• Journal of Broadcast Engineering
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• v.16 no.5
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• pp.797-800
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• 2011

The objective of MPEG Media Transport (MMT), which is on going standard, is to develop efficient delivery of media over packet based networks in an adaptive, progressive, download/streaming fashion over various IP based networks, including terrestrial, satellite and cable broadcast networks. In this paper we introduce utilization of signal strength information based on Cross Layer Design(CLD) to efficient multimedia delivery over wireless network in which in practice the wireless conditions can vary significantly. Many recent studies have shown that a significant improvement in wireless video throughput can be achieved by utilizing signal strength information on CLD [1][2]. Despite of its usefulness, however, it was difficult to employ signal strength information in rate adaptation applications due to different representation of signal strength information for each underlying wireless network. To that end, we proposed syntax and semantics of signal strength information in such a way that the information can be interpreted in the unified way. The proposed signal strength information was proposed for the MMT standardization.

• Journal of Communications and Networks
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• v.13 no.5
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• pp.518-524
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• 2011

A high assurance IP encryption (HAIPE) compliant protocol accelerator is proposed for military networks consisting of red (or classified) networks and black (or unclassified) networks. The boundary between red and black sides is assumed to be protected via a HAIPE device. However, the IP layer encryption introduces challenges for bandwidth on demand satellite communication. The problems experienced by transmission control protocol (TCP) over satellites are well understood: While standard modems (on the black side) employ TCP performance enhancing proxy (PEP) which has been shown to work well, the HAIPE encryption of TCP headers renders the onboard modem's PEP ineffective. This is attributed to the fact that under the bandwidth-on-demand environment, PEP must use traditional TCP mechanisms such as slow start to probe for the available bandwidth of the link (which eliminates the usefulness of the PEP). Most implementations recommend disabling the PEP when a HAIPE device is used. In this paper, we propose a novel solution, namely broadband HAIPE-embeddable satellite communications terminal (BHeST), which utilizes dynamic network performance enhancement algorithms for high latency bandwidth-on-demand satellite links protected by HAIPE. By moving the PEP into the red network and exploiting the explicit congestion notification bypass mechanism allowed by the latest HAIPE standard, we have been able to regain PEP's desired network enhancement that was lost due to HAIPE encryption (even though the idea of deploying PEP at the modem side is not new). Our BHeST solution employs direct video broadcast-return channel service (DVB-RCS), an open standard as a means of providing bandwidth-on-demand satellite links. Another issue we address is the estimation of current satellite bandwidth allocated to a remote terminal which is not available in DVBRCS. Simulation results show that the improvement of our solution over FIX PEP is significant and could reach up to 100%. The improvement over the original TCP is even more (up to 500% for certain configurations).

• 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 Journal of Korean Institute of Communications and Information Sciences
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• v.34 no.4B
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• pp.368-376
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• 2009

Over recent years, the development of multimedia devices has meant that a wider multimedia streaming service can be supported, and there are now many ways in which TV channels can communicate with different terminals. Generally, scalable video streaming is known to provide more efficient channel capacity than simulcast video streaming. Simulcast video streaming requires a large network bandwidth for all resolutions, but scalable video streaming needs only one flow for all resolutions. On the contrary, to preserve the same video quality, SVC(Sealable Video Coding) needs a higher bit-rate than AVC(non-layered Video Coding) due to the coding penalty($10%{\sim}30%$). In previous research, scalable video streaming has been compared with simulcast video streaming for network channel capacity, in two-user simulation environments. The simulation results show that the channel capacity of SVC is $16{\sim}20%$ smaller than AVC, but scalable video streaming is not efficient because of the limit of the present network framework. In this paper, we propose a new network framework with a new router using EDE(Extraction Decision Engine) and SVC Extractor to improve network performance. In addition, we compare the SVC environment in the proposed framework with previous research on the same way subject. The proposed network framework shows a channel capacity 50%(maximum) lower than that found in previous research studies.

