• Journal of KIISE
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• v.43 no.7
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• pp.827-833
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• 2016

HTTP Adaptive Streaming (HAS) is a technique that adapts its video quality to network conditions for providing Quality of Experience. In the HAS approach, a video content is encoded at multiple bitrates and the encoded video content is divided into several video segments. A HAS player estimates the network bandwidth and adjusts the video bitrate based on estimated bandwidth. However, the segment scheduler in the conventional HAS player requests video segments periodically without considering TCP. If the waiting duration for the next segment request is quite long, the TCP connection can be initialized and it restarts slow-start. Slow-start causes the reduction in TCP throughput and consequentially leads to low-quality video streaming. In this study, we propose a TCP-aware segment scheduling scheme to improve performance of HAS service. The proposed scheme adjusts request time for the next video request to prevent initialization of TCP connection and also considers the point of scheduling time. The simulation proves that our scheme improves the Quality of Service of the HAS service without buffer underflow issue.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.33 no.3B
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• pp.144-151
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• 2008

In this paper, we propose a video streaming server that guarantees a certain level of quality when a server should serve video streaming service to multiple heterogenous clients simultaneously with TCP transport. If each heterogeneous client requests video streaming service in according to its own requirement such as bitrate of content and these requests are accepted by a server, then TCP flows for each video streaming session fairly share limited uplink bandwidth of the server. At this time, because TCP's bandwidth fair-share characteristics can result in bandwidth shrinkage of higher bitrate video streaming session, the client of higher bitrate video may suffer sluggish playback which is related with streaming QoS degradation. To tackle this problem, our proposed server system uses multiple TCP connections adaptively for each video streaming session depending on the anticipated status of the client playout buffer. Simulation results show that our proposed algorithm can successfully reduce the occurrence of playout buffer underrun and enhance streaming quality for whole video clients.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.334-336
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• 2005

최근 오디오나 비디오 스트리밍과 같은 멀티미디어 트래픽이 증가하고 있다. 이러한 트래픽들은 패킷을 전달하는데 대부분 UDP(User Datagram Protocol)기반의 RTP(Realtime Transport Protocol)를 사용한다. 하지만 UDP기반의 RTP는 기본적으로 혼잡제어 메커니즘이 없으며 현재 인터넷의 주요 트래픽인 TCP(Transmission Control Protocol)와의 형평성을 보장하지 않는다는 문제점을 갖는다. 본 논문에서는 스트리밍 트래픽의 TCP 친화적인 전송률 조절 기법으로 TF-RTP(TCP-Friendly RTP)를 제안하였다. TF-RTP는 네트워크 상태가 혼잡하여 패킷 손실이 발생할 경우, 개선된 파라미터들을 사용하여 경쟁하는 TCP의 전송률을 보다 정확하게 계산하여 스트리밍 트래픽의 전송률을 조절함으로써 경쟁하는 TCP 트래픽과 친화적으로 동작하며 네트워크 대역폭을 보다 공평하게 사용하게 된다. 실험을 통해 제안한 TF-RTP가 TCP의 전송률을 보다 정확하게 계산하며 TCP 친화성, 공평성 측면에서도 성능 개선을 보임을 확인할 수 있었다.

• Journal of KIISE:Computing Practices and Letters
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• v.14 no.2
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• pp.206-210
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• 2008

The most noticeable aspect of progressive streaming is the media playback during its download through TCP to avoid a lengthy wait for a content to finish downloading. By employing TCP, it is usually possible to detect lost packets by using the checksum and sequence numbering functions of TCP Thereafter, we can recover the lost packets by the retransmission function of TCP. However, there must remain enough amount of media data in the recipient buffer in order to guarantee seamless media playback even during retransmission. In this paper, we propose an efficient algorithm for determining the initial buffering time before start of playback to guarantee seamless playback during retransmission considering the probability of client buffer under-flow. The effectiveness of the proposed algorithm will be proved through extensive simulation results.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2003.10a
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• pp.837-840
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• 2003

Recently the multimedia streaming traffic such as digital audio and video in the Internet has increased tremendously. Unlike TCP, the UDP protocol, which has been used to transmit streaming traffic through the Internet, does not apply any congestion control mechanism to regulate the data flow through the shared network. If this trend is let go unchecked, these traffic will effect the performance of TCP, which is used to transport data traffic, and may lead to congestion collapse of the Internet. To avoid any adverse effort on the current Internet functionality, A study on a new protocol of modification or addition of some functionality to existing transport protocol for transmitting streaming traffic in the Internet is needed. TCP-frienly congestion control mechanism is classified with window-based congestion control scheme and rate-based congestion control scheme. In this paper, we propose an algorithm for improving the transmitting rate on a hybrid TCP-friendly congestion control scheme combined with widow-based and rate-based congestion control for multimedia streaming in the internet.

