Network coding-based survivability techniques for multi-hop wireless networks

Multi-hop Wireless Networks (MWN) have drawn a lot of attention in the last decade, and will continue to be a hot and active research area in the future also. MWNs are attractive because they require much less effort to install and operate (compared to wired networks), and provide the network users with the flexibility and convenience they need. However, with these advantages comes a lot of challenges. In this work, we focus on one important challenge, namely, network survivability or the network ability to sustain failures and recover from service interruption in a timely manner. Survivability mechanisms can be divided into two main categories; Protection and restoration mechanisms. Protection is usually favored over restoration because it usually provides faster recovery. However, the problem with traditional protection schemes is that they are very demanding and consume a lot of network resources. Actually, at least 50% of the used resources in a communication session are wasted in order to provide the destination with redundant information, which can be made use of only when a network failure or information loss occurs. To overcome this problem and to make protection more feasible, we need to reduce the used network resources to provide proactive protection without compromising the recovery speed. To achieve this goal, we propose to use network coding. Basically, network coding allows intermediate network nodes to combine data packets instead of just forwarding them as is, which leads to minimizing the consumed network resources used for protection purposes. In this work we give special attention to the survivability of many-to-one wireless flows, where a set of N sources are sending data units to a common destination T. Examples of such many-to-one flows are found in Wireless Mesh Networks (WMNs) or Wireless Sensor Networks (WSNs). We present two techniques to provide proactive protection to the information flow in such communication networks. First, we present a centralized approach, for which we derive and prove the sufficient and necessary conditions that allows us to protect the many-to-one information flow against a single link failure using only one additional path. We provide a detailed study of this technique, which covers extensions for more general cases, complexity analysis that proves the NP-completeness of the problem for networks with limited min-cuts, and finally performance evaluation which shows that in the worst case our coding-based protection scheme can reduce the useful information rate by 50% (i.e., will be equivalent to traditional protection schemes). Next, we study the implementation of the previous approach when all network nodes have single transceivers. In this part of our work we first present a greedy scheduling algorithm for the sources transmissions based on digital network coding, and then we show how analog network coding can further enhance the performance of the scheduling algorithm. Our second protection scheme uses deterministic binary network coding in a distributed manner to enhance the resiliency of the Sensors-to-Base information flow against packet loss. We study the coding efficiency issue and introduce the idea of relative indexing to reduce the coding coefficients overhead. Moreover, we show through a simulation study that our approach is highly scalable and performs better as the network size and/or number of sources increases. The final part of this work deals with unicast communication sessions, where a single source node S is transmitting data to a single destination node T through multiple hops. We present a different way to handle the survivability vs. bandwidth tradeoff, where we show how to enhance the survivability of the S-T information flow without reducing the maximum achievable S-T information rate. The basic idea is not to protect the bottleneck links in the network, but to try to protect all other links if possible. We divide this problem into two problems: 1) pre-cut protection, which we prove it to be NP-hard, and thus, we present an ILP and a heuristic approach to solve it, and 2) post-cut protection, where we prove that all the data units that are not delivered to T directly after the min-cut can be protected against a single link failure. Using network coding in this problem allows us to maximize the number of protected data units before and after the min-cut

CASPR: Judiciously Using the Cloud for Wide-Area Packet Recovery

We revisit a classic networking problem -- how to recover from lost packets in the best-effort Internet. We propose CASPR, a system that judiciously leverages the cloud to recover from lost or delayed packets. CASPR supplements and protects best-effort connections by sending a small number of coded packets along the highly reliable but expensive cloud paths. When receivers detect packet loss, they recover packets with the help of the nearby data center, not the sender, thus providing quick and reliable packet recovery for latency-sensitive applications. Using a prototype implementation and its deployment on the public cloud and the PlanetLab testbed, we quantify the benefits of CASPR in providing fast, cost effective packet recovery. Using controlled experiments, we also explore how these benefits translate into improvements up and down the network stack

