• The Transactions of the Korean Institute of Electrical Engineers P
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• v.65 no.4
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• pp.326-334
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• 2016

In this paper for developing and implementing the realtime HD level video streaming transmitter which is operated on the multi-platform in all network and client environment compared to the exist video live streaming transmitter. We design the realtime HD level video streaming transmitter supporting the multi-platform using the TMS320DM386 video processor of T.I company and then porting the Linux kernel 2.6.29 and implementing the RTSP(Real Time Streaming Protocol)/RTP(Real Time Transport Protocol), HLS(Http Live Streaming), RTMP(Real Time Messaging Protocol) that can support the multi-platform of video stream protocol of the received equipments (smart phone, tablet PC, notebook etc.). For proving the performance of developed video streaming transmitter, we make the testing environment for testing the performance of streaming transmitter using the notebook, iPad, android Phone, and then analysis the received video in the client displayer. In this paper, we suggest the developed the Realtime HD(High Definition) level Video Streaming transmitter performance data values higher than the exist products.

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

• KIPS Transactions on Computer and Communication Systems
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• v.10 no.5
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• pp.137-144
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• 2021

The IETF has recently been standardizing the QUIC protocol for HTTP/3 services. It is noted that HTTP/3 uses QUIC as the underlying protocol, whereas HTTP/1.1 and HTTP/2 are based on TCP. Differently from TCP, the QUIC uses 0-RTT or 1-RTT transmissions to reduce the connection establishment delays of TCP and SCTP. Moreover, to solve the head-of-line blocking problem, QUIC uses the multi-streaming feature. In addition, QUIC provides various features, including the connection migration, and it is available at the Chrome browser. In this paper, we analyze the performance of QUIC for HTTP-based web and streaming services by comparing with the existing TCP and Streaming Control Transmission Protocol (SCTP) in the network environments with different link delays and packet error rates. From the experimental results, we can see that QUIC provides better throughputs than TCP and SCTP, and the gaps of performances get larger, as the link delays and packet error rates increase.

• ETRI Journal
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• v.28 no.4
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• pp.411-424
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• 2006

This paper presents a multimedia streaming platform for efficiently transmitting MPEG-4 content over IP networks. The platform includes an MPEG-4 compliant streaming server and client, supporting object-based representation of multimedia scenes, interactivity, and advanced encoding profiles defined by the ISO standard. For scalability purposes, we employ an application-layer multicast scheme for media transmission using overlay networks. The overlay network, governed by the central entity of the network distribution manager, is dynamically deployed according to a set of pre-defined criteria. The overlay network supports both broadcast delivery and video-on-demand content. The multimedia streaming platform is standards-compliant and utilizes widespread multimedia protocols such as MPEG-4, real-time transport protocol, real-time transport control protocol, and real-time streaming protocol. The design of the overlay network was architected with the goal of transparency to both the streaming server and the client. As a result, many commercial implementations that use industry-standard protocols can be plugged into the architecture relatively painlessly and can enjoy the benefits of the platform.

• ETRI Journal
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• v.33 no.3
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• pp.382-392
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• 2011

This paper proposes the streaming radio frequency identification (RFID) protocol to support robust data streaming in a passive communication, which is extended from the ISO18000-6 Type C RFID standard. By observing and modeling the unique bit error behavior through detailed analysis in this paper, we found that performance is significantly limited by inaccurate and unstable link frequencies as well as low SNR which are inevitable for passive devices. Based on the analysis, we propose a simple and efficient protocol to adaptively insert extra error control sequences in a packet for tolerating tough link condition while maximizing the throughput and preserving the minimal implementation cost. To evaluate effectiveness of our proposal in real-time streaming applications, we experimented on real-time H.264 video streaming and prototyped the system on FPGA. To our best knowledge, our paper is the first work to take analytical approach for maximizing the throughput and demonstrate the possibility of the realtime multimedia streaming transmission in the passive RFID system.

• Journal of the Korea Institute of Information Security & Cryptology
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• v.26 no.6
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• pp.1505-1511
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• 2016

Live streaming is a method to provide media service by sending recoded media to a user's video player. In order to provide video and audio contents in real-time for a large number of users simultaneously, live streaming compatible protocols such as RTMP (Real Time Messaging Protocol), HLS (Http Live Streaming), are required. In this paper, we analyzed vulnerability of paid live streaming services with the captured packets from the applications used by six major OTT (over-the-top) companies in Korea supporting live streaming services. We found that streaming channels were not encrypted and access control mechanisms were not properly used. Thus, guest users can freely use paid live streaming services.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.9 no.3
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• pp.1035-1056
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• 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 the Korea Institute of Information and Communication Engineering
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• v.13 no.8
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• pp.1571-1579
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• 2009

This paper proposes a P2P(Peer-to-Peer) streaming protocol which allows various QoS(Qualify of Service) requirements of video streaming to be supported for various personal IPTV applications in ubiquitous Internet environments. The proposed P2P streaming protocol takes fairly short startup delay, guarantees end-to-end delay bound and bandwidth requirements, and supports reliability level of video streaming for an IPTV application. The QoS-based P2P streaming protocol can properly use not only the advantages of scalability, availability, and ubiquity of P2P streaming but also complement its disadvantages of performance and reliability so that various types of personal IPTV applications can be properly implemented in the existing broadband Internet environments.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.12 no.5
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• pp.275-281
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• 2012

A streaming service is considered as an important fact to listen musics at mobile device in the popular smart-phone era. An apple's HTTP Live Streaming method is used in iOS and a RTSP(Real-time Streaming Protocol) method is used in Android OS. A Wowza Media Sever will be used to support both apple and android mobile streaming. This paper describes a design and implementation of smart-phone app. to record user's songs by userself with smart-phone recording function, upload to the streaming server and to service real-time mobile streaming play for an user's selected song and provide song sharing SNS functions.

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