Raptorq-Based Multihop File Broadcast Protocol

The objective of this thesis is to describe and implement a RaptorQ broadcast protocol application layer designed for use in a wireless multihop network. The RaptorQ broadcast protocol is a novel application layer broadcast protocol based on RaptorQ forward error correction. This protocol can deliver a file reliably to a large number of nodes in a wireless multihop network even if the links have high loss rates. We use mixed integer programming with power balance constraints to construct broadcast trees that are suitable for implementing the RaptorQ-based broadcast protocol. The resulting broadcast tree facilitates deployment of mechanisms for verifying successful delivery. We use the Qualcomm proprietary RaptorQ software development kit library as well as a Ruby interface to implement the protocol. During execution, each node operates in one of main modes: source, transmitter, or leaf. Each mode has five different phases: STARTUP, FINISHING (Poll), FINISHING (Wait), FINISHING (Extra), and COMPLETED. Three threads are utilized to implement the RaptorQ-based broadcast protocol features. Thread 1 receives messages and passes them to the receive buffer. Thread 2 evaluates the received message, which can be NORM, POLL, MORE, and DONE, and passes the response message to the send buffer. Thread 3 multicasts the content of the send buffer. Results obtained by testing the implementation of the RaptorQ-based broadcast protocol demonstrate that efficient and reliable distribution of files over multihop wireless networks with a high link loss rates is feasible

Layer-Aware Forward Error Correction for Mobile Broadcast of Layered Media

The bitstream structure of layered media formats such as scalable video coding (SVC) or multiview video coding (MVC) opens up new opportunities for their distribution in Mobile TV services. Features like graceful degradation or the support of the 3-D experience in a backwards-compatible way are enabled. The reason is that parts of the media stream are more important than others with each part itself providing a useful media representation. Typically, the decoding of some parts of the bitstream is only possible, if the corresponding more important parts are correctly received. Hence, unequal error protection (UEP) can be applied protecting important parts of the bitstream more strongly than others. Mobile broadcast systems typically apply forward error correction (FEC) on upper layers to cope with transmission errors, which the physical layer FEC cannot correct. Today's FEC solutions are optimized to transmit single layer video. The exploitation of the dependencies in layered media codecs for UEP using FEC is the subject of this paper. The presented scheme, which is called layer-aware FEC (LA-FEC), incorporates the dependencies of the layered video codec into the FEC code construction. A combinatorial analysis is derived to show the potential theoretical gain in terms of FEC decoding probability and video quality. Furthermore, the implementation of LA-FEC as an extension of the Raptor FEC and the related signaling are described. The performance of layer-aware Raptor code with SVC is shown by experimental results in a DVB-H environment showing significant improvements achieved by LA-FEC. © 2011 IEEE.Hellge, C.; Gómez Barquero, D.; Schierl, T.; Wiegand, T. (2011). Layer-Aware Forward Error Correction for Mobile Broadcast of Layered Media. IEEE Transactions on Multimedia. 13(3):551-562. doi:10.1109/TMM.2011.2129499S55156213

Maximizing Cache Value For Distributing Content Via Small Cells In 5G

The objective of this thesis is to develop an algorithm for distributing content at small cells in 5G, wherein small cells (SCs) are set up instead of macro cells to serve mobile users. Users may simultaneously access between one and some maximum number of SCs, the actual number of SCs being drawn from a distribution. The source library is located away from these sites, and it stores files as a collection of RaptorQ-encoded symbols. By using RaptorQ symbols a very large number of distinct encoding symbols can be generated and a collection of received encoded symbols slightly larger than the number of source symbols can be used to recover the original file with linear time complexity. A number of these encoded symbols should be placed in SC caches to facilitate efficient download. The main objective of this thesis is to develop an algorithm of low complexity to determine the number of encoded symbols of each file that should be cached at each SCs as a function of cache size. Each file is characterized by the number of encoded symbols required to reconstruct the file at the user equipment and its download preference probability. The optimization problem considered is to determine the symbols distribution for a set of files stored in the library in order to minimize backhaul. The objective function is the average value of storing a set of encoded symbols per file download, constrained by available cache memory. Parameters are the files’ preference probabilities and sizes, coverage areas probabilities, the total number of files in the library, and the cache capacity. This study contributes to the literature by developing an n log n algorithm to solve the optimization problem, extending previous results from constant files size for all files to arbitrary actual files sizes for all files, and extending distribution portions from continuous fractions of files to integer number of symbols

