Accurate estimation of the signal power received at a given location can be obtained through Ray Tracing (RT) propagation model. Nevertheless, processes involved in RT are extremely complex and not adapted for Radio Frequency (RF) mapping purposes. Particularly, the reception test process, responsible for validating received rays at given locations, contributes to over 98% of the overall RT complexity in large-scale scenarios. In this paper, we introduce a novel RT algorithm able to divide by a factor of 50 the RF map rendering time while limiting the difference in bitrate estimation to less than 1Mbps on average as compared to existing approaches. Our algorithm integrates hardware acceleration techniques for RT, proposes a new modeling of the terrain in the form of a triangular mesh, and introduces a new reception test able to efficiently calculate the signal power received per each triangle of the mesh. Our simulations show how these processes together can reduce the complexity of RF map generation, hence allowing the algorithm to scale to large-scale and precise scenarios.

Ray Tracing is an electromagnetic wave propagation modeling approach used for accurate generation of Quality of Service (QoS) maps in mobile networks. Due to its complexity, current implementation of Ray Tracing fails to generate such maps in wide areas. In this paper, we propose an optimization to Ray Tracing able to accurately generate QoS maps in a reasonable time. Using a site-specific ray launching technique and an alternative to the reception test process, we divide by almost 1200 the execution time of Ray Tracing with less than 2% of memory usage as compared to baseline solutions.

Ray Tracing is a propagation modelling approach that accurately estimates the signal power received by end users while taking into account the details of the environment in their vicinity. This accuracy is at the cost of high computational load and high memory consumption due to the heavy computation performed by processes such as Ray Generation. In this paper, we introduce a site-specific ray generation technique able to generate up to 1 million rays within 5 seconds and a root mean square error for bandwidth estimation within 2 Mbps. Depending on the location of the antenna and the coverage area, our technique gives the minimum possible number of rays required in order to estimate end-users' signal power received and their download bitrate.

Traditional loss-based Congestion Control Algorithms (CCAs) suffer from performance issues over wireless networks mostly due to their inability to distinguish wireless random losses from congestion losses. Different loss discrimination algorithms have been proposed to tackle this issue but they are not efficient for 4G networks since they do not consider the impact of various link layer mechanisms such as adaptive modulation and coding and retransmission techniques on congestion in LTE Radio Access Networks (RANs). We propose MELD (MEC-based Edge Loss Discrimination), a novel server-side loss discrimination mechanism that leverages recent advancements in Multi-access Edge Computing (MEC) services to discriminate packet losses based on real-time RAN statistics. Our approach collects the relevant radio information via MEC's Radio Network Information Service and uses it to correctly distinguish random losses from congestion losses. Our experimental study made with the QUIC transport protocol shows over 80% higher goodput when MELD is used with NewReno and 8% higher goodput when used with Cubic.

To handle the ever growing demand of resource intensive experiments distributed, network emulation tools such as Mininet and Maxinet have been proposed. They automatically allocate experimental resources. In this work, we show that resources are poorly allocated, leading to resource overloading and hence to dubious experimental results.This is why we propose and implement a new placement module for distributed emulation. Our algorithms take into account both link and node resources and minimize the number of physical hosts needed to carry out the emulation. Through extensive numerical evaluations, simulations, and actual experiments, we show that our placement methods outperform existing ones and allowing to re-establish trust in experimental results.

5G enhanced Mobile broadband (eMBB) aims to provide users with a peak data rate of 20 Gbps in the Radio Access Network (RAN). However, since most Congestion Control Algorithms (CCAs) rely on startup and probe phases to discover the bottleneck bandwidth, they cannot quickly utilize the available RAN bandwidth and adapt to fast capacity changes without introducing large delay increase, especially when multiple flows are sharing the same Radio Link Control (RLC) buffer. To tackle this issue, we propose RAPID, a RAN-aware proxy-based flow control mechanism that prevents CCAs from overshooting more than the available RAN capacity while allowing near optimal link utilization. Based on analysis of up-to-date radio information using Multi-access Edge Computing (MEC) services and packet arrival rates, RAPID is able to differentiate slow interactive flows from fast download flows and allocate the available bandwidth accordingly. Our experiments with concurrent Cubic and BBR flows show that RAPID can reduce delay increase by a factor of 10 to 50 in both Line-of-Sight (LOS) and Non-LOS (NLOS) conditions while preserving high throughput.

Many companies and organizations are moving their applications from on-premises data centers to the cloud. The cloud infrastructures can potentially provide an infinite amount of computation (e.g., Elastic Compute) and storage (e.g., Simple Service Storage). In addition, all cloud providers propose different offers: IaaS, PaaS, and SaaS. This demo focuses on the IaaS services, presenting a simple tool to measure the network delay in a virtual infrastructure built entirely in the cloud. These measurements are useful for organizations that are moving current applications to, or creating new applications in, the cloud, but have requirements on the maximum, or average, network delay that these applications can tolerate. We present CloudTrace, a simple CLI tool that creates regional and multiregional experiments to measure delay, using Amazon AWS.

In this article, we introduce a novel dynamic controller assignment algorithm targeting connected vehicle services and applications, also known as Internet of Vehicles (IoV). The proposed approach considers a hierarchically distributed control plane, decoupled from the data plane, and uses vehicle location and control traffic load to perform controller assignment dynamically. We model the dynamic controller assignment problem as a multi-agent Markov game and solve it with cooperative multi-agent deep reinforcement learning. Simulation results using real-world vehicle mobility traces show that the proposed approach outperforms existing ones by reducing control delay as well as packet loss.

Terrestrial infrastructure-based wireless networks do not always guarantee their resources will be shared uniformly by nodes in vehicular networks. This is due mainly to the uneven and dynamic geographical distribution of vehicles and path loss effects. In this paper, we leverage multiple fifth-generation (5G) unmanned aerial vehicles (UAVs) to enhance fairness in network resource allocation among vehicles by positioning UAVs on-demand as “flying communication infrastructure”. We propose a deep reinforcement learning (DRL) approach to determine UAVs’ position to improve network resource allocation fairness and efficiency while considering the UAVs’ flying range, communication range, and energy constraints. We use a parametric fairness function to attain a number of resource allocation objectives ranging from maximizing the total throughput of vehicles, maximizing minimum throughput, and achieving proportional bandwidth allocation. Simulation results show that the proposed DRL approach to UAV positioning can improve network resource allocation according to the targeted fairness objective.

