• Title/Summary/Keyword: QoS guarantee

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Improvement of F-GCRA Algorithm for ATM-GFR Service (ATM-GFR 서비스를 위한 F-GCRA 알고리즘 개선)

  • Park, In-Yong
    • The KIPS Transactions:PartC
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    • v.13C no.7 s.110
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    • pp.889-896
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    • 2006
  • ATM Forum has defined a guaranteed frame rate (GFR) service to serve Internet traffic efficiently. The GFR service provides virtual connections (VCs) for minimum cell rate (MCR) guarantees and allows them to fairly share the residual bandwidth. And ATM Forum has recommended a frame-based generic cell rate algorithm (F-GCRA) as a frame classifier, which determines whether an Am cell is eligible to use the guaranteed bandwidth in a frame level. An ATM switch accommodates cells in its buffer or drops them in a frame level according to current buffer occupancy. A FIFO shared buffer has so simple structure as to be feasibly implemented in switches, but has not been able to provide an MCR guarantee for each VC without buffer management based on per-VC accounting. In this paper, we enhance the F-GCRA frame classifier to guarantee an MCR of each VC without buffer management based on per-VC accounting. The enhanced frame classifier considers burstness of TCP traffic caused by congestion control algorithm so as to enable each VC to use its reserved bandwidth sufficiently. In addition, it is able to alleviate the unfairness problem in usage of the residual bandwidth. Simulation results show that the enhanced frame classifier satisfies quality of services (QoSs) of the GFR service for the TCP traffic.

Preceding Error Recovery Algorithm for Multimedia Stream in the Tree-based Multicast Environments (트리기반 멀티캐스트 환경에서 멀티미디어 스트림을 위한 선행에러복구 방안)

  • Kim, Ki-Young;Yoon, Mi-Youn;Shin, Young-Tae
    • The KIPS Transactions:PartC
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    • v.10C no.3
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    • pp.345-354
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    • 2003
  • IP Multicast is required of more little network resources than one in unicast. Furthermore, reliable multicast has been researched for supporting reliability at IP Multicast mechanism. Although these studies are carried out, they only have focused on general data. In other words, in case that realtime packet, they can not support reliability since they do not consider realtime properties such as dependency of interframe and playback in time. Besides, we also request to support scalability because we are based on Mobile IP network together with internet. Thus, we need a mechanism to guarantee reliability and scalability of realtime stream data. In this paper, we propose PER (Preceding Error Recovery) that reflect characteristics of the realtime data, especially for H.323. PER provides scalable reliability because it is based on tree-based multicast basically and helps to support scalable relibility as reducing control packet and recovers stream buffer space from underflow status as soon as possible. PER shows much better scalable and reliable than existing works.

A Study on Wireless Data Quality Measurement Method for u-Healthcare Service in WiBro Environment (WiBro 환경에서의 u-Healthcare 서비스를 위한 무선데이터 품질 측정 방안 연구)

  • Shim, Jae-Sung;Yun, Sung-Yeol;Park, Seok-Cheon
    • Journal of the Korea Academia-Industrial cooperation Society
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    • v.13 no.2
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    • pp.834-841
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    • 2012
  • In order to the mobile service uses the wireless terminal including the development of the wire wireless network and Smart phone, and etc. use including mobile office, which and etc. increases. Because in order that the u-Healthcare service using this appeared before the footlights and the existing quality measurement reference considered the speed, error rate, and etc. just, guarantees the stability of the u-Healthcare, the quality control by service are necessary. In this paper, the quality measurement reference by mobile service considering the radio environment as the method for satisfying the quality guarantee of the u-Healthcare mobile service user and user needs was presented. The WiBro u-Healthcare wireless data service quality based system in the end user perspective was established through the main performance index and entrepreneur case presented in the international standardization institute including 3GPP, WiMAX forum, GSMA, and etc. through the related research and the validity of the quality index establishment was presented according to each service.