Raptorqp2P: Maximize The Performance Of P2P File Distribution With Raptorq Coding

BitTorrent is the most popular Peer-to-Peer (P2P) file sharing system widely used for distributing large files over the Internet. It has attracted extensive attentions from both network operators and researchers for investigating its deployment and performance. For example, recent studies have shown that under steady state, its rarest first scheme with the tit-for-tat mechanism can work very effectively and make BitTorrent near optimal for the generic file downloading process. However, in practice, the highly dynamic network environment, especially the notorious user churns prevalently existing in most peer-to-peer systems, can severely degrade the downloading performance. In this thesis, we first study on the limitations of BitTorrent under dynamic network environments, focusing on two scenarios where with our preliminary modeling and analysis, we clearly identify how network dynamics and peer churns can significantly degrade the performance. With these findings, we further propose a novel protocol named RaptorQP2P, which is based on RaptorQ coding, to overcome the limitations of current BitTorrent design and maximize the performance of P2P file distribution. The new protocol features two levels of RaptorQ encoding. At the top layer, the entire file is RaptorQ encoded to yield a collection of source blocks and repair blocks, and then each source and repair block is RaptorQ encoded independently to yield a collection of source symbols and repair symbols for the block. The symbols are independently transferred among the peers and when a sufficient number of distinct symbols for a particular block have been received, whether source or repair, the block can be reconstructed. The file can be reconstructed using a sufficient arbitrary number of distinct blocks. Our results show that RaptorQP2P can well handle the network dynamics as well as peer churns and significantly shorten the downloading completion time by up to 41.4% with excellent scalability on both file size and user population

Understanding the characteristics of Internet traffic and designing an efficient RaptorQ-based data transport protocol for modern data centres

This thesis is the amalgamation of research on efficient data transport protocols for data centres and a comprehensive and systematic study of Internet traffic, which came as a result of the need to understand traffic patterns and workloads in modern computer networks. The first part of the thesis is on the development of efficient data transport pro- tocols for data centres. We study modern data transport protocols for data centres through large scale simulations using the OMNeT++ simulator. We developed and experimented with an OMNeT++ model of NDP. This has led to the identification of limitations of the state of the art and the formulation of research questions with respect to data transport protocols for modern data centres. The developed model includes an implementation of a Fat-tree topology and per-packet ECMP load bal- ancing. We discuss how we integrated the model with the INET Framework and validated it by running various experiments that test different model parameters and components. This work revealed limitations of NDP with respect to efficient one-to-many and many-to-one communication in data centres, which led to the de- velopment of SCDP, a novel and general-purpose data transport protocol for data centres that, in contrast to all other protocols proposed to date, natively supports one-to-many and many-to-one data communication, which is extremely common in modern data centres. SCDP does so without compromising on efficiency for short and long unicast flows. SCDP achieves this by integrating RaptorQ codes with receiver-driven data transport, in-network packet trimming and Multi-Level Feed- back Queuing (MLFQ); (1) RaptorQ codes enable efficient one-to-many and many- to-one data transport; (2) on top of RaptorQ codes, receiver- driven flow control, in combination with in-network packet trimming, enable efficient usage of network re- sources as well as multi-path transport and packet spraying for all transport modes. Incast and Outcast are eliminated; (3) the systematic nature of RaptorQ codes, in combination with MLFQ, enable fast, decoding-free completion of short flows. We extensively evaluated SCDP in a wide range of simulated scenarios with realistic data centre workloads. For one-to-many and many-to-one transport sessions, SCDP performs significantly better than NDP. For short and long unicast flows, SCDP performs equally well or better compared to NDP. In the second part of the thesis, we extensively study Internet traffic. Getting good statistical models of traffic on network links is a well-known, often-studied problem. A lot of attention has been given to correlation patterns and flow duration. The distribution of the amount of traffic per unit time is an equally important but less studied problem. We study a large number of traffic traces from many different networks including academic, commercial and residential networks using state-of-the-art statistical techniques. We show that the log-normal distribution is a better fit than the Gaussian distribution. We also investigate a second, heavy- tailed distribution and show that its performance is better than Gaussian but worse than log-normal. We examine anomalous traces which are a poor fit for all tested distributions and show that this is often due to traffic outages or links that hit maximum capacity. Stationarity tests showed that the traffic is stationary at some range of aggregation times. We demonstrate the utility of the log-normal distribution in two contexts: predicting the proportion of time traffic will exceed a given level (for link capacity estimation) and predicting 95th percentile pricing. We also show the log-normal distribution is a better predictor than Gaussian orWeibull distributions

