• Journal of Digital Convergence
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• v.12 no.8
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• pp.1-18
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• 2014

Recently, many of the mobile network operators or telcos are introducing the LTE service in order to effectively cope with an explosive increasing mobile traffics due to an expansion of the use of smart phones. The 1.8GHz, 2.6GHz, and 800MHz band classes are most widely used for LTE. In particular, the 1.8GHz band class is the most useful one in terms of the reusability of the existing (2G) network, global harmonization, bandwidth, eco-system of equipments and devices, and so on. In recent years, major countries in the world have allocated the 1.8GHz band spectrum in a wide bandwidth unit suitable for the upcoming LTE-Advanced service. This paper surveyed the 1.8GHz band spectrum allocation policies of the 12 OECD countries, including Republic of Korea. From the survey, we have found that they rebuilt or refarmed the existing holders' bands, recovered the public (i.e., military)-use bands, and allocated the bands in a wide bandwidth and in an equal or similar size.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37C no.9
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• pp.829-840
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• 2012

Recently, in network communication environments, it is changing very fast from wired to wireless. The open shortest path firtst (OSPF), one of link state routing protocols, mainly used in wired networks, is the routing method to select optimal traffic path as identifying the link state of neighbor routers. The traditional OSPF cost functions performs with first fixed cost permanently, unless the router link is changed. However, in wireless networks, the performance of links show big difference by other environment factors. The bit error rate (BER), a parameter which can quite affect link state in wireless networks, is not considered in the traditional OSPF cost functions. Only a link bandwidth is considered in the traditional OSPF cost functions. In this paper, we verify the various parameters which can affect link performance, whether it is permissible to use as the parameter of proposed cost functions. To propose new cost functions, we use the effective bandwidth. This bandwidth is calculated by proposed formula using the BER of the network link and link bandwidth. As applied by the proposed triggering condition, the calculated effective bandwidth decrease the unstable of network by generating less link state update messages in wireless networks that frequently changes the link state. Simulation results show that the proposed cost functions significantly outperforms the traditional cost functions in wireless networks in terms of the services of VoIP and data transmission.

• The KIPS Transactions:PartC
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• v.14C no.6
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• pp.519-524
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• 2007

One of the most important issues on the sensor network with resource limited sensor nodes is prolonging the network lifetime by effectively utilizing the limited node energy. The most representative mechanism to achieve a long lived sensor network is the clustering mechanism which can be further classified into the single hop mode and the multi hop mode. The single hop mode requires that all sensor nodes in a cluster communicate directly with the cluster head(CH) via single hop md, in the multi hop mode, sensor nodes communicate with the CH with the help of other Intermediate nodes. One of the most critical factors that impact on the performance of the existing multi hop clustering mechanism is the cluster size and, without the assumption on the uniform node distribution, finding out the best cluster size is intractable. Since sensor nodes in a real sensor network are distributed non uniformly, the fixed size mechanism may not work best for real sensor networks. Therefore, in this paper, we propose a new dynamic size multi hop clustering mechanism in which the cluster size is determined according to the distance from the sink to relieve the traffic passing through the CHs near the sink. We show that our proposed scheme outperforms the existing fixed size clustering mechanisms by carrying out numerical analysis and simulations.

• The Transactions of the Korea Information Processing Society
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• v.3 no.4
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• pp.947-960
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• 1996

The primary goal for developing high performance ATM switching systems is to minimized the probability of cell loss, cell delay and deterioration of throughput. ATM switching element that is the most suitable for this purpose is the shared buffer memory switch executed by common random access memory and control logic. Since it is difficult to manufacture VLIS(Very Large Scale Integrated circuit) as the number of input ports increased, the used of switching module method the realizes 32$\times$32, 150 Mb/s switch utilizing 8$\times$8, 600Mb/s os 16$\times$16, 150Mb/s unit switch is latest ATM switching technology for small and large scale. In this paper, buffer capacity satisfying total-memory-reduction effect by buffer sharing in a shared buffer memory switch are analytically evalu ated and simulated by computer with cell loss level at traffic conditions, and also features of switching network utilizing the switching module methods in small and large-capacity ATM switching system is analized. Based on this results, the structure in outline of 32$\times$32(4.9Gb/s throughput), 150Mb/s switches under research in many countries is proposed, and eventually, switching-network structure for ATM switching system of small and large and capacity satisfying with above primary goals is suggested.

