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.

A science is defined by a set of encyclopedic knowledge related to facts or phenomena following rules or evidenced by experimentally-driven observations. Computer Science and in particular computer networks is a relatively new scientific domain maturing over years and adopting the best practices inherited from more fundamental disciplines. The design of past, present and future networking components and architectures have been assisted, among other methods, by experimentally-driven research and in particular by the deployment of test platforms, usually named as testbeds . However, often experimentally-driven networking research used scattered methodologies, based on ad-hoc, small-sized testbeds , producing hardly repeatable results. We believe that computer networks needs to adopt a more structured methodology, supported by appropriate instruments, to produce credible experimental results supporting radical and incremental innovations. This paper reports lessons learned from the design and operation of test platforms for the scientific community dealing with digital infrastructures. We introduce the SLICES initiative as the outcome of several years of evolution of the concept of a networking test platform transformed into a scientific instrument. We address the challenges, requirements and opportunities that our community is facing to manage the full research-life cycle necessary to support a scientific methodology.

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.

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.

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.

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.

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.

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.

Video streaming represents a large fraction of Internet traffic. Surprisingly, little is known about the network characteristics of this traffic. In this paper, we study the network characteristics of the two most popular video streaming services, Netflix and YouTube. We show that the streaming strategies vary with the type of the application (Web browser or native mobile application), and the type of container (Silverlight, Flash, or HTML5) used for video streaming. In particular, we identify three different streaming strategies that produce traffic patterns from non-ack clocked ON-OFF cycles to bulk TCP transfer. We then present an analytical model to study the potential impact of these streaming strategies on the aggregate traffic and make recommendations accordingly.

In this paper, we show how to exploit real-time communication applications to determine the IP address of a targeted user. We focus our study on Skype, although other real-time communication applications may have similar privacy issues. We first design a scheme that calls an identified-targeted user inconspicuously to find his IP address, which can be done even if he is behind a NAT. By calling the user periodically, we can then observe the mobility of the user. We show how to scale the scheme to observe the mobility patterns of tens of thousands of users. We also consider the linkability threat, in which the identified user is linked to his Internet usage. We illustrate this threat by combining Skype and BitTorrent to show that it is possible to determine the filesharing usage of identified users. We devise a scheme based on the identification field of the IP datagrams to verify with high accuracy whether the identified user is participating in specific torrents. We conclude that any Internet user can leverage Skype, and potentially other real-time communication systems, to observe the mobility and filesharing usage of tens of millions of identified users.

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.

Network latency and packet loss are considered to be an important requirement for realistic evaluation of Peer-to-Peer protocols. Dedicated clusters, such as Grid'5000, do not provide the variety of network latency and packet loss rates that can be found in the Internet. However, compared to the experiments performed on testbeds such as PlanetLab, the experiments performed on dedicated clusters are reproducible, as the computational resources are not shared. In this paper, we perform experiments to study the impact of network latency and packet loss on the time required to download a file using BitTorrent. In our experiments, we observe a less than 15% increase on the time required to download a file when we increase the round-trip time between any two peers, from 0 ms to 400 ms, and the packet loss rate, from 0% to 5%. Our main conclusion is that the underlying network latency and packet loss have a marginal impact on the time required to download a file using BitTorrent. Hence, dedicated clusters such as Grid'5000 can be safely used to perform realistic and reproducible BitTorrent experiments.

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.

Protection of data against long fading time is one of the greatest challenges posed by a satellite delivery system offering multimedia services to mobile devices like DVB-SH. To deal with this challenge several enhancements and modifications of the existing terrestrial mobile TV (DVB-H) are being considered. These solutions provide the required protection depth but they don't take into account the specificity of mobile handheld devices such as power consumption, memory constraints and chipsets implementation costs. In this paper, we propose an innovative algorithm (called Multi Burst Sliding Encoding or MBSE) that extends the DVB-H intra-burst (MPE-FEC) protection to an inter-burst protection so that complete burst losses could be recovered while taking into account the specificity of mobile handheld devices. Based on a clever organisation of the data, our algorithm allows to provide protection against long term fading while still using RS code implemented in DVB-H chipsets. We evaluate the performance of MBSE by both theoretical analysis as well as intensive simulations and experiments. The results also show good performance in terms of protection, battery and memory saving. The MBSE is now under standardisation and it is considered by the DVB Forum as the main solution for the DVB-SH class terminals.

