• Journal of the Institute of Electronics Engineers of Korea TC
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• v.42 no.12
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• pp.17-24
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• 2005

Traditional Internet applications such as FIP and E-mail are increasingly sharing bandwidth with newer, more demanding applications such as Web browsing, IP telephony, video conference and online games. These new applications require Quality of Service (QoS), in terms of delay, loss and throughput that are different from QoS requirements of traditional applications. Unfortunately, current Active Queue Management (AQM) approaches offer monolithic best-effort service to all Internet applications regardless of the current QoS requirements. This paper proposes and evaluates a new AQM technique, called MCDT that provides dynamic and separated buffer threshold for each Applications, those are FTP and e-mail on TCP traffic, streaming services on tagged UDP traffic, and the other services on untagged UDP traffic. Using a new QoS metric, our simulations demonstrate that MCDT yields higher QoS in terms of the delay variation and a packet loss than RED when there are heavy UDP traffics that include streaming applications and data applications. MCDT fits the current best-effort Internet environment without high complexity.

• 한국정보통신설비학회:학술대회논문집
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• 2007.08a
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• pp.289-292
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• 2007

FTTH (Fiber to the Home) is constructed with optical devices and optical cables from central office (CO) to subscribers. With FTTH, it is possible to provide service platform in order to meet the customers' emerging demands for more bandwidth/security and diverse IP multimedia applications including high definition (HD) video streaming service. FTTH can provide enough bandwidth from 100Mbps to several Gbps per subscriber. It is also most appropriate network architecture for convergence of communication and broadcasting. So, FTTH is thought the final destination of telecom operators, since it is the fastest and most reliable solution so far and various IP multimedia services are possible through it. In this paper, newly attempted technologies of FTTH construction for apartment complexes using CWDM-PON system are discussed.

• Journal of the Korea Society of Computer and Information
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• v.21 no.7
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• pp.17-22
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• 2016

In this paper we propose a novel QoE control scheme based on Service Level Agreement (SLA) which enables to guarantee the user perceived service quality of various high-quality mobile multimedia services such as mobile video streaming and real-time network games to a certain level. For this purpose, a SLA control structure is adopted to provision QoE, and a novel resource management scheme gets coupled with this control structure. Simulation results show that our proposed scheme can decrease the average delay and increase the total throughput by adjusting SLA.

• International Journal of Contents
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• v.8 no.1
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• pp.47-54
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• 2012

The dynamic adaptation between various service environments using the application profiles for the service continuity is a key issue of the profile-based service continuity framework (PSCF) for N-screen service using next generation networks. PSCF offers an optimized service framework for providing continuous user services, which are multimedia video streaming, educational broadcasting, game, etc., using the various devices that are not restricted by the service environment of the user. This paper specifies the functional model of PSCF, service scenario and explains the experimental results of the service continuity for N-screen service using PSCF.

• The Transactions of The Korean Institute of Electrical Engineers
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• v.66 no.3
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• pp.575-580
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• 2017

In these days, the big-size and high resolution multimedia file is widely used through networks. To transfer and service effectively, the internet network technology is necessary to substitute broadcasting. Normally Content Delivery Network(CDN) is widely used in conventional internet for multimedia services. But it has a small bandwidth to service. So to solve this problems, many researchers have suggest the protocol for download, content distribution/saving, server synchronization, caching, pushing rate, and streaming etc. But all of these has some defects like low resolution, packets loss and delay, real application implementations etc. So, this paper suggests a new method of network protocol based on P2P collaboration and user's content using information. And it evaluated the performance of suggested method. As the results, it showed the effectiveness of 4 performances indices : download speed, decreasing rate of connected user in same time, adaptive hit ratio, traffic decreasing rate.

