• Journal of the Korea Society for Simulation
• /
• v.19 no.4
• /
• pp.89-98
• /
• 2010

In the heterogeneous system of various wired or wireless network with IP-based backbone, the continuities of agreedon QoS for multimedia services should be guaranteed regardless of network types and terminal mobility through seamless vertical handover. This paper proposes a QoS provisioning mechanism called D-ISHO which guarantees the continuities of agreed-on QoS and seamless for multimedia services by considering both such characteristics as delay, loss rate and jitter per each service and such status as available band-width, call arrival rate and data transmission rate during the vertical handover. Simulation is done for performance analysis with the measure of handover failure rate and packet loss rate.

• Proceedings of the Korean Information Science Society Conference
• /
• 2001.10c
• /
• pp.451-453
• /
• 2001

오늘날 네트워크의 대역특이 커지고 동시에 실시간 처리를 요하는 다양한 멀티미디어 애플리케이션들이 생성되고 있다. 그러나 문제는 고속 LAN 환경에서 많은 사용자들이 멀티미디어 애플리케이션들을 비롯한 다양한 네트워크 서비스들을 사용하고 있지만, WAN 환경으로의 선로로 전송하는데 있어서의 차별화가 없다는 것이다 따라서 경성 실시간(hard real time) 처리를 요하는 멀티미디어 데이터 들의 시간 제한을 지켜줄 수 없고, 비교적 지연시간의 제약을 받지 않는 HTML, FTP, e-Mail, 등의 연성 실시간(soft real time) 처리를 요하는 애플리케이션들에 대해 불필요한 대역폭 낭비를 일으킨다. 이러한 문제를 최소화하기 위해 본 논문에서는 엔터프라이즈 네트워크 등과 같은 고속 네트워크 망을 사용하는 환경에서 다양한 멀티미디어 데이터 패킷들을 고정적으로 할당된 대역폭에 따라 우선적으로 서비스되도록 하여 지연시간 제한을 최대한 보장해 주기 위한 리눅스 상에서 구현된 QoServer 개발 기술을 소개한다.

• Journal of Korea Multimedia Society
• /
• v.13 no.4
• /
• pp.594-600
• /
• 2010

Many studies are DDoS in Internet network, but the study is the fact that is not enough in a voice network. Therefore, we designed the extended TRW algorithm that was a DDoS attack traffic detection algorithm for the voice network which used an IP data network to solve upper problems in this article and evaluated it. The algorithm that is proposed in this paper analyzes TRW algorithm to detect existing DDoS attack in Internet network and, design connection and end connection to apply to a voice network, define probability function to count this. For inspect the algorithm, Set a threshold and using NS-2 Simulator. We measured detection rate by an attack traffic type and detection time by attack speed. At the result of evaluation 4.3 seconds for detection when transmitted INVITE attack packets per 0.1 seconds and 89.6% performance because detected 13,453 packet with attack at 15,000 time when transmitted attack packet.

• Journal of the Korean Society for Railway
• /
• v.12 no.6
• /
• pp.902-908
• /
• 2009

TETRA is the standard of Digital Trunked Radio System developed by the ETSI (European Telecommunications Standards Institute). It is currently adopted as Electric Power IT Wireless Backbone Network in Korea, and a national enterprise is going on for versatile utilization of TETRA. To use TETRA wireless network, TETRA modem is very necessary such that performance measurements are very crucial for each TETRA modem by various manufacturers. In this paper, PED (Protocol Evaluation Data) is suggested for PD performance measurement in D-TRS wireless network environment. The performance measurements for different data lengths and transmission intervals are done using TG (Traffic Generator) on Test Bed. The data size is increased by 10 bytes from 10-byte to 400-byte, and it is measured 1,000 times for each transmission interval of 0.5, 1.0, and 1.5 seconds. Based on the transmission time measured, average transmission speed and MER (Message Error Rate) are derived for TETRA Modem performance measurement. Two TMR880i's of EADS are used for TETRA modem, and SwMI (Switching and Management Infrastructure) of EADS is used for switching system in this paper.

