• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.4 no.6
• /
• pp.1152-1168
• /
• 2010

Recently, due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services such as the mobile IPTV. In such a wireless broadband network, transmitting a continuous stream of multimedia data is difficult to achieve due to mobile stations (MSs) movement. Providing Quality of Service (QoS) for multimedia video streaming applications requires the server and/or client to be network-aware and adaptive. Therefore, in order to deploy a mobile IPTV service in wireless broadband networks, offering users efficient wireless resource utilization and seamlessly offering user perceived QoS are important issues. In this paper, we propose a new adaptive streaming scheme, called MARC (Mobile-aware Adaptive Rate Control), which adjusts the quality of bit-stream and transmission rate of video streaming based on the wireless channel status and network status. The proposed scheme can control the rate of multimedia streaming to be suitable for the wireless channel status by using awareness information of the wireless channel quality and the mobile station location. The proposed scheme can provide a seamless multimedia playback service in wireless broadband networks in addition to improving the QoS of multimedia streaming services. The proposed MARC scheme alleviates the discontinuity of multimedia playback and allocates a suitable client buffer to the wireless broadband network. The simulation results demonstrate the effectiveness of our proposed scheme.

• Journal of Digital Contents Society
• /
• v.7 no.4
• /
• pp.251-256
• /
• 2006

In this paper, We proposed joint-modulation CDMA system for efficient streaming transmission. and, We analyzed joint-modulation CDMA system using nakagami fading model and impulsive interference model for transmission channel environment. Also, it can being compensate the performance degradation by using MRC diversity scheme and BCH coding scheme.

• Journal of International Society for Simulation Surgery
• /
• v.4 no.1
• /
• pp.17-20
• /
• 2017

Purpose Interactive 3D visualization system through remote data transmission over heterogeneous network is growing due to the improvement of internet based real time streaming technology. Materials and Methods The current internet's IP layer has several weaknesses against IP spoofing or IP sniffing type of network attacks since it was developed for reliable packet exchange. In order to compensate the security issues with normal IP layer, we designed a remote medical visualization system, based on Virtual Private Network. Results Particularly in hospital, if there are many surgeons that need to receive the streaming information, too much load on the gateway can results in deficit of processing power and cause the delay. Conclusion End to end security through the network method would be required.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.2A
• /
• pp.146-156
• /
• 2002

This paper presents an adaptive H.263+ rate control algorithm for streaming video applications under the networks supporting bandwidth renegotiation, which can communicate with end-users to accommodate their time-varying bandwidth requests during the data transmission. That is, the requests of end-users can be supported adaptively according to the availability of the network resources, and thus the overall network utilization can be improved simultaneously. They are especially suitable for the transmission of non-stationary video traffics. The proposed rate control algorithm communicates with the network to renegotiate the required bandwidth fort the underlying video which are measured based on the motion change information, and choose their control strategies according to the renegotiation results. Unlike most conventional algorithms that control only the spatial quality by adjusting quantization parameters, the proposed algorithm treats both the spatial and temporal qualities at the same time to enhance human visual perceptual quality. Experimental results are provided to demonstrate that the proposed rate control algorithm can achieve superior performance to the conventional ones with low computational complexity under the networks supporting bandwidth renegotiation.

• KSII Transactions on Internet and Information Systems (TIIS)
• /
• v.9 no.3
• /
• pp.1035-1056
• /
• 2015

The efficiency of live Peer-to-Peer (P2P) streaming protocols depends on the appropriateness and the management abilities of their underlying overlay multicast. While a tree overlay structure confines transmission delays efficiently by maintaining deterministic delivery paths, an overlay mesh structure provides adequate resiliency to peers dynamics and easy maintenance. On the other hand, content freshness, playback fluidity and streaming continuity are still challenging issues that require viable solutions. In this paper, we propose a Hybrid Live P2P Streaming Protocol (HLPSP) based on a hybrid overlay multicast that integrates the efficiency of both the tree and mesh structures. Extensive simulations using OMNET++ are conducted to investigate the efficiency of HLPSP in terms of relevant performance metrics, and position HLPSP with respect to DenaCast the enhanced version of the well-known CoolStreaming protocol. Simulation results show that HLPSP outperforms DenaCast in terms of startup delay, end-to-end delay, play-back delay and data loss.

