• The Journal of Korean Institute of Communications and Information Sciences
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• v.28 no.9A
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• pp.702-709
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• 2003

In this paper, we propose an efficient and QoS-aware MAC scheduling algorithm for Bluetooth, which considers both throughput and delay performance of each Master-Slave pair in scheduling decisions, and thus, attempts to maximize overall performance. The proposed algorithm, MTDPP (Modified Throughput-Delay Priority Policy), makes up for the drawbacks of T-D PP (Throughput-Delay Priority Policy) proposed in [6] and improves the performance. Since Bluetooth employs a master-driven TDD based scheduling algorithm, which is basically operated with the Round Robin policy, many slots may be wasted by POLL or NULL packets when there is no data waiting for transmission in queues. To overcome this link wastage problem, several algorithms have been proposed. Among them, queue state-based priority policy and low power mode-based algorithm can perform with high throughput and reasonable fairness. However, their performances may depend on traffic characteristics, i.e., static or dynamic, and they require additional computational and signaling overheads. In order to tackle such problems, we propose a new scheduling algorithm. Performance of our proposed algorithm is evaluated with respect to throughput and delay. Simulation results show that overall performances can be improved by selecting suitable parameters of our algorithm.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.37B no.9
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• pp.806-813
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• 2012

In a cloud computing environment, various solutions were introduced to provide the service to users such as Infrastructure as a Service (IaaS), Platform as a Service (PaaS), Software as a Service (SaaS) and Desktop as a Service (DaaS). Nowadays, Mobile as a Service (MaaS) to provide the mobility in a cloud environment. In other words, users must have access to data and applications even when they are moving. Thus, to support the mobility to a mobile Thin-Client is the key factor. Related works to support the mobility for mobile devices were Mobile IPv6 and Proxy Mobile IPv6 which showed performance drawbacks such as packet loss during hand-over which could be very critical when collaborating with cloud computing environment. The proposed model in this paper deploys middleware and replica servers to support the data transmission among cloud and PMIPv6 domain. It supports efficient mobility during high-speed movement as well as high-density of mobile nodes in local mobility anchor. In this paper, through performance evaluation, the proposed scheme shows the cost comparison between previous PMIPv6 and verifies its significant efficiency.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.46 no.3
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• pp.48-58
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• 2009

Geo-graphical routing protocols as GPSR are known to be very suitable and useful for vehicular ad-hoc networks. However, a corresponding node can include some stale neighbor nodes being out of a transmission range, and the stale nodes are pone to get a high priority to be a next relay node in the greedy mode. In addition, some useful redundant information can be eliminated during planarization in the recovery mode. This paper deals with a new recovery mode, the Greedy Border Superiority Routing(GBSR), along with an Adaptive Neighbor list Management(ANM) scheme. Each node can easily treat stale nodes on its neighbor list by means of comparing previous and current Position of a neighbor. When a node meets the local maximum, it makes use of a border superior graph to recover from it. This approach improve the packet delivery ratio while it decreases the time to recover from the local maximum. We evaluate the performance of the proposed methods using a network simulator. The results shown that the proposed protocol reveals much better performance than GPSR protocol. Please Put the of paper here.

• Journal of the Korea Society of Computer and Information
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• v.19 no.12
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• pp.227-237
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• 2014

The purpose of this paper is to implement a multiple bio-signal central monitoring system based on textile fabrics flexible platform which can obtain and monitor bio signals(heart rate, body temperature, electrocardiography, electromyogram) of workers in special working environments and additional situational information (3-axis acceleration, temperature, humidity, illumination, surrounding image). This system can prevent various accidents that may occur in the remote work environment and provide fast and efficient response by detecting workers' situations in real-time. For it, the textile fabrics flexible platform was made as innerwear or outerwear so that it does not interfere with workers' performance while collecting bio-signal and situational information, and obtained information is sent to the central monitoring system through wireless communication. The central monitoring system is based on wireless medical telemetry service of WMTS (Wireless Medical Telemetry Service); can monitor from 2 to 32 people simultaneously; and was designed so that it can be expanded. Also, in this study, to verify performance of the WMTS communication model, packet transmission rates were compared according to the distance.

• Journal of KIISE:Information Networking
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• v.32 no.1
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• pp.20-28
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• 2005

TCP(Transmission Control Protocol) is one of the most important Internet protocols, which is widely used in wireless networks as well as wired networks. However, when TCP is deployed for wireless networks, it takes severe performance degradation because TCP was designed for wired network. To overcome this drawback, a number of TCP variants have been proposed in the literature. However, most previous schemes did not consider TCP enhancement over heterogeneous networks. In heterogeneous networks, an mobile node (MN) may move from one access network to another(i.e., vertical handover). In the case of vertical handover, an MN experiences a TCP performance degradation caused by the packet loss and the sudden change of link characteristics between two different access networks. In this work, we investigate the TCP performance degradation occurred in vortical handover across heterogeneous networks. First, we have conducted the measurement study over GPRS-WLAN testbed. In the measurement study. we observed the TCP performance degradation in the case of handover from WLAN to GPRS. In order to study more different TCP behaviors during vertical handover, we performed comprehensive simulations using a network simulator 2(ns-2). Based on measurement and simulation results, we investigated how to improve TCP performance in vertical handover and we concluded that the existing mechanisms cannot be perfect solutions and new mechanisms are strongly required.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.8
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• pp.1631-1636
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• 2015

