• The Journal of the Korea Contents Association
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• v.10 no.1
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• pp.46-58
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• 2010

The user valence of VoIP services with SIP protocol is increasing rapidly because of cheap communication cost and its conveniency. But attacker can easily modify the packet contents of SIP protocol as SIP header is transmitted by using UDP methods in text form. The reason is that SIP protocols does not provide an authentication function on the transmission session. Therefore, existing SIP protocol is very weak on SIP Packet Flooding attack etc. In order to solve like this kinds of SIP vulnerabilities, we used SIP status codes under the monitoring module for detecting SIP Flooding attacks and additionally proposed an advanced protocol where the authentication and security function is strengthened about SIP packet. We managed SIP session spontaneously in order to strengthen security with SIP authentication function and to solve the vulnerability of SIP protocol. The proposed mechanism can securely send SIP packet to solves the security vulnerability with minimum traffic transmission. Also service delay in SIP proxy servers will be minimized to solve the overload problem on SIP proxy server.

• The Journal of The Korea Institute of Intelligent Transport Systems
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• v.15 no.2
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• pp.102-110
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• 2016

After Sewol ferry disaster occurred in April 2014, Korean government began investing to deploy LTE-based public safety network until the year of 2017. In order to reduce the operating and capital costs, resource sharing scheme among public safety network and commercial LTE networks is considered as one of the viable approaches. This thesis proposes a method of allocating bandwidth of public safety network based on various priorities required for disaster scenarios and stages in a resource sharing environment. In order to obtain the highest efficiency, we formulate the bandwidth allocation problem as a Fractional Knapsack Problem. Greedy algorithm was applied to solve the problem. For performance evaluation, we created several disaster scenarios and set suitable parameters for each scenario based on a disaster manual. The proposed method is compared with two typical methods, which are Class-based bandwidth allocation and Uniform bandwidth allocation. The results showed that the better performance in terms of the sum of the values and the amount of lost bytes.

• 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.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.4 no.3
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• pp.537-548
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• 2000

ATM is the effective information transmission method which multiply statistical and can accept effective the traffic of the various transmission rate. However, it can happen excessive cell loss probability and cell delay when it have temporarily overload. Therefore, it is required the effective traffic control and network resource management for which guarantee QoS(Quality of Service) in terms of users and bandwidth utilization maximization in terms of networks. In this paper, we proposed POBP(PushOut BP) scheme which mixed pushout scheme with BP(Back Pressure), reactive control scheme recommended at the ATM Forum, to guarantee QoS in two stages ATM switch networks. In proposed method, we have to understand exactly using bandwidth information in real-time and become traffic shaping. Thorough these processes, we can not only use effectively unused bandwidth, but also guarantee the fair bandwidth utilization and then can improve cell-loss possibility happened by congestion states in two stage ATM networks.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.957-965
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• 1999

그룹간 공동 작업이나 화상 회의와 같은 그룹 통신의 수요가 늘어나면서 멀티캐스팅을 이용한 효율적인 데이타 전송에 대한 요구가 증가하고 있다. 특히 오디오나 비디오 데이타와는 달리 공동 문서 작업이나 그룹간 문서 전송을 위한 데이타 처리를 위해서는 어느 정도의 시간 손실이 있더라도 신뢰성을 보장할 수 있는 멀티캐스트 프로토콜이 요구된다. 그러나 멀티캐스트 전송에서의 신뢰성을 보장하기 위하여 손실 패킷에 대한 재전송 패킷이 전체 그룹으로 멀티캐스트 되는 것은 네트워크 상에 트래픽을 폭증시키는 요인이 된다. 이에 본 논문에서는 지역 그룹에서의 오류 회복을 위한 재전송 제한 기법을 제안하여 손실 패킷의 재전송 과정에서 발생하는 트래픽의 폭주를 제어하고자 한다. 이것은 재전송 패킷이 중복없이 다중 수신자에게 전송될 수 있도록 하여 그룹 내의 재전송 패킷의 양을 줄이고 필요없는 중복 패킷이 네트워크의 효율을 저하시키는 것을 방지하고자 하는 데 그 목적이 있다. Abstract As the size and the geographic span of communication groups increases, efficient data transmission schemes using Multicast service become more and more essential. Especially, unlike audio and video applications, for some collaborative applications and other data delivery components which require ordered and lossless delivery of data, Reliable Multicast Service is needed to ensure consistent presentation across multiple views. Thus error recovery by retransmission of loss data is provided in order to guarantee the reliability of multicast transmission protocol. However, redundant retransmission packets by multicast may cause traffic implosion on the Internet and it can be aggravated with continuous retransmission.This paper describes a Restricted Retransmission Mechanism as an error recovery method of multicast service in a local group, which can handle traffic implosion in retransmission process. It reduces redundant retransmission packets flowing into a local group and supports reliable multicast transmission. The goal of this mechanism is to reduce retransmission packets and decrease the load for group members and networks.