• Journal of Communications and Networks
• /
• v.13 no.3
• /
• pp.277-285
• /
• 2011

Real-time multimedia streaming over the Internet is rapidly increasing with the popularity of user-created contents, Web 2.0 trends, and P2P (peer-to-peer) delivery support. While many homes today are broadband-enabled, the quality of experience (QoE) of a user is still limited due to frequent interruption of media playout. The vulnerability of TCP (transmission control protocol), the popular transport-layer protocol for streaming in practice, to the packet losses, retransmissions, and timeouts makes it hard to deliver a timely and persistent flow of packets for online multimedia contents. This paper presents TCP-real-time online multimedia environment (ROME), a novel transport-layer framework that allows the establishment and coordination of multiple many-to-one TCP connections. Between one client with multiple home addresses and multiple co-located or distributed servers, TCP-ROME increases the total throughput by aggregating the resources of multiple TCP connections. It also overcomes the bandwidth fluctuations of network bottlenecks by dynamically coordinating the streams of contents from multiple servers and by adapting the streaming rate of all connections to match the bandwidth requirement of the target video.

• Journal of KIISE:Information Networking
• /
• v.37 no.5
• /
• pp.363-373
• /
• 2010

The existing QoS mechanisms for video streaming are short of the consideration for various user environments and the characteristic of streaming applying programs. In order to overwhelm this problem, studies on the video streaming protocols exploiting scalable video coding (SVC), which provide spatial, temporal, and qualitative scalability in video coding, are progressing actively. However, these protocols also have the problem to deepen network congestion situation, and to lower fairness between other traffics, as they are not equipped with congestion control mechanisms. SVC based streaming protocols also have the problem to overlook the property of videos encoded in SVC, as the protocols transmit the streaming simply by extracting the bitstream which has the maximum bit rate within available bandwidth of a network. To solve these problems, this study suggests TCP-friendly network adaptive SVC streaming(T-NASS) protocol which considers both network status and SVC bitstream property. T-NASS protocol extracts the optimal SVC bitstream by calculating TCP-friendly transmission rate, and by perceiving the network status on the basis of packet loss rate and explicit congestion notification(ECN). Through the performance estimation using an ns-2 network simulator, this study identified T-NASS protocol extracts the optimal bitstream as it uses TCP-friendly transmission property and perceives the network status, and also identified the video image quality transmitted through T-NASS protocol is improved.

• Journal of Internet Computing and Services
• /
• v.17 no.4
• /
• pp.19-26
• /
• 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:Computing Practices and Letters
• /
• v.8 no.3
• /
• pp.328-335
• /
• 2002

When we want to transfer video streaming data through computer networks, we will have to be allocated a larger bandwidth compared to a general application. In general, it causes a serious network overload inevitably due to the limited bandwidth. In this paper, in order to resolve the problem, we haute taken a method for transmitting video streaming data using RTP and RTCP. With RR(Receiver Report) packet in RTCP we will test it to check whether the traffic in a network has occurred or not. If it happened, we haute tried to reduce the overall network traffic by dynamically changing the quantization factor of the Motion JPEG that is one of the encoding styles in JMF. When the ratio of the average of transmission for each session to the average of overall transmission is greater than 7%, we should adjust the amount of data to be transmitted for each session to reach the session mean values. The experimental results show that the proposed method taken here reduces the overload effectively and therefore improves the efficiency for transmitting video streaming data.

• Journal of KIISE:Information Networking
• /
• v.37 no.6
• /
• pp.483-488
• /
• 2010

Due to the prevalence of various mobile devices and wireless broadband networks, there has been a significant increase in interest and demand for multimedia streaming services. Moreover, the user can service the participatory video broadcasting service in the mobile device and it can be used to deliver the real-time news and more variety information in the user side. Live multimedia service of user participation should consider not only the video quality but also the delay bounds and continuity of video playback for improving the user perceived QoS (Quality of Service) of streaming service. In this paper, we propose an adaptive rate control scheme, called DeBuG (Delay Bounds Guaranteed), to guarantee the delay bounds and continuity of video playback for the real-time streaming in mobile devices. In order to provide those, the proposed scheme has a quality adaptation function based on the transmission buffer status and network status awareness. It also has a selective frame dropper, which is based on the media priority, before the transmission video frames. The simulation results demonstrate the effectiveness of our proposed scheme.