• Journal of KIISE:Information Networking
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• v.29 no.6
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• pp.722-737
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• 2002

As the number of multimedia applications increases, various protocols and architectures have been proposed to provide QoS(Quality of Service) guarantees in the Internet. Most of these techniques, though, bear inherent contradiction between the scalability and the capability of providing QoS guarantees. In this paper, we propose a protocol, named DQSP(Dynamic QoS Support Protocol), which provides the dynamic resource allocation and admission control for QoS guarantees in a scalable way. In DQSP, the core routers only maintain the per source-edge router resource allocation state information. Each of the source-edge routers maintains the usage information for the resources allocated to itself on each of the network links. Based on this information, the source edge routers perform admission control for the incoming flows. For the resource reservation and admission control, DQSP does not incur per flow signaling at the core network, and the amount of state information at the core routers depends on the scale of the topology instead of the number of user flows. Simulation results show that DQSP achieves efficient resource utilization without incurring the number of user flow related scalability problems as with IntServ, which is one of the representative architectures providing end-to-end QoS guarantees.

• Journal of information and communication convergence engineering
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• v.10 no.3
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• pp.231-236
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• 2012

For real-time wireless sensor network applications, it is essential to provide different levels of quality of service (QoS) such as reliability, low latency, and fault-tolerant traffic control. To meet these requirements, an (m,k)-firm based real-time routing protocol has been proposed in our prior work, including a novel local transmission status indicator called local DBP (L_DBP). In this paper, a fault recovery scheme for (m,k)-firm real-time streams is proposed to improve the performance of our prior work, by contributing a delay-aware forwarding candidates selection algorithm for providing restricted redundancy of packets on multipath with bounded delay in case of transmission failure. Each node can utilize the evaluated stream DBP (G_DBP) and L_DBP values as well as the deadline information of packets to dynamically define the forwarding candidate set. Simulation results show that for real-time service, it is possible to achieve both reliability and timeliness in the fault recovery process, which consequently avoids dynamic failure and guarantees meeting the end-to-end QoS requirement.

• Proceedings of the Korean Society of Computer Information Conference
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• 2008.06a
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• pp.55-62
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• 2008

최근 네트워크 고도화로 유무선 통합된 협업 서비스를 고객에게 제공한다. 서비스 제공자의 다양한 네트워크 서비스 상품이 앞으로 추가 출시될 예정이며 이에 따른 서비스 품질은 매우 중요한 이슈가 될 것이다. 특히 멀티미디어 서비스는 프리미엄 서비스로 서비스 제공자는 고객에게 서비스 품질을 보증해야만 된다. 초고속 인터넷 서비스 상품으로 인터넷 전화기, 인터넷 다운 속도 등 다양한 인터넷 서비스 품질 측정을 제공하고 있으나 전용선에 대한 인터넷 실시간 멀티미디어 서비스는 서비스 품질 측정이 초기 상태이다. 따라서 본 논문에서는 멀티미디어 서비스를 위한 서비스 품질 측정 지표, 종단간 서비스 품질 측정 방안, 고객에게 원활한 서비스를 제공하기 위한 에이전트 시스템 설계, 서비스 제공자가 이미 보유한 NMS/EMS에 에이전트 시스템 연동 방안을 제안한다.

• The Transactions of the Korea Information Processing Society
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• v.6 no.9
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• pp.2512-2523
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• 1999

In this paper, we present the design and implementation results of RTP(Real-time Transport Protocol) which is introduced to provide end-to-end delivery service for multimedia data with real-time characteristics and to provide QoS(quality of service) monitoring services. Conventional RTP communication modules have some problems such as poor reusability and inefficiency. Reusability problem stems from the inherent characteristics of RTP that is framework is deliberately not complete, and so, RTP should have been designed and implemented in application-imbedded from. The RTP communication module, proposed in this paper, is designed and implemented in application-independent from in order to be used in all kinds of high level applications independent of their architectures and functions.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.11 no.5
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• pp.2499-2522
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• 2017

Multi-hop networks are a low-setup-cost solution for enlarging an area of network coverage through multi-hop routing. Carrier sense multiple access with collision avoidance (CSMA/CA) is frequently used in multi-hop networks. Multi-hop networks face multiple problems, such as a rise in contention for the medium, and packet loss under heavy-load, saturated conditions, which consumes more bandwidth due to re-transmissions. The number of re-transmissions carried out in a multi-hop network plays a major role in the achievable quality of service (QoS). This paper presents a statistical, analytical model for the end-to-end delay of contention-based medium access control (MAC) strategies. These strategies schedule a packet before performing the back-off contention for both differentiated heterogeneous data and homogeneous data under saturation conditions. The analytical model is an application of Markov chain theory and queuing theory. The M/M/1 model is used to derive access queue waiting times, and an absorbing Markov chain is used to determine the expected number of re-transmissions in a multi-hop scenario. This is then used to calculate the expected end-to-end delay. The prediction by the proposed model is compared to the simulation results, and shows close correlation for the different test cases with different arrival rates.