There has been a considerable increase in the demand for high quality mobile video streaming services, while at the same time, the video traffic volume is expected to grow exponentially. Consequently, maintaining high quality of experience (QoE) and saving network resources are becoming crucial challenges to solve. In this paper, we propose a name-based mobile streaming scheme that allows efficient video content delivery by exploiting a smart pulling mechanism designed for information-centric networks (ICNs). The proposed mechanism enables fast packet loss recovery by leveraging in-network caching and coding. Through an experimental evaluation of our mechanism over an open wireless testbed and the Internet, we demonstrate that the proposed scheme leads to higher QoE levels than classical ICN and TCP-based streaming mechanisms.

ISP networks are taking a leap forward thanks to emerging technologies such as Software Defined Networking (SDN) and Network Function Virtualization (NFV). Efficient algorithms considered too hard to be put in practice on legacy networks now have a second chance to be considered again. In this context, we rethink the ISP network dimensioning problem with protection against Shared Risk Link Group (SLRG) failures. We consider a path-based protection scheme with a global rerouting strategy in which, for each failure situation, we may have a new routing of all the demands. Our optimization task is to minimize the needed amount of bandwidth. We develop a scalable mathematical model that we handle using the Column Generation technique. We show the effectiveness of our methods and demonstrate the feasibility of our approach using Mininet.

In this paper, we propose a network planning and provisioning framework that optimizes deployment cost in C-RAN based 5G networks. Our framework is based on a Mixed Integer Quadratically Constrained Programming (MIQCP) model which optimizes ``virtualized'' 5G service chain deployment cost while performing adequate provisioning to address user demand and performance requirements. We use two realistic scenarios to showcase that our framework can be applied to different types of deployments and discuss the computational cost and scalability of our solution.

In this paper, we build upon our prior work on D2-ITS, a flexible and extensible framework to dynamically distribute network control to enable message dissemination in Intelligent Transport Systems (ITS), and extend it with handover and load balancing capabilities. More specifically, D2-ITS' new handover feature allows a controller to automatically "delegate" control of a vehicle to another controller as the vehicle moves. Control delegation can also be used as a way to balance load among controllers and ensure that required application quality of service is maintained. We showcase D2-ITS' handover and load- balancing features using the Mininet-Wifi network simula- tor/emulator. Our preliminary experiments show D2-ITS' ability to seamlessly handover control of vehicles as they move.

We propose D2-ITS, a flexible and extensible framework to dynamically distribute network control to enable message dissemination in Intelligent Transport Systems (ITS). By decoupling the control-from the data plane, D2-ITS leverages network programmability to address ITS scalability, delay intolerance and decentralization. It uses a distributed control plane based on a hierarchy of controllers that can dynamically adjust to environment- and network conditions in order to satisfy ITS application requirements. We demonstrate the benefits of D2-ITS through a proof-of-concept prototype using the ns-3 simulation platform. Results indicate lower message delivery latency with minimal additional overhead.

New applications where anyone can broadcast video are becoming very popular on smartphones. With the advent of high definition video, ISP providers may take the opportunity to propose new high quality broadcast services to their clients. Because of its centralized control plane, Software Defined Networking (SDN) seems an ideal way to deploy such a service in a flexible and bandwidth-efficient way. But deploying large scale multicast services on SDN requires smart group membership management and a bandwidth reservation mechanism to support QoS guarantees that should neither waste bandwidth nor impact too severely best effort traffic. In this paper, we propose a Network Function Virtualization based solution for Software Defined ISP networks to implement scalable multicast group management. Then, we propose the Lazy Load balancing Multicast (L2BM) routing algorithm for sharing the network capacity in a friendly way between guaranteed-bandwidth multicast traffic and best-effort traffic. Our implementation of the framework made on Floodlight controllers and Open vSwitches is used to study the performance of L2BM.

Owing to the rapid growth in high-quality video streaming over the Internet, preserving high-level robustness against data loss and low latency, while maintaining higher data transmission rates, is becoming an increasingly important issue for high-quality real-time delay-sensitive streaming. In this paper, we propose a low latency, low loss streaming mechanism, L4C2, specialized for high-quality delay-sensitive streaming. With L4C2, nodes in the network estimate the acceptable delay and packet loss probability in their uplinks, aiming at retrieving lost data packets from in-network cache and/or coded data packets using in-network coding within an acceptable delay, by extending the Content-Centric Networking (CCN) approach. Further, L4C2 naturally provides multiple paths and multicast technologies to efficiently utilize network resources while sharing network resources fairly with competing data flows by adjusting the video quality when necessary. We validate through comprehensive simulations that L4C2 achieves a high success probability of data transmission considering the acceptable one-way delay, and higher QoE while suppressing the interest and redundant data traffic than the proposed multipath congestion control mechanism in CCN.

Software-Defined Wireless Networking (SDWN) is an emerging approach based on decoupling radio control functions from the radio data plane through programmatic interfaces. Despite diverse ongoing efforts to realize the vision of SDWN, many questions remain open from multiple perspectives such as means to rapid prototype and experiment candidate software solutions applicable to real-world deployments. To this end, emulation of SDWN has the potential to boost research and development efforts by re-using existing protocol and application stacks while mimicking the behavior of real wireless networks. In this article, we provide an in-depth discussion on that matter focusing on the Mininet-WiFi emulator design to fill a gap in the experimental platform space. We showcase the applicability of our emulator in an SDN wireless context by illustrating the support of a number of use cases aiming to address the question on how far we can go in realistic SDWN experiments, including comparisons with the results obtained in a wireless testbed. Finally, we discuss the ability to replay packet-level and radio signal traces captured in the real testbed toward a virtual yet realistic emulation environment in support of SDWN research.

New applications where anyone can broadcast high quality video are becoming very popular. Internet services providers (ISPs) may take the opportunity to propose new high quality multicast services to their clients. Because of its centralized control plane, software defined networking (SDN) enables the deployment of such a service in a flexible and bandwidth-efficient way. But deploying large-scale multicast services on SDN requires smart group membership management and a bandwidth reservation mechanism with QoS guarantees that should neither waste bandwidth nor impact too severely best effort traffic. In this paper, we propose: 1) a scalable multicast group management mechanism based on a network function virtualization approach for software defined ISP networks to implement and deploy multicast services on the network edge and 2) the lazy load balancing multicast (L2BM) routing algorithm for sharing the core network capacity in a friendly way between guaranteed-bandwidth multicast traffic and best-effort traffic and that does not require costly real-time monitoring of link utilization. We have implemented the mechanism and algorithm, and evaluated them both in a simulator and a testbed. In the testbed, we experimented the group management at the edge and L2BM in the core with an open vSwitch-based QoS framework and evaluated the performance of L2BM with an exhaustive set of experiments on various realistic scenarios. The results show that L2BM outperforms other state-of-the art algorithms by being less aggressive with best-effort traffic and accepting about 5%-15% more guaranteed-bandwidth multicast join requests.

