Retransmissions on the DLC layer are essential to ABR service providing the low CLR (cell loss ratio) over the unreliable wireless channel with high bit error rate. In the wireless ATM, the DLC layer below ATM layer performs the retransmission and reordering of the cells to recover the cell loss over the wireless channel and by doing so, the effect of the wireless channel characteristics with high bit error rate can be minimized on the ATM layer which is designed under the assumption of the low bit error rate. We propose, in this paper, the schemes to reflect the changes of the transmission rate over the wireless channel on the ABR rate control. Proposed scheme can control the source rate to the changes of the transmission rate over the wireless channel and reduce the required buffer size in the AP (access point). In the simulation, we assume that the DLC layer can inform the ATM layer of the wireless channel quality as good or bad. Our simulation results show that the proposed schemes require the smaller buffer size compared with the existing scheme, enhanced dynamic max rate control algorithm (EDMRCA). It is also shown that the scheme with the intelligent DLC which adjusts the rate to the wireless channel quality not only provides the low CLR with smaller buffer requirement but also improves the throughput by utilizing the wireless bandwidth more efficiently.
As the full-motion videos in MPEG are widely available nowadays, an editor that could easily edit such kind of media data is required to develop various multimedia applications. In order to concatenate and apply a transition effect to two video streams encoded in MPEG, they should be decoded first since there are dependencies in the frames in MPEG-encoded video stream. Since this decode-edit-encode process requires a huge amount of computing/storage resources, a new editing scheme that could apply various transition effects to MPEG video streams directly while keeping the quality of video data is strongly required. This paper proposes a new editing scheme that could apply three transition effects (such as fade-in, fade-out, and dissolve) to MPEG video streams in a compressed domain. In the proposed scheme, an extension of previous method in which the frames are partially decompressed and transition effects are applied is adopted for I- and P-frames. In addition, a new processing scheme for B-frame that could apply the transition effects in DCT domain directly using an approximation of motion compensation based on the motion vector to reference frames. Since this processing scheme could apply the transition effects in a compressed domain directly, the editing process could be speed-up about $3{\sim}4$ times faster than previous decode-edit-encoding method while keeping the quality of video data as good as the source data. The proposed scheme could be used to build a software-only MPEG video editing system that helps to edit MPEG video data even on a low-cost desk-top computer.
The delivery of multimedia that efficiently adapts its bit-rate to changing network characteristics and conditions is one of the important challenging tasks in the design of today's real-time multimedia streaming systems such as IPTV, Mobile IPTV and so on. In these work, the primary focus is on network congestion, to improve network stability and inter-protocol fairness. However, these existing works have problems which do not support QoE (Quality of Experience), because they did not consider essential characteristics of contents playback such as the media continuity. In this paper, we propose a novel rate control scheme for improving the QoE of multimedia streaming service in the Internet congestion, called NCAR (Network and Client-Aware Rate control), which is based on network-aware congestion control and client-aware flow control scheme. Network-aware congestion control of the NCAR offers an improving reliability and fairness of multimedia streaming, and reduces the rate oscillation as well as keeping high link utilization. Client-aware flow control of NCAR offers a removing the media discontinuity and a suitable receiver buffer allocation, and provides a good combination of low playback delay. Simulation results demonstrate the effectiveness of our proposed schemes.
With growing population of internet and web applications, distributed environment is considered to be the standard architecture of application. A network management systems(NMS) is necessary to control and monitor the complex network resources for providing and sharing the heft quality service. We recognize the NMS as a standard infrastructure for supporting efficient networking and a separate commercial applications. We believe every resource including software, hardware and environment for the network management should be separated from special protocols, vendors and applications. Therefore, We need a standard network management system that is efficient and consistent because of the heterogeous network features. In regards to software development, software reuse through assembling and extending the reusable elements such as patterns and components assures to realize the best productivity and quality The component based development(CBD) methodology that can assemble black box though well defined interfaces makes it possible to develop easer and quicker applications and is proved as the best software development solution involved in construction, selection and assembly of components. In this thesis, we describe the architecture for the network management and identify, define and design the components through analysis and design in the network management domain and Identified components mapped to the component architecture. We also specify the component development and design and implement the component for developing the network management. Implemented components apply to the component repository system that register, retrieve and understand the components. We analyze, design and implement the entire network management system based on configuration, connection, performance and fault management through the pre-developed components.