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.

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.

The end-to-end delay among a set of peers can be easily and scalably estimated using a landmark approach Peers' coordinates are deduced from delay measurements to a small number N of landmarks L{L1,..., Ln}. We wonder whether it is feasible to use such distributed solution for bandwidth estimation. This requires that each peer determines its bandwidth vector by measuring the direct and reverse bandwidth on its path with each landmark. The bandwidth vectors of two peers are then used to estimate the bandwidth between them in both ways. For a couple of peers, we denote by (i) direct path the network path that joins them directly using IP routing, and by (ii) indirect path the path that joins them via a landmark node. N indirect paths (N being the number of landmarks) are assigned to each direct path. Our idea is to estimate the bandwidth of the direct path using those of the indirect paths. The indirect paths that have more links common with the direct path, are assigned more weight in the estimation function. We consider different estimation functions and we study the impact of the landmarks' locations on the accuracy of the estimations.

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

In this paper, an enhanced scalable proximity model is described. This work investigates the bandwidth estimation problem and it is impact on the quality of service of the file sharing application. The file transfer application over the TCP protocol is considered and it can be encountered in the emerging file sharing P2P applications or in the replicated Web server context. Since the impact of the bandwidth estimation is the main concern, the large TCP transfer due to its sensitivity to the parameter is considered. Thus considering the delay alone for proximity characterization is far from being optimal for large file transfer applications

The replication of digital content in overlay networks makes the identification of the best server an interesting problem. In this paper, our aim is to reduce the content transfer time, which we define as the time required to download a digital content by a client using TCP. Our scheme consists of ranking the servers from the best one to the worst one based on a metric that corresponds to a prediction of the content transfer time from each server. Our prediction function considers the critical performance parameters that have an impact on the quality of the transfer, such as the load of the servers and the characteristics of the path between the client and the servers. Once the servers are ranked, the client can download the content on point-to-point from the best server, or in parallel from a subset of servers at the top of the list (best servers). Our experimental results show that our approach for identifying the best server(s) outperforms the existing classical solutions. Moreover, these results show that the proposed metric predicts the transfer time of a content with around 96% accuracy.

As mobile users cross the border of two adjacent domains with on-going sessions, their re-authentication causes a significant impact on inter-domain handoff latency as it requires remote contact with the authentication server across domains, making it difficult to employ current authentication protocols. This paper focuses on the cross-domain authentication over wireless local area networks (WLANs) that minimizes the need for remote access. We analyze the security requirements suggested by the IEEE 802.11i authentication standard, and consider additional requirements to help reduce the authentication latency without compromising the level of security. We propose an enhanced protocol called the Mobility-adjusted Authentication Protocol (MAP) that performs mutual authentication and hierarchical key derivation with minimal handshakes, relying on symmetric cryptographic functions. We also present security context nodes (SCNs) that handle security contexts in conjunction with MAP, which allows for avoiding continuous remote contact with the home authentication server. In contrast to Kerberos which favors inter-realm authentication, MAP achieves a 26% reduction of authentication latency without degrading the level of security.

Satellites are expected to have an important role in providing the Internet protocol (IP) multicast service to complementing next-generation terrestrial networks. In this paper, we focus on the deployment of IP multicast over the next generation of digital video broadcasting-based geosynchronous earth orbit satellites supporting multiple spot beams and on-board switching technologies. We propose a new encapsulation scheme optimized for IP multicast, which has two distinct modes enabling two alternative on-board switching approaches: the self-switching and the label-switching. We also detail a set of mechanisms and protocols for ground stations, as well as for the on-board processor to allow an efficient multicast forwarding in this type of environment, while reducing the load of control and data messages in the satellite segment, and building efficient multicast delivery trees reaching only the spot beams containing at least one member of the corresponding multicast session. To integrate satellite links in the terrestrial Internet, we present satellite multicast adaptation protocol (SMAP), a protocol which is implemented in satellite stations to process incoming protocol independent multicast-sparse mode (PIM-SM) messages sent by terrestrial nodes to the satellite system. SMAP helps to update the tables required for the mapping between IP packets and MPEG-2 data segments, their switching on board the satellite, and their filtering at the satellite receivers.

