• Journal of Broadcast Engineering
• /
• v.10 no.4 s.29
• /
• pp.548-556
• /
• 2005

In wireless network environments, wireless channels are characterized by time-varying fading and interference conditions, which may lead to burst packet corruptions and delay variation. This can cause severe quality degradation of streaming media. To guarantee successful transmission of media over the hostile wireless networks, where channel conditions are highly fluctuating, a flexible and network-adaptive transport method is required. Thus, we propose a network-adaptive transport error control consisting of packet-level interleaved FEC and delay-constrained ARQ, which acts as an application-level transport method of streaming media to alleviate burst packet losses while adapting to the changing channel condition in wireless networks. The performances of the proposed network-adaptive transport error control, general error control schemes, and hybrid schemes are evaluated by a developed simulator at the transport-level and video quality of streaming media. Simulation results show that the proposed mechanism provides the best overall performance among compared other schemes in terms of the transport-level performance of error control and the performance of video quality for streaming media.

• Journal of the Korea Institute of Information and Communication Engineering
• /
• v.17 no.7
• /
• pp.1618-1626
• /
• 2013

Virtualization service processes all operation including the data creation, storing, and disposal in a server and transmits processed data as the streaming media form. Therefore, client can use the same environment as the traditional desktop environment without considering the type of device. Virtualization service should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived Quality of Service(QoS) of streaming service. In this paper, we propose a network-adaptive transmission architecture that focuses on guaranteeing QoS requirements for virtualization service. In order to provide those, the proposed architecture have the transmission rate adaptation function based on available bandwidth and the content bit-rate control function based on sender buffer state. Through each function, proposed architecture guarantee the delay bounds and continuity of virtualization contents playback. The simulation results show that proposed network-adaptive transmission architecture provides a improve performance of throughput and transmission delay.

• IEMEK Journal of Embedded Systems and Applications
• /
• v.10 no.6
• /
• pp.353-362
• /
• 2015

In order to provide smart devices with high quality multimedia streaming services, an adaptive streaming technique over HTTP has been received much attention recently and the Dynamic Adaptive Streaming over HTTP (DASH) standard has been established. In DASH, however, the technique to select an appropriate quality of multimedia based on the performance metrics measured in a smart device might have some difficulties to reflect the capabilities of other neighboring smart devices and dynamic network conditions in real time. To solve the problem, this paper proposes a novel software agent approach, called DASH agent (DA), which gathers and analyzes the device capabilities and dynamic network conditions in real time and finally determines the highest achievable quality of segment to meet the best Quality of Experience (QoE) in current situations. The simulation results show that our approach provides higher quality of multimedia segments with less frequency of quality changes to lower quality of multimedia segments.

• Journal of the Institute of Electronics Engineers of Korea SD
• /
• v.43 no.2 s.344
• /
• pp.60-75
• /
• 2006

The MPEG-21 Multimedia Framework initiative aims to support a wide range of networks and devices in the delivery and consumption of multimedia resources. One of the primary goals of MPEG-21 is universal multimedia access (UMA) through Digital Item Adaptation (DIA), which supports multimedia streaming to heterogeneous devices ensurung the same readability and seamlessness. We pioneer power saving of luminal devices with MPEG-21 DIA, so that the MPEG-21 DIA can also be used to support power saving, even though the framework is not primarily designed for power reduction and only limited power awareness is defined by DIA. We introduce several power-saving techniques conforming to MPEG-21 DIA specifications and show the dependency relation among introduced techniques. We achieve energy savings of up to $66\%$ in hand-held multimedia devices with minor QoS (quality of service) degradation.

• Journal of KIISE:Computing Practices and Letters
• /
• v.16 no.12
• /
• pp.1141-1153
• /
• 2010

This paper introduces ONLIS(Overlay Multicast Network for Live IPTV Service), a novel overlay multicast network optimized to deliver live broadcast IPTV stream. We analyzed IPTV reference model of ITU-T IPTV standardization group in terms of network and stream delivery from the source networks to the customer networks. Based on the analysis, we divide IPTV reference model into 3 networks; source network, core network and access network, ION(Infrastructure-based Overlay Multicast Network) is employed for the source and core networks and PON(P2P-based Overlay Multicast Network) is applied to the access networks. ION provides an efficient, reliable and stable stream distribution with very negligible delay while PON provides bandwidth efficient and cost effective streaming with a little tolerable delay. The most important challenge in live P2P streaming is to reduce end-to-end delay without sacrificing stream quality. Actually, there is always a trade-off between delay & stream quality in conventional live P2P streaming system. To solve this problem, we propose two approaches. Firstly, we propose DSPT(Distributed Streaming P2P Tree) which takes advantage of combinational overlay multicasting. In DSPT, a peer doesn't fully rely on SP(Supplying Peer) to get the live stream, but it cooperates with its local ANR(Access Network Relay) to reduce delay and improve stream quality. When RP detects bandwidth drop in SP, it immediately switches the connection from SP to ANR and continues to receive stream without any packet loss. DSPT uses distributed P2P streaming technique to let the peer share the stream to the extent of its available bandwidth. This means, if RP can't receive the whole stream from SP due to lack of SP's uploading bandwidth, then it receives only partial stream from SP and the rest from the ANR. The proposed distributed P2P streaming improves P2P networking efficiency.

