• Journal of Internet Computing and Services
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• v.17 no.6
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• pp.9-15
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• 2016

In this paper, we propose a smart HLS(HTTP Live Stream) platform with dynamic single buffering for the best transmission of adaptive video bit-rates. This smart HLS can optimizes the channel transition zapping-time with the monitoring of bandwidth between HLS server and OTT(Over The Top) client. This platform is designed through the control of video stream due to proper multi-bitrates and bandwidths. This proposed OTT can decode the live and VOD(Video On Demand) videos with the buffering of optimumal bitrate. And, the HLS can be cooperated with a smart OTT, and segmented for the m3u8 files of H.265 MPEG-2 TS(Transport Stream) videos. As a resullt, this single buffer based smart OTT can transmit optimal videos with the maximum data buffering according to the adaptive bit-rate depending on the network bandwidth efficiency and the decoded VOD video, too.

• Journal of the Korea Society of Computer and Information
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• v.7 no.4
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• pp.188-198
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• 2002

In this paper, we propose an active network to support reliable data transmission in the mobile ad-hoc network. The active network uses DSR(Dynamic Source Routing) protocol as its basic routing protocol, and uses source and destination nodes as key active nodes. For reliable improvement the source node is changed to source active node to add function that its buffer to store the last data with the flow control for data transmission per destination node. The destination node is changed to destination active node to add function that it requests the re-transmission for data that was not previously received by the destination active node with the flow control for data reception per source active node As the result of evaluation. we found the proposed active network guaranteed reliable data transmission with almost 100% data reception rate for slowly moving mobile ad-hoc network and with more 95% data reception rate, which is improvement of 3.5737% reception rate compared with none active network, for continuously fast moving mobile ad-hoc network.

• The Transactions of the Korea Information Processing Society
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• v.4 no.7
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• pp.1770-1780
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• 1997

Our problem is improvement of image quality because it is inevitable cell loss of image data when traffic congestion occurs. If cells are discarded indiscriminately in transmission of MPEG video data, it occurs severe degradation in quality of service(QOS). In this paper, to solve this problem, we propose two method. The first, we analyze the traffic characteristics of an MPEG encoder and generate high priority and low priority data stream. During network congestion, only the least low priority cells are dropped, and this ensures that the high priority cells are successfully transmitted, which, in turn, guarantees satisfactory QoS. In this case, the prioritization scheme for the encoder assigns components of the data stream to each priority level based on the value of a parameter ${\beta}$. The second, Number of high priority cells are increased when value of ${\beta}$ is large. It occurs the loss of high priority cell in the congestion. To prevent it, this paper is regulated to data stream rate as buffer occupancy with UPC controller. Therefore, encoder's bandwidth can be calculated renegotiation of the encoder and networks. In this paper, the encoder's bandwidth requirements are characterized by a usage parameter control (UPC) set consisting of peak rate, burstness, and sustained rate. An adaptive encoder rate control algorithm at the Networks Interface Card(NIC) computes the necessary UPC parameter to maintain the user specified quality of service. Simulation results are given for a rate-controlled VBR video encoder operating through an ATM network interface which supports dynamic UPC. These results show that dynamic bandwidth renegotiation of prioritized data stream could provided bandwidth saving and significant quality gains which guarantee high priority data stream.

• Journal of Internet Computing and Services
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• v.17 no.4
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• pp.19-26
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• 2016

In this paper, the dynamic buffering based smart OTT platform was proposed, and analyzed for adaptive bit-rate video delivery with the optimization of HLS (HTTP Live Streaming). This platform consists of the software platform between sever and client which detects the bandwidth capacity, and adjusts the quality of the streaming for multiple bit-rates resolutions. In order to apply adaptive buffering, two buffers are added to the basic HLS player, and each buffer is responsible for constantly buffering a previous and the next channels relative to the current channel. This adaptive transmitting with smart OTT platform is superior to delivering a static video file at a single buffering, because the video stream of adaptive double buffers can be switched streaming according to client's available network speed. As a result, this proposed smart OTT can be cooperated to the application of HLS server with segmented H.265 MPEG-2 TS video & m3u8 files with its information based on the optimized transmission channel state of live and VOD, and applied to PLC transmission, too.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.8
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• pp.930-938
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• 1999