The Software-Defined Networking approach permits to realize new policies. In OpenFlow in particular, a controller decides on behalf of the switches which forwarding rules must be installed and where. However with this flexibility comes the challenge of the computation of a rule allocation matrix meeting both high-level policies and the network constraints such as memory or link capacity limitations. Nevertheless, in many situations (e.g., data-center networks), the exact path followed by packets does not severely impact performances as long as packets are delivered according to the endpoint policy. It is thus possible to deviate part of the traffic to alternative paths so to better use network resources without violating the endpoint policy. In this paper, we propose a linear optimization model of the rule allocation problem in resource constrained OpenFlow networks with relaxing routing policy. We show that the general problem is NP-hard and propose a polynomial time heuristic, called OFFICER, which aims to maximize the amount of carried traffic in under-provisioned networks. Our numerical evaluation on four different topologies shows that exploiting various paths allows to increase the amount of traffic supported by the network without significantly increasing the path length.

Nowadays, content retrieval is marking the Internet usage. User communications are no longer tied up to host interconnection. Information Centric Networking (ICN) models are proposed to cope with these changes. The new paradigm redesigns the Internet architecture to bring out content to the first level. Over the last decade, many key projects have proposed a large solution spectrum to rebuilt networking primitives focused on the content. One important and direct challenge of this shift is the large amount of routing states due to identifying contents rather than hosts. In this paper, we focus especially on DONA, one of the first ICN architecture, and analyse the required memory space to store routing states. Our study shows that today's technologies are not able to satisfy the content routing needs. Thus, we propose an enhancement of DONA called BADONA to deal with this problem. It uses the Bloom filter to drastically reduce the usage of the memory space. Finally, we evaluate our proposal performances to underscore its contribution.

Motivated by the internets of the future, which will likely be considerably larger in size as well as highly heterogeneous and decentralized, we propose Decentralize-SDN, D-SDN, a framework that enables not only physical- but also logical distribution of the Software-Defined Networking (SDN) control plane. D-SDN accomplishes network control distribution by defining a hierarchy of controllers that can “match” an internet's organizational- and administrative structure. By delegating control between main controllers and secondary controllers, D-SDN is able to accommodate administrative decentralization and autonomy.It incorporates security as an integral part of the framework. This paper describes D-SDN and presents two use cases, namely network capacity sharing and public safety network services.

We propose to demonstrate Direct Code Execution (DCE), a framework that enables to execute nearly unmodified applications and Linux Kernel code jointly with the ns-3 simulator. DCE allows therefore fully deterministic reproducibility of network experiments. DCE also supports larger scale scenarios than real-time emulators by using simulation time dilatation. In this demonstration, we will showcase two main scenarios: (1) a basic example describing how to integrate in DCE the Data Center TCP (DCTCP) Linux kernel patch, and then how to customize this protocol and run it on different scenarios; (2) a more advanced use case demonstrating how to benefit from DCE to build a rich and realistic evaluation environment for Software Defined Wireless Networks based on Open vSwitch and the NOX SDN controller.

We describe the first capability, to our knowledge, to execute nearly unmodified applications and Linux kernel code in the context of a widely-used open source discrete event networking simulator (ns-3). We propose Direct Code Execution (DCE), a framework that dramatically increases the number of available protocol models and realism available for ns-3 simulations. DCE meets the goals recently proposed for fully reproducible networking research and runnable papers, with the added benefits of 1) the ability of completely deterministic reproducibility, 2) the scalability that simulation time dilation offers, 3) capabilities supporting automated code coverage analysis, and 4) improved debuggability via execution within a single address space. In this paper, we describe in detail DCE, report on packet processing benchmarks and showcase key features of the framework with different use cases. We reproduce a previously published Multipath TCP (MPTCP) experiment and highlight how code coverage testing can be automated by showing results achieving 55-86% coverage of the MPTCP implementation. Then we demonstrate how network stack debugging can be easily performed and reproduced across a distributed system. Our first benchmarks are promising and we believe this framework can benefit the network community by enabling realistic, reproducible experiments and runnable papers.

This paper proposes a novel approach to capacity sharing in hybrid networked environments, i.e., environments that consist of infrastructure-based as well as infrastructureless networks. The proposed framework is based on Software-Defined Networking (SDN) and provides flexible, efficient, and secure capacity sharing solutions in a variety of hybrid network scenarios. In this paper, we describe the challenges raised by capacity sharing in hybrid networks, describe our framework in detail and how it addresses these challenges, and discuss implementation issues. To the best of our knowledge, this is the first SDN-based capacity sharing solution that targets hybrid networks and that incorporates security as an integral part of the proposed approach.

Content-Centric Networking (CCN) is designed for efficient content dissemination and supports caching contents on the path from content providers to content consumers to improve user experience and reduce costs. However, this strategy is not optimal inside a domain. In this paper, we propose a solution to improve caching in CCN using a Software-Defined Networking approach.

Emerging challenged networks require new protocols and strategies to cope with a high degree of mobility, high delays and unknown, possibly non-existing routes within the network. Researchers have proposed different store-carry-and-forward protocols for data delivery in challenged networks. These have been complemented with appropriate drop and scheduling policies that deal with the limitations of the nodes' buffers and the limited duration of opportunistic encounters in these networks. Nevertheless, the vast majority of these protocols and strategies are designed for end-to-end transmissions. Yet, a paradigm shift from the traditional way of addressing the endpoints in the network has been occurring towards content-centric networking. To this end, we present CEDO, a content-centric dissemination algorithm for challenged networks. CEDO aims at maximizing the total delivery-rate of distributed content in a setting where a range of contents of different popularity may be requested and stored, but nodes have limited resources. It achieves this by maintaining a delivery-rate utility per content that is proportional to the content miss rate and that is used by the nodes to make appropriate drop and scheduling decisions. This delivery-rate utility can be estimated locally by each node using unbiased estimators fed by sampled information on the mobile network obtained by gossiping. Both simulations and theory suggest that CEDO achieves its set goal, and outperforms a baseline LRU-based policy by 72%, even in relatively small scenarios. The framework followed by CEDO is general enough to be applied to other global performance objectives as well.

Delivery of multimedia digital content through wireless networks poses non-trivial performance issues because of scarcity and volatility of the wireless medium. The goal of this study is to analyze the impact of multipath fading and interference on multicast video streaming in an indoor LOS (Line of Sight) wireless environment using off-the-shelf fixed WiFi equipment. Multipath fading experienced by the receivers is represented by empirically estimating ricean K-factor from packet traces captured at the receivers. Degradation in the video quality incurred during transmission is quantified by measuring video quality metric (VQM). To quantify the impact of channel interference, controlled traffic is generated on a channel adjacent to the selected channel. We use human movements to alter the depth of signal fading at the receivers. Our experiments show that packet loss increased two times in the face of multipath fading and almost three times in the face of channel interference. The measurements presented in this paper can help users to monitor and maintain the quality of their wireless networks and optimize the use of limited network resources.