Fast Multi-Phase Packet Classification Architecture using Internal Buffer and Single Entry Caching (내부 버퍼와 단일 엔트리 캐슁을 이용한 다단계 패킷 분류 가속화 구조)

  • Kang, Dae-In;Park, Hyun-Tae;Kim, Hyun-Sik;Kang, Sung-Ho
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.44 no.9
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    • pp.38-45
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    • 2007
  • With the emergence of new applications, packet classification is essential for supporting advanced internet applications, such as network security and QoS provisioning. As the packet classification on multiple-fields is a difficult and time consuming problem, internet routers need to classify incoming packet quickly into flows. In this paper, we present multi-phase packet classification architecture using an internal buffer for fast packet processing. Using internal buffer between address pair searching phase and remained fields searching phases, we can hide latency from the characteristic that search times of source and destination header fields are different. Moreover we guarantee the improvement by using single entry caching. The proposed architecture is easy to apply to different needs owing to its simplicity and generality.

A Session Allocation Algorithm for Fair Bandwidth Distribution of Multiple Shared Links (다중 공유 링크들의 공정한 대역폭 분배를 위한 세션할당 알고리즘)

  • Shim, Jae-Hong;Choi, Kyung-Hee;Jung, Gi-Hyun
    • The KIPS Transactions:PartC
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    • v.11C no.2
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    • pp.253-262
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    • 2004
  • In this paper, a session allocation algorithm for a switch with multiple shared links is proposed. The algorithm guarantees the reserved bandwidth to each service class and keeps the delay of sessions belonging to a service class as close as possible even if the sessionsare allocated to different shared links. To support these qualities of services, a new scheduling model for multiple shared links is defined and a session allocation algorithm to decide a shared link to be allocated to a new session on the connection establishmentis developed based on the model. The proposed heuristic algorithm allocates a session to a link including the subclass with the shortest (expected) delay that subclasses of the service class the session belongs to will experience. Simulation results verify that a switch with multiple shared links hiring the proposed algorithm provides service classes with fairer bandwidth allocation and higher throughput, and guarantees reserved bandwidth better than the switch hiring other session algorithms. It also guarantees very similarservice delay to the sessions in the same service class.

A Low Latency Handoff Scheme with Lossless Remote Subscription for Real-time Multimedia Communications in Mobile IP Environments (모바일 IP환경에서의 실시간 멀티미디어 통신을 위한 무손실 원격지 가입 저 지연 핸드오프)

  • Kim Ho-cheal;Kim Young-tak
    • Journal of KIISE:Information Networking
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    • v.31 no.6
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    • pp.620-632
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    • 2004
  • IP is not suitable for mobile nodes by network-based routing because mobile nodes are dynamically change their network attachment point. Mobile-IP is an IETF standard providing continuous access to the Internet for mobile nodes, but it has the triangle routing problem. Also it has a performance degradation problem by massive packet loss during layer 3 handoff of mobile nodes. Especially, two IETF multicast support schemes for Mobile-IP do not guarantee the quality of realtime multimedia services because they have several problems such as long routing path, packet duplication(hi-directional tunneling) and massive packet loss(remote subscription). In this paper, we propose a lossless remote subscription scheme that guarantees the quality of realtime multimedia services over Mobile-IP. From the result of simulation, we verified that the proposed scheme in this paper can reduce the delay time of remote subscription by the effect of the low latency handoff scheme that is extended to apply to the multicast group management and it requires only 0.58% buffer spaces of the previously proposed lossless remote subscription scheme.

Real-time traffic service in network with DiffServ (DiffServ 기반 네트워크에서의 실시간 트래픽 서비스)

  • Joung, Jin-No
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.32 no.1B
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    • pp.53-60
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    • 2007
  • We investigate the end-to-end delay bounds in large scale networks with Differentiated services (DiffServ) architecture. It is generally understood that networks with DiffServ architectures, where packets are treated according to the class they belong, can guarantee the end-to-end delay for packets of the highest priority class, only in lightly utilized cases. We focus on tree networks, which are defined to be acyclic connected graphs. We obtain a closed formula for delay bounds for such networks. We show that, in tree networks, the delay bounds exist regardless of the level of network utilization. These bounds are quadratically proportional to the maximum hop counts in heavily utilized networks; and are linearly proportional to the maximum hop counts in lightly utilized networks. Considering that tree networks, especially the Ethernet networks are being accepted more and more for access networks as well as provider networks, we argue that based on these delay bounds DiffServ architecture is able to support real time applications even for a large network. Throughout the paper we use Latency-Rate (LR) server model, with which it has proven that FIFO and Strict Priority are LR servers to each flows in certain conditions.

