• KSII Transactions on Internet and Information Systems (TIIS)
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• v.8 no.11
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• pp.3775-3788
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• 2014

With the increasing number of the wireless and mobile networks, the way that people use the Internet has changed substantively. Wireless multimedia services, such as wireless video streaming, mobile video game, and mobile voice over IP, will become the main applications of the future wireless Internet. To accommodate the growing volume of wireless data traffic and multimedia services, cognitive radio (CR) and Information-Centric Network (ICN) have been proposed to maximize the utilization of wireless spectrum and improve the network performance. Although CR and ICN have high potential significance for the future wireless Internet, few studies have been conducted on collaborative operations of CR and ICN. Due to the lack of infrastructure support in multi-hop ad hoc CR networks, the problem is more challenging for video streaming services. In this paper, we propose a Cross-layer Video Streaming Mechanism (CLISM) for Cognitive Radio Ad Hoc Information Centric Networks (CRAH-ICNs). The CLISM included two distributed schemes which are designed for the forwarding nodes and receiving nodes in CRAH-ICNs. With the cross-layer approach, the CLISM is able to self-adapt the variation of the link conditions without the central network controller. Experimental results demonstrate that the proposed CLISM efficiently adjust video transmission policy under various network conditions.

• The KIPS Transactions:PartC
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• v.10C no.4
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• pp.479-484
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• 2003

In the current Internet, it is difficult for an Internet Phone to guarantee the QoS due to variable network conditions such as packet loss rate, delay and bandwidth. In addition, the QoS of an Internet Video Phone is more hard to guarantee because of video data. In this paper, we investigate application-level QoS control schemes that can adapt to variable network conditions, and describe an error control scheme and a congestion control scheme. Based on these QoS control schemes, we have designed and implemented an Internet Video Phone System that supports adaptive audio and video delivery. Through experiments, we found that the Internet Video Phone can reduce the packet loss rate considerably as well as adjust the transmission rate considering other TCP flows.

• ETRI Journal
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• v.32 no.6
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• pp.843-853
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• 2010

A successful deployment of multimedia applications over wireless environments entails improving the quality of service (QoS), not only from a technical point of view, but also considering the quality of experience (QoE) from the final user's perception. Although objective QoE measure models avoid the difficulties of subjective surveys, subjective QoE assessments are essential to understand the way users evaluate the QoS. In this work, we study the effect of a wide range of parameters on the QoE of VVoIP applications in a real wireless scenario. Through a complete statistical analysis of users' ratings, we identify the following facts. Although the use of VVoIP in wireless networks does not yet represent an advantage for users, there are great expectations for all applications under study, and with greater popularity comes higher expectations. It is easier for respondents to identify good behavior than poor behavior. Whereas the respondents' frequency of Internet use does not impact on the scores, respondents' gender does. Finally, the most determining parameters of quality from a user's perspective were instability, video quality, voice distortion, usefulness, and graphical interface.

• The Journal of the Korea institute of electronic communication sciences
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• v.10 no.1
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• pp.21-26
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• 2015

In this paper, we propose a system which can detect the smoke in real time from the high-quality IP camera. For real-time processing, open directly the RTSP streams transmitted from the IP camera using the library FFmpeg as opening a video file. To recognize smoke, color information and morphological characteristics of smoke, as well as the dynamic characteristics of the smoke also considered for candidate regions. To combine the characteristics of the various smoke effectively, the Adaboost algorithm, was used as the boosting algorithm finally. Through the experiments with input videos from IP camera, the proposed algorithms were useful to detect smokes.

• Proceedings of the Korea Information Processing Society Conference
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• 2011.04a
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• pp.1188-1191
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• 2011

IPTV 방송에서 인터넷 기반 멀티캐스트는 일대다 또는 다대다 통신을 위한 차세대 중요한 서비스로 주목 받고 있다. 멀티캐스트는 네트워크 또는 애플리케이션 레벨에서 서비스할 수 있다. IP 멀티캐스트는 소스노드에서 라우터로 데이터그램을 보내면 라우터가 이를 복제하여 수신노드들에게 전달해 주는 네트워크레벨 서비스로 네트워크 자원을 효율적 사용할 수 있다. 그러나 네트워크에 IP 멀티캐스트 라우터가 설치되어야 하는 등 여러 문제로 인해 널리 사용되지 못하고 있다. 따라서 대안으로 애플리케이션 레벨에서의 오버레이 멀티캐스트가 주목 받고 있다. 오버레이 멀티캐스트는 종단호스트가 라우터처럼 동작하는 것으로 비록 IP 멀티캐스트에 비해서 링크 사용율과 지연 값이 높아질 수 있지만, IP 멀티캐스트의 현실적인 적용의 어려움을 해결할 수 있는 장점을 가지고 있다. 본 논문에서는 IP 멀티캐스트가 제공되지 않는 네트워크에서 효율적인 SVC(Scalable Video Coding) 멀티미디어 방송서비스와 SSENS(SVC Stream Exchange Network Server) 라우팅을 위한 MST(Minimum Spanning tree)를 목적으로 하는 오버레이 멀티캐스트 트리를 구현하는 알고리즘을 설계한다. 제안된 알고리즘의 성능 분석을 위해 시뮬레이션을 통해 패킷 손실 측면에서 Prim 알고리즘에 비해 평균 패킷 손실율이 40% 가까이 향상됨을 증명하였다.