• 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.

• Journal of KIISE
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• v.41 no.11
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• pp.974-981
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• 2014

These days, use of multimedia streaming service and demand of QoS (Quality of Service) improvement have been increased because of development of network. QoS of streaming service is influenced by a jitter, delay, throughput, and loss rate. For guaranteeing these factors which are influencing QoS, the role of transport layer is very important. But existing TCP which is a typical transport layer protocol increases the size of congestion window slowly and decreases the size of a congestion window drastically. These TCP characteristic have a problem which cannot guarantee the QoS of UHD multimedia streaming service. In this paper, we propose a router buffer based congestion control method for improving the QoS of UHD streaming services. Our proposed scheme applies congestion window growth rate differentially according to a degree of congestion for preventing an excess of available bandwidth and maintaining a high bandwidth occupied. Also, our proposed scheme can control the size of congestion window according to a change of delay. After comparing with other congestion control protocols in the jitter, throughput, and loss rate, we found that our proposed scheme can offer a service which is suitable for the UDH streaming service.

• Journal of KIISE:Information Networking
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• v.36 no.3
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• pp.191-197
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• 2009

One of popular applications in the Internet is the multimedia streaming services such as IPTV or You Tube in which supporting quality of service (QoS) is an important issue. One of widely adopted rate control scheme is TCP-friendly rate control (TFRC) which shows comparable performance with TCP in term of throughput and lower variation of throughput over time. On the other hand, devices with multiple interfaces are emerging in the market. However, it has not proposed to exploit multiple interfaces simultaneously for multimedia streaming services with TFRC. In this paper, we propose a multimedia streaming algorithm with TFRC which exploits multiple interfaces to guarantee the quality of service. We show that the proposed scheme shows better performance than those with a single interface in terms of throughput and communication costs.

• Proceedings of the Korean Information Science Society Conference
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• 2005.07a
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• pp.361-363
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• 2005

최근 컴퓨터 기술의 발전과 더불어 급속하게 발전하는 네트워크 기술이 실생활에 보급되고 네트워크를 통한 멀티미디어 데이터의 교환이나 전송과 같은 서비스들이 활성화 되면서 멀티미디어 데이터가 점점 대용량화 다양화되고 있다. 기존의 멀티미디어 서비스는 정해진 서버로부터 미리 데이터를 받아서 보는 다운로드 서비스가 대부분이었고 스트리밍 서비스라 할지라도 사용자의 기호나 원하는 요구사항에는 미치지 못 하였다. 지금까지 스트리밍 데이터를 전송 하는 프로토콜로 주로 UDP(User Datagram Protocol)를 사용하였다. 하지만 UDP는 혼잡제어를 하지 않으며 현재 인터넷의 주요 트래픽인 TCP(Transmission Control Protocol) 트래픽은 혼잡제어를 한다. 그래서 UDP에 의한 스트리밍 서비스는 TCP 트래픽의 전송률을 저하시키며 더 나아가 네트워크의 전체의 성능을 저하시키는 요인이 될 수 있다. 본 논문에서는 IETF(Internet Engineering Task Force)에서 제정한 실시간 스트리밍 데이터 서비스를 위한 표준인 RTP(Real-Time Transport Protocol)/RTCP(Real-Time Transport Control Protocol)를 적용하여 RTCP의 정보를 가지고 현재 네트워크 상태를 판단하고 스트리밍 서비스를 할 때 데이터의 전송률은 TCP 친화적인 전송률로 조절하는 스트리밍 서비스를 구현하였다.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2015.10a
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• pp.791-793
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• 2015

Recently, UCC (User Created Contents) and VoD (Video on Demand), and multimedia content are growing, IP-TV, Smart TV, OHTV (Open Hybrid TV) various services such as multi platform (Multi-platform) environment, services and QoS issues. To solve this problem, the network efficiently, and improve the quality of content is necessary for the system. In this paper, the network of channels State and transmission of multimedia data based on dynamic resource usage, TCP and UDP, Adaptive dual-streaming system used for design and analysis. In addition, the existing TCP and UDP streaming system using a single protocol for analysis and verification of the effectiveness of the difference between and. This is a disaster, and medical/first aid system will be utilized in the field of feed, are ubiquitous.