Ubiquity of portable computing devices coupled with wide availability of wireless communication present new impor- tant opportunities for applications involving media-rich content dissemination. However, as access networks become increasingly more heterogeneous, seamless data delivery across internets consisting of a variety of network technology becomes a real challenge. In this demonstration, we showcase a system that enables content dissemination over heterogeneous internets consisting of wired, infrastructure-based and infrastructure-less wireless networks that may be prone to intermittent connectivity. Using an efficient, yet flexible buffer management scheme, we are able to address application-specific performance requirements such as average delay, delivery probability, energy efficiency, etc. Our system uses the Message Delivery in Heterogeneous, Disruption-prone Networks (MeDeHa) [2]) framework to deliver messages across a heterogeneous internet coupled with History-Based Scheduling and Drop (HBSD) buffer management [1] as a way to optimize resources provided by opportunistic networks. MeDeHa, which is described in detail in [2], provides seamless data delivery over interconnecting networks of different types, i.e., infrastructure-based and infrastructure-less networks. MeDeHa's comprehensive approach to bridging infrastructure-based and infrastructureless networks also copes with intermittent connectivity. For this demonstration, we showcase a "complete stack" solution featuring, from to top to bottom, the DTN2 "bundle" layer, HBSD as an "external router" to DTN2, and MeDeHa, which handles message delivery. We have implemented, on a Linux-based testbed, (i) the MeDeHa framework, (ii) the HBSD [3] external router for the DTN2 [4] architecture.

In this paper, we present a naming scheme for heterogeneous networks composed of infrastructure-based and infrastructure-less networks where nodes may be subject to intermittent connectivity. The proposed scheme aims at decoupling object identification from location and is designed to operate with status-quo Internet routing. We showcase the proposed naming scheme implemented on the NS-3 network simulator and demonstrate that nodes are able to receive messages in both infrastructure-based and infrastructure-less networks despite frequent disconnections and changing location identifiers (i.e., IP address), while visiting different networks.

Communication networks, whether they are wired or wireless, have traditionally been assumed to be connected at least most of the time. However, emerging applications such as emergency response, special operations, smart environments, VANETs, etc. coupled with node heterogeneity and volatile links (e.g. due to wireless propagation phenomena and node mobility) will likely change the typical conditions under which networks operate. In fact, in such scenarios, networks may be mostly disconnected, i.e., most of the time, end-to-end paths connecting every node pair do not exist. To cope with frequent, long-lived disconnections, opportunistic routing techniques have been proposed in which, at every hop, a node decides whether it should forward or store-and-carry a message. Despite a growing number of such proposals, there still exists little consensus on the most suitable routing algorithm(s) in this context. One of the reasons is the large diversity of emerging wireless applications and networks exhibiting such “episodic” connectivity. These networks often have very different characteristics and requirements, making it very difficult, if not impossible, to design a routing solution that fits all. In this paper, we first break up existing routing strategies into a small number of common and tunable routing modules (e.g. message replication, coding, etc.), and then show how and when a given routing module should be used, depending on the set of network characteristics exhibited by the wireless application. We further attempt to create a taxonomy for intermittently connected networks. We try to identify generic network characteristics that are relevant to the routing process (e.g., network density, node heterogeneity, mobility patterns) and dissect different “challenged” wireless networks or applications based on these characteristics. Our goal is to identify a set of useful design guidelines that will enable one to choose an appropriate routing protocol for the application or network in hand. Finally, to demonstrate the utility of our approach, we take up some case studies of challenged wireless networks, and validate some of our routing design principles using simulations.

We address the issue of asserting the accuracy of coordinates advertised by nodes of Internet coordinate systems during distance estimations. Indeed, some nodes may lie deliberately about their coordinates to mount various attacks against applications and overlays. Our proposed method consists in two steps: 1) establish the correctness of a node's claimed coordinate (which leverages our previous work on securing the coordinates embedding phase using a Surveyor infrastructure); and 2) issue a time limited validity certificate for each verified coordinate. Validity periods are computed based on an analysis of coordinate inter-shift times observed on PlanetLab, and shown to follow a long-tail distribution (lognormal distribution in most cases, or Weibull distribution otherwise). The effectiveness of the coordinate certification method is validated by measuring the impact of a variety of attacks on distance estimates.

As a result of the wide spread of new efficient mobile communication devices, users are more and more willing to find on these devices (PDAs, Smart-phones, etc) applications similar to the ones they are using on their desktop computers. We go one step in this direction and propose in this demo a software we have designed and implemented to enable content sharing among spontaneous communities of mobile users using wireless multihop connections. Our contribution consists in designing, implementing and testing a tracker-less Bit-Torrent like application adapted to wireless ad-hoc networks. The proposed architecture includes two principal components: a membership management service and a content sharing service. As classical tracker-based Bit-Torrent membership management and peer discovery are unfeasible in ad hoc networks, we design the membership management service as a distributed tracker overlay that connects peers involved in the same sharing session. Using the membership information provided by the tracker overlay, the content sharing service schedules the data transfer connections among the session members by leveraging the multihop routing feature of wireless ad-hoc networks. Our test-bed composed of PDAs and smart-phones equipped with WIFI adapters and Windows Mobile 6 operating system is used to validate the feasibility of our application and to evaluate its performance.

Communication networks are traditionally assumed to be connected. However, emerging wireless applications such as vehicular networks, pocket-switched networks, etc., coupled with volatile links, node mobility, and power outages, will require the network to operate despite frequent disconnections. To this end, opportunistic routing techniques have been proposed, where a node may store-and-carry a message for some time, until a new forwarding opportunity arises. Although a number of such algorithms exist, most focus on relatively homogeneous settings of nodes. However, in many envisioned applications, participating nodes might include handhelds, vehicles, sensors, etc. These various "classes” have diverse characteristics and mobility patterns, and will contribute quite differently to the routing process. In this paper, we address the problem of routing in intermittently connected wireless networks comprising multiple classes of nodes. We show that proposed solutions, which perform well in homogeneous scenarios, are not as competent in this setting. To this end, we propose a class of routing schemes that can identify the nodes of "highest utility” for routing, improving the delay and delivery ratio by four to five times. Additionally, we propose an analytical framework based on fluid models that can be used to analyze the performance of various opportunistic routing strategies, in heterogeneous settings.