In today's Internet a user can be adversely affected by other users that overload the router. To address this problem, routers need to provide flow isolation. In this paper, we present MuxQ, a new queue management mechanism that provides a high degree of isolation without using per-flow queuing. MuxQ protects the multiplexing function of the router buffer by progressively controlling the allocation of buffer space in a FIFO queue. The allocation decision is based on state information of only a limited number of flows: the flows that do currently have packets in the queue. We evaluate MuxQ by simulation and show that it performs vastly better than the classical drop-tail. By using a very simple algorithm MuxQ provides reasonable flow isolation.

In this work, we propose a new active queue management mechanism called MFQ (multicast fair queuing) for multicast flows. MFQ is independent of the inter-multicast fairness policy and it interacts with a multicast bandwidth allocation module that can implement either a multicast fairness policy or a multicast pricing model and tries using a single FIFO queue to achieve the expected allocation which may depend on the number of downstream receivers. We perform simulations to evaluate the performance of MFQ for different bandwidth allocation schemes. Results obtained suggest that MFQ achieves the expected allocation for non-responsive multicast flows.

Despite a decade of research and development, multicast has not yet been deployed on a global scale. Among the difficulties with the current infrastructure are inter-multicast fairness support, multicast congestion control, and multicast routing. We focus on the inter-multicast fairness issue which addresses the way how the network resources are shared between competing multicast flows. We propose a simple and scalable single FIFO queue-based active queue management mechanism called MFQ (multicast fair queuing) to achieve the desired inter-multicast fairness. MFQ interacts with an external multicast bandwidth allocation module which implements a pre-defined inter-multicast fairness function. To guarantee a fine-grained packet queuing/dropping, MFQ uses a novel bandwidth sharing notion, called multicast allocation layer (MAL). Based on this notion, MFQ classifies multicast packets into layers and adjusts their weights in order to provide a bandwidth sharing being as close as possible to that given by the fluid model algorithm. Simulation results demonstrate that MFQ achieves the expected allocation for both responsive and non-responsive multicast flows. We validate our findings by analyzing the impact of network and groups dynamics on the expected bandwidth allocation and comparing it to that obtained by MFQ. Furthermore, we show that MFQ converges very fast to a stable state and easily adapts itself to the dynamic change of the flows weights.

As the satellite technology will be one of the main components of the next generation Internet (NGI), a naturally occurring question concerns the feasibility of providing an efficient satellite-based Internet access. In this paper, we are interested in the problem of the planning of satellite-terrestrial hybrid networks for the multicast transfer. We first develop an architectural model for such type of networks, then we define the problem and we propose a planning heuristic that determines the suboptimal positions of satellite uplinks in a given terrestrial network. The proposed method minimizes the mean cost (the number of branches) of multicast trees for the unreliable multicast and the mean packet transfer delay for the reliable multicast. In addition, we develop a configuration policy of PIM-SM in hybrid networks concerning the choice of the list of rendezvous points (RPs) and the switching from the RP-routed tree to the shortest path tree. The obtained results demonstrate the ability of the proposed methods (the planning heuristic and the PIM-SM configuration policy) to improve the multicast performance criteria and to determine effectively the satellite uplink positions using PIM-SM combined with UDLR (unidirectional link routing).

Finding an appropriate end-to-end congestion control scheme for each type of flow, such as real-time or multicast flows, may be difficult. But it becomes even more complex to have these schemes be friendly among themselves and with TCP. The assistance of routers within the network for fair bandwidth sharing among the flows is therefore helpful. However, most of the existing mechanisms that provide this fair sharing imply complex buffer management and maintaining flow state in the routers. We propose to realize this fair bandwidth sharing without per-flow state in the routers, using only a trivial queueing discipline. Packets are tagged near the source, depending on the nature of the flow. In the core of the network, routers use FIFO queues, and simply drop the packet with the highest tag value in case of congestion. Contrarily to other stateless fair queueing algorithms in the core routers, we do not try to maintain instantaneous flow rates equal. Instead, we take into account the responsiveness nature of the flows, and adjust loss rates such that average rates are equal. The novel approach of our scheme, called TUF, tag-based unified fairness, not only improves the overall fairness but enables us to maintain it in realistic environments, with non-negligible round trip times or bursty traffic, where other schemes fail. The corresponding cost is the need for models of the end to-end responsive natures of the flows.

