• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4C
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• pp.353-364
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• 2002

To prevent the performance degradation of TCP due to packet loss in the smooth handoff by the route optimization extension of Mobile If protocol, a buffering of packets at a base station is needed. A buffering of packets at a base station recovers the packets dropped during the handoff by forwarding the buffered packets at the old base station to the mobile user. But, when the mobile user moves to a new foreign network which is connected to a congested router, the buffered packets forwarded by the old base station are dropped and the link utilization performance degraded due to increased congestion by the forwarded packets. In this paper, when the mobile user moves to a new foreign network which is connected to a congested router, Ive propose a packet forwarding control scheme required far the old base station to improve TCP performance in mobile networks. The old base station forwards or discards the buffered packets during handoff by proposed packet forwarding control scheme based on congestion states of RED(Random Early Detection) at the congested router. Simulation results slow that link utilization performance can be improved by applying proposed packet forwarding control scheme.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.32 no.4B
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• pp.227-233
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• 2007

According to the recent development of network technology, the demands of users are diversified and the needs of multimedia traffic are increasing. In general, UDP(User Datagram Protocol) traffic is used to transport multimedia data, which satisfied the real-time and isochronous characteristics. UDP traffic competes with TCP traffic and incur the network congestion. However, TCP traffic performs network congestion control but does not consider the receiver's status. Thus, it is not appropriate in case of streaming services. In this paper, we solve a fairness problems and proposed a network algorithm based on RTP/RTCP(Real-time Transport Protocol/Realtime Transport Control Protocol) in view of receiver status. The POBA(Priority Ordered Buffer Algorithm), which applies priorities in the receiver's buffer and networks, shows that it provides the appropriate environment for streaming services in view of packet loss ratio and buffer utilization of receiver's buffer compared with the previous method.

• The Transactions of the Korea Information Processing Society
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• v.7 no.4
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• pp.1211-1216
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• 2000

Asynchronous transfer mode (ATM) can be efficiently used to transport packet data services. The switching system will support voice and packet data services simultaneously from end to end applications. To guarantee quality of service (QoS) of the offered services, source rateot send packet data is needed to control the network overload condition. Most existing control algorithms are shown to provide the threshold-based feedback control technique. However, real-time voice calls can be dynamically connected and released during data services in the network. If the feedback control information delays, quality of the serviced voice can be degraded due to a time delay between source and destination in the high speed link. An adaptive algorithm based on the optimal least mean square error technique is presented for the predictive feedback control technique. The algorithm attempts to predict a future buffer size from weight (slope) adaptation of unknown functions, which are used fro feedback control. Simulation results are presented, which show the effectiveness of the algorithm.

• Proceedings of the Korean Information Science Society Conference
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• 2012.06d
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• pp.109-111
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• 2012

본 논문에서는 동적으로 현재화면의 최적밝기 정보를 구해 이 값을 이용하여 LCD 장치를 제어하고, 이를 통해 소모전력을 감소시키는 기법을 제안한다. 논문에서 제안된 기법은 프레임버퍼 이미지를 통해 얻은 현재 화면의 이미지 정보를 분석하여 계산된 현재 화면의 최적의 휘도 정보를 찾고 이 값에 맞는 LCD 장치의 최적 Backlight를 계산하는 과정을 거쳐 얻어진 최종 Backlight 레벨로 LCD의 Backlight를 제어하는 방법으로 소모 전력을 절감하는 기법이다. 이 기법을 적용하여 실험을 통해서 약 10%정도의 개선효과를 보였음을 밝힌다.

• Proceedings of the Korean Information Science Society Conference
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• 1999.10c
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• pp.635-637
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• 1999

기존의 인터넷 기반 데이터 서비스는 'Best-effort' 정책으로 멀티미디어 서비스를 위한 트래픽 제어(traffic control) 기능을 제공하지 못한다. 그러므로 실시간 데이터 전송을 기반으로 제공되는 VOD(Video-on-Demand) 서비스는 사용자가 원하는 QoS(Quality of Service)를 제공할 수 없다. 따라서 인터넷 상에서 트래픽 제어 기능을 갖춘 중개자를 두어 VOD 서비스를 제공하는 방법에 대해 제안한다. 중개자의 주요 기능은 비디오 서버와 클라이언트 사이에 비디오 데이터의 버퍼링 기능을 이용하여 실시간 트래픽 제어를 수행한다. 이런 기능을 가진 중개자를 에이전트라 부르고, 에이전트에 대한 설계 및 구현에 대해 기술한다.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.527-539
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• 2001

