• Title/Summary/Keyword: Packet Loss

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A Route Repair Scheme for Reducing DIO Poisoning Overhead in RPL-based IoT Networks (RPL 기반 IoT 네트워크에서 DIO Poisoning 오버헤드를 감소시키는 경로 복구 방법)

  • Lee, Sung-Jun;Chung, Sang-Hwa
    • Journal of KIISE
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    • v.43 no.11
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    • pp.1233-1244
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    • 2016
  • In the IoT network environments for LLNs(Low power and Lossy networks), IPv6 Routing Protocol for Low Power and Lossy networks(RPL) has been proposed by IETF(Internet Engineering Task Force). The goal of RPL is to create a directed acyclic graph, without loops. As recommended by the IETF standard, RPL route recovery mechanisms in the event of a failure of a node should avoid loop, loop detection, DIO Poisoning. In this process, route recovery time and control message might be increased in the sub-tree because of the repeated route search. In this paper, we suggested RPL route recovery method to solve the routing overhead problem in the sub-tree during a loss of a link in the RPL routing protocol based on IoT wireless networks. The proposed method improved local repair process by utilizing a route that could not be selected as the preferred existing parents. This reduced the traffic control packet, especially in the disconnected node's sub tree. It also resulted in a quick recovery. Our simulation results showed that the proposed RPL local repair reduced the recovery time and the traffic of control packets of RPL. According to our experiment results, the proposed method improved the recovery performance of RPL.

Two Flow Control Techniques for Teleconferencing over the Internet (인터넷상에서 원격회의를 위한 두 가지 흐름 제어 기법)

  • Na, Seung-Gu;Go, Min-Su;An, Jong-Seok
    • Journal of KIISE:Computer Systems and Theory
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    • v.26 no.8
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    • pp.975-983
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    • 1999
  • 최근 네트워크의 속도가 빨라지고 멀티미디어 데이터를 다루기 위한 기술들이 개발됨에 따라 많은 멀티미디어 응용 프로그램들이 인터넷에 등장하고 있다. 그러나 이들 응용프로그램들은 수신자에게 전송되는 영상.음성의 품질이 낮기 때문에 기대만큼 빠르게 확산되지 못하고 있다. 영상.음성의 품질이 낮은 이유는 현재 인터넷이 실시간 응용프로그램이 요구하는 만큼 빠르고 신뢰성 있게 데이터를 전송할 수 없기 때문이다. 현재 인터넷의 내부구조를 바꾸지 않고 품질을 높이기 위해 많은 연구들이 진행되고 있는데 그 중 하나는 동적으로 변화하는 인터넷의 상태에 맞게 멀티캐스트 트래픽의 전송율을 조절하는 종단간의 흐름제어이다. 본 논문은 기존의 흐름제어 기법인 IVS와 RLM의 성능을 개선시키기 위한 두 가지 흐름제어 기법을 소개한다. IVS는 송신자가 주기적으로 측정된 네트워크 상태에 따라 전송율을 일정하게 조절한다. 송신자가 하나의 데이타 스트림을 생성하는 IVS와는 달리 RLM에서는 송신자가 계층적 코딩에 의하여 생성된 여러개의 데이타 스트림을 전송하고 각 수신자는 자신의 네트워크 상태에 맞게 데이타 스트림을 선택하는 기법이다. 그러나 IVS는 송신자가 전송율을 일정하게 증가시키고, RLM은 각자의 네트워크 상태를 고려하지 않고 임의의 시간에 하나 이상의 데이타 스트림을 받기 때문에 성능을 저하시킬 수 있다. 본 논문에서는 TCP-like IVS와 Adaptive RLM이라는 두 가지 새로운 기법을 소개한다. TCP-like IVS는 송신자가 전송율을 동적으로 결정하고, Adaptive RLM은 하나 이상의 데이타 스트림을 받기 위해 적당한 시간을 선택할 수 있다. 본 논문에서는 시뮬레이션을 통해 여러 가지 네트워크 구조에서 두 가지 방식이 기존의 방식에 비하여 더욱 높은 대역폭 이용율과 10~20% 정도 적은 패킷손실율을 이룬다는 것을 보여준다.Abstract Nowadays, many multimedia applications for the Internet are introduced as the network gets faster and many techniques manipulating multimedia data are developed. These multimedia applications, however, do not spread widely and are not fast as expected at their introduction time due to the poor quality of image and voice delivered at receivers. The poor quality is mainly attributed to that the current Internet can not carry data as fast and reliably as the real-time applications require. To improve the quality without modifying the internal structure of the current Internet, many researches are conducted. One of them is an end-to-end flow control of multicast traffic adapting the sending rate to the dynamically varying Internet state. This paper proposes two flow-control techniques which can improve the performance of the two conventional techniques; IVS and RLM. IVS statically adjusts the sending rate based on the network state periodically estimated. Differently from IVS in which a sender produces one single data stream, in RLM a sender transmits several data streams generated by the layered coding scheme and each receiver selects some data streams based on its own network state. The more data streams a receiver receives, the better quality of image or voice the receiver can produce. The two techniques, however, can degrade the performance since IVS increases its sending rate statically and RLM accepts one more data stream at arbitrary time regardless of the network state respectively. We introduce two new techniques called TCP-like IVS and Adaptive RLM; TCP-like IVS can determine the sending rate dynamically and Adaptive RLM can select the right time to add one more data stream. Our simulation experiments show that two techniques can achieve better utilization and less packet loss by 10-20% over various network topologies.

