Network Coding Meets TCP: Theory and Implementation

The theory of network coding promises significant benefits in network performance, especially in lossy networks and in multicast and multipath scenarios. To realize these benefits in practice, we need to understand how coding across packets interacts with the acknowledgment (ACK)-based flow control mechanism that forms a central part of today's Internet protocols such as transmission control protocol (TCP). Current approaches such as rateless codes and batch-based coding are not compatible with TCP's retransmission and sliding-window mechanisms. In this paper, we propose a new mechanism called TCP/NC that incorporates network coding into TCP with only minor changes to the protocol stack, thereby allowing incremental deployment. In our scheme, the source transmits random linear combinations of packets currently in the congestion window. At the heart of our scheme is a new interpretation of ACKs-the sink acknowledges every degree of freedom (i.e., a linear combination that reveals one unit of new information) even if it does not reveal an original packet immediately. Thus, our new TCP ACK rule takes into account the network coding operations in the lower layer and enables a TCP-compatible sliding-window approach to network coding. Coding essentially masks losses from the congestion control algorithm and allows TCP/NC to react smoothly to losses, resulting in a novel and effective approach for congestion control over lossy networks such as wireless networks. An important feature of our solution is that it allows intermediate nodes to perform re-encoding of packets, which is known to provide significant throughput gains in lossy networks and multicast scenarios. Simulations show that our scheme, with or without re-encoding inside the network, achieves much higher throughput compared to TCP over lossy wireless links. We present a real-world implementation of this protocol that addresses the practical aspects of incorporating network coding and decoding with TCP's wind ow management mechanism. We work with TCP-Reno, which is a widespread and practical variant of TCP. Our implementation significantly advances the goal of designing a deployable, general, TCP-compatible protocol that provides the benefits of network coding.National Science Foundation (U.S.) (Grant CNS-0627021)National Science Foundation (U.S.) (Grant CNS-0721491)National Science Foundation (U.S.) (Grant CCF-0915922)United States. Defense Advanced Research Projects Agency (Subcontract 18870740-37362-C)United States. Defense Advanced Research Projects Agency (Subcontract 060786)United States. Defense Advanced Research Projects Agency (Subcontract 069145)United States. Defense Advanced Research Projects Agency (Contract N66001-06-C-2020)Space and Naval Warfare Systems Center San Diego (U.S.) (Contract N66001- 08-C-2013

An optimized framework for header suppression of real time IPV6 traffic in multiprotocol label switching (MPLS) networks.

Pensuisan Label Multiprotokol (MPLS) dengan IPv6 telah dinyatakan oleh Pasukan Petugas Kejuruteraan Internet (IETF) sebagai mampu diskalakan dan sangat sesuai untuk jenis-jenis trafik yang berlainan seperti VoIP dan Video. Namun, kepala IP yang besar melahirkan overhed kepala yang berlebihan dalam rangkaian MPLS, mengakibatkan kesesakan trafik lalu menjejaskan prestasi rangkaian tulang belakang. Multiprotocol Label Switching (MPLS) with IPv6 has been defined by the Internet Engineering Task Force (IETF) as highly scalable and well suited for different types of traffic such as VoIP and Video. However, large IP headers create excessive header overhead in a MPLS network leading to traffic congestion degrading the backbone network performance

Scalable reliable on-demand media streaming protocols

This thesis considers the problem of delivering streaming media, on-demand, to potentially large numbers of concurrent clients. The problem has motivated the development in prior work of scalable protocols based on multicast or broadcast. However, previous protocols do not allow clients to efficiently: 1) recover from packet loss; 2) share bandwidth fairly with competing flows; or 3) maximize the playback quality at the client for any given client reception rate characteristics. In this work, new protocols, namely Reliable Periodic Broadcast (RPB) and Reliable Bandwidth Skimming (RBS), are developed that efficiently recover from packet loss and achieve close to the best possible server bandwidth scalability for a given set of client characteristics. To share bandwidth fairly with competing traffic such as TCP, these protocols can employ the Vegas Multicast Rate Control (VMRC) protocol proposed in this work. The VMRC protocol exhibits TCP Vegas-like behavior. In comparison to prior rate control protocols, VMRC provides less oscillatory reception rates to clients, and operates without inducing packet loss when the bottleneck link is lightly loaded. The VMRC protocol incorporates a new technique for dynamically adjusting the TCP Vegas threshold parameters based on measured characteristics of the network. This technique implements fair sharing of network resources with other types of competing flows, including widely deployed versions of TCP such as TCP Reno. This fair sharing is not possible with the previously defined static Vegas threshold parameters. The RPB protocol is extended to efficiently support quality adaptation. The Optimized Heterogeneous Periodic Broadcast (HPB) is designed to support a range of client reception rates and efficiently support static quality adaptation by allowing clients to work-ahead before beginning playback to receive a media file of the desired quality. A dynamic quality adaptation technique is developed and evaluated which allows clients to achieve more uniform playback quality given time-varying client reception rates

