This paper introduces a data-driven model for generating synthetic satellite users based on population and geographical factors. The model applies the logarithmic opinion pool to combine complementary perspectives on user distribution, such as population density and rural predominance. This principled aggregation captures realistic spatial disparities in connectivity and improves the representativeness of satellite constellation simulations. The approach supports the development of adaptive non-terrestrial network (NTN) strategies aligned with both orbital topology and terrestrial demand.

We propose a fully decentralized, adaptive routing system that exploits the dynamic topology of Walker Delta Low Earth Orbit (LEO) constellations to achieve efficient traffic distribution and improved reliability without requiring global link-state information. The system addresses scalability challenges in mega-constellations, where congestion, limited ground infrastructure, and uneven user distribution degrade service quality in high-demand regions. Although inter-satellite links provide diverse routing paths, the lack of multibeam support in some terminals complicates routing and can increase path length during cross-plane communication. Our evaluation of a Hop-by-Hop based strategy in Walker Delta constellations shows that it achieves performance comparable to optimal congestion-minimization algorithms and outperforms source-routing methods, while maintaining near-minimal hop counts for low-latency communication.

The availability of abundant solar energy is driving renewed interest in orbital computing infrastructures hosted on Sun-Synchronous Orbit (SSO) constellations. However, these constellations suffer from limited instantaneous Earth coverage and constrained direct-to-ground, which is clearly incompatible with the connectivity and responsiveness requirements of datacenter clients. In this paper, we propose a hybrid architecture where SSO satellites offload traffic through Low Earth Orbit (LEO) Walker constellations—such as OneWeb and Starlink—which act as a dynamic relay layer toward the ground. We design and evaluate dynamic inter-orbital link topologies enabling SSO-to-Walker connectivity and analyze their performance in terms of link availability, relay opportunities, path continuity, and network resilience. Our results show that leveraging Walker constellations as relays significantly improves SSO systems, effectively compensating for their coverage limitations.

Advances in satellite miniaturization and launch cost reduction have enabled the deployment of increasingly large nanosatellite swarms in Low Earth Orbit. However, the ground infrastructure supporting these missions has not evolved at the same pace: many operators rely on a single ground station that can serve only one satellite at a time and must observe a non-negligible reconfiguration delay between consecutive contacts. This creates a critical bottleneck for missions requiring rapid offloading of large observation datasets within a few orbital passes.

The study finds that in urban environments, the performance of non-cross-layer routing protocols (AODV, OLSR, DSDV) becomes nearly identical, contradicting earlier research that ranked their effectiveness. Using a new building-aware loss model in ns-3, we show that realistic urban conditions, unlike traditional open-field models, cause end-to-end delay to increase up to tenfold, with high variability due to unstable links. We also investigate whether adjusting parameters in open-field models could mimic urban connectivity. While this tuning matched some network connectivity metrics, it failed to replicate more complex concepts linked with propagation dynamics of cities, proving that open-field models cannot accurately simulate urban networks without fundamental changes.

Routing in nanosatellites swarms presents distinct challenges, including variable node availability, constrained bandwidth, and dynamic topology. Strategies like delay-tolerant networking (DTN) can be advantageous, as they adapt to intermittent connectivity by storing and forwarding data when connections are established. Moreover, geographic routing protocols that exploit satellite positions can improve efficiency, while machine learning approaches may optimize routing decisions based on changing network conditions. What about hybrid approaches that may combine some of these methods? Basically, the crucial question is where to begin. The primary challenge for nanosatellites network designers is to determine which routing strategies to test prior to deployment. Given the vast number of existing routing protocols, testing all of them is not possible. This problem motivates the present study, which share the authors' experiences on selecting the most suitable routing algorithms for a given nanosatellites swarm. In particular, the study reports how the use of graph theory metrics helps in restricting the set of routing algorithms to be considered for network characterization and protocol selection.