Performance Analysis of Packet CDMA R-ALOHA for Multi-media Integration in Cellular Systems with Adaptive Access Permission Probability

  • Kyeong Hur;Eom, Doo-Seop;Tchah, Kyun-Hyon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.25 no.12B
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    • pp.2109-2119
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    • 2000
  • In this paper, the Packet CDMA Reservation ALOHA protocol is proposed to support the multi-traffic services such as voice and videophone services with handoff calls, high-rate data and low-rate data services efficiently on the multi-rate transmission in uplink cellular systems. The frame structure, composed of the access slot and the transmission slot, and the proposed access permission probability based on the estimated number of contending users for each service are presented to reduce MAI. The assured priority to the voice and the videophone handoff calls is given through higher access permission probability. And through the proposed code assignment scheme, the voice service can be provided without the voice packet dropping probability in the CDMA/PRMA protocols. The code reservation is allowed to the voice and the videophone services. The low-rate data service uses the available codes during the silent periods of voice calls and the remaining codes in the codes assigned to the voice service to utilize codes efficiently. The high-rate data service uses the assigned codes to the high-rate data service and the remaining codes in the codes assigned to the videophone service. Using the Markov-chain subsystem model for each service including the handoff calls in uplink cellular systems, the steady-state performances are simulated and analyzed. After a round of tests for the examples, through the proposed code assignment scheme and the access permission probability, the Packet CDMA Reservation ALOHA protocol can guarantee the priority and the constant QoS for the handoff calls even at large number of contending users. Also, the data services are integrated efficiently on the multi-rate transmission.

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SE-CAC: A Novel Call Admission Control Scheme for Multi-service IDMA Systems

  • Ge, Xin;Liu, Gongliang;Mao, Xingpeng;Zhang, Naitong
    • KSII Transactions on Internet and Information Systems (TIIS)
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    • v.5 no.5
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    • pp.1049-1068
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    • 2011
  • In this paper a simple and effective call admission control (CAC) scheme is proposed for the emerging interleave-division multiple-access (IDMA) systems, supporting a variety of traffic types and offering different quality of service (QoS) requirements and priority levels. The proposed scheme is signal-to-interference-plus-noise ratio (SINR) evolution based CAC (SE-CAC). The key idea behind the scheme is to take advantage of the SINR evolution technique in the process of making admission decisions, which is developed from the effective chip-by-chip (CBC) multi-user detection (MUD) process in IDMA systems. By virtue of this semi-analytical technique, the MUD efficiency can be estimated accurately. Additionally, the computational complexity can be considerably reduced. These features make the scheme highly suitable for IDMA systems, which can combat intra-cell interference efficiently with simple CBC MUD. Analysis and simulation results show that compared to the traditional CAC scheme considering MUD efficiency as a constant, the proposed SE-CAC scheme can guarantee high power efficiency and throughput for multimedia traffic even in heavy load conditions, illustrating the high efficiency of CBC MUD. Furthermore, based on the SINR evolution, the SE-CAC can make accurate estimation of available resource considering the effect of MUD, leading to low outage probability as well as low blocking and dropping probability.

An Architecture of ISP-based P2P IPTV Services and Its Characteristics (계층 구조형 ISP 기반 P2P IPTV 서비스 구조 및 특성)

  • Sung, Moo-Kyung;Han, Chi-Moon
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.35 no.4B
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    • pp.659-669
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    • 2010
  • The P2P IPTV will create a new paradigm for the Internet services. However, it cannot guarantee the reliability of their server and QoS because of using common Internet users(peers) for SIP server or relay server, though the infrastructural cost is low. This paper proposes the ISP-based P2P IPTV architecture which can solute the limitations of conventional P2P-based IPTV. In this model, ISP can build P2P overlay network with ISP servers and directly manage each server needed for session connection. So, the servers have higher performance and better reliability than previous one. Besides, robustness is improved because each sever is set by P2P overlay network. To evaluate the characteristics of the ISP-based P2P IPTV architecture, we simulate it for some parameters which are end-to-end streaming delay time, connection delay time and traffic amount. We compared the proposed architecture with the conventional P2P architecture about video service and confirmed that the performance of ISP-based P2P IPTV is better than conventional P2P based IPTV.