• Proceedings of the Korean Institute of Information and Commucation Sciences Conference
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• 2010.10a
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• pp.643-645
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• 2010

The Internet-based multimedia services such as IPTV is being expanded with the development of technology to support the convergence of broadcasting and telecommunications technology for the control seems to be growing larger. Especially for the real-time TV broadcast multicast control technology to support the authentication and resource control, in addition to the technology services that enhance the value of technology for a variety of services in both directions seems to be developed. And, Internet-based transmission system transmit the video content for the video content delivery using RTP(Real Time Transport Protocol). Standardization body, IETF(Internet Engineering Task Force) within the RTP, according to a variety of audio and video formats only transmission format(RTP Payload Format) Establish a separate standard and scalable video content "RTP Payload Format for SVC(Switched Virtual Connection) Video" the standardization is currently processing. In this paper we are improving the quality of broadcasting and telecommunication systems, so that the upper layer by the application can react adaptively to the existing MPEG2-TS and RTP who are provided by a variety of content applied to a variety of devices consumers ETE(End- to-End) QoS(Quality of Service) for enhance the system who was designed and implemented.

• Journal of the Institute of Electronics Engineers of Korea TC
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• v.40 no.7
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• pp.261-272
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• 2003

An Ethernet PON(Passive Optical Network) is an economical and efficient access network that has received significant research attention in recent years. A MAC(Media Access Control) protocol of PON , the next generation access network, is based on TDMA(Time Division Multiple Access) basically and can classify this protocol into a fixed length slot assignment method suitable for leased line supporting QoS(Quality of Service) and a variable length slot assignment method suitable for LAN/MAN with the best effort. For analyzing the performance of these protocols, we design an Ethernet PON model using OPNET tool. To establish the maximum efficiency of a network, we verify a MAC protocol and determine the optimal number of ONUs(Optical Network Unit) that can be accepted by one OLT(Optical Line Terminal) and propose the suitable buffer size of ONU based on analyzing the end-to-end Ethernet delay, queuing delay, throughput, and utilization.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.17 no.8
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• pp.2188-2208
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• 2023

Quality of Service (QoS) is a critical feature of Wireless Sensor Networks (WSNs) with routing algorithms. Data packets are moved between cluster heads with QoS using a number of energy-efficient routing techniques. However, sustaining high scalability while increasing the life of a WSN's networks scenario remains a challenging task. Thus, this research aims to develop an energy-balancing component that ensures equal energy consumption for all network sensors while offering flexible routing without congestion, even at peak hours. This research work proposes a Gravitational Blackhole Search Optimised splay tree routing framework. Based on the splay tree topology, the routing procedure is carried out by the suggested method using three distinct steps. Initially, the proposed GBSO decides the optimal route at initiation phases by choosing the root node with optimum energy in the splay tree. In the selection stage, the steps for energy update and trust update are completed by evaluating a novel reliance function utilising the Parent Reliance (PR) and Grand Parent Reliance (GPR). Finally, in the routing phase, using the fitness measure and the minimal distance, the GBSO algorithm determines the best route for data broadcast. The model results demonstrated the efficacy of the suggested technique with 99.52% packet delivery ratio, a minimum delay of 0.19 s, and a network lifetime of 1750 rounds with 200 nodes. Also, the comparative analysis ensured that the suggested algorithm surpasses the effectiveness of the existing algorithm in all aspects and guaranteed end-to-end delivery of packets.

• Journal of Internet Computing and Services
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• v.8 no.1
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• pp.25-31
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• 2007

The development of efficient Quality of Service (QoS) routing algorithms in high speed networks is very difficult since divergent services require various quality conditions, If the QoS parameter we concern is to measure the delay on that link, then the routing algorithm obtains the Least Delay (LD) path, Meanwhile, if the parameter is to measure of the link cast, then it calculates the Least Cost (LC) path. The Delay Constrained Least Cast (DCLC) path problem of the mixed issues on LD and LC has been shown to be NP-hard. The path cost of LD path is relatively mere expensive than that of LC path, and the path delay of LC path is relatively higher than that of LD path in DCLC problem. In this paper. we propose the algorithm based on estimated path for the DCLC problem and investigate its performance, It employs a new parameter which is probabilistic combination of cost and delay, We have performed empirical evaluation that compares our proposed algorithm with the DCUR in various network situations.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.9B
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• pp.1272-1280
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• 2010

WFQ (Weighted Fair Queuing) provides not only fairness among traffic flows in using bandwidth but also guarantees the Quality of Service (QoS) that individual flow requires, which is why it has been applied to the resource reservation protocol (RSVP)-capable router. The RSVP allocates an enough resource to satisfy both the rate and end-to-end delay requirements of the flow in condition of no packet loss, and the WFQ guarantees those QoS requirements with the allocated resource. In a practice, however, most QoS-guaranteed services, specially the Voice of IP, allow a few percent of packet loss, so it is strongly desired that the RSVP and WFQ make the best use of this allowable packet loss. This paper enhances the WFQ to allow packet loss and investigates its performance. The performance evaluation showed that allowing the packet loss of 0.4% can improve the flow admission capability by around 40 percent.