21 research outputs found

    Robust streaming in delay tolerant networks

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    Delay Tolerant Networks (DTN) do not provide any end to end connectivity guarantee. Thus, transporting data over such networks is a tough challenge as most of Internet applications assume a form of persistent end to end connection. While research in DTN has mainly addressed the problem of routing in various mobility contexts with the aim to improve bundle delay delivery and data delivery ratio, little attention has been paid to applications. This paper investigates the support of streaming-like applications over DTN. We identify how DTN characteristics impact on the overall performances of these applications and present Tetrys, a transport layer mechanism, which enables robust streaming over DTN. Tetrys is based on an on the fly coding mechanism able to ensure full reliability without retransmission and fast in-order bundle delivery in comparison to classical erasure coding schemes. We evaluate our Tetrys prototype on real DTN connectivity traces captured from the Rollerblading tour in Paris. Simulations show that on average, Tetrys clearly outperforms all other reliability schemes in terms of bundles delivery service

    A packet error recovery scheme for vertical handovers mobility management protocols

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    Mobile devices are connecting to the Internet through an increasingly heterogeneous network environment. This connectivity via multiple types of wireless networks allows the mobile devices to take advantage of the high speed and the low cost of wireless local area networks and the large coverage of wireless wide area networks. In this context, we propose a new handoff framework for switching seamlessly between the different network technologies by taking advantage of the temporary availability of both the old and the new network technology through the use of an "on the fly" erasure coding method. The goal is to demonstrate that our framework, based on a real implementation of such coding scheme, 1) allows the application to achieve higher goodput rate compared to existing bicasting proposals and other erasure coding schemes; 2) is easy to configure and as a result 3) is a perfect candidate to ensure the reliability of vertical handovers mobility management protocols. In this paper, we present the implementation of such framework and show that our proposal allows to maintain the TCP goodput(with a negligible transmission overhead) while providing in a timely manner a full reliability in challenged conditions

    On-the-Fly Coding for Real-Time Applications

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    Although ironically it does not offer any real-time guarantee, Internet is a popular solution to support multimedia time-constrained applications (e.g. VoIP, Video Conferencing, ...). Following this trend, this paper focuses on the performance of these applications by studying the benefit of using a novel reliability concept which aims at signifi- cantly improving the performance of these time constrained applications over lossy best-effort networks. This reliability mechanism emerged from several recent works from both network and coding theories. Its principle is to integrate feedbacks in an on-the fly coding scheme in order to optimize the trade-off ”packet decoding delay” vs ”throughput”. We present the first evaluations of this mechanism for VoIP and video-conferencing applications for various erasure channels. Compared to classic block-based erasure codes, the results show significant gains in terms of quality observed by the user for both applications

    eCMT-SCTP: Improving Performance of Multipath SCTP with Erasure Coding Over Lossy Links

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    Performance of transport protocols on lossy links is a well-researched topic, however there are only a few proposals making use of the opportunities of erasure coding within the multipath transport protocol context. In this paper, we investigate performance improvements of multipath CMT-SCTP with the novel integration of the on-the-fly erasure code within congestion control and reliability mechanisms. Our contributions include: integration of transport protocol and erasure codes with regards to congestion control; proposal for a variable retransmission delay parameter (aRTX) adjustment; performance evaluation of CMT-SCTP with erasure coding with simulations. We have implemented the explicit congestion notification (ECN) and erasure coding schemes in NS-2, evaluated and demonstrated results of improvement both for application goodput and decline of spurious retransmission. Our results show that we can achieve from 10% to 80% improvements in goodput under lossy network conditions without a significant penalty and minimal overhead due to the encoding-decoding process

    A packet error recovery scheme for vertical handovers mobility management protocols

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    Mobile devices are connecting to the Internet through an increasingly heterogeneous network environment. This connectivity via multiple types of wireless networks allows the mobile devices to take advantage of the high speed and the low cost of wireless local area networks and the large coverage of wireless wide area networks. In this context, we propose a new handoff framework for switching seamlessly between the different network technologies by taking advantage of the temporary availability of both the old and the new network technology through the use of an “on the fly” erasure coding method. The goal is to demonstrate that our framework, based on a real implementation of such coding scheme, 1) allows the application to achieve higher goodput rate compared to existing bicasting proposals and other erasure coding schemes; 2) is easy to configure and as a result 3) is a perfect candidate to ensure the reliability of vertical handovers mobility management protocols. In this paper, we present the implementation of such framework and show that our proposal allows to maintain the TCP goodput (with a negligible transmission overhead) while providing in a timely manner a full reliability in challenged conditions