With the advance of wireless communication technology in terrestrial area, underwater communication is also evolving very fast from a simple point-to-point transmission to an elaborate networked communications. Underwater acoustic channel has quite different features comparing with the terrestrial radio channel in terms of propagation delay, Doppler shift, multipath, and path loss. Thus, existing technologies developed for terrestrial communication might not work properly in underwater channel. Especially medium access control (MAC) protocols which highly depend on propagation phenomenon should be newly designed for underwater network. CSMA/CA has drawn lots of attention as a candidate of underwater MAC protocol, since it is able to resolve a packet collision and the hidden node problem. However, a received signal could be degraded by the interferences from the nodes locating outside the receiver's propagation radius. In this paper, we study the effects of interference on the CSMA/CA based underwater network. We derived the SNR with the interference using the sonar equation and analyzed the degradation of the RTS/CTS effects. These results are compared with the terrestrial results to understand the differences. Finally we summarized the design considerations in CSMA/CA based underwater network.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.41 no.5
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• pp.555-564
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• 2016

This paper proposes a time slot allocation scheme for military patrol environments. This proposal comes from analysis of traffic properties in a military patrol environment. In the near future, robots are expected to explore enemy grounds and measure threat, taking the place of human patrol. In order to control such robots, control messages must be extremely accurate. One mistake from the control center could cause a tragedy. Thus, high reliability must be guaranteed. Another goal is to maintain a continual flow of multimedia data sent from patrol robots. That is, QoS (Quality of Service) must be guaranteed. In order to transmit data while fulfilling both attributes, the per-path based centralized TDMA slot allocation scheme is recommended. The control center allocates slots to robots allowing synchronization among robots. Slot allocation collisions can also be avoided. The proposed scheme was verified through the ns-3 simulator. The scheme showed a higher packet delivery ratio than the algorithm in comparison. It also performed with shorter delay time in the downlink traffic transmission scenario than the algorithm in comparison.

• Journal of Broadcast Engineering
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• v.15 no.2
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• pp.205-216
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• 2010

In this paper, a method of MPEG-2 Transport Stream (TS) multiplexing for Ultra HDTV (UHDTV) and its design and implementation as a SW tool is described. In practice, UHD video may be divided into several HD videos and each video is encoded in parallel. Therefore, it is necessary to synchronize and multiplex multiple bitstreams encoding each HD video for transmitting and storing UHD video. In this paper, it is assumed that 4 HD videos partitioning a UHD spatially are encoded as H.264/AVC and two 5.0 channel audios are encoded by AC-3. Therefore, 4 H.264/AVC elementary streams (ESs) and 2 AC-3 ESs is mainly considered in the TS multiplexing of UHD. For the carriage of H.264/AVC and AC-3 over MPEG-2 TS, PES packetization and TS multiplexing are designed and implemented based on the extended specification of the MPEG-2 Systems and ATSC (Digital audio compressed standard), respectively. The implemented UHD TS multiplexing tool emulates real time HW operation in the time unit corresponding to the duration of one TS packet transmission in a given TS rate. In particular, in order to satisfy the timing model, the buffers defined in the TS System Target Decoder (T-STD) are monitored and their statuses are considered in the scheduling of TS multiplexing. For UHD multiplexing, two kinds of multiplexing structures, which are UHD re-multiplexing and UHD program multiplexing, are implemented and their strength and weakness are investigated. The developed UHD TS multiplexing tool is tested and verified in terms of the syntax and semantics conformance and functionalities by using a commercial analyzer and real-time presentation tools.

• Journal of KIISE:Information Networking
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• v.33 no.4
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• pp.333-342
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• 2006

The cdma2000 lxEV - DO mobile communication system provides broadcast and multicast services (BCMCS) to meet an increasing demand from multimedia data services. The servicing of video streams over a BCMCS network must, however, face a challenge from the unreliable and error-prone nature of the radio channel. The BCMCS network uses Reed-Solomon coding integrated with the MAC protocol for error recovery. We analyze this coding technique and show that it is not effective in the case of slowly moving mobiles. To improve the playback quality of an MPEG-4 FGS video stream, we propose the Hybrid error recovery scheme, which combines Reed-Solomon with ARQ, using slots which are saved by reducing the Reed-Solomon coding overhead. The target packets to be retransmitted are prioritized by a utility function to reduce the packet error rate in the application layer within a fixed retransmission budget. This is achieved by considering of the map of the error control block at each mobile node. The proposed Hybrid error recovery scheme also uses the characteristics of MPEG-4 FGS (fine granularity scalability) to improve the video quality even when conditions are adverse: slow-moving nodes and a high error rate in the physical channel.

• Journal of the Institute of Electronics Engineers of Korea CI
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• v.39 no.6
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• pp.53-67
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• 2002

The multimedia synchronization is one of the key issues to be resolved in order to provide a good quality of multimedia related services, such as Video on Demands(VoD), Lecture on Demands(LoD), and tele-conferences. In this paper, we introduce an adaptive multimedia synchronization algorithm using the level of buffers and load of servers, which are modeled and analyzed by ExSpect, a Petri net based simulation tool. In the proposed algorithm, the audio and video buffers are divided to 5 different levels, and the pre-defined play-out speed controller tries to make the buffer level to be normal in different temporal relations between multimedia streams using buffer levels and server loads. Because each multimedia packet is played by the pre-defined play-out speed, the media data can be reproduced within the permissible limit of errors while preserving the level of buffers to be normal. The proposed algorithm is able to handle and support various communication restrictions between providers and users, and offers little jitter play-out to many users in networks with the limited transmission capability. The performance of the developed algorithm is analyzed in various network conditions using a Petri net simulation tool.