Abstract The design of efficient IEEE 802.11 physical (PHY) rate adaptation algorithms is a challenging research topic and usually the issues surrounding their implementations on real 802.11 devices are not disclosed. In this paper, we identify and evaluate the key parameters to design such algorithms. We then present a survey on existing PHY rate adaptation mechanisms and discuss their advantages and drawbacks. We also propose three new 802.11 PHY rate adaptation mechanisms, named adaptive auto rate fallback (AARF), closed loop adaptive rate allocation (CLARA), and adaptive multi‐rate retry (AMRR). AARF, proposed for low‐latency systems, has low complexity and obtains similar performance than RBAR in stationary and non‐fading wireless channels. CLARA is a culmination of the best attributes of the transmitter‐based ARF and RBAR control mechanisms with additional practical features such as adaptive fragmentation to improve multipath‐fading channel sensing and to provide feedback control signaling. AMRR is designed for high‐latency systems; it has been implemented and evaluated on an AR5212‐based device. Experimentation results show more than 20% performance improvement in throughput over the default algorithm implemented in the AR5212 MADWIFI driver. Copyright © 2008 John Wiley & Sons, Ltd.

The physical rate adaptation in 802.11 is a deeply investigated, though still open issue. Since 802.11 uses the random access distributed coordination function (DCF) mechanism to access the medium, collisions can occur when two or more stations want to transmit data simultaneously. The challenge of rate adaptation schemes is to adapt the physical transmission rate based on channel-related losses, i.e. collisions should not influence the choice of the rate. In this paper we propose a new rate adaptation algorithm that behaves like auto rate fallback (ARF), but makes use of the RTS/CTS handshake, when necessary, to decide whether the physical transmission rate should be changed. Main advantages of this algorithm are its simple implementation and the good performance it attains in presence of collisions. We evaluate the performance of this new rate adaptation algorithm, comparing it with other well known algorithms, by using the new NS-3 simulator.

In this paper, we present an efficient message delivery mechanism that enables distribution/dissemination of messages in an internet connecting heterogeneous networks and prone to disruptions in connectivity. We call our protocol MeDeHa (pronounced "medea") for Message Delivery in Heterogeneous, Disruption-prone Networks. MeDeHa is complementary to the IRTF's Bundle Architecture: while the Bundle Architecture provides storage above the transport layer in order to enable interoperability among networks that support different types of transport layers, MeDeHa stores data at the link layer addressing heterogeneity at lower layers (e.g., when intermediate nodes do not support higher-layer protocols). MeDeHa also takes advantage of network heterogeneity (e.g., nodes supporting more than one network) to improve message delivery. For example, in the case of IEEE 802.11 networks, participating nodes may use both infrastructure- and ad hoc modes to deliver data to otherwise unavailable destinations. Another important feature of MeDeHa is that there is no need to deploy special-purpose nodes such as message ferries, data mules, or throwboxes in order to relay data to intended destinations, or to connect to the backbone network wherever infrastructure is available. The network is able to store data destined to temporarily unavailable nodes for some time depending upon existing storage as well as quality-of-service issues such as delivery delay bounds imposed by the application. We evaluate MeDeHa via simulations using indoor scenarios (e.g. convention centers, exposition halls, museums etc.) and show significant improvement in delivery ratio in the face of episodic connectivity. We also showcase MeDeHa's support for different levels of quality-of-service through traffic differentiation and message prioritization.

This paper addresses the issue of the security of Internet Coordinate Systems,by proposing a general method for malicious behavior detection during coordinate computations. We first show that the dynamics of a node, in a coordinate system without abnormal or malicious behavior, can be modeled by a Linear State Space model and tracked by a Kalman filter. Then we show, that the obtained model can be generalized in the sense that the parameters of a filtercalibrated at a node can be used effectively to model and predict the dynamic behavior at another node, as long as the two nodes are not too far apart in the network. This leads to the proposal of a Surveyor infrastructure: Surveyor nodes are trusted, honest nodes that use each other exclusively to position themselves in the coordinate space, and are therefore immune to malicious behavior in the system.During their own coordinate embedding, other nodes can thenuse the filter parameters of a nearby Surveyor as a representation of normal, clean system behavior to detect and filter out abnormal or malicious activity. A combination of simulations and PlanetLab experiments are used to demonstrate the validity, generality, and effectiveness of the proposed approach for two representative coordinate embedding systems, namely Vivaldi and NPS.

Communication networks (wired or wireless) have traditionally been assumed to be connected at least most of the time. However, emerging applications such as emergency response, special operations, smart environments, VANETs, etc. coupled with node heterogeneity and volatile links will likely change the typical conditions under which networks operate. In fact, in such scenarios, networks may be mostly disconnected. To cope with frequent, long-lived disconnections, opportunistic routing techniques have been proposed in which, at every hop, a node decides whether it should either forward and/or store-and-carry a message. As a result, a number of message replicas may be created and routed independently ("spraying"). Most opportunistic routing schemes to-date perform greedy replication handing over a copy of a message to the first nodes encountered. Yet, in a network with heterogeneous nodes, where some nodes may be much "better" relays than others, such greedy schemes may waste valuable message replicas (and thus energy, storage space, etc.) on "useless" relays. For this reason, we propose the idea of utility-based replication, where some fitness or utility function is maintained for all nodes in a distributed fashion, and a small budget of message replicas is allocated according to this utility only to the fittest nodes. We describe a number of variations using different utility functions, and show that an improvement of up to 5-6× in delay can be achieved over greedy algorithms.

Multicast is an efficient paradigm for transmitting data from a sender to a group of receivers. According to the IEEE 802.11 standard, the multicast service is defined as an unreliable service, that is, it does not include the use of ACK frames. Furthermore, different to the unicast service, the multicast service makes use of a single rate out of the various rates included in the basic service set defined by the IEEE 802.11 standard. Even though various proposals have recently appeared in the literature addressing these issues, none of them has come out with a structured set of control mechanisms taking into account the varying conditions characterising the wireless channels as well as the requirements of various applications. A novel cross-layer auto rate selection multicast mechanism for multi-rate wireless LANs, namely auto rate selection for multicast, capable of adapting the data transmission to the varying conditions of the channel and taking into account the characteristics of various applications, is introduced. The simulation results show that our proposal outperforms the IEEE 802.11 standard and the mechanisms recently proposed in the literature.