Low-frequency radio interferometry is crucial to understanding the universe and its very early days. Unfortunately, most of the current instruments are ground-based and thus impacted by the interferences massively produced by the Earth. To alleviate this issue, scientific missions aim at using Moon-orbiting nano-satellite swarms as distributed radio-telescopes in outer space, keeping them out of Earth interference range. However, swarms of nano-satellites are systems with complex dynamics and need to be appropriately characterized to achieve their scientific mission. This paper presents a methodology based on graph theory for characterizing the swarm network system by computing graph theory metrics around three properties: the node density, network connectivity and ISL availability. We show that these properties are well-suited for highlighting a possible heterogeneity in a network and adapt a routing strategy accordingly. This work is the first milestone in defining the best-suited routing strategy within the swarm from the derived network properties.

We address the problem of partitioning a network of nano-satellites to fairly distribute the network load under energy consumption constraints. The study takes place in a context where a swarm of nano-satellites orbits the Moon and works as, but not limited to, a distributed radio-telescope for low-frequency radio interferometry. During an interferometry mission, each nano-satellite collects observation data, then shares them with the other swarm members to compute a global image of space. However, the simultaneous transmission of large volumes of data can cause communication issues by overloading the radio channel, leading to potential packet loss. In this context, we investigate three division algorithms based on graph sampling techniques. We prove that random walk-based algorithms overall perform the best in terms of conservation of graph properties and fairness for group sizes down to 10% of the original graph.

We investigate the feasibility of using machine learning methods for predicting the Quality of Experience (QoE) of end users in the context of video streaming over satellite networks. To achieve this, we analyzed QoE and traffic data from 2,400 YouTube video sessions over emulated geosynchronous (GSO) satellite links. The objective is to determine whether existing learning methods, originally developed for wired or mobile networks, can be adapted to accurately predict key QoE factors over SATCOM. We particularly investigate a specific existing framework, which achieves outstanding performance in predicting resolution and initial delay. However, we point out some discrepancies in their hypothesis, leading to optimistic forecasting results. We then refine their methodology to ensure a complete independence between training and test datasets, leading to a fairer QoE video streaming forecast over satellite networks.

Most modern congestion control algorithms, that aim to optimize delay and throughput, exploit more metrics than the sole packet loss congestion information. These additional metrics are mostly based on the round trip time evolution and allow congestion controls to reach better performance, in particular on wireless and cellular links as demonstrated by Copa, BBR, or REMY. Basically, these metrics allow congestion control to estimate the queuing level of the path and its evolution, to assess the presence of congestion. Actually, a good estimation of this level obviously prevents congestion losses, but also allows assessing a ratio of error link losses among the whole observed losses. The consistency and accuracy of these metrics are key to good congestion control performance, and this explains, for instance, the good performance of Copa currently in production at Facebook. However, these metrics remain challenging and the quest of an accurate and practical estimation seems complex. This paper investigates how a novel deep learning algorithm, known as Attention, can help in assessing queuing evolution and status on an end-to-end path. Among others, we focus on the evolution of the total time spent by packets in the buffers, which is the key metric of Copa. The results unequivocally demonstrate a better accuracy of this metric used by Copa.

This study investigates the importance of choosing the first (respectively last) hop to access (respectively to exit) a Low Earth Orbit (LEO) satellite constellation, which is of upmost importance for the LEO routing performance. Usually, basic routing strategies connect a ground station to its nearest satellite, and this strategy does not always lead to the optimal routing path. We propose to select this first/last satellites within a subset of k-nearest satellites. After performing routing simulations over one of the next-generation satellite constellations, preliminary results show that this in/out hop selection strategy leads to a better link capacity usage and a lower data loss rate, allowing a faster TCP bulk data transfer.

We introduce a XOR-based source routing (XSR) scheme as a novel approach to enable fast forwarding and low-latency communications. XSR uses linear encoding operation to both 1) build the path labels of unicast and multicast data transfers; 2) perform fast computational efficient routing decisions compared to standard table lookup procedure without any packet modification all along the path. XSR specifically focuses on decreasing the complexity of forwarding router operations. This allows packet switches (e.g, link-layer switch or router) to perform only simple linear operations over a binary vector label which embeds the path. XSR provides the building blocks to speed up the forwarding plane and can be applied to different data planes such as MPLS or IPv6. Compared to recent approaches based on modular arithmetic, XSR computes the smallest label possible and presents strong scalable properties allowing to be deployed over any kind of core vendor or datacenter networks. At last but not least, the same computed label can be used interchangeably to cross the path forward or reverse in the context of unicast communication.

How many times have ever asked yourself: "Can I trust my satellite experiments' outcome?". Performing experiments on real satellite system can either be (1) costly, as the radio resource may be scarce or (2) not possible, as you can hardly change the waveforms transmitted by the satellite platform. Moreover, assessing user applications QoE can hardly be done using only simulated environments while the QoS modeling of a satellite system can often lead to non-conclusive or ambiguous results. The aim of this paper is to bring out representative solutions allowing the networking community to drive consistent experiments using open-source tools. To this end, we compare Mininet and OpenSAND satellite emulator to a real satellite access provided by CNES. We consider VoIP traffic to analyze the trade-off between reliability of the results, ease of use and reproducibility of the experiments.

We define a novel core network router scheduling architecture to carry and isolate time constrained and elastic traffic flows from best-effort traffic. To date, one possible solution has been to implement a core DiffServ network with standard fair queuing and scheduling mechanisms as proposed in the well-known “A Differentiated Services Code Point (DSCP) for Capacity-Admitted Traffic” from RFC5865. This architecture is one of the most selected solutions by internet service provider for access networks (e.g. Customer-Premises Equipment or satellite PEP). In this study, we argue that the proposed standard implementation does not allow to efficiently quantify the reserved capacity for the AF class. By using a novel credit based shaper mechanism called Burst Limiting Shaper, we show that we can provide the same isolation for the time constrained EF class while better quantifying the part allocated to the AF class.

End-to-end (E2E) reliable communication in Delay Tolerant Network (DTN) is a challenging task due to long delay and frequent link disruptions. To enable reliability, the IETF is currently looking at strategies to integrate erasure coding mechanisms inside DTN architecture. The objective is to extend the ability of the existing DTN bundle fragmentation mechanism to support cases where bundles have a high probability of being lost. To date, discussions agree that an intermediate node can re-encode bundles, leaving all decoding process at the destination node in order to let intermediate node operations be as simple as possible. We propose to study and analyze possible re-encoding strategies at intermediate nodes using an on-the-fly coding paradigm. We also investigate how re-encoding and acknowledgment strategies based on this coding scheme would enable E2E reliable communication. Finally, we propose an adaptive mechanism with low complexity that deals with both re-routing events and network dynamics which are common in the context of DTN. Simulation results show that re-encoding at the relay and the adaptive mechanism allows a significant reduction in terms of network overhead injected by erasure codes while ensuring the E2E reliability.

This paper introduces Rest, a reliable streaming protocol for lossy networks. Rest ensures full reliability while recovering losses as soon as possible thanks to the proactive injection of redundancy packets encoded following an on-the-fly scheme. It dynamically adapts the sending of codes depending on the estimation of the packet error rate with periodic acknowledgments to limit feedback dependency and protocol overhead. Results show that data are smoothly delivered to the receiving application with minimum overhead when errors are uniform. For systems with limited processing capacity, we propose to use a bounded encoding window to deliver data more uniformly while limiting decoding matrices size. We study the performance of Rest under different network conditions and highlight the underlying trade-offs behind each system parameter. We show that an optimal acknowledgement frequency can be estimated to minimize overhead while meeting system requirements in terms of delivery delay and computational power.

We present a new delay-based transport protocol named FLOWER, that aims at providing a Lower-than-Best-Effort (LBE) service. The objective is to propose an alternative to the Low Extra Delay Background Transport (LEDBAT) widely deployed within the official BitTorrent client. Indeed, besides its intra-fairness problem, known as latecomer unfairness, LEDBAT can be too aggressive against TCP, making it ill suited for providing LBE services over certain networks such as constrained wireless networks. By using a fuzzy controller to modulate the sending rate, FLOWER aims to solve LEDBAT issues while fulfilling the role of a LBE protocol. Our simulation results show that FLOWER can carry LBE traffic in network scenarios where LEDBAT cannot while solving the latecomer unfairness problem. Finally, the presented algorithm is simple to implement and does not require complex computation that would prevent its deployment.

This paper proposes "LD-Tetrys" (Late Decoding Tetrys), a solution based on an on-the-fly erasure code that attempts to solve the problem of late decoded packets usually considered as lost by the video decoder. LD-Tetrys has the following advantages: i) it drastically improves the trade-off between throughput and quality without modifying the codecs or adding complexity at the encoder side ii) it allows a simple but robust configuration. The only cost is a minor modification of the decoding process and a slight increase in the video decoding complexity. Last but not least, LD-Tetrys requires a much smaller playout buffer to obtain the same perceived video quality, bringing benefits for interactive applications.

Most of the efforts to characterize DTN routing are focused on the trade-off between delivery ratio and delay. Buffer occupancy is usually not considered a problem and most of the related work assumes infinite buffers. In the present work, we focus on the drop ratio for message forwarding considering finite buffers. We model message drops with a continuous time Markov chain (CTMC). To the best of our knowledge, there is no previous work with such approach. We focus on the worst case with 1-packet buffers for message forwarding in homogeneous inter-contact times (ICT) and 2-class heterogeneous ICT. Our main contribution is to link the encounter rate(s) with the drop ratio. We show that the modeled drop ratio fits simulation results obtained with synthetic traces for both cases.

The increasing heterogeneity and asymmetry in wireless network environments makes QoS guarantees in terms of delays and throughput a challenging task. In this paper, we study a novel scheduling algorithm for multipath transport called Delay Aware Packet Scheduling (DAPS) which aims to reduce the receiver's buffer blocking time considered as a main parameter to enhance the QoS in wireless environments. We develop an analytical model of maximum receiver's buffer blocking time and extend the DAPS algorithm considering implementation issues. Performance evaluations based on ns-2 simulations highlight the enhanced QoS that DAPS can provide. With reference to the classical multipath transport protocol CMT-SCTP, we observe a significant reductions of the receiver's buffer occupancy, down by 77%, and the application delay, down by 63%.

The Low Extra Delay Background Transport (LEDBAT) protocol is a recently standardized protocol that aims to offer a scavenger service (i.e. the goal is to exploit the remaining and unused capacity of a link). LEDBAT is a delay-based protocol mainly defined by two parameters: a target queuing delay and a gain. The RFC 6817 provides guidelines to configure both parameters that strongly impact on the LEDBAT behavior in terms of fairness with other protocols. However, these guidelines are questioned by several studies as they might lead to the generation of a non-LBE (Less-than-Best-Effort) traffic. This paper explores the set of optimal parameters allowing LEDBAT protocol to effectively perform as an LBE traffic. We conclude that the optimal couple of target and decrease gain is (5 ms; 10). However, we observe that the aggregated use of optimized LEDBAT sources still disturb the overall traffic performance and that the exponential backoff is not an answer to this issue. As a result, we believe that additional strategies to limit the number of LEDBAT flows are required for integrating this protocol at a large scale.

International audience

Performance of transport protocols on lossy links is a well researched topic, however there are only a few proposals which make use of the opportunities presented by the use of erasure coding within the transport layer, particularly in the multipath transport protocol context. In this paper, we investigate the improvements in the performance of multipath SCTP transport protocol brought by the novel integration of the on-the-fly erasure code into the congestion control and reliability mechanisms of CMT-SCTP. Our contributions include: integration of transport protocol and erasure codes with regards to congestion control and reliability mechanisms; proposal for a variable retransmission delay parameter in the sender (aRTX) adjustment; evaluation of the performance of CMT-SCTP with erasure coding with simulations. Our results show that we can achieve from 10% to 80% improvements in application goodput under lossy multipath network conditions without a significant penalty i.e. with a minimal (10%) overhead due to the encoding-decoding process. We further evaluate the performance of video streaming using an equivalent of partially reliable CMT-SCTP with erasure coding and again demonstrate a solid performance improvement for our proposal, compared to CMT-SCTP. Finally, we provide an analytical evaluation of CMT-SCTP with erasure codes and validate this with simulation results.

International audience

Compared to wired networks, Delay/Disruption Tolerant Networks (DTN) are challenging to monitor due to their lack of infrastructure and the absence of end-to-end paths. This work studies the feasibility, limits and convergence of monitoring such DTNs. More specifically, we focus on the efficient monitoring of intercontact time distribution (ICT) between DTN participants. Our contribution is two-fold. First we propose two schemes to sample data using monitors deployed within the DTN. In particular, we sample and estimate the ICT distribution. Second, we evaluate this scheme over both simulated DTN networks and real DTN traces. Our initial results show that (i) there is a high correlation between the quality of sampling and the sampled mobility type, and (ii) the number and placement of monitors impact the estimation of the ICT distribution of the whole DTN.

A new class of erasure codes for delay-constraint applications, called on-the-fly coding, have recently been introduced for their improvements in terms of recovery delay and achievable capacity. Despite their promising characteristics, little is known about the complexity of the systematic and non-systematic variants of this code, notably for live multicast transmission of multimedia content which is their ideal use case. Our paper aims to fill this gap and targets specifically the metrics relevant to mobile receivers with limited resources: buffer size requirements and computation complexity of the receiver. As our contribution, we evaluate both code variants on uniform and bursty erasure channels. Results obtained are unequivocal and demonstrate that the systematic codes outperform the non-systematic ones, in terms of both the buffer occupancy and computation overhead.

We present a solution to evaluate the performance of transport protocols as a function of link layer reliability schemes (i.e. ARQ, FEC and Hybrid ARQ) applied to satellite physical layer traces. As modelling such traces is complex and may require approximations, the use of real traces will minimise the potential for erroneous performance evaluations resulting from imperfect models. Our Trace Manager Tool (TMT) produces the corresponding link layer output, which is then used within the ns-2 network simulator via the additionally developed ns-2 interface module. We first present the analytical models for the link layer with bursty erasure packets and for the link layer reliability mechanisms with bursty erasures. Then, we present details of the TMT tool and our validation methodology, demonstrating that the selected performance metrics (recovery delay and throughput efficiency) exhibit a good match between the theoretical results and those obtained with TMT. Finally, we present results showing the impact of different link layer reliability mechanisms on the performance of TCP Cubic transport layer protocol.

We propose a novel DTN routing algorithm, called DQN, specifically designed for quasi-deterministic networks with an application to satellite constellations. We demonstrate that our proposal efficiently forwards the information over a satellite network derived from the Orbcomm topology while keeping a low replication overhead. We compare our algorithm against other well-known DTN routing schemes and show that we obtain the lowest replication ratio without the knowledge of the topology and with a delivery ratio of the same order of magnitude than a reference theoretical optimal routing.

This article proposes a stochastic model to obtain the end-to-end delay law between two nodes of a Delay Tolerant Network (DTN). We focus on the commonly used Binary Spray and Wait (BSW) routing protocol and propose a model that can be applied to homogeneous or heterogeneous networks (i.e. when the inter-contact law parameter takes one or several values). To the best of our knowledge, this is the first model allowing to estimate the delay distribution of Binary Spray and Wait DTN protocol in heterogeneous networks. We first detail the model and propose a set of simulations to validate the theoretical results.

Video streaming for heterogeneous types of devices, where nodes have different devices characteristics in terms of computational capacity and display, is usually handled by encoding the video with different qualities. This is not well suited for Peer-To-Peer (P2P) systems, as a single peer group can only share content of the same quality, thus limiting the peer group size and efficiency. To address this problem, several existing works propose the use of Multiple Descriptions Coding (MDC). The concept of this type of video codec is to split a video in a number of descriptions which can be used on their own, or aggregated to improve the global quality of the video. Unfortunately existing MDC codes are not flexible, as the video is split in a defined number of descriptions. In this paper, we focus on the practical feasibility of using a Fountain MDC code with properties similar to existing Fountain erasure codes, including the ability to create any number of descriptions when needed (on the fly). We perform simulations using selected pictures to assess the feasibility of using these codes, knowing that they should improve the availability of the video pieces in a P2P system and hence the video streaming quality. We observe that, although this idea seems promising, the evaluated benefits, demonstrated by the PSNR values, are limited when used in a real P2P video streaming system.