• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.12
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• pp.108-116
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• 2009

In this paper, we propose a Ethernet based audio distribution system with a new L1/L2 hybrid switching scheme, and evaluate its performance. The proposed scheme not only offers guaranteed low latency and jitter characteristics that are essentially required for the distribution of high-quality uncompressed audio traffic, and but also provide an efficient transmission of data traffic on the Ethernet environment. The audio distribution system with a proposed scheme consists of a master node and a number of relay nodes, and all nodes are mutually connected as a daisy-chain topology through up and downlinks. The master node generates an audio frame for each cycle of 125us, and the audio frame has 24 time slotted audio channels for carrying stereo 24 channels of 16-bit PCM sampled audio. On receiving the audio frame from its upstream node via the downlink, each intermediate node inserts its audio traffic to the reserved time slot for itself, then relays again to next node through its physical layer(L1) transmission - repeating. After reaching the end node, the audio frame is loopbacked through the uplink. On repeating through the uplink, each node makes a copy of audio slot that node has to receive, then play the audio. When the audio transmission is completed, each node works as a normal L2 switch, thus data frames are switched during the remaining period. For supporting this L1/L2 hybrid switching capability, we insert a glue logic for parsing and multiplexing audio and data frames at MII(Media Independent Interlace) between the physical and data link layers. The proposed scheme can provide a good delay performance and transmission efficiency than legacy Ethernet based audio distribution systems. For verifying the feasibility of the proposed L1/L2 hybrid switching scheme, we use OMNeT++ as a simulation tool with various parameters. From the simulation results, one can find that the proposed scheme can provides outstanding characteristics in terms of both jitter characteristic for audio traffic and transmission efficiency of data traffics.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.13 no.1
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• pp.67-74
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• 2009

This paper presents a method of improving the service availability by distributing the traffic of voice/data calls over the multi-layer cells in a mobile communication network. The traffic model is described and the call handling performance is analyzed. In our method, a fast moving call is moved to and serviced in the upper layer cell. A call is also moved upward when an overflow occurs. But unlike other methods, the call that is moved upward in the overflow case is the one which has the longest sojourn time in the cell. Moreover, when the call that was moved upward due to overflow condition stays longer than a certain period of time in the upper layer cell, the system moves the call back to the lower layer in order to save the more expensive resources of the upper layer cell. Call handling performance of this method evaluated from M/M/C/K models shows clear improvement with respect to call blocking probability and forced termination probability.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.10A
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• pp.965-973
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• 2007

We analyzed that the loss of data in TCP protocol based wireless networks caused by overlapped responses in bi-directional nodes that were resulted in out of the data sequence. This loss can be prevented by using revised TCP rate control algorithm and the performance of throughput can also be improved. The rate control algorithm is applied when the congestion happens between nodes while traffic packets are retransmitting in TCP bandwidth. In addition to applying the rate control algorithm, we determine the number of system clients in bandwidth and the average of pausing time between transmitting serial files to produce a competitive level so that an efficient performance of rapid retransmitting for the loss of multi-packets. This paper discusses the improvement of congestion control in that the decrease of the loss, firstly, as ensuring an efficient connection rate and, secondly, as using sliding window flow control.

• Journal of the Korea Academia-Industrial cooperation Society
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• v.14 no.4
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• pp.1976-1981
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• 2013

In this paper, we propose a frequency-domain interference cancellation algorithm for direct-sequence spread spectrum systems. In the previously proposed frequency-domain interference cancellation algorithms that protocol defines the rules concerning the collection of means of Transmission Control Protocol (TCP: Transmission Control Protocol) is the most widely used in the transport layer. Two-way traffic through the network path to the same end-to-end transfer of data in the opposite direction between pairs of nodes are infused with two or more TCP connection using the network traffic patterns from the exchanger and routers share results of approval. Per-flow input/output structure of matter using the LTS online reaction when evaluated as this is the most important factor. TCP-MT when the connection duration is one of the largest performance gains.