• Journal of Digital Convergence
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• v.9 no.5
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• pp.225-233
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• 2011

Internet telephony is an Internet service which supports voice telephone using VoIP technology on the IP-based Internet. It has some advantages in that voice telephone services can be accompanied with multimedia services such as video communication and messaging services. Recently, the introduction of smart phones has led to a growth in social networking services and thus, the research and development of Internet telephony has been actively progressed and has the potential to become a replacement for the telephone service that is currently being used. In this paper we designed and implemented a recording system which records voice data of SIP-based Internet telephone's voice calls. It is developed on the linux system and has some features such as audio mixing of two in/out voice channels, live packet sniffing, and the ability to transfer mixed audio files to the log file server. These functions are implemented using various open source softwares. Afterwards, this VoIP recording system will be applied as a base technology to advanced services like a VoIP-based call center system.

• The KIPS Transactions:PartA
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• v.12A no.7 s.97
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• pp.627-636
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• 2005

This paper presents a PC cluster-based distributed video on demand (VOD) server that minimizes the load of the interconnection network by adopting channel bonding-based MVIA and the interval cache algorithm Video data is distributed to the disks of each server node of the distributed VOD server and each server node receives the data through the interconnection network and sends it to clients. The load of the interconnection network increases because of the large volume of video data transferred. We adopt two techniques to reduce the load of the interconnection network. First, an Msupporting channel bonding technique is adopted for the interconnection network. n which is a user-level communication protocol that reduces the overhead of the TCP/IP protocol in cluster systems, minimizes the time spent in communicating. We increase the bandwidth of the interconnection network using the channel bonding technique with MThe channel bonding technique expands the bandwidth by sending data concurrently through multiple network cards. Second, the interval cache reduces traffic on the interconnection network by caching the video data transferred from the remote disks in main memory Experiments using the distributed VOD server of this paper showed a maximum performance improvement of $30\%$ compared with a distributed VOD server without channel bonding-based MVIA and the interval cache, when used with a four-node PC cluster.

• Journal of Broadcast Engineering
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• v.14 no.6
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• pp.796-806
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• 2009

SVC (Scalable Video Coding) is a new video coding standard to provide convergence media service in heterogeneous environments with different networks and diverse terminals through spatial-temporal-quality combined flexible scalabilities. This paper presents the performance analysis on packet loss in the delivery of SVC over IP networks and an efficient adaptation method to packet loss caused by buffer overflow. In particular, SVC with MGS (Medium Grained Scalability) as well as spatial and temporal scalabilities is addressed in the consideration of packet-based adaptation since finer adaptation is possible with a sufficient numbers of quality layers in MGS. The effect on spatio-temporal quality due to the packet loss of SVC with MGS is evaluated. In order to minimize quality degradation resulted by packet loss, the proposed adaptation of MGS based SVC first sets adaptation unit of AU (Access Unit) or GOP corresponding to allowed delay and then selectively discards packets in order of importance in terms of layer dependency. In the experiment, the effects of packet loss on quantitative qualities are analyzed and the effectiveness of the proposed adaptation to packet loss is shown.

• Journal of Broadcast Engineering
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• v.16 no.2
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• pp.305-318
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• 2011

Since switched digital video (SDV) provides specific programs only to the subscribers who request the programs, SDV has attracted considerable interest of MSOs for bandwidth efficiency. In North America, MSOs service over 2.3 million households with the SDV for cable networks. In Korea, since demand of HD program, high-speed Internet, VoD, and VoIP is noticeably rising, the SDV is considered as the alternative for bandwidth saving and efficient managing. In this paper, the characteristics, operating structure, and bandwidth saving of the SDV are discussed and technical requirements for the SDV are also introduced. The channel switching performance and stability of the SDV are analyzed through the test-bed.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2013.05a
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• pp.345-348
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• 2013

This paper provides a method for compression and transmission of on-chip bus data. As the data traffic on on-chip buses is rapidly increasing with enlarged video resolutions, many video processor chips suffer from a lack of bus bandwidth and their IP cores have to wait for a longer time to get a bus grant. In multimedia data such as images and video, the adjacent data signals very often have little or no difference between them. Taking advantage of this point, this paper develops a simple bus data compression method to improve the chip performance and presents its hardware implementation. The method is applied to a Video Codec - 1 (VC-1) decoder chip and reduces the processing time of one macro-block by 13.6% and 10.3% for SD and HD videos, respectively.