Efficient Virtual Machine Migration for Mobile Cloud Using PMIPv6 (모바일 클라우드 환경에서 PMIPv6를 이용한 효율적인 가상머신 마이그레이션)

  • Lee, Tae-Hee;Na, Sang-Ho;Lee, Seung-Jin;Kim, Myeong-Eeob;Huh, Eui-Nam
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.37B no.9
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    • pp.806-813
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    • 2012
  • In a cloud computing environment, various solutions were introduced to provide the service to users such as Infrastructure as a Service (IaaS), Platform as a Service (PaaS), Software as a Service (SaaS) and Desktop as a Service (DaaS). Nowadays, Mobile as a Service (MaaS) to provide the mobility in a cloud environment. In other words, users must have access to data and applications even when they are moving. Thus, to support the mobility to a mobile Thin-Client is the key factor. Related works to support the mobility for mobile devices were Mobile IPv6 and Proxy Mobile IPv6 which showed performance drawbacks such as packet loss during hand-over which could be very critical when collaborating with cloud computing environment. The proposed model in this paper deploys middleware and replica servers to support the data transmission among cloud and PMIPv6 domain. It supports efficient mobility during high-speed movement as well as high-density of mobile nodes in local mobility anchor. In this paper, through performance evaluation, the proposed scheme shows the cost comparison between previous PMIPv6 and verifies its significant efficiency.

A Study on Scheme to Support QoS using Differentiated Services in MPLS Network (MPLS 망에서 Differentiated Services를 이용한 QoS 지원 방안에 관한 연구)

  • Park, Chun-Kwan;Jeon, Byung-Chun
    • Journal of IKEEE
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    • v.5 no.2 s.9
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    • pp.136-145
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    • 2001
  • As with appearing new applications that requires QoS guarantee such as VoIP, VPN in Internet, problems of IP QoS has been one of most important issues in next-generation Internet. IETF has proposed integrated services model(Int-Serv) and differentiated service(Diff-Serv) to supply IP QoS in Internet. Int-Serv model uses the state information of each IP flow, so satisfies QoS according to traffic characteristics, but increases the amount of flow state information with increasing flow number. Diff-Serv model uses PHP(Per Hop Behavior), and there are well-defined classes to provide differentiated traffics with different services according to delay and loss sensitivity. Diff-Serv model can provide diverse services in Internet because of having no the state and signal information of each flow. As MPLS uses the packet forwarding technique based-on label, it implements the traffic engineering in the networks easily. The MPLS can set up the path with different traffic parameters, and assign each path to particular Class of Services. Therefore it is possible to support the Diff-Serv model with well-defined classes. In this paper we investigate the performance improvement of Diff-Serv function in the MPLS network to guarantee class of services in Internet.