Our main focus here is to study the IP multicast service behaviour over the next-generation of satellite-terrestrial hybrid networks for which we develop a general architecture and we enumerate their main characteristics. We first focus on the Internet Group Membership Protocol (IGMP) feedback implosion problem and we present the exponential feedback raise that able to provide sufficiently stable expectation values across a large group size. Then, we propose and compare different approaches that can be used to enable the multicast in the on-board satellite switch. We show that the approach based on maintaining a multicast beams table (MBT) containing the list of spot beams concerned by each multicast group is the most suitable approach because it allows an efficient and transparent integration of satellite links in the Internet. In fact, the on-board switch monitors the IGMP reports and queries sent over the satellite link and update the MBT entries. We also turn our attention to multicast routing protocols. We present some undesirable behaviour of DVMRP, PIM-DM, and PM-SM. For DVMRP and PIM-DM, we identify some configurations where the satellite receivers may receive duplicated packets and we propose a method to overcome this problem. We then develop a configuration policy of PIM-SM in hybrid networks concerning the choice of the list of rendezvous point (RPs) and the switching from the RP-routed tree to the shortest path tree.

Group communication supports information transfer between a set of participants. It is becoming more and more relevant in distributed environments. For distributed or replicated data, it provides efficient communication without overloading the network. For some types of multimedia applications, it is the only way to control data transmission to group members. This paper surveys protocol functions and mechanisms for data transmission within a group, from multicast routing problems up to end-to-end multipoint transmission control. We provide a bibliography which is organized by topic.

A protocol compiler takes as input an abstract specification of a protocol and generates an implementation of that protocol. Protocol compilers usually produce inefficient code both in terms of code speed and code size. We show that the combination of two techniques makes it possible to build protocol compilers that generate efficient code. These techniques are: (i) the use of a compiler that generates from the specification a unique tree-shaped automation (rather than multiple independent automata) and (ii) the use of optimization techniques applied at the automation level, i.e., on the branches of the trees. We have developed a protocol compiler that uses both these techniques. The compiler takes as the input a protocol specification written in the synchronous language Esterel. The specification is compiled into a unique automation by the Esterel front end compiler. The automation is then optimized and converted into C code by our protocol optimizer called HIPPCO. HIPPCO improves the code performance and reduces the code size by simultaneously optimizing the performance of the common path and optimizing the size of the uncommon path. We evaluate the gain expected with our approach on a real-life example, namely a working subset of the TCP protocol generated from an Esterel specification. We compare the protocol code generated with our approach to that derived from the standard BSD TCP implementation. The results are very encouraging. HIPPCO-generated code executes up to 25% fewer instructions than the BSD code for input packet processing while only increasing the code size by 25%.

A protocol compiler takes as input an abstract specification of a protocol and generates an implementation of that protocol. Protocol compilers usually produce inefficient code both in terms of code speed and code size. In this paper, we show that the combination of two techniques makes it possible to build protocol compilers that generate efficient code. These techniques are i) the use of a compiler that generates from the specification a unique tree-shaped automaton (rather than multiple independent automata), and ii) the use of optimization techniques applied at the automaton level, i.e. on the branches of the trees.We have developed a protocol compiler that uses both these techniques. The compiler takes as input a protocol specification written in the synchronous language Esterel. The specification is compiled into a unique automaton by the Esterel front end compiler. The automaton is then optimized and converted into C code by our protocol optimizer called HIPPCO. HIPPCO improves code performance and reduces code size by simultaneously optimizing the performance of the common path and optimizing the size of the uncommon path. We evaluate the gain expected with our approach on a real-life example, namely a working subset of the TCP protocol generated from an Esterel specification. We compare the protocol code generated with our approach to that derived from the standard BSD TCP implementation. The results are very encouraging. HIPPCO-generated code executes up to 25 % fewer instructions than the BSD code for input packet processing while maintaining comparable code size.

The design issues of a multimedia transport protocol, TP5, which achieves the multiplexing of both real time and normal data flows at the transport layer are discussed. The use of a single transport connection to carry both data flows is proposed. Two schemes for the support of the synchronization of these data flows are presented. The protocol TP5 implementing the above mechanisms is described. It is shown that with the use of the described synchronization schemes, the TP5 protocol minimizes the control of the real time traffic while still controlling the normal data flow.>