• Journal of the Korea Society of Computer and Information
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• v.12 no.3
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• pp.185-194
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• 2007

WiBro has become intentionally standardize as IEEE 802.16e. This WiBro service has been started by a portable internet at home as well as abroad. In this paper, an offender hacker do not direct attack on system on system that It marched an attack directly in damage system because a place oneself in mobile station of portable internet WiBro and avoid to attack hacker's system. At this time, a mobile make use of network inspection policy for back-tracking based on log data. Used network log audit, and presented TCP/IP bases at log bases as used algorithm, the SWT technique that used Thumbprint Algorithm. Timing based Algorithm, TCP Sequence number. Study of this paper applies algorithm to have been progressed more that have a speed to be fast so that is physical logical complexity of configuration of present Internet network supplements a large disadvantage, and confirm an effective back-tracking system. result of research of this paper contribute to realize a back-tracking technique in ubiquitous in WiBro internet network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.4C
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• pp.343-355
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• 2003

Fair bandwidth allocations at routers protect adaptive flows from non-adaptive ones and may simplify end-to end congestion control. However, traditional fair bandwidth allocation mechanisms, like Weighted Fair Queueing and Flow Random Early Drop, maintain state, manage buffera and perform packet scheduling on a per-flow basis. These mechanisms are more complex and less scalable than simple FIFO queueing when they are used in the interi or of a high-speed network. Recently, to overcome the implementation complexity problem and address the scalability and robustness, several fair bandwidth allocation mechanisms without per-flow state in the interior routers are proposed. Core-Stateless Fair Queueing and Rainbow Fair Queuing are approximates fair queueing in the core-stateless networks. In this paper, we proposed simple Layered Fair Queueing (SLFQ), another core-stateless mechanism to approximate fair bandwidth allocation without per-flow state. SLFQ use simple layered scheme for packet labeling and has simpler packet dropping algorithm than other core-stateless fair bandwidth allocation mechanisms. We presente simulations and evaluated the performance of SLFQ in comparison to other schemes. We also discussed other are as to which SLFQ is applicable.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.3B
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• pp.183-190
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• 2003

The explosion of the Internet has brought an acute need for efficient operation, which becomes to develop several technologies based on optical networks. By matching merging technology using wavelength multiplexing, an efficient administration with limited wavelength can provide scalability of network. The merging in optical layer has limitation of devices since the merging can not be performed in this layer. Hence the merging must be implemented in electronic layer. When the merging is implemented, the delay time would be increased in OXC (Optical Cross Connection), but the improvement of throughput rate will be expected in the process of explosive traffic due to reduction of optical channel supporting large bandwidth. In this thesis, we proposed an OXC of dual module having a merging function. We considered the optimum merging point with the largest in system performance and confirmed results using the simulation.

• Journal of the Korea Computer Industry Society
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• v.3 no.1
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• pp.85-94
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• 2002

EMRCA(Explicit Max_min Rate Control Algorithm) switch, which has been proposed in the ATM(Asychronous Transfer Mode) standard, controls the ABR(Available Bit Rate) service traffic in the ATM networks. The ABR service class of ATM networks uses a feedback control mechanism to adapt to varying link capacities. The VS/VD(Virtual Source/Virtual Destination) technique offers the possibility to segment the otherwise end-to-end ABR control loop into separate loops. The improved feedback delay and the control of ABR traffic inside closed segments provide a better performance and QoS(Quality of Service) for ABR connections with respect to throughput, delay, and jitter. This paper is study of an ABR VS/VD flow control method. Linear control theory offers the means to derive correct choices of parameters and to assess performance issues, like stability of the system, during the design phase. The performance goals are a high link utilization, fair bandwidth distribution and robust operation in various environments, which are verified by discrete event simulations. The major contribution of this work is the use of linear control theory to model and design an ABR flow control method tailored for the special layout of a VS/VD switch, the simulation shows that this techniques better than conventional method.