Multicast video streaming over multirate wireless LANs imposes strong demands on video codecs and the underlying network. It is not sufficient that only the video codec or only the underlying protocols adapt to changes in the wireless link quality. Research efforts should be applied in both and in a synchronized way. Cross layer design is a new paradigm that addresses this challenge by optimizing communication network architectures across traditional layer boundaries. This paper presents cross-layer architecture for adaptive video multicast streaming over multirate wireless LANs where layer-specific information is passed in both directions, top-down and bottom-up. The authors jointly consider three layers of the protocol stack: the application, data link and physical layers. The authors analyze the performance of the proposed architecture and extensively evaluate it via simulations. The results show that the real-time video quality of the overall system can be greatly improved by cross-layer signaling.

Internet coordinate-based systems are poised to become an important service to support overlay construction and topology-aware applications. Indeed, through network distance embedding into an appropriate geometric space, such systems allow for accurate network distance estimations with low overhead. However, coordinate systems often rely on good cooperation between nodes for correct coordination and assume that information reported by probed nodes is correct. In this paper, we identify various attacks against coordinate embedding systems and show their effectiveness on two representative positioning systems, namely Vivaldi and NPS. Our study demonstrates that these attacks can seriously disrupt the operations of these systems and therefore the virtual networks and applications relying on them for distance measurements. Through simulations of different potential scenarios where malicious nodes provide biased coordinate information and delay measurement probes, we quantify the effects of attack strategies that aim to (i) introduce disorder in the system, (ii) fool honest nodes to move far away from their correct positions and (iii) isolate particular target nodes in the system through collusion. Our findings confirm the susceptibility of the coordinate systems to such attacks.

Recent proposals in multicast overlay networks have demonstrated the importance of exploiting underlying network topology data to construct efficient overlays. While they avoid virtual coordinates embedding and fixed landmarks measurements, these topology-aware proposals often rely on incremental and periodic refinements to improve each node's position in the delivery tree. We claim that there are barriers for the scalability of existing overlay multicast protocols. In fact, periodical refinement and control processes induce additional overhead and high communication cost. On the other hand, users attending a video conferencing session or an event broadcast expect an acceptable quality as soon as they join the multicast session. It is then important to overcome an efficiency problem from which almost all current overlay multicast proposals suffer. This problem is the long convergence time to reach a stabilized quality state in the overlay delivery tree. We propose a novel overlay multicast tree construction scheme, called LCC : Locate, Cluster and Conquer, designed to address the aforementioned scalability and efficiency issues. The scheme consists in two phases: a selective locating phase and an overlay construction phase. Using partial knowledge of location-information for participating nodes, the selective locating phase algorithm consists in locating the closest existing set of nodes (cluster) in the overlay for a newcomer. It allows then to avoid initially randomly-connected structures without using virtual coordinates system embedding nor fixed landmarks measurements. Then, on the basis of this locating process, the overlay construction phase consists in building and managing a topology-aware clustered hierarchical overlay.

Recent proposals in multicast overlay construction have demonstrated the importance of exploiting underlying network topology. However, these topology-aware proposals often rely on incremental and periodic refinements to improve the system performance. These approaches are therefore neither scalable, as they induce high communication cost due to refinement overhead, nor efficient because long convergence time is necessary to obtain a stabilized structure. In this paper, we propose a highly scalable locating algorithm that gradually directs newcomers to their a set of their closest nodes without inducing high overhead. On the basis of this locating process, we build a robust and scalable topology-aware clustered hierarchical overlay scheme, called LCC. We conducted both simulations and PlanetLab experiments to evaluate the performance of LCC. Results show that the locating process entails modest resources in terms of time and bandwidth. Moreover, LCC demonstrates promising performance to support large scale multicast applications

While the deployment of WiFi networks continue to grow at an explosive rate, the multicast multimedia delivery service on WiFi compliant devices is still in its early stage of development. The real culprit is the IEEE 802.11 MAC protocol, and in particular, the absence of feedback mechanism when multicast is used. Recently, a leader-based mechanism has been proposed to overcome this problem. In this paper, we measure the characteristics of the legacy multicast transmission mechanism and analyze its flaws. Then, we study the performance of the leader-based approach and compare its performance with the standard multicast service. The analysis is done on a large set of measurements made with our wireless testbed. Such measurements are an important complement to previous simulation studies and help in the design of the best mechanism to replace the faulty legacy multicast mechanism. Our study confirms that the leader-based mechanism outperforms the standard open-loop multicast mechanism while keeping fairness among other traffic.

Network coding is a new transmission paradigm that proved its strength in optimizing the usage of network resources. In this paper, we evaluate the gain from using network coding for file sharing applications running on top of wireless mesh networks. With extensive simulations carried out on a simulator we developed specifically for this study, we confirm that network coding can improve the performance of the file sharing application, but not as in wired networks. The main reason is that nodes over wireless cannot listen to different neighbors simultaneously. Nevertheless, one can get more from network coding if the information transmission is made more diverse inside the network. We support this argument by varying the loss rate over wireless links and adding more sources.

We consider the medium access control (MAC) layer for very high-speed wireless LANs, which is designed to support rich multimedia applications such as high-definition television. In such networks, the physical (PHY) layer data rate is proposed to exceed 216 Mbps. The legacy MAC layer, however, greatly restricts the performance improvement due to its overhead. It has been shown that MAC utilizes less than 20% of the transportation ability provided by the PHY layer. To mitigate this inefficiency, we propose an aggregation with fragment retransmission (AFR) scheme, which supports transmissions of very large frames and partial retransmissions in the case of errors. Aggregation allows for increased performance despite per-transmission overhead while partial retransmission alleviates the risk of losing the entire frame. Extensive simulations show that AFR fundamentally outperforms the legacy MAC protocol. It is particularly effective for applications with high data rates and large packet sizes such as HDTV and high-rate UDP traffic. For applications with very low data rates and small packet sizes such as voice over IP, AFR performs slightly better.

The software approach to developing Digital Signal Processing (DSP) applications brings some great features such as flexibility, re-usability of resources and easy upgrading of applications. However, it requires long and tedious tests and verification phases because of the increasing complexity of the software applications. This implies the need of a software programming environment capable of putting together DSP modules and providing facilities to debug, verify and validate the code. The objective of the work is to provide such facilities as simulation and verification for developing DSP software applications. This led us to developing an extension toolkit, EPSPECTRA, built upon PSPECTRA, one of the first toolkits available to design basic software radio applications on standard PC workstations. In this paper, we first present EPSPECTRA, an ESTEREL-based extension of PSPECTRA that makes the design and implementation of portable DSP applications easier. It allows the drastic reduction of testing and verification time while requiring relatively little expertise in formal verification methods. Second, we demonstrate the use of EPSPECTRA, taking as an example the radio interface part of a GSM base station. We also present the verification procedures for the three safety properties of the implementation programs which have complex control-paths. These have to obey strict scheduling rules. In addition, EPSPECTRA achieves the verification of the targeted application since the same model is used for the executable code generation and for the formal verification.