Streaming Video Performance and Enhancements in Resource-Constrained Wireless Networks

Streaming video is an increasingly popular application in wireless networks. The concept of a live streaming video yields several enticing possibilities: real-time video conferencing, television broadcasting, pay-per-view movie streaming, and more. These ideas have already been explored via the internet and have met with mixed success, largely due to the shortcomings of the underlying network. Taking streaming video to wireless networks, then, poses several significant challenges. Wireless networks are inherently more susceptible to failures and data corruption due to their unstable communications medium. This volatility suggests serious drawbacks for any implementation of streaming video. Video frame errors, jitter, and even complete sync loss are entirely conceivable in a wireless environment. Many of these issues have been undertaken and several approaches to mediation or even solution of these problems are underway. This thesis proposes to use advanced simulation techniques to properly exhaustively permute many vital parameters within a UMTS network and uncover, if they exist, bottlenecks in UMTS performance under considerable network load. This is accomplished via a described testing plan with simulation environment. Additionally this thesis proposes a new UDP-like transport layer specially optimized for streaming media over resource-constrained networks, tested to work with significant improvements under the UMTS cellular networking system. Finally this thesis provides several innovative new methods in the furtherance of the field of streaming media research in resourceconstrained and cellular environments. Overall this thesis makes several important contributes to an exciting and ever-growing field of active research and discussion

Modélisation et distribution adaptatives de grandes scènes naturelles

Cette thèse traite de la modélisation et la diffusion de grandes scènes 3D naturelles. Nous visons à fournir des techniques pour permettre à des utilisateurs de naviguer à distance dans une scène 3D naturelle, tout en assurant la cohérence botanique et l'interactivité. Tout d'abord, nous fournissons une technique de compression multi-résolution, fondée sur la normalisation, l'instanciation, la décorrélation, et sur le codage entropique des informations géometriques pour des modèles de plantes. Ensuite, nous étudions la transmission efficace de ces objets 3D. L'algorithme de paquétisation proposé fonctionne pour la plupart des représentations multi-résolution d'objet 3D. Nous validons les techniques de paquétisation par des expériences sur un WAN (Wide Area Network), avec et sans contrôle de congestion (Datagram Congestion Control Protocol). Enfin, nous abordons les questions du streaming au niveau de la scène. Nous optimisons le traitement des requêtes du côté serveur en fournissant une structure de données adaptée et nous préparons le terrain pour nos travaux futurs sur l'évolutivité et le déploiement de systèmes distribués de streaming 3D. ABSTRACT : This thesis deals with the modeling and the interactive streaming of large natural 3D scenes. We aim at providing techniques to allow the remote walkthrough of users in a natural 3D scene ensuring botanical coherency and interactivity.First, we provide a compact and progressive representation for botanically realistic plant models. The topological structure and the geometry of the plants are represented by generalized cylinders. We provide a multi-resolution compression scheme, based on standardization and instantiation, on difference-based decorrelation, and on entropy coding. Then, we study efficient transmission of these 3D objects. The proposed packetization scheme works for any multi-resolution 3D representation. We validate our packetization schemes with extensive experiments over a WAN (Wide Area Network), with and without congestion control (Datagram Congestion Control Protocol). Finally, we address issues on streaming at the scene-level. We optimize the viewpoint culling requests on server-side by providing an adapted datastructure and we prepare the ground for our further work on scalability and deployment of distributed 3D streaming systems

A Survey on Data Plane Programming with P4: Fundamentals, Advances, and Applied Research

With traditional networking, users can configure control plane protocols to match the specific network configuration, but without the ability to fundamentally change the underlying algorithms. With SDN, the users may provide their own control plane, that can control network devices through their data plane APIs. Programmable data planes allow users to define their own data plane algorithms for network devices including appropriate data plane APIs which may be leveraged by user-defined SDN control. Thus, programmable data planes and SDN offer great flexibility for network customization, be it for specialized, commercial appliances, e.g., in 5G or data center networks, or for rapid prototyping in industrial and academic research. Programming protocol-independent packet processors (P4) has emerged as the currently most widespread abstraction, programming language, and concept for data plane programming. It is developed and standardized by an open community and it is supported by various software and hardware platforms. In this paper, we survey the literature from 2015 to 2020 on data plane programming with P4. Our survey covers 497 references of which 367 are scientific publications. We organize our work into two parts. In the first part, we give an overview of data plane programming models, the programming language, architectures, compilers, targets, and data plane APIs. We also consider research efforts to advance P4 technology. In the second part, we analyze a large body of literature considering P4-based applied research. We categorize 241 research papers into different application domains, summarize their contributions, and extract prototypes, target platforms, and source code availability.Comment: Submitted to IEEE Communications Surveys and Tutorials (COMS) on 2021-01-2