• The KIPS Transactions:PartC
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• v.12C no.2 s.98
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• pp.191-200
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• 2005

The brilliant achievements in computers and the internet technology make it easy for users to get useful information. But at the same time, the damages caused by intrusions and denial of service attacks are getting more worse. Specially because denial of service attacks by internet worm incapacitate computers and networks, we should draw up a disposal plan against it. So far many rule-based intrusion detection systems have been developed, but these have the limits of these ability to detect new internet worms. In this paper, we propose a system to detect intrusion and generate detection rule against scan-based internet worm, paying attention to the fact that internet worms scan network to infect hosts. The system detects internet worms using detection rule. And if it detects traffic causing by a new scan-based internet worm, it generates new detection nile using traffic information that is gathered. Therefore it can response to new internet worms early. Because the system gathers packet payload, when it is being necessary only, it can reduce system's overhead and disk space that is required.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.16 no.1
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• pp.139-145
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• 2016

Dumbbell is the most basic topology that can be used in almost all kind of network experiment within it or just by little expansion. While Transmission Control Protocol TCP is the basic protocol that is used for the connectivity among networks and stations. TCP major and basic goal is to provide path and services to different applications for communication. For that reason TCP has to transfer a lot of data through a communication medium that cause serious congestion problem. To calculate the congestion problem, different kind of pre-cure solutions are developer which are Loss Based Variant and Delay Based Variant. While LBV keep track of the data that is going to be passed through TCP protocol, if the data packets start dropping that means congestion occurrence which notify as a symptom, TCP CUBIC use LBV for notifying the loss. Similarly the DBV work with the acknowledgment procedure that is used in when data ACK get late with respect to its set data rate time, TCP COMPOUND/VAGAS are examples of DBV. Many algorithms have been purposed to control the congestion in different TCP variants but the loss of data packets did not completely controlled. In this paper, the congestion control algorithms are implemented and corresponding results are analyzed in Dumbbell topology, it is typically used to analyze the TCP traffic flows. Fairness of throughput is evaluated for different TCP variants using network simulator (NS-2).

• Journal of Korea Multimedia Society
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• v.11 no.7
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• pp.1007-1015
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• 2008

In order to provide real-time data from sensors and instruments at manufacturing processes on web, we proposed a communication service model based on XML(eXtensible Markup Language). HTML(Hyper Text Markup Language) is inadequate for describing real-time data from manufacturing plants while it is suitable for display of non-real-time multimedia data on web. For applying XML-based web service of process data in Intranet environment, real-time performance of communication services was evaluated to provide the system design criteria. XML schema for the data presentation was proposed and its communication performance was evaluated by simulation in terms of transmission delay due to increased message length and processing delay for transformation of raw data into defined format. For transformation of raw data into XML format, we proposed two structures: one is the scheme where transformation is done at an SCC(Supervisory Control Computer) after receiving real-time data from instruments. the other is the scheme where transformation is carried out at instruments before the data are transmitted to the SCC. Performances of two structures were evaluated on a testbed under various conditions such as six packet sizes and offered loads of 20%, 50% and 80%, respectively. Test results show that proposed schemes are applicable to the systems in Ethernet 100BaseT network if total message traffic is less than 7 Mbps.

• Journal of the Korea Institute of Information and Communication Engineering
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• v.24 no.4
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• pp.515-521
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• 2020

Recently, data traffic has exploded due to development of various industries, which causes problems about losing of efficiency and overloaded existing networks. To solve these problems, network slicing, which uses a virtualization technology and provides a network optimized for various services, has received a lot of attention. In this paper, we propose a resource allocation method using credit value. In the method using the clustering technology, an operation for selecting a cluster is performed whenever an allocation request for various services occurs. On the other hand, in the proposed method, the credit value is set by using the residual capacity and balancing so that the slice request can be processed without performing the operation required for cluster selection. To prove proposed method, we perform processing time and balancing simulation. As a result, the processing time and the error factor of the proposed method are reduced by about 13.72% and about 7.96% compared with the clustering method.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.7B
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• pp.463-471
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• 2007

Because of MANET's dynamic characteristic, the hidden node problem can happen. Thus it must use with distributed channel access. In Ad hoc networks, carrier sense multiple access with collision avoidance(CSMA/CA) is one of the most widely used medium access control(MAC) schemes for asynchronous data traffics. However, CSMA/CA could not guarantee the quality of multimedia traffics. CSMA is a contention based protocol. Therefor once a node gets a channel, it can momopolze. Thus the fairness problem with channel starvation will happen. We will propose a new MAC protocol to guarantee QoS for multimedia data in ad hoc networks.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.42 no.1
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• pp.128-139
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• 2017

Vehicular Ad hoc Network (VANET) is one of technologies to realize various ITS services for safe driving and efficient traffic control. However, data delivery in VANETs is complicated due to high mobility and unreliable wireless transmission. In this paper, we develop a novel forwarding scheme to deliver packets in a reliable and timely manner. The proposed forwarding scheme uses traffic statistics to predict the encounter of two vehicles, and optimize its forwarding decision by taking into consideration the probability of successful transmission between them at the encounter place. We evaluate our scheme through simulations and show that our proposed scheme provides reliable data delivery in VANETs.