• The Journal of the Institute of Internet, Broadcasting and Communication
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• v.14 no.4
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• pp.39-44
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• 2014

IMS(IP Multi-Media Subsystem) base on the IP service platform which can offer multimedia as the voice, audio, video, and data is service platform. In 3G mobile communication in the early day, IMS had a suggestion for supporting to multimedia service in the 3GPP. But now It is broadly substituting in the IPTV, wire phone company and it is substituted in internet platform base on the soft-switch in currently. Especially nowadays, 4G LTE in a mobile communication company is rapidly growing in market. Therefore, in this study, we had designed to the main prosser that can admit to 1,000 user over and SIP gateway which can link the IMS 코어 that can link SIP Device which adopt the standard protocol on the SIP.

• Proceedings of the Korean Society of Broadcast Engineers Conference
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• 2010.07a
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• pp.126-129
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• 2010

본 논문에서는 패킷 오류가 발생하는 IP 망을 통해 비디오 전송 서비스를 제공하기 위한 조건적 ARQ 기반의 오류 제어기법을 제안한다. IP 망에서 패킷 손실이 발생하면 수신 단말은 서버에게 손실된 패킷의 재전송을 요청할 수 있다. 그러나 망 상황과 접속 단말 개수를 고려하지 않고 무조건적인 재전송을 요청 할 경우 손실된 패킷이 복호화에 필요한 시점까지 도착 할 것인지가 보장되지 않을 뿐더러, 오히려 불필요한 재전송 요청에 의해 망에서의 트래픽 량을 가중시킬 수 있다. 따라서 망 상황을 고려하여 손실된 패킷이 재전송에 의해 적절한 시간 내에 도착이 가능한지를 판단하여 조건적으로 ARQ를 요청하는 기술이 요구된다. 본 논문에서는 기존의 RFC 문서에서 권고하는 재전송 관련 메카니즘들을 효과적으로 활용하여 표준기술 기반의 조건적 ARQ 오류제어 기법을 제안한다. 제안된 기법에서는 단말에서 추정된 망 상황을 기반으로 조건적으로 재전송 요청 여부를 결정하고, 더 나아가서 유효한 재전송 요청 시점을 결정하여 보다 효율적인 ARQ 가 적용될 수 있도록 한다. 제안된 조건적 ARQ 기반의 오류 제어 기법의 성능을 검증하기 위하여 패킷 손실 네트워크 환경을 NIST-Net을 활용하여 구축하며 실험 결과를 통해 제안된 기법의 우수한 오류 강인 성능을 확인한다.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.39B no.7
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• pp.467-476
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• 2014

Recently, various services which use the fused form of broadcasting and communication are being offered over the Internet globally. To provide various interactive services through connected smart TV on wired or wireless network, the seamless multimedia service is key feature. In this paper, Dynamic Adaptive Streaming over HTTP (DASH)is adopted to develop cost-effective adaptive streaming system for Open Hybrid TV(OHTV) service. This system can provide seamless adaptive streaming service for IP VOD, Video Clips, Ad insertion which are defined in OHTV.

• 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 Society of Computer Information Conference
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• 2016.07a
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• pp.71-72
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• 2016

본 논문에서는 사설방송이 필요한 상황에서 HD급 비디오카메라 등의 영상촬영장비 및 영상출력장비의 출력을 송출하고, 송출된 영상콘텐츠를 수신하여 기존의 HDTV 또는 컴퓨터로 전송할 수 있는 휴대가 가능한 방송 송수신 시스템을 제안한다. 즉 개발된 장비를 조합하여 기존의 공청망 또는 LAN을 이용하여 영상콘텐츠를 송수신하는 것으로써, 공청을 위한 선로가 포설되지 않았어도 기존의 LAN을 활용하여 영상콘텐츠를 송수신할 수 있다.

• Proceedings of the Korean Institute of Communication Sciences Conference
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• 2006.07a
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• pp.402-402
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• 2006