• Journal of the Korea Institute of Information and Communication Engineering
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• v.17 no.12
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• pp.2826-2834
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• 2013

In mobile ad hoc networks (MANET), power-saving technology of mobile nodes is divided into transmit power control (TPC), power-saving mode (PSM), and routing. TPC and PSM are operated in physical layer but the routing is managed in network layer, so the design of a joint algorithm is needed to provide better performance. Therefore, in this paper, we propose a joint power-saving and routing algorithm for maximizing the network lifetime while satisfying the end-to-end data rate in ad hoc networks. The proposed algorithm first applies the TPC or PSM to reduce the power consumption of mobile nodes and then performs the routing by considering the decided node lifetime in order to maximize the path lifetime. Simulation results show that the proposed algorithm maximize the lifetime while satisfying the required rate according to the number of mobile nodes and the level of interference.

• The Journal of the Korea institute of electronic communication sciences
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• v.7 no.3
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• pp.491-497
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• 2012

Network coding is that the nodes can combine and mix the packets rather than merely forward them. Therefore, network coding is expected to improve throughput and channel efficiency in the wireless network. Relevant researches have been carried out to adapt network coding to wireless multi-hop network. In this paper, we designed the network coding for bidirectional traffic service in routing layer and IP layer of Ad-hoc network. From the simulation result, the traffic load and the end to end distance effect the performance of the network coding. As end to end distance and the traffic load become larger, the gain of network coding become more increased.

• Journal of Internet Computing and Services
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• v.8 no.1
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• pp.25-31
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• 2007

The development of efficient Quality of Service (QoS) routing algorithms in high speed networks is very difficult since divergent services require various quality conditions, If the QoS parameter we concern is to measure the delay on that link, then the routing algorithm obtains the Least Delay (LD) path, Meanwhile, if the parameter is to measure of the link cast, then it calculates the Least Cost (LC) path. The Delay Constrained Least Cast (DCLC) path problem of the mixed issues on LD and LC has been shown to be NP-hard. The path cost of LD path is relatively mere expensive than that of LC path, and the path delay of LC path is relatively higher than that of LD path in DCLC problem. In this paper. we propose the algorithm based on estimated path for the DCLC problem and investigate its performance, It employs a new parameter which is probabilistic combination of cost and delay, We have performed empirical evaluation that compares our proposed algorithm with the DCUR in various network situations.

• Journal of KIISE:Information Networking
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• v.34 no.5
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• pp.348-359
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• 2007

The conventional mobile VPN services assumed the mobile communications occur between the MN in foreign networks and the CN in the home network. However, if a MN wants to communicate with another MN in a foreign network, it could degrade the performance of the mobile VPN service because of the triangular routing problem. In this paper, we propose a route optimization mechanism based on the mobile VPN using an x-HA allocated by diameter MIP in order to support the efficient communication between the mobile VPN users in foreign networks. The i-HA maintains the VPN-TIA as well as the x-HoA as the CoAs to solve the security problem and to provide an efficient route optimization simultaneously. Moreover, we proposed revised IPSec tunnel configuration to reduce the IPSec tunnel overheads at a MN when the MN communicates with several MNs in the foreign networks at the same time. The VPN server, a security management entity in the home network, notifies an additional IPSec tunnel establishment between the x-HAs where the communication peers are registered. The simulation result showed that the proposed scheme decreases the end-to-end packet delay time and improves the throughput after the handoff compared to the existing mechanism.

• 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 KIISE:Information Networking
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• v.29 no.5
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• pp.501-509
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• 2002

When using TCP over a mobile network, TCP responds to a handoff by invoking a congestion control algorithm, thereby resulting in a degraded end-to-end performance in a mobile network. In this paper, two schemes are proposed, TCP-MD and TCP-R. TCP-MD can detect the movement of s mobile host early on, whereas TCP-R can force the source to freeze data transmission during registration. The proposed schemes maintain end-to-end TCP semantics, making it possible to fully interoperate with the existing infrastructure. Only a small change is required in the mobile host, plus the implementation is simple because some Mobile IP messages are used to notify the handoff, eliminating the need for any additional messages. Simulations confirmed that the proposed schemes give an excellent performance under various environments.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.30 no.12C
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• pp.1239-1247
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• 2005

It is important to improve TCP performance in Self-similar TCP network where signalling between the same end nodes through bidirectional traffic routes. In wireless link, the traffic limitation pattern occurred in two or more TCP connections is applied into MPEC video control as multi time-interval congestion control. For TCP update variable, we extend TCP and perform as function call, and we study a method of relating TCP with LTS module controlling with the information type that is overcoming the limit of feedback loop determined by RTT. For comparison, we measure the TCP throughput without LTS and verify the fairness by means of meta control. The improved TCP performance is shown by that the number of connections of traffic congestion control increases when RTT increases.

• The KIPS Transactions:PartC
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• v.14C no.2
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• pp.163-170
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• 2007

The next generation communication environment consists of various wireless access networks with distinct features that are configured as an overlay topology. In the network environments, the frequency of hand overs should be minimized and the error propagation should be solved in order to provide high-quality multimedia services to mobile users. Therefore, we propose an performance enhancement approach, based on mSCTP, that provides high quality multimedia services to mobile users by ameliorating the error propagation problem. We utilizes the following four functions: 1) the separation of transmission paths according to the types of frames. 2) retransmission strategy to minimize the loss rate of frames, 3) Foced vertical handover execution by utilizing bicasting, 4) using the stability period in order to reduce the effect of the ping pong phenomenon. The simulation results show that the proposed approach provides seamless multimedia service to mobile users by achieving error resilience.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.18 no.1
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• pp.43-46
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• 2018

This paper has designed a wideband bandpass filter (WBSF : Wide Band Stop Filter) using a stepped impedance resonator (SIR : Stepped Impedance Resonator) with improved performance and improved hairpin coupling structure. The SIR WBSF is small in size and has the advantage of having excellent bandstop characteristics. The designed BSF has a structure in which a quadrangular shaped hairpin of a / 4 length is arranged symmetrically on the upper and lower sides of the input and output transmission lines. The input and output terminals were terminated at 50 ohms for system applications. The center frequency of the SIR WBSF is 6.3 GHz, which is the second harmonic of 3.15 GHz. The designed filter has a 3dB bandwidth of 2.9 GHz and a transmission coefficient ($S_{21}$) of 33.2 dB. The reflection coefficient ($S_{11}$) at the center frequency is 0.106 dB. The application field is used for fixed microwave relay stations, fixed satellite and earth stations, and fixed satellite communications. The overall size is $20mm{\times}10mm$.

• The KIPS Transactions:PartC
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• v.11C no.7 s.96
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• pp.905-916
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• 2004

Self-organizing hierarchical ad hoc network (SOHAN) is a new ad-hoc network architecture designed to improve the scalability properties of conventional 'flat' ad hoc networks. This network architecture consists of three tiers of ad-hoc nodes, i.e.. access points, forwarding nodes and mobile nodes. This paper presents a topology discovery and routing protocol for the self-organization of SOHAN. We propose a cross-layer path metric based on link quality and MAC delay which plays a key role in producing an optimal cluster-based hierarchical topology with high throughput capacity. The topology discovery protocol provides the basis for routing which takes place in layer 2.5 using MAC addresses. The routing protocol is based on AODV with appropriate modifications to take advantage of the hierarchical topology and interact with the discovery protocol. Simulation results are presented which show the improved performance as well as scalability properties of SOHAN in terms of through-put capacity, end-to-end delay, packet delivery ratio and control overhead.