Permitted Rate(TPR) for aggregate TCP flows. To reduce the excessive use of greedy TCP flows by notifying droppings of their IN packets constantly to them without a decrease in the link utilization, according to the TPR, the ARD marker performs random early fair remarking and dropping of their excessive IN packets at the aggregate flow level. Thus, the throughput of a TCP flow no more depends on only the sporadic and unfair OUT packet droppings at the RIO buffer in the core router. Then, the ARD marker regulates the packet transmission rate of each TCP flow to the contract rate by increasing TCP fairness, without a decrease in the link utilization.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.25 no.4B
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• pp.692-699
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• 2000

We build and analyze some types of queueing model to discuss capacity dimensioning of access links of a LAN interworking unit connected to the Internet backbone network. We assume that the IWU has a FIFO buffer to transmit IP packets to the Internet through the backbone. In order to analyze the system, we use a Poisson process and an MMPP process as input traffic models of IP packets and we use a general service time distribution as a service time model. But we use both an exponential service time and a deterministic service time in numerical examples for simple and efficient performance comparisons. As performance measures, we obtain the packet loss probability and the mean packet delay. We present some numerical results to show the effect of arrival rate, buffer size and link capacity on packet loss and mean delay.

• Journal of the Optical Society of Korea
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• v.18 no.5
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• pp.466-471
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• 2014

An asynchronous optical packet-switching (AOPS) node with load-balancing capability can achieve better performance in reducing the high packet-loss ratio (PLR) and time delay caused by unbalanced traffic. This paper proposes a novel dynamic load-balancing algorithm for an AOPS node with limited buffer and without wavelength converters, and considering the data flow distribution and service levels. By calculating the occupancy state of the output ports, load state of the input ports, and priorities for data flow, the traffic is balanced accordingly. Simulations demonstrate that asynchronous variant data packets and output traffic can be automatically balanced according to service levels and the data flow distribution. A PLR of less than 0.01% can be achieved, as well as an average time delay of less than 0.46 ns.