Adaptive Modeling and Distribution of Large Natural Scenes

This thesis deals with the modeling and the interactive streaming of large natural 3D scenes. We aim at providing techniques to allow the remote walkthrough of users in a natural 3D scene ensuring botanical coherency and interactivity.First, we provide a compact and progressive representation for botanically realistic plant models. The topological structure and the geometry of the plants are represented by generalized cylinders. We provide a multi-resolution compression scheme, based on standardization and instantiation, on difference-based decorrelation, and on entropy coding. Then, we study efficient transmission of these 3D objects. The proposed packetization scheme works for any multi-resolution 3D representation. We validate our packetization schemes with extensive experiments over a WAN (Wide Area Network), with and without congestion control (Datagram Congestion Control Protocol). Finally, we address issues on streaming at the scene-level. We optimize the viewpoint culling requests on server-side by providing an adapted datastructure and we prepare the ground for our further work on scalability and deployment of distributed 3D streaming systems

Joint Source-Channel Coding for Image Transmission over Underlay Multichannel Cognitive Radio Networks

The increasing prominence of wireless applications exacerbates the problem of radio spectrum scarcity and promotes the usage of Cognitive Radio (CR) in wireless networks. With underlay dynamic spectrum access, CRs can operate alongside Primary Users, the incumbent of a spectrum band, as long as they limit the interference to the Primary Users below a certain threshold. Multimedia streaming transmissions face stringent Quality of Services constraints on top of the CR interference constraints, as some packets in the data stream have higher levels of importance and are the most vulnerable to packet loss over the channel. This raises a need for Unequal Error Protection (ULP) for multimedia streams transmissions, in which the channel encoder assigns different amount of error correction to different parts of the data stream, thereby protecting more the most valuable parts of the stream from packet loss problems. This research presents an end-to-end system setup for image transmission, utilizing ULP as part of a Joint Source-Channel Coding scheme over a multichannel CR network operating through underlay dynamic spectrum access. The setup features a Set Partitioning in Hierarchical Trees (SPIHT) source encoder, and Reed-Solomon forward error correction channel coding, and uses their properties to devise an ULP framework that maximizes the quality of the received image

MPLS AND ITS APPLICATION

Real-time and multimedia applications have grown enormously during the last few years. Such applications require guaranteed bandwidth in a packet switched networks. Moreover, these applications require that the guaranteed bandwidth remains available when a node or a link in the network fails. Multiprotocol Label Switching (MPLS) networks cater to these requirements without compromising scalability. Guaranteed service and protection against failures in an MPLS network requires backup paths to be present in the network. Such backup paths are computed and installed at the same time a primary is provisioned. This thesis explains the single-layer restoration routing by placing primary as well as backup paths in MPLS networks. Our focus will be on computing and establishing backup paths, and bandwidth sharing along such backup paths. We will start by providing a quick overview of MPLS routing. We will identify the elements and quantities that are significant to the understanding of MPLS restoration routing. To this end, we will introduce the information locally stored at MPLS nodes and information propagated through routing protocols, in order to assist in efficient restoration routing. L2VPNs and VPLS will also be covered in the end of this thesis. In the end SDN (software defined networks) will be introduced

Multicast communication support over satellite networks

In this dissertation, we focus on providing multicast communication support over satellite networks. We investigate the possible performance enhancements in terms of the throughput, capacity, and scalability of a Ka-band, multiple spot-beam satellite communication system that supports unicast and multicast services. The role satellite systems play in today's communication infrastructure is changing rapidly, fueled by the technological advance in the design of new satellite systems, and by the new multimedia service applications, such as on-demand multimedia content delivery, distance learning, and distributed software updates that would benefit from the wide-area coverage, direct and ubiquitous access capability of the satellite systems. These applications require concurrent transmission of the same content to multiple users. In order for multicasting-based services to grow over satellite networks, there must be an incentive to deploy them. We address the problem of user heterogeneity that occurs when multicast users that are located across several different spot-beam locations experience different channel conditions. We propose a novel power allocation scheme for smoothing out the heterogeneity experienced by the multicast groups, while making sure that unicast users get a fair share of system resources as well. Our power allocation scheme would benefit from user feedback in determining the channel conditions. However, collecting feedback from a large set of users is a challenging task in satellite systems, since access to the uplink bandwidth is to be shared between several users, and the resources are usually limited. We introduce a novel algorithm that reduces the volume of feedback information that is to be transmitted over the satellite segment of the network, while maintaining that the relevant information is collected in a timely manner. Finally, we focus our attention to the potential benefits of integrating packet level forward error correction coding to packet delivery for reliable multicast services over satellite networks. Forward error protection helps recover corrupted data, and minimizes the need for retransmissions over the satellite channel. We investigate the use of a special form of forward error correcting (FEC) code and couple it with an adaptive control mechanism to dynamically adjust the number of encoding packets forwarded to the users