• Journal of Advanced Navigation Technology
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• v.13 no.5
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• pp.669-676
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• 2009

HTTP or audio/video streaming services are good candidates for future centralized navigation system and in order to provide stability for such services, service providers use a cluster of web servers. In this paper, we provide the criteria for web server cluster design of vehicular users with consideration of differentiated access per different user classes. Several feasible scenarios are examined and their performance analysis using queueing theory is presented to provide the foundation for web server cluster design using traffic load balancer. Through the thorough analysis, efficient criteria for traffic load balancer design is derived. In order to satisfy users' service requirements, priority services controlled by traffic load balancer are considered and analyzed. We also provide the evaluation of the accuracy of the analytical model through simulation.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.31 no.7B
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• pp.608-618
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• 2006

The increasing popularity of multimedia streaming services introduces new challenges in content distribution. Especially, it is important to provide the QoS guarantees as they are increasingly expected to support the multimedia applications. The service providers can improve the performance of multimedia streaming by caching the initial segment (prefix) of the popular streams at proxies near the requesting clients. The proxy can initiate transmission to the client while requesting the remainder of the stream from the server. In this paper, in order to apply the prefix caching service based on IETF's RTSP environment to the wireless networks, we propose the effective RTSP handling scheme that can adapt to the radio situation in wireless network and reduce the cutting phenomenon. Also, we propose the traffic based caching algorithm (TSLRU) to improve the performance of caching proxy. TSLRU classifies the traffic into three types, and improve the performance of caching proxy by reflecting the several elements such as traffic types, recency, frequency, object size when performing the replacement decision. In simulation, TSLRU and RTSP handling scheme performs better than the existing schemes in terms of byte hit rate, hit rate, startup latency, and throughput.

• The Journal of the Korea Contents Association
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• v.12 no.4
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• pp.10-19
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• 2012

Seamless streaming of multimedia content that ensures Quality of Service over heterogeneous networks has been a desire for many multimedia services, for which the multimedia contents should be adapted to usage environments such as network characteristics, terminal capabilities, and user preferences. Scalability in video coding is a good feature to meet the requirement of heterogeneous networks. In this paper, we propose a dynamic adaptation scheme of H.264/AVC SVC bit-stream using the MPEG-21 Digital Item Adaptation (DIA) tool. MPEG-21 DIA framework provides systematic solutions in choosing an adaptation operation to given conditions and supports interoperable video adaptation. The experiment results show that the proposed adaptation scheme provides QoS-enabled delivery and consumption of SVC with time-varying constraints of network, terminal, and user preference, in a robust and efficient way. In particular, the proposed adaptation scheme is proved to work well with very low delay under the condition that the variation rate of the given network bandwidth is upto 62%.

• KIPS Transactions on Computer and Communication Systems
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• v.1 no.1
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• pp.29-34
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• 2012

Wireless multicast service can be used for video streaming service to save the network resources by sending the same popular multimedia contents to a group of users at once. For better multimedia streaming multicast service, we propose a quality-aware rate adaptation (QARA) scheme for scalable video multicast in rate adaptive wireless networks. In QARA, transmission rate is determined depending on the content's type and users' channel conditions. First, the base layer is transmitted by a low rate for high reliability. That means we provide basic service quality to all users. On the contrary, the transmission rate for enhancement layer is adapted by using channel condition feedback from a randomly selected node. So, the enhancement layer frames in a multimedia content is sent with various transmission rates. Therefore, each node can be provided with differentiated quality services. Consequently, QARA is capable of serving heterogeneous population of mobile nodes. Moreover, it can utilize network resources more efficiently. Our simulation results show that QARA outperforms utilization of the available transmission rate and reduces the data transmission time.

• Journal of KIISE:Computing Practices and Letters
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• v.10 no.6
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• pp.460-471
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• 2004

Since the RTP is suitable for real-time data transmission in multimedia services like VoD, AoD, and VoIP, it has been adopted as a real-time transport protocol by RTSP, H.323, and SIP. Even though the RTP protocol stack for embedded systems has been in great need for efficient support of multimedia services, such a stack has not been developed yet. In this paper, we explain embeddedRTP which supports the RTP protocol stack at the kernel level so that it is suitable for embedded systems. Since embeddedRTP is designed to reside in the UBP module, existing applications which rely ell TCP/IP services can proceed the same as before, while applications which rely on the RTP protocol stack can request HTP services through embeddedRTp API. EmbeddedRTP stores transmitted RTP packets into per session packet buffer, using the packet's port number and multimedia session information. Communications between applications and embeddedRTP is performed through system calls and signal mechanisms. Additionally, embeddedRTP API makes it possible to develop applications more conveniently. Our performance test shows that packet-processing speed of embeddedRTP is about 7.5 times faster than that oi VCL RTP for multimedia streaming services on PDA in spite that its object code size is reduced about by 58% with respect to UCL RTP's.