• Journal of Broadcast Engineering
• /
• v.16 no.5
• /
• pp.722-737
• /
• 2011

In this paper, we propose an estimation of network jitter that occurs when transmitting TCP packets containing MPEG-2 TS in progressive streaming service over wired or wireless Internet networks. Based on the estimated network jitter size, we can calculate required de-jitter buffering time to absorb the network jitter at the receiver side. For this purpose, by exploiting the PCR timestamp existing in the TS packet header, we create a new timestamp information that is marked in the optional field of TCP packet header to estimate the network jitter. By using the proposed de-jitter buffering scheme, it is possible to employ the conventional T-STD buffer model without any modification in the progressive streaming service over IP networks. The proposed method can be applicable to the recently developed international standard, MPEG DASH (dynamic adaptive streaming over HTTP) technology.

• The Journal of Korean Institute of Communications and Information Sciences
• /
• v.27 no.9C
• /
• pp.836-841
• /
• 2002

Packet scheduling is an essential function to guarantee a quality of service by differentiating services in the Internet. Scheduling algorithms that have been suggested so far can be devided into Round-Robin methods and Fair Queueing methods Round-Robin methods have the advantage of high-speed processing through simple implementations, while Fair Queueing methods offer accurate services. Fair queueing algorithms, however, have problems of computational overheads and implementation complexity as their schedulers manage the states of every flow. This paper suggests a new method in which each flow performs the calculation in a distributed way to decide the service order. Our algorithm significantly reduces the scheduler's computational overheads while providing the same level of accuracy with the previous Fair Queueing algorithms.

• Journal of Broadcast Engineering
• /
• v.14 no.6
• /
• pp.796-806
• /
• 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 KIISE:Information Networking
• /
• v.36 no.5
• /
• pp.425-436
• /
• 2009

In recent years, the user demands have increased for multimedia service of high quality over the broadband convergence network. These rising demands for high quality multimedia service led the popularization of various user terminals and large scale display equipments, which needs a variety type of QoS (Quality of Service). In order to support demands for QoS, numerous research projects are in progress both from the perspective of network as well as end system; For example, at the network perspective, QoS guaranteeing by improving of internet performance such as Active Queue Management, while at the end system perspective, SVC (Scalable Video Coding) encoding scheme to guarantee media quality. However, existing AQM algorithms have problems which do not guarantee QoS, because they did not consider the essential characteristics of video encoding schemes. In this paper, it is proposed to solve this problem by deploying the TS- AQM (Temporal Scalability Active Queue Management) which employs the differentiated packet dropping for dependency of the temporal level among the frames, based on SVC encoding characteristics by exploiting the TID (Temporal ID) field of the SVC NAL unit header. The proposed TS-AQM guarantees multimedia service quality through video decoding reliability for SVC streaming service, by differentiated packet dropping when congestion exists.

• Proceedings of the Korea Information Processing Society Conference
• /
• 2001.10b
• /
• pp.1319-1322
• /
• 2001

본 논문에서는 WAN 환경에서의 선도시험망 기반 QoS 테스트베드를 구성하고 차등화 서비스에 의한 QoS 의 성능을 측정하였다. 전송율, RTT, 패킷 손실, FTP 소요시간을 대상으로 QoS 보장 서비스와 베스트-에포트 서비스에 대한 성능을 비교 분석하였다. 모든 측정대상에 대하여 QoS 보장 트래픽이 좋은 성능을 보여주었으며, 특히 멀티미디어 어플리케이션의 비디오 트래픽에 대해서도 QoS 보장 서비스에 의한 전송서비스가 고품질의 영상을 제공하였다.

• Electronics and Telecommunications Trends
• /
• v.18 no.5 s.83
• /
• pp.73-80
• /
• 2003

인터넷 서비스에 대한 수요가 양적 팽창의 단계에서 서비스 고품질화 및 다양화 등에 대한 질적 변화의 단계로 전환하는 환경 아래 국내외 통신 사업자들은 시장경쟁력을 가지고 차별화된 품질 서비스를 인터넷 사용자에게 직접 제공하는 등 다양한 노력을 추구하고 있다. 이와 같은 노력의 한 방안으로 각 통신 사업자 및 전문업체들은 품질측정 사이트를 제공함으로써 상품별 속도기상도, 지연시간 및 패킷손실률 등 다양한 품질 서비스를 가입자가 직접 느끼고, 만족하게 할 수 있도록 다양한 방법을 강구하고 있다. 본 논문에서는 국내외 통신 사업자 및 전문업체들의 품질측정 사이트들을 통해 제공되는 다양한 품질측정 서비스 활동들을 비교 및 분석하고, 이를 바탕으로 추후 나아갈 품질측정 사이트의 서비스 항목들을 제시한다.