A block ACK (BTA) scheme has been proposed in IEEE 802.11e to improve medium access control (MAC) layer performance. It is also a promising technique for next-generation high-speed wireless LANs (WLANs) such as IEEE 802.11n. We present a theoretical model to evaluate MAC saturation throughput of this scheme. This model takes into account the effects of both collisions and transmission errors in a noisy channel. The accuracy of this model is validated by NS-2 simulations.

We present an analytical model that accounts for the positions of stations with respect to the access point (AP) while evaluating the performance of the 802.11 MAC layer. Our work is based on the Bianchi model where the performance of the 802.11 MAC layer is computed using a discrete time Markov chain, but where all stations are implicitly assumed to be located at the same distance from the AP. In our model, given the position of one station, we compute its saturation throughput while considering the positions of other concurrent stations. Further, our model provides the total saturation throughput of the medium. We solve the model numerically and we show that the saturation throughput per station is strongly dependent not only on the station's position, but also on the positions of the other stations. Results confirm that a station achieves a higher throughput when it is closer to the AP, but bring out that there is a distance threshold above which the throughput decrease is fast and significant. When a station is far from the AP compared to the other stations, it contends for the bandwidth not used by the other stations. We believe that our model is a good tool to dimension 802.11 wireless access networks and to study their capacities and their performances.

Abstract It is well known that the medium access control (MAC) layer is the main bottleneck for the IEEE 802.11 wireless LANs. Much work has been done on performance analysis of the 802.11 MAC. However, most of them assume that the wireless channel is error free. In this paper, we investigate the saturation throughput performance achieved at the MAC layer, in both congested and error‐prone channels. We provide a simple and accurate analytical model to calculate the MAC throughput. The model is validated through extensive simulation results. Our results show that channel errors have a significant impact on the system performance. Copyright © 2005 John Wiley & Sons, Ltd.

The emerging widespread use of real-time multimedia applications over wireless networks makes the support of quality of service (QoS) a key problem. In this paper, we focus on QoS support mechanisms for IEEE 802.11 wireless ad-hoc networks. First, we review limitations of the upcoming IEEE 802.11e enhanced DCF (EDCF) and other enhanced MAC schemes that have been proposed to support QoS for 802.11 ad-hoc networks. Then, we describe a new scheme called adaptive fair EDCF that extends EDCF, by increasing the contention window during deferring periods when the channel is busy, and by using an adaptive fast backoff mechanism when the channel is idle. Our scheme computes an adaptive backoff threshold for each priority level by taking into account the channel load. The new scheme significantly improves the quality of multimedia applications. Moreover, it increases the overall throughput obtained both in medium and high load cases. Simulution results show that our new scheme outperforms EDCF and other enhanced schemes. Finally, we show that the adaptive fair EDCF scheme achieves a high degree of fairness among applications of the same priority level.

Today, three different physical (PHY) layers for the IEEE 802.11 WLAN are available (802.11a/b/g); they all provide multi-rate capabilities. To achieve a high performance under varying conditions, these devices need to adapt their transmission rate dynamically. While this rate adaptation algorithm is a critical component of their performance, only very few algorithms such as Auto Rate Fallback (ARF) or Receiver Based Auto Rate (RBAR) have been published and the implementation challenges associated with these mechanisms have never been publicly discussed. In this paper, we first present the important characteristics of the 802.11 systems that must be taken into account when such algorithms are designed. Specifically, we emphasize the contrast between low latency and high latency systems, and we give examples of actual chipsets that fall in either of the different categories. We propose an Adaptive ARF (AARF) algorithm for low latency systems that improves upon ARF to provide both short-term and long-term adaptation. The new algorithm has very low complexity while obtaining a performance similar to RBAR, which requires incompatible changes to the 802.11 MAC and PHY protocol. Finally, we present a new rate adaptation algorithm designed for high latency systems that has been implemented and evaluated on an AR5212-based device. Experimentation results show a clear performance improvement over the algorithm previously implemented in the AR5212 driver we used.

We present a media-oriented mechanism for selecting the appropriate transmission mode in 802.11-based wireless LANs (WLANs). The main goal of this mechanism is to improve the effective throughput for transporting loss-tolerant multimedia traffic over a WLAN by taking into account both the application characteristics and the physical channel conditions. In particular, the proposed cross-layer mechanism exploits the robustness of multimedia coding by allowing packets with corrupted payloads reach the receiving application. The sending application specifies its quality of service requirements (data rate, BER tolerance, etc.), and the receiver selects the best transmission mode (transmission rate, modulation scheme, FEC scheme) while taking into account the time-varying channel conditions. We discuss the modifications needed for the control and data-packet headers to implement our approach in the framework of the IEEE 802.11 standards. We use ns2 simulations to contrast our scheme with an existing 802.11 rate selection algorithm. The results indicate that the proposed cross-layer approach achieves up to 5 Mbps increase in throughput and 20-meter increase in the coverage range. Furthermore, by disabling FEC from some of the standard transmission modes, we show that the goodput of loss-tolerant applications can be improved significantly.

This paper describes and analyzes SCORE, a scalable multicast-based communication protocol for large-scale virtual environments (LSVE) on the Internet. Today, many of these applications have to handle an increasing number of participants and deal with the difficult problem of scalability. We propose an approach at the transport layer, using multiple multicast groups and multiple agents. This approach involves the dynamic partitioning of the virtual environment into spatial areas and the association of these areas with multicast groups. It uses a method based on the theory of planar point processes to determine an appropriate cell size, so that the incoming traffic at the receiver side remains with a given probability below a sufficiently low threshold. We evaluate the performance of our scheme and show that it allows to significantly improve the participants' satisfaction while adding very low overhead.

Abstract Quality‐of‐service (QoS) is a key problem of today's IP networks. Many frameworks (IntServ, DiffServ, MPLS etc.) have been proposed to provide service differentiation in the Internet. At the same time, the Internet is becoming more and more heterogeneous due to the recent explosion of wireless networks. In wireless environments, bandwidth is scarce and channel conditions are time‐varying and sometimes highly lossy. Many previous research works show that what works well in a wired network cannot be directly applied in the wireless environment. Although IEEE 802.11 wireless LAN (WLAN) is the most widely used IEEE 802.11 wireless LAN (WLAN) standard today, it cannot provide QoS support for the increasing number of multimedia applications. Thus, a large number of 802.11 QoS enhancement schemes have been proposed, each one focusing on a particular mode. This paper summarizes all these schemes and presents a survey of current research activities. First, we analyze the QoS limitations of IEEE 802.11 wireless MAC layers. Then, different QoS enhancement techniques proposed for 802.11 WLAN are described and classified along with their advantages/drawbacks. Finally, the upcoming IEEE 802.11e QoS enhancement standard is introduced and studied in detail. Copyright © 2004 John Wiley & Sons, Ltd.