The wireless asynchronous transfer mode (ATM) system has the advantage of providing the broadband services with various quality-of-service requirements to the mobile terminal efficiently by utilizing the ATM technology developed for the wired ATM system. The available bit rate (ABR) service among various ATM services utilizes the available bandwidth remaining in the ATM link, which allows the efficient bandwidth usage. During the handoff of the mobile terminal, however, the queue length in the access point (AP) which resides in the boundary of the wired ATM network and the wireless ATM network may increase abruptly. In this paper, we propose a scheme which prevents the buffer-overflow and buffer-underflow in the AP during the handoff of the wireless ABR connection in the wireless ATM system using binary feedback rate-based ABR traffic control. This scheme controls the source's cell generation rate during both handoff period and some time interval after the completion of the handoff procedure. The simulation results show that the proposed scheme prevents the buffer-overflow and buffer-underflow. The proposed scheme can contribute to increasing the throughput of the wireless ABR service during handoff by preventing the buffer overflow and underflow during handoff period.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.44 no.8
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• pp.51-61
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• 2007

For provisioning reliable data transmission, the 3GPP RLC specification adopted a selective-repeat ARQ scheme assisted by a window-based admission control scheme. In the 3GPP ARQ, which is a member of the selective-repeat ARQ clan, inheres the reordering problem A long and irregular reordering time results in the degradation of throughput and delay performance, and may incur the overflow of the reordering buffer. Furthermore, the reordering time must be regulated to meet the requirements of some services which are loss-sensitive and delay-sensitive as well. Perceiving the reordering hazard, we propose an alternative, identified as snowball scheme, to the original admission control scheme of the 3GPP ARQ with aiming at deflating the occupancy of the reordering buffer. A unique feature of the snowball scheme is to reject a new DATA PDU if it is non-adjacent to any DATA PDU sojourning at the reordering buffer. Such an intentional rejection apparently reduces the occupancy of the reordering buffer while it may deteriorate the throughput and delay performance. Developing an analytical approximation method, we investigate the effect of snowball scheme on the saturated occupancy and throughput. Also, we, using a simulation method, evaluate the peak occupancy, normalized throughput and average delay in the practical environment. From the simulation results, we reveal that the snowball scheme is able to enhance occupancy performance as well as throughput performance compared with the original admission control scheme of the 3GPP ARQ.

• Journal of the Institute of Electronics Engineers of Korea SD
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• v.44 no.6 s.360
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• pp.77-82
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• 2007

In the existing frame-layer rate control for H.264, buffer status and content complexity are used improperly, causing quality fluctuations of high-motion video at low bit rates and high frame rates (under 19.2 kbps at 30 fps). We propose an enhanced H.264 frame-layer rate control scheme to obtain steady video quality and stable buffer management. Experimental results showed that the proposed scheme performed better than existing schemes.

• The KIPS Transactions:PartC
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• v.12C no.6 s.102
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• pp.793-798
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• 2005

We focus on the RP (Rendezvous Point) system model in the multicast network based on BcN (Broadband Convergence Network) integrating broadcasting, telecommunication and Internet with one. Based on the condition multiple queues with different service and single server, when the arrivals tome in group with the site of the group geometrically distributed, we define the relationship between incoming arrival rate and corresponding buffer size. We also investigate the Profit according to both Service Provider and Network Operator Then we make a decision whether a new service request is accepted or not based on given interning rate range.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.40 no.7
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• pp.1330-1338
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• 2015

In wireless communication network, the amount of packet data for the video streaming in the playout buffer of the receiver is changed with time according to network condition. If the amount of packet data is less than a specific buffer amount, the buffer underflow problem is generated. On the contrary, if the amount of packet data is more than a given buffer amount, the buffer overflow problem is occurred. When the playout of the video streaming is processed, these buffer underflow and overflow problems cause stop and skip phenomenons and then provide the discontinuity of playout. Therefore, to solve the buffer underflow and overflow problems of the video streaming in wireless communication network, This paper analyzes the combined effect of Token Bucket scheme, which controls the bursty traffic, and AMP(Adaptive Media Playout) scheme, which adaptively changes the playout speed of receiver. Through simulation, we found that the combination of Token Bucket and AMP schemes provides the superiority in terms of the occurrence number of buffer underflow and overflow, the stop duration time and the number of removed frames generated by underflow and overflow, and PSNR.