Performance Modelling and Optimisation of Multi-hop Networks

A major challenge in the design of large-scale networks is to predict and optimise the total time and energy consumption required to deliver a packet from a source node to a destination node. Examples of such complex networks include wireless ad hoc and sensor networks which need to deal with the effects of node mobility, routing inaccuracies, higher packet loss rates, limited or time-varying effective bandwidth, energy constraints, and the computational limitations of the nodes. They also include more reliable communication environments, such as wired networks, that are susceptible to random failures, security threats and malicious behaviours which compromise their quality of service (QoS) guarantees. In such networks, packets traverse a number of hops that cannot be determined in advance and encounter non-homogeneous network conditions that have been largely ignored in the literature. This thesis examines analytical properties of packet travel in large networks and investigates the implications of some packet coding techniques on both QoS and resource utilisation. Specifically, we use a mixed jump and diffusion model to represent packet traversal through large networks. The model accounts for network non-homogeneity regarding routing and the loss rate that a packet experiences as it passes successive segments of a source to destination route. A mixed analytical-numerical method is developed to compute the average packet travel time and the energy it consumes. The model is able to capture the effects of increased loss rate in areas remote from the source and destination, variable rate of advancement towards destination over the route, as well as of defending against malicious packets within a certain distance from the destination. We then consider sending multiple coded packets that follow independent paths to the destination node so as to mitigate the effects of losses and routing inaccuracies. We study a homogeneous medium and obtain the time-dependent properties of the packet’s travel process, allowing us to compare the merits and limitations of coding, both in terms of delivery times and energy efficiency. Finally, we propose models that can assist in the analysis and optimisation of the performance of inter-flow network coding (NC). We analyse two queueing models for a router that carries out NC, in addition to its standard packet routing function. The approach is extended to the study of multiple hops, which leads to an optimisation problem that characterises the optimal time that packets should be held back in a router, waiting for coding opportunities to arise, so that the total packet end-to-end delay is minimised

Random Linear Network Coding for 5G Mobile Video Delivery

An exponential increase in mobile video delivery will continue with the demand for higher resolution, multi-view and large-scale multicast video services. Novel fifth generation (5G) 3GPP New Radio (NR) standard will bring a number of new opportunities for optimizing video delivery across both 5G core and radio access networks. One of the promising approaches for video quality adaptation, throughput enhancement and erasure protection is the use of packet-level random linear network coding (RLNC). In this review paper, we discuss the integration of RLNC into the 5G NR standard, building upon the ideas and opportunities identified in 4G LTE. We explicitly identify and discuss in detail novel 5G NR features that provide support for RLNC-based video delivery in 5G, thus pointing out to the promising avenues for future research.Comment: Invited paper for Special Issue "Network and Rateless Coding for Video Streaming" - MDPI Informatio

Multimedia delivery in the future internet

The term “Networked Media” implies that all kinds of media including text, image, 3D graphics, audio and video are produced, distributed, shared, managed and consumed on-line through various networks, like the Internet, Fiber, WiFi, WiMAX, GPRS, 3G and so on, in a convergent manner [1]. This white paper is the contribution of the Media Delivery Platform (MDP) cluster and aims to cover the Networked challenges of the Networked Media in the transition to the Future of the Internet. Internet has evolved and changed the way we work and live. End users of the Internet have been confronted with a bewildering range of media, services and applications and of technological innovations concerning media formats, wireless networks, terminal types and capabilities. And there is little evidence that the pace of this innovation is slowing. Today, over one billion of users access the Internet on regular basis, more than 100 million users have downloaded at least one (multi)media file and over 47 millions of them do so regularly, searching in more than 160 Exabytes1 of content. In the near future these numbers are expected to exponentially rise. It is expected that the Internet content will be increased by at least a factor of 6, rising to more than 990 Exabytes before 2012, fuelled mainly by the users themselves. Moreover, it is envisaged that in a near- to mid-term future, the Internet will provide the means to share and distribute (new) multimedia content and services with superior quality and striking flexibility, in a trusted and personalized way, improving citizens’ quality of life, working conditions, edutainment and safety. In this evolving environment, new transport protocols, new multimedia encoding schemes, cross-layer inthe network adaptation, machine-to-machine communication (including RFIDs), rich 3D content as well as community networks and the use of peer-to-peer (P2P) overlays are expected to generate new models of interaction and cooperation, and be able to support enhanced perceived quality-of-experience (PQoE) and innovative applications “on the move”, like virtual collaboration environments, personalised services/ media, virtual sport groups, on-line gaming, edutainment. In this context, the interaction with content combined with interactive/multimedia search capabilities across distributed repositories, opportunistic P2P networks and the dynamic adaptation to the characteristics of diverse mobile terminals are expected to contribute towards such a vision. Based on work that has taken place in a number of EC co-funded projects, in Framework Program 6 (FP6) and Framework Program 7 (FP7), a group of experts and technology visionaries have voluntarily contributed in this white paper aiming to describe the status, the state-of-the art, the challenges and the way ahead in the area of Content Aware media delivery platforms