미래의 통합 이동 통신망은 멀티미디어 데이터 서비스를 하기 위해 작은 셀로 망이 구성될 것이다. 셀이 작은 경우, 이동 호스트의 셀간 이동이 상대적으로 증가해서 잦은 핸드오프 처리로 서비스 중단이 빈번히 발생할 수 있을 뿐만 아니라 자원의 낭비도 많아지게 된다. 따라서 이런 문제를 해결하기 위해 이동 호스트의 이동성을 예측해서 멀티캐스트 연결을 구성해서 사전 핸드오프 처리를 하는 SGMH방법을 제안한다. SGMH방법의 성능 평가를 위해 핸드오프 처리과정의 데이터 및 제어 정보의 흐름을 분석하고, 망 종속 인자들을 도입해서 버퍼 오버헤드와 서비스 중단 시간 면에서 기존 핸드오프 방법들과 비교했다. 비교 분석한 결과는 제안한 방법이 기존의 방법보다 기지국에서의 버퍼 오버헤드뿐만 아니라 서비스 중단 시간을 줄일 수 있음을 보였다.Abstract For mobile multimedia data services, future mobile communication networks will consist of small cells. In case of smaller cell size, the number of user movements between cells will increase and more network resources will be consumed for processing frequent handoffs. This processing delay will cause frequent connection service disruptions. To solve these problems, a multicast-based handoff method ,called SGMH(Sub-Group Multicast-based Handoff), is suggested. SGMH method estimates future movements of mobile hosts and setups a dynamic multicast connection tree for procesing handoffs. For performance evaluation, we represent flows of data and control messages as timing diagrams and introduce several network related factors. In terms of buffer overhead and service disruption time we compared the SGMH method to other methods. The results show that SGMH method has better performance in that it can reduce consumption of network resources and minimize service disruption time.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.27 no.4C
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• pp.295-307
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• 2002

The IEEE 1394 protocol is a de facto standard in multimedia digital home network. It supports several advanced features such as hot plugging, dynamic network reconfiguration, isochronous transmission and so on. Since the IEEE 1394 was adapted in the field of multimedia service with QoS guarantee, back bone network protocol with reel-time digital instrumentation and control sub networks, and physical layer protocol for real-time middleware such as real-time CORBA, the additional real-time features has been required in device driver implementation because of the necessity of the predictability enhancement. To guarantee the real-time features, the device driver of the IEEE 1394 should support the priority based packet processing and also need a isochronous buffer management mechanism to deal with the periodic isochronous communication. In this paper, we proposed a new software architecture of the IEEE 1394 device driver for supporting the real-time characteristics such as priority based packet processing, priority based scheduling and so on.

• Journal of KIISE:Information Networking
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• v.35 no.5
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• pp.409-414
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• 2008

In mobile ad hoc networks, most of on demand routing protocols such as DSR and AODV do not deal with traffic load during the route discovery procedure. To solve the congestion and achieve load balancing, many protocols have been proposed. However, the existing load balancing schemes has only considered avoiding the congested route in the route discovery procedure or finding an alternative route path during a communication session. To mitigate this problem, we have proposed a new scheme which considers the packet bursting mechanism in congested nodes. The proposed packet bursting scheme, which is originally introduced in IEEE 802.11e QoS specification, is to transmit multiple packets right after channel acquisition. Thus, congested nodes can forward buffered packets promptly and minimize bottleneck situation. Each node begins to transmit packets in normal mode whenever its congested status is dissolved. We also propose two threshold values to define exact overloaded status adaptively; one is interface queue length and the other is buffer occupancy time. Through an experimental simulation study, we have compared and contrasted our protocol with normal on demand routing protocols and showed that the proposed scheme is more efficient and effective especially when network traffic is heavily loaded.