• The Transactions of the Korea Information Processing Society
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• v.3 no.7
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• pp.1727-1738
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• 1996

This paper proposes a extended service filtering technique to solve overload control problems inService Con-trol Poin(SCP) due to Televoting (VOT) services with heavy traffic characteristics. Also, this paper compares this new technique with existing IN overload control lechniques, and calculate steady state call blocking probabilities in In under overload conditions. The proposed technique considers SCP overload and IN CS-1services such as VOT service that need to use the specialized resources of IP. This technique uses first an activating step which SCP sends service filtering request to SSP and IP for the start of service filtering. Then, in the filtering step SSP sendas filtering results to SCP periodically or each N-calls.Also,when filtering time-out timer expired or call counter value exceeded maximum number of calls,SSP stops service filtering,and sends ser-vice filteing response to SCP in the deactivating step. The application of this techique to VOT service defined in IN CS-1 service is investigated and analyzed by using an analytical VOT service model.In order to apply this technique to VOT service, this paper also discusses VOT service and the corresponding N architecture including Service Switching Point(SSP),SCP and IP. Then this paper also constructs an analytical VOT service model,and calculates SCP and SSP/IP(circuit)call blocking probablities.With the application of this new technique,this paper shows good performance enhancements in SCP call blocking probability.

• Journal of KIISE:Information Networking
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• v.34 no.2
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• pp.110-119
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• 2007

Mobile IPv6 (MIPv6) enables a mobile node (MN) to maintain its connectivity with a correspondent node (CN) while changing its point of attachment. In MIPv6, packets sent from a CN to a MN during handover are lost. Several mechanisms including FMIPv6 and HMIPv6 have been proposed in order to minimize packet loss. However, such mechanisms still suffer from performance degradation due to not only packet loss but also out-of-sequence packets. In this paper, we propose I-FHMIPv6 to resolve packet loss as well as the out-of-sequence packet problem. In I-FHMIPv6, the flush message is newly defined in order to notify a home agent (HA) or CN of the fact that the binding cache entry of a MN is about to be updated. A MN receiving the flush message can know that there is no more packets transmitted via the previous route, which resolve the out-of-sequence packet problem. Moreover, with the proposed mechanism, we can minimize packet loss by integrating FMIPv6 and HMIPv6 efficiently. I-FHMIPv6 is evaluated by performing simulations, and the simulation results show that I-FHMIPv6 outperforms FMIPv6 and HMIPv6.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.9A
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• pp.737-748
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• 2005

Two flexible admission control schemes for integrated voice and stream-type data services are proposed in DS-CDMA systems. Most Previous studies on admission control have focused on integration of short, bursty Packet-type data services and conventional voice services. However, stream-type data services with a relatively long service holding time are expected to be a considerable portion of data traffic in future generation cellular systems. Scheme I is a basic scheme that accommodates both voice and data services with full bandwidth. However, voice services are given priority over data services using the duration difference between the holding times for these services. Scheme ll uses a different method to efficiently give priority to voice services over stream-type data services. An additional interference margin for voice services is provided by suppressing interference from stream-type data services according to voice access requests and a varying interference status. Performance of the two schemes is evaluated by developing Markovian models. Numerical results show that the voice capacity is highly sensitive to the service holding time of data services while the performance measures of data services are not highly sensitive. Scheme H is a significant improvement over Scheme I for accommodating voice and stream-type data services

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

• Journal of IKEEE
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• v.2 no.1 s.2
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• pp.122-133
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• 1998

The CL network fer CBDS(Connectionless Broadband Data Service) defined by ITU-(International Telecommunication Union-Telecommunication) consists of CL(ConnectionLess) servers which route frames and links which connect between CL user and CL server or between CL servers[1][2]. In this configuration, two kinds of congestion, CL link congestion and CL server congestion may occur. Therefore this paper has proposed the scheme that can deal with the above congestion state of LAN traffic efficiently in interworking LANs in ATM network. Proposed schemes ale link-by-link scheme and hybrid scheme. Link-by-link scheme uses the ABR feedback loops of AIM layer between CL user and CL server or between CL servers, and the header translation table carrying out the routing function of frame. And hybrid scheme is the scheme that adds end-to-end loop to link-by-link scheme and then uses RM message using CLNAP-PDU(ConnectionLess Network Access Protocol - Protocol Data Units) of CL layer. These schemes can be implemented easily because of using the ABR loop of ATM layer, end-to-end loop of CL layer, and the routing function of CL server. In this paper the characteristic and effects of two schemes in CL server bas been compared and analyzed.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.38 no.9
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• pp.12-20
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• 2001

The ATM-PON has an important meaning at next generation access network because the equipment transmits various types of user traffic with a single platform through the passive optical splitter. Ranging is a technology to place all ONUs at the same virtual distance in order to provide the synchronization for the upstream signal in the ATM-PON based on the Time Division Multiplexing (TDM). In this paper, we review the merits of ATM PON and related protocol for the PON operation. We summarize and analyze the steps about the ranging protocol based on ITU-T G.983.1 and implement a simulator that can simulate the ranging procedure based on our modeling. In this paper, we can investigate time requirements of G.983.1 by using the simulator and find out the wasted rate of the bandwidth during the ranging procedure in order to know the influence for ONUs while ATM-PON is in-service. Also, we show that a new ranging scheme reducing window size can avoid the degradation of quality of service of ONUs in service with simulation.