Shin Sun Young;Shin Sang-Chul;Moon Tae Hee;Sohn So Young
Journal of KIISE:Information Networking
/
v.32
no.2
/
pp.220-235
/
2005
Internet Service Quality has been constantly the center of attention to Internet Service Providers. The KII project plan (Korea Information Infrastructure) aims to build broadband backbone networks mainly through optical fiber cables and ATM switches and to Provide the government ministries, local authorities and non-profit organizations with high-speed, highly capable broadband access to these networks at reasonable rates. The KB service model, however, is different from other Internet Service models much in the same way SP (Service Provider) and NP (Network Provider) differ from each other. In this paper, we evaluate KII service according to various customer satisfaction indicators under the methodologies and categories as put forth in both the ACSI (American Customer Satisfaction Index) and SEM (Structural Equation Model). We use a structural equation model (SEM) to demarcate the Korea Information Infrastructure Customer Satisfaction Index (KIICSI) in relation to network service quality. The results of our study suggest some strategies for the KII Project need to be modified and effectively implemented in order to increase the satisfaction level of the KII customers.
Kang Kyung-Tae;Cho Yong-Jin;Cho Yong-Woo;Cho Jin-Sung;Shin Heon-Shik
Journal of KIISE:Information Networking
/
v.33
no.4
/
pp.333-342
/
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.
Kang, Eun Sook;Tark, Kwan-Chul;Lee, Taewha;Kim, In Sook
Quality Improvement in Health Care
/
v.9
no.2
/
pp.116-133
/
2002
Background : It is very common in Korea to take care of non-acute patients in an acute setting, due to the lack of long-term facilities. Long term hospitalization increase medical expenses and decreases the bed utilization, which can affect the urgent and emergent admissions, and eventually jeopardize the hospital financially. In this study, strategies for effective transfers to the lower levels of care, and to decrease the length of stay were presented by surveying and analyzing the patient's knowledge of the transfer needs, and the willingness to transfer those whose hospital length of stay was more than 30days. Method : The survey is subject to a group of 251 patients who have been hospitalized over 30 days in a general hospital in Seoul. Excluding those that were in the Intensive Care Unit and psychiatric ward, 214 in-patients were used as participants. They were surveyed from April 9, 2002 to April 17, 2002. One hundred and thirty seven out of 214 were responded which made the response rate 64%. Data were analyzed by SAS and SPSS. Result : Multi-variable Logistic Regression Analysis showed a significant effect in medical expenses, knowledge of referral system and the information of the receiving hospital. The financial burden in medical expenses made the patient 10.7 times more willing to be transferred, knowledge of the referral system made them 5 times more willing to be transferred, and the information of receiving hospital makes 6.5 times more willing to be transferred. Reasons for willing to be transferred to a lower level of care were the phase of physical therapy, the distance from home, the attending physician's advice and being unable to be treated as an out patient. Reasons for refusing to be transferred were the following. The attending physician's competency, not being ready to be discharged, not trusting the receiving hospital's competency due to the lack of information, or never hearing about the referring system by the attending physician. Conclusion : Based on this, strategies for the effective transfer to the lower levels of care were suggested. It is desirable for the attending physician to be actively involved by making an effort to explain the transfer need, and referring to the Healthcare Coordinating Center, which can help the patient make the right decision. Nationwide networking for the referral system is the another key factor that may need to be suggested as an alternative to decrease the medical expenses. Collaborating with the Home Health Agency for the early discharge planning and the Social Service Department for financial aid are also needed. It is recommended that the hospital should expedite the transfer process by prioritizing the cost and the information as medical expenses, knowledge of referring system and the information of the receiving hospital, are the most important factors to the willingness to transfer to a lower level of care.
