• ETRI Journal
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• v.27 no.6
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• pp.777-787
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• 2005

In this paper, we propose an urgency- and efficiency-based wireless packet scheduling (UEPS) algorithm that is able to schedule real-time (RT) and non-real-time (NRT) traffics at the same time while supporting multiple users simultaneously at any given scheduling time instant. The UEPS algorithm is designed to support wireless downlink packet scheduling in an orthogonal frequency division multiple access (OFDMA) system, which is a strong candidate as a wireless access method for the next generation of wireless communications. The UEPS algorithm uses the time-utility function as a scheduling urgency factor and the relative status of the current channel to the average channel status as an efficiency indicator of radio resource usage. The design goal of the UEPS algorithm is to maximize throughput of NRT traffics while satisfying quality-of-service (QoS) requirements of RT traffics. The simulation study shows that the UEPS algorithm is able to give better throughput performance than existing wireless packet scheduling algorithms such as proportional fair (PF) and modified-largest weighted delay first (M-LWDF), while satisfying the QoS requirements of RT traffics such as average delay and packet loss rate under various traffic loads.

• Journal of the Korean Operations Research and Management Science Society
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• v.22 no.3
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• pp.145-162
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• 1997

This paper an analytical model for estimating packet waiting times at stations in the DQDB network, where the most upstream station is saturated. This model is useful in comparing the extreme unfairness which downstream stations experience due to their geographical locations in accessing the medium. Each station is modeled as an M/G/1, where the service time is defined to be the time a packet spends in the transmission buffer. The service time is decomposed into five components, and in turn, the first and second moment of each component are derived in three different modes of operation. Simulation experiments are presented for model validation and results are discussed.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.24 no.10B
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• pp.1852-1860
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• 1999

This paper studied performance of the AAL2 for variable rate real time services in ATM network with discrete-time simulation model. In this simulation, input parameters are packet fill delay for AAL2 PDU generation, guard time for ATM cell generation, burstness and number of channels. Though variation of the above mentioned parameters, we obtained end-to end delay variations and throughput, analyzed performance effect of the each parameter for voice packet service.

• IE interfaces
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• v.4 no.1
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• pp.63-69
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• 1991

In this paper, the Go-Back-N ARQ protocol with decoding in communication network is considered. The time delay and throughput are respectively analyzed as a function of window size and decoding time out. Packets arrive continuously at the decoder, and are stored in a buffer if the decoder is busy upon its arrival. The decoder devotes no more than a time-out period of predetermined length to the decoding of any single packet. If packet decoding is completed within that period, the packet leaves the system. Otherwise, it is retransimitted and its decoding starts anew. The time delay and throughput are obtained using recursive formula and difference equation. An appropriate time out and window size that satisfies the grade of service can be determined.

• IEIE Transactions on Smart Processing and Computing
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• v.5 no.3
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• pp.169-177
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• 2016

To realize high-speed intrusion detection by accommodating many regular expression (regex)-based signatures and growing network link capacities, we propose the Service TimE-Aware Load-balancing (STEAL) algorithm. This work is motivated from the observation that utilization of a many-core network intrusion detection system (NIDS) is influenced by unfair computational distribution among many-core NIDS nodes. To avoid such unfair computational distribution, STEAL is designed to dynamically distribute a large volume of traffic among many-core NIDS nodes based on packet service time, which is represented by the deep packet time in many-core NIDS nodes. From experiments, we show that compared to the commonly used load-balancing algorithm based on arrival rate, STEAL increases the number of received packets (i.e., decreases the number of dropped packets) in many-core NIDS. Specifically, by integrating an open source NIDS (i.e. Bro) with STEAL, we show that even under attack-dominant traffic and with many signatures, STEAL can rapidly improve the performance of many-core NIDS to realize high-speed intrusion detection.

• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.136-141
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• 2007

The frequency accuracy and phase alignment is necessary for ensuring the quality of service (QoS) for applications such as voice, real-time video, wireless hand-off, and data over a converged access medium at the telecom network. As telecom networks evolve from circuit to packet switching, proper synchronization algorithm should be meditated for IP networks to achieve performance quality comparable to that of legacy circuit-switched networks. The Time of Packet (ToP) specified in IEEE 1588 is able to synchronize distributed clocks with an accuracy of less than one microsecond in packet networks. But, The ToP can be affected by impairments of a network such as packet delay variation. This paper proposes the efficient method to minimize the expectable delay variation when ToP synchronizes the distributed clocks. The simulation results are presented to demonstrate the improved performance case when the efficient ToP transmit algorithm is applied.

• Journal of Korea Multimedia Society
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• v.17 no.12
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• pp.1453-1460
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• 2014

WFQ (Weighted Fair Queuing) provides not only fairness among traffic flows in using bandwidth but also guarantees the Quality of Service (QoS) that individual flow requires, which is why it has been applied to the resource reservation protocol (RSVP)-capable router. The RSVP allocates an enough resource to satisfy both the rate and end-to-end delay requirements of the flow in the condition of no packet loss, and the WFQ scheduler guarantees those QoS requirements with the allocated resource. In practice, however, most QoS-guaranteed services allow a degree of packet loss, especially from 0.1% to 3% for Voice over IP. This paper discovers that the packet loss rate of each traffic flow is determined by only its time-stamp adjustment value, and then enhances the WFQ to provide a differentiated packet loss guarantee under general traffic conditions in terms of both traffic characteristics and QoS requirements. The performance evaluation showed that the proposed WFQ could increase the utilization of bandwidth by 8~11%.

• The Journal of the Korea Contents Association
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• v.5 no.3
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• pp.201-208
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• 2005

To guarantee the quality of service for user request, we should consider various kinds of things. The important thing of QoS is that response time of service is transparently suggested 'to network users. We can know the response time of service using the information of network latency, system latency, and software component latency, In this paper, we carried out the modeling of network latency and analyzed the effects of packets loss to the network latency, Also, we showed the effectiveness of modeling using the NS-2. This research can help to provide the effective methods in case of SLA(Service Level Agreement) agreement between service provider and user.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.05a
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• pp.414-418
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• 2003

VoIP is a core technology used to transmit both voice and data in an integrated packet' form. Within this technology, SIP is the signaling protocol used for 'real time' call services; particularly those where H323 is used. Yet, when considering the needs of mobile users, it is essential we integrate VoIp within the mobile technology so the mobile host is able to receive the 'packet' transported and by, and connected to, any available internet-address. For all this to occur, we need to improve Network Delay by reducing transmission problems associated with mobile services. If we are to obtain an optimal service then we must reduce any network delays which may arise from joining Mobile IP and VoIp services. This paper, therefore, considers how, unlike previous research, these delays may be improved through the use of the signaling technology\ulcorner SIP. It also considers how this research may be introduced into current wired and wireless integrated services enabling them to use the IP 'packet'.

• 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.