Fulcrum: Flexible Network Coding for Heterogeneous Devices

Producción CientíficaWe introduce Fulcrum, a network coding framework that achieves three seemingly conflicting objectives: 1) to reduce the coding coefficient overhead down to nearly n bits per packet in a generation of n packets; 2) to conduct the network coding using only Galois field GF(2) operations at intermediate nodes if necessary, dramatically reducing computing complexity in the network; and 3) to deliver an end-to-end performance that is close to that of a high-field network coding system for high-end receivers, while simultaneously catering to low-end receivers that decode in GF(2). As a consequence of 1) and 3), Fulcrum has a unique trait missing so far in the network coding literature: providing the network with the flexibility to distribute computational complexity over different devices depending on their current load, network conditions, or energy constraints. At the core of our framework lies the idea of precoding at the sources using an expansion field GF(2 h ), h > 1, to increase the number of dimensions seen by the network. Fulcrum can use any high-field linear code for precoding, e.g., Reed-Solomon or Random Linear Network Coding (RLNC). Our analysis shows that the number of additional dimensions created during precoding controls the trade-off between delay, overhead, and computing complexity. Our implementation and measurements show that Fulcrum achieves similar decoding probabilities as high field RLNC but with encoders and decoders that are an order of magnitude faster.Green Mobile Cloud project (grant DFF-0602-01372B)Colorcast project (grant DFF-0602-02661B)TuneSCode project (grant DFF - 1335-00125)Danish Council for Independent Research (grant DFF-4002-00367)Ministerio de Economía, Industria y Competitividad - Fondo Europeo de Desarrollo Regional (grants MTM2012-36917-C03-03 / MTM2015-65764-C3-2-P / MTM2015-69138-REDT)Agencia Estatal de Investigación - Fondo Social Europeo (grant RYC-2016-20208)Aarhus Universitets Forskningsfond Starting (grant AUFF-2017-FLS-7-1

THE INFLUENCE OF THE PACKET SIZE ON END TO END DELAY OF VIDEO DATA CODED WITH RAPTORQ CODES AND NETWORK CODES IN VEHICULAR ADHOC NETWORKS

The transmission of video files in Vehicular Adhoc Networks (VANETs) has become very prevalent as commuters prefer video data during travel. The delay with which the data is received becomes very significant as video packets received after their scheduled deadlines become useless. The performance of the network may significantly be reduced on such packet drops especially with mobile networks. This work aims at the reduction of end to end delay of video packets by applying the two techniques- Network Coding (NC) and RaptorQ (RQ) codes. The techniques are implemented in four VANET scenarios and an extensive analysis is done by varying the packet sizes during the transmission of three files of various sizes. The End to End Delay (EED) and Packet Delivery Ratio (PDR) are measured and plotted for all scenarios. The results show the influence of packet size on these parameters considered and the suitability of the techniques applied. The observations also show that RQ proves better for smaller files and NC suits better when the file size increases

Error and Congestion Resilient Video Streaming over Broadband Wireless

In this paper, error resilience is achieved by adaptive, application-layer rateless channel coding, which is used to protect H.264/Advanced Video Coding (AVC) codec data-partitioned videos. A packetization strategy is an effective tool to control error rates and, in the paper, source-coded data partitioning serves to allocate smaller packets to more important compressed video data. The scheme for doing this is applied to real-time streaming across a broadband wireless link. The advantages of rateless code rate adaptivity are then demonstrated in the paper. Because the data partitions of a video slice are each assigned to different network packets, in congestion-prone wireless networks the increased number of packets per slice and their size disparity may increase the packet loss rate from buffer overflows. As a form of congestion resilience, this paper recommends packet-size dependent scheduling as a relatively simple way of alleviating the buffer-overflow problem arising from data-partitioned packets. The paper also contributes an analysis of data partitioning and packet sizes as a prelude to considering scheduling regimes. The combination of adaptive channel coding and prioritized packetization for error resilience with packet-size dependent packet scheduling results in a robust streaming scheme specialized for broadband wireless and real-time streaming applications such as video conferencing, video telephony, and telemedicine