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Audio Stream Delivery Using AMR(Adaptive Multi-Rate) Coder with Forward Error Correction in the Internet (인터넷 환경에서 FEC 기능이 추가된 AMR음성 부호화기를 이용한 오디오 스트림 전송)

  • 김은중;이인성
    • The Journal of Korean Institute of Communications and Information Sciences
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    • v.26 no.12A
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    • pp.2027-2035
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    • 2001
  • In this paper, we present an audio stream delivery using the AMR (Adaptive Multi-Rate) coder that was adopted by ETSI and 3GPP as a standard vocoder for next generation IMT-2000 service in which includes combined sender (FEC) and receiver reconstruction technique in the Internet. By use of the media-specific FEC scheme, the possibility to recover lost packets can be much increased due to the addition of repair data to a main data stream, by which the contents of lost packets can be recovered. The AMR codec is based on the code-excited linear predictive (CELP) coding model. So we use a frame erasure concealment for CELP-based coders. The proposed scheme is evaluated with ITU-T G.729 (CS-ACELP) coder and AMR - 12.2 kbit/s through the SNR (Signal to Noise Ratio) and the MOS (Mean Opinion Score) test. The proposed scheme provides 1.1 higher in Mean Opinion Score value and 5.61 dB higher than AMR - 12.2 kbit/s in terms of SNR in 10% packet loss, and maintains the communicab1e quality speech at frame erasure rates lop to 20%.

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An Enhanced Fast Handover for Proxy MIPv6 Scheme for Efficient Mobile Environment of The Future Network (미래네트워크의 효율적인 모바일 환경 구축을 위한 향상된 Fast Handover for Proxy MIPv6 기법)

  • Go, Kwang-Sub;Jung, Ui-Seok;Mun, Young-Song
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.48 no.1
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    • pp.84-91
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    • 2011
  • To develop the new network, the future network architecture is studied. Since the mobile devices are also advanced, they need for the mobility protocols. The one of the protocols, Fast handovers for proxy MIPv6(PFMIPv6) has studied by the Internet Engineering Task Force(IETF). Since PFMIPv6 adopts the entities and the concepts of fast handovers for MIPv6(FMIPv6) in proxy MIPv6(PMIPv6), it reduces the packet loss. Although the conventional scheme has proposed that it cooperated with an Authentication, Authorization and Accounting (AAA) infrastructure for authentication of a mobile node in PFMIPv6, it has the drawbacks such as high signaling cost and long handover latency. To reduce the signaling cost and the handover latency, we propose an enhanced authentication scheme in Fast handover for Proxy MIPv6. The proposed scheme reduces the handover latency and the signaling cost because the registration procedure and the authentication procedure are simultaneously performed. We also compare the proposed scheme with the conventional scheme in terms of the signaling cost and the handover latency.

Simple Mobility Management Protocol Based on P2P for Global IP Mobility Support (글로벌 IP 이동성 지원을 위한 P2P 기반 간단한 이동성 관리 프로토콜)

  • Chun, Seung-Man;Nah, Jae-Wook;Park, Jong-Tae
    • Journal of the Institute of Electronics Engineers of Korea TC
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    • v.48 no.12
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    • pp.17-27
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    • 2011
  • Most of the previous mobility management protocols such as IETF MIPv4/6 and its variants standardized by the IETF do not support global seamless handover because they require partially changes of the existing network infrastructure. In this article, we propose a simple mobility management protocol (SMMP) which can support global seamless handover between homogeneous or heterogeneous wireless networks. To do this, the SMMP employs separate location management function as DMMS to support global user and service mobility and the bidirectional tunnels are dynamically constructed to support seamless IP mobility by using the IEEE MIH extension server, which is extended the IEEE 802.21 MIH standards. The detailed architecture and functions of the SMMP have been designed. Finally, the mathematical analysis and the simulation have been done. The performance results show the proposed SMMP outperforms the existing MIPv6 and HMIPv6 in terms of handover latency, packet loss, pear signal noise ratio (PSNR).