    On the Limit of Fountain MDC Codes for Video Peer-To-Peer Networks

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    Video streaming for heterogeneous types of devices, where nodes have different devices characteristics in terms of computational capacity and display, is usually handled by encoding the video with different qualities. This is not well suited for Peer-To-Peer (P2P) systems, as a single peer group can only share content of the same quality, thus limiting the peer group size and efficiency. To address this problem, several existing works propose the use of Multiple Descriptions Coding (MDC). The concept of this type of video codec is to split a video in a number of descriptions which can be used on their own, or aggregated to improve the global quality of the video. Unfortunately existing MDC codes are not flexible, as the video is split in a defined number of descriptions. In this paper, we focus on the practical feasibility of using a Fountain MDC code with properties similar to existing Fountain erasure codes, including the ability to create any number of descriptions when needed (on the fly). We perform simulations using selected pictures to assess the feasibility of using these codes, knowing that they should improve the availability of the video pieces in a P2P system and hence the video streaming quality. We observe that, although this idea seems promising, the evaluated benefits, demonstrated by the PSNR values, are limited when used in a real P2P video streaming system

    Tetrys : Un mécanisme de fiabilisation polyvalent

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    Actuellement, une succession d’erreurs qui ne peut être masquée par le mécanisme de fiabilisation (code à effacement) doit attendre au minimum un RTT pour être corrigée, ce qui n’est souvent pas satisfaisant pour les application temps-réel. Les concepts apportés par la théorie du codage réseau (Network Coding) permettent aujourd’hui de combler le fossé entre fiabilisation et fiabilité totale en s'abstrayant du concept d’ARQ. Cet article présente un mécanisme innovant nommé Tetrys dont l’une des caractéristiques est de pouvoir reconstruire les pertes dans un temps paramétrable et indépendant du RTT. A notre meilleure connaissance, c'est la première fois que les propriétés temps réel d'un tel mécanisme sont énoncées et étudiées. Intuitivement, les applications ciblées sont celles nécessitant une fiabilité totale avec contrainte de délai. Il s’avère qu’à taux de redondance égal, des applications telles que la VoIP et la vidéo-conférence sont bien plus performantes lorsque les flux sont protégés par le mécanisme Tetrys que par les mécanismes FEC ou H-ARQ classiques. Après un rapide rappel des points clés relatifs à FEC et H-ARQ, nous décrivons le principe de Tetrys et montrons son possible déploiement. Nous comparons les performances de FEC, H-ARQ et Tetrys du point de vue applicatif à l'aide d’un prototype et suivant des métriques de délai et dans le cadre de la VoIP, de qualité de la transmission (MOS)

    When on-the-fly erasure code makes late video decoding happen

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    In this paper, we proposed the late decoding solution based on Tetrys (called LD-Tetrys) to deal with delay constrained applications. Our analysis showed that LD-Tetrys fits well to the requirements for late video decoding while the other schemes (e.g., FEC, HARQ) do not. We also developed an evaluation framework which is independent of video codec and network topology. Simulation results acknowledge that LD-Tetrys' performance is better than the normal decoding with Tetrys, the original Tetrys. Furthermore, LD-Tetrys consistently outperforms the traditional block based erasure codes such as AL-FEC in terms of video quality. For future work, we are working on the theoretical modeling and analysis. We also expect to perform extensive experiments to obtain a complete evaluation

    On-the-fly erasure coding for real-time video applications

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    This paper introduces a robust point-to-point transmission scheme: Tetrys, that relies on a novel on-the-fly erasure coding concept which reduces the delay for recovering lost data at the receiver side. In current erasure coding schemes, the packets that are not rebuilt at the receiver side are either lost or delayed by at least one RTT before transmission to the application. The present contribution aims at demonstrating that Tetrys coding scheme can fill the gap between real-time applications requirements and full reliability. Indeed, we show that in several cases, Tetrys can recover lost packets below one RTT over lossy and best-effort networks. We also show that Tetrys allows to enable full reliability without delay compromise and as a result: significantly improves the performance of time constrained applications. For instance, our evaluations present that video-conferencing applications obtain a PSNR gain up to 7dB compared to classic block-based erasure codes
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