• Journal of the Institute of Electronics Engineers of Korea SP
• /
• v.39 no.6
• /
• pp.672-686
• /
• 2002

현재 단대단(End-to-End) 비디오 서비스의 질을 높이기 위해 많은 연구가 각 계층별로 진행되고 있다. 전송계층에서의 오류 제어(Error Control), 네트워크계층에서의 QoS (Quality of Service)모델, 표현 및 응용계층에서의 오류 강인성(Error resilience)/오류 은닉(Error concealment) 등이 연구 개발되고 있다. 그러나 계층 간의 연관성이 높은 부분에서의 통합을 통한 성능향상에 관한 연구는 그 필요성과 효율에 비해 아직도 미흡하다. 본 논문은 QoS 서비스 모델하에서의 적응적 FEC(Forward Error Correction) 적용 및 우선순위에 따른 비디오패킷(VP ,VideoPacket)을 통하여 효율적인 계층화 비디오 스트리밍을 단대단 QoS성능의 향상에 관점을 맞추어 제안한다. 제안하는 방식은 최소 화질 보장과 같은 효율에서 보다 적은 가격에서의 서비스를 제공할 수 있다. 이를 위하여 통합형 서비스(IntServ, IS, Integrated Service) 의 자원예약을 사용하는 방법과 높은 가격의 자원 예약을 사용하지 않는 차별화 서비스(DiffServ, DS, Differentiated Service)를 적용했으며 이에 보장형 서비스의 특징을 공통을 가지기 위해 계층화 FEC를 적용하였으며 적절한 가격의 조절을 위하여 비디오패킷을 통한 데이터 분할을 적용하였다. 본 논문은 또한 최종 사용자의 만족도를 PSNR(Picture Signal to Noise Ration)과 PSNR에서 표현하지 못하는 부분의 평가를 위해 손상프레임율(DFR, Damaged Frame Ratio)과 오류프레임율(EFR,Error Frame Ratio)을 제안 이를 통해 평가하고자 한다. 제안하는 방식의 실험 결과는 비디오 코딩계층과 전송 계층, 네트워크 계층의 결합된 성능이며 이는 또한 화질의 개선뿐만 아니라 사용자의 가격문제에 대하여서도 비교 분석하였다.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.32 no.11A
• /
• pp.1106-1112
• /
• 2007

As a demand for high speed uplink packet services increases, the WCDMA enhanced uplink, also known as high speed uplink packet access (HSUPA), has been specified in release 6 by 3GPP. This HSUPA will provide various types of multimedia services, such as real-time video streaming, gaming, VoIP, and FTP. Generally, the performance of HSUPA is dominated by scheduling policy. Therefore, it is required to design a scheduling algorithm considering the traffic characteristics to provide QoS guaranteed services in various traffic environments. In this paper, we propose a scheduling algorithm considering the traffic characteristics to guarantee QoS in a mixed traffic environment. Finally, the performance of the proposed scheduling algorithm is evaluated in terms of average packet delay, packet delay jitter, and system throughput using a system level simulator.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.31 no.8C
• /
• pp.777-785
• /
• 2006

H.264 is the prominent video coding standard in various applications such as real-time video streaming and digital multimedia broadcasting, since it provides enhanced compression performance, error resilience tools, and network adaptation. Since compressed video stream is vulnerable to packet loss, error resilience and error concealment(EC) tools are essential for the transmission of video over the Internet. In this paper, we first propose a simple temporal EC method that improves the EC performance for intra-frame in H.264 when the amount of motion is relatively small. Then we propose a new hybrid EC method for intra-frame in H.264, which combines the spatial EC and temporal EC adaptively. The simulations are performed in packet-lossy environments, and the proposed hybrid EC method shows about 0.5-4dB PSNR improvement compared to the conventional spatial EC method that is used for intra-frame in H.264.

• Journal of Broadcast Engineering
• /
• v.24 no.6
• /
• pp.948-955
• /
• 2019

As interests in 360-degree VR (Virtual Reality) video services enormously grow, compression and streaming technologies for VR video data have been rapidly developed. Quality Emphasized Region (QER) based streaming scheme has been developed as a kind of viewport-adaptive 360-degree video streaming technology for maintaining immersive experience and reducing bandwidth waste. For selecting a QER corresponding to the user's gaze coordinate, QER-based streaming scheme requires the calculation of Quality Emphasis Center (QEC) distance and signaling message delivery for requesting QER switching. QEC distance calculations require high computational complexity because of repeated calculations as many times as the number of QERs. Furthermore, the signaling message interval results in a trade-off relationship between efficient bandwidth usage and flexible QER switching. In this paper, we propose an improved QER selection algorithm based on MMT protocol to solve this problem. The proposed algorithm could achieve computational complexity reduction by using preprocessed QER_ID_MAP. Also, flexible QER switching could be achieved, as well as efficient bandwidth utilization by an adaptive adjustment of the signaling interval.

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