In the case of acquisition and displaying medical Images such as ultrasonography and endoscopy on VGA monitor of PC system, image degradation of tear-drop appears through scan conversion. In this study, we compare several methods which can solve this degradation and implement the hardware system that resolves this problem in real-time with PC. It is possible to represent high quality image display and real-time processing and acquisition with specific de-interlacing device and PCI bridge on our hardware system. Image quality is improved remarkably on our hardware system. It is implemented as PC-based system, so acquiring, saving images and describing text comment on those images and PACS networking can be easily implemented.metabolism. All images were spatially normalized to MNI standard PET template and smoothed with 16mm FWHM Gaussian kernel using SPM96. Mean count in cerebral region was normalized. The VOls for 34 cerebral regions were previously defined on the standard template and 17 different counts of mirrored regions to hemispheric midline were extracted from spatially normalized images. A three-layer feed-forward error back-propagation neural network classifier with 7 input nodes and 3 output nodes was used. The network was trained to interpret metabolic patterns and produce identical diagnoses with those of expert viewers. The performance of the neural network was optimized by testing with 5~40 nodes in hidden layer. Randomly selected 40 images from each group were used to train the network and the remainders were used to test the learned network. The optimized neural network gave a maximum agreement rate of 80.3% with expert viewers. It used 20 hidden nodes and was trained for 1508 epochs. Also, neural network gave agreement rates of 75~80% with 10 or 30 nodes in hidden layer. We conclude that artificial neural network performed as well as human experts and could be potentially useful as clinical decision support tool for the localization of epileptogenic zones.
The most important issue in providing multimedia traffic on a mobile computing environments is to guarantee the mobile host(client) with consistent QoS(Quality of Service). However, the QoS negotiated between the client and network in one cell may not be honored due to client mobility, causing hand-offs between cells. In this paper, a call admission control mechanism is proposed to provide consistent QoS guarantees for multimedia traffics in a mobile computing environment. Each cell can reserve fractional bandwidths for hand-off calls to its adjacent cells. It is important to determine the right amount of reserved bandwidth for hand-off calls because the blocking probability of new calls may increase if the amount of reserved bandwidth is more than necessary. An adaptive bandwidth reservation based on an MPP(Mobility Pattern Profile) and a 2-tier cell structure has been proposed to determine the amount of bandwidth to be reserved in the cell and to control dynamically its amount based on its network condition. We also propose a call admission control based on this bandwidth reservation and "next-cell prediction" scheme using an MPP. In order to evaluate the performance of our call admission control mechanism, we measure the metrics such as the blocking probability of our call admission control mechanism, we measure the metrics such as the blocking probability of new calls, dropping probability of hand-off calls, and bandwidth utilization. The simulation results show that the performance of our mechanism is superior to that of the existing mechanisms such as NR-CAT1, FR-CAT1, and AR-CAT1.
Multimedia applications are much more sensitive to QoS(Quality of Service) than text based ones due to their data continuity. In order to provide a fast moving MH(Mobil Host) using multimedia application with a consistent QoS,an efficient call admission mechanism is in need. This paper proposes the 2SCA(2-Step Call Admission) scheme based on cal admission scheme using pripority to guarantee the consistent QoS for mobile multimedia applications. A calls of MH are classified new calls, hand-off calls, and QoS upgrading calls. The 2SCA is composed of the basic call admission and advanced call admission; the former determines the call admission based on bandwidth available in each cell and the latter determines the call admission by applying DTT(Delay Tolerance Time), PQeueu(Priority Queue), and UpQueue(Upgrade Queue) algorithm according to the type of each call blocked at the basic call admission stage. In order to evaluate the performance of our mechanism, we measure the metrics such as the dropping probability of new calls, dropping probability of hand-off calls, and bandwidth utilization. The result shows that the performance of our mechanism is superior to that of existing mechanisms such as CSP(Complete Sharing Policy), GCP(Guard Channel Policy) and AGCP(Adaptive Guard Channel Policy).
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