MAC-Layer Error Control for Real-Time Broadcasting of MPEG-4 Scalable Video over 3G Networks (3G 네트워크에서 MPEG-4 스케일러블 비디오의 실시간 방송을 위한 실행시간 예측 기반 MAC계층 오류제어)

  • Kang, Kyungtae;Noh, Dong Kun
    • Journal of the Korea Society of Computer and Information
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    • v.19 no.3
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    • pp.63-71
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    • 2014
  • We analyze the execution time of Reed-Solomon coding, which is the MAC-layer forward error correction scheme used in CDMA2000 1xEV-DO broadcast services, under different air channel conditions. The results show that the time constraints of MPEG-4 cannot be guaranteed by Reed-Solomon decoding when the packet loss rate (PLR) is high, due to its long computation time on current hardware. To alleviate this problem, we propose three error control schemes. Our static scheme bypasses Reed-Solomon decoding at the mobile node to satisfy the MPEG-4 time constraint when the PLR exceeds a given boundary. Second, dynamic scheme corrects errors in a best-effort manner within the time constraint, instead of giving up altogether when the PLR is high; this achieves a further quality improvement. The third, video-aware dynamic scheme fixes errors in a similar way to the dynamic scheme, but in a priority-driven manner which makes the video appear smoother. Extensive simulation results show the effectiveness of our schemes compared to the original FEC scheme.

Design and Analysis of Multiple Mobile Router Architecture for In-Vehicle IPv6 Networks (차량 내 IPv6 네트워크를 위한 다중 이동 라우터 구조의 설계와 분석)

  • Paik Eun-Kyoung;Cho Ho-Sik;Choi Yang-Hee
    • The Journal of The Korea Institute of Intelligent Transport Systems
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    • v.2 no.2 s.3
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    • pp.43-54
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    • 2003
  • As the demand for ubiquitous mobile wireless Internet grows, vehicles are receiving a lot of attention as new networking platforms. The demand for 4G all-IP networks encourages vehicle networks to be connected using IPv6. By means of network mobility (NEMO) support, we can connect sensors, controllers, local ,servers as well as passengers' devices of a vehicle to the Internet through a mobile router. The mobile router provides the connectivity to the Internet and mobility transparency for the rest of the mobile nodes of an in-vehicle nv6 network. So, it is .important for the mobile router to assure reliable connection and a sufficient data rate for the group of nodes behind it. To provide reliability, this paper proposes an adaptive multihoming architecture of multiple mobile routers. Proposed architecture makes use of different mobility characteristics of different vehicles. Simulation results with different configurations show that the proposed architecture increases session preservation thus increases reliability and reduces packet loss. We also show that the proposed architecture is adaptive to heterogeneous access environment which provide different access coverage areas and data rates. The result shows that our architecture achieves sufficient data rates as well as session preservation.

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A Congestion Control Algorithm for the fairness Improvement of TCP Vegas (TCP Vegas의 공정성 향상을 위한 혼잡 제어 알고리즘)

  • 오민철;송병훈;정광수
    • Journal of KIISE:Information Networking
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    • v.31 no.3
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    • pp.269-279
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    • 2004
  • The most important factor influencing the robustness of the Internet Is the end-to-end TCP congestion control. However, the congestion control scheme of TCP Reno, the most popular TCP version on the Internet, employs passive congestion indication. It makes worse the network congestion. Recently, Brakmo and Peterson have proposed a new version of TCP, which is named TCP Vegas, with a fundamentally different congestion control scheme from that of the Reno. Many studies indicate that the Vegas is able to achieve better throughput and higher stability than the Reno. But there are two unfairness problems in Vegas. These problems hinder the spread of the Vegas in current Internet. In this paper, in order to solve these unfairness problems, we propose a new congestion control algorithm called TCP PowerVegas. The existing Vegas depends mainly only on the rtt(round trip time), but the proposed PowerVegas use the new congestion control scheme combined the Information on the rtt with the information on the packet loss. Therefore the PowerVegas performs the congestion control more competitively than the Vegas. Thus, the PowerVegas is able to solve effectively these unfairness problems which the Vegas has experienced. To evaluate the proposed approach, we compare the performance among PowerVegas, Reno and Vegas under same network environment. Using simulation, the PowerVegas is able to achieve better throughput and higher stability than the Reno and is shown to achieve much better fairness than the existing Vegas.