Bluetooth is an emerging standard for short range, low cost, low power wireless access technology. The Bluetooth technology is just starting to appear on the market and there is an urgent need to enable new applications with real time constraints to run on top of Bluetooth devices. The Bluetooth Specification proposes a Round Robin scheduler as possible solution for scheduling the transmissions in a Bluetooth Piconet. However, this basic scheme performs badly under asymmetric traffic conditions. Recently, several polling schemes have been proposed to improve performance on asymmetric transmissions and to support bandwidth guarantee. However, there is no solution available to support both delay and bandwidth guarantees required by real time applications. In this paper, we present FPQ, a new polling algorithm for Bluetooth Piconet that supports both delay and bandwidth guarantees and aims to remain fair and efficient with asymmetric flow rates. We present an extensive set of simulations and provide performance comparisons with other polling algorithms. Our performance study indicates that FPQ, while supporting flow rate and maximum delay QoS requests, outperforms Deficit Round Robin in term of delays by at least 10% in all cases, sometimes by more than 30% to 50%. Moreover, FPQ was designed to take the specifics of Bluetooth into consideration, in particular the low complexity required for cheap implementation.

The IEEE 802.11 medium access control (MAC) protocol provides a contention-based distributed channel access mechanism for mobile stations to share the wireless medium, which may introduce a lot of collisions in case of overloaded active stations. Slow contention window (CW) decrease scheme is a simple and efficient solution for this problem. In this paper, we use an analytical model to compare the slow CW decrease scheme to the IEEE 802.11 MAC protocol. Several parameters are investigated such as the number of stations, the initial CW size, the decrease factor value, the maximum backoff stage and the coexistence with the RequestToSend and ClearToSend (RTS/CTS) mechanism. The results show that the slow CW decrease scheme can efficiently improve the throughput of IEEE 802.11, and that the throughput gain is higher when the decrease factor is larger. Moreover, the initial CW size and maximum backoff stage also affect the performance of slow CW decrease scheme.

This paper describes an adaptive service differentiation scheme for QoS enhancement in IEEE 802.11 wireless ad-hoc networks. Our approach, called adaptive enhanced distributed coordination function (AEDCF), is derived from the new EDCF introduced in the upcoming IEEE 802.11e standard. Our scheme aims to share the transmission channel efficiently. Relative priorities are provisioned by adjusting the size of the contention window (CW) of each traffic class taking into account both applications requirements and network conditions. We evaluate through simulations the performance of AEDCF and compare it with the EDCF scheme proposed in the 802.11e. Results show that AEDCF outperforms the basic EDCF, especially at high traffic load conditions. Indeed, our scheme increases the medium utilization ratio and reduces for more than 50% the collision rate. While achieving delay differentiation, the overall goodput obtained is up to 25% higher than EDCF. Moreover, the complexity of AEDCF remains similar to the EDCF scheme, enabling the design of cheap implementations.

No abstract available.

Advances in processor and analog-to-digital conversion technology have made the software approach an increasingly attractive alternative for implementing radio-based systems. For mobile telephony base stations, the advantages with the new architecture are obvious: great cost savings by using one transceiver per base transceiver station (BTS) instead of one per channel, tremendous flexibility by moving system-specific parameters to the digital part, and allowing the support of a wide range of modulation and coding schemes. This paper considers the software implementation of a GSM BTS, and analyzes the performance of each of its radio interface modules. The performance of each software module is evaluated using both a % CPU metric and a processor-independent metric based on SPEC benchmarks. The results can be used to dimension systems, e,g., to estimate the number of software-based GSM channels that can be supported by a given processor configuration, and to predict the impact of future processor enhancements on BTS capacity. Two novel aspects of this work are the portability of the software modules and the platform-independent evaluation of their computational requirements.

We describe the software implementation of a GSM base station and consider the performance of each of its radio interface components. The performance of each software module is evaluated using both a % CPU metric and a processor-independent metric based on SPEC benchmarks. These results can be used to dimension systems, e.g., to estimate the number of GSM channels that can be supported by a given processor configuration, and to predict the impact of future processor enhancements on base station capacity.

Anecdotal evidence suggests that the quality of many videoconferences in the Internet is mediocre because of high packet loss rates. This makes it important to design and implement mechanisms that minimize packet loss and its impact in video (and audio) applications. There are two such types of mechanisms. Rate control mechanisms attempt to minimize the amount of packet loss by matching the bandwidth requirements of a video flow to the capacity available in the network. However, they do not prevent packet loss altogether. Error control mechanisms attempt to minimize the visual impact of lost packets at the destinations. We provide motivation for using error control mechanisms based on forward error correction (FEC) and packet reconstruction. We examine a specific mechanism, and evaluate its cost as well as the benefit expected from using it. This mechanism can be augmented to obtain a joint source/channel coding scheme suitable for both the current and the future integrated services Internet.

This paper describes the INRIA videoconferencing system (IVS), a low bandwidth tool for real-time video between workstations on the Internet using UDP datagrams and the IP multicast extension. The video coder-decoder (codec) is a software implementation of the ITU-T recommendation H.261 originally developed for the integrated services digital network (ISDN). Our focus is on adapting this codec for the Internet environment. We propose a packetization scheme, an error control scheme and an output rate control scheme that adapts the image coding process based on network conditions. This work shows that it is possible to maintain videoconferences with reasonable quality across packet-switched networks without requiring special support from the network such as resource reservation or admission control.

Datagram networks such as the Internet do not provide guaranteed resources such as bandwidth or guaranteed performance measures such as maximum delay. One way to support packet video in these networks is to use feedback mechanisms that adapt the output rate of video coders based on the state of the network. The authors present one such mechanism. They describe the feedback information, and how it is used by the coder control algorithm. They also examine how the need to operate in a multicast environment impacts the design of the control mechanism. This mechanism has been implemented in the H.261 video coder of IVS. IVS is a videoconference system for the Internet developed at INRIA. Experiments indicate that the control mechanism is well suited to the Internet environment. In particular, it makes it possible to establish and maintain quality videoconferences even across congested connections in the Internet. Furthermore, it prevents video sources from swamping the resources of the Internet, which could lead to unacceptable service to all users of the network.>

We describe a mechanism for scalable control of multicast continuous media streams. The mechanism uses a novel probing mechanism to solicit feedback information in a scalable manner and to estimate the number of receivers. In addition, it separates the congestion signal from the congestion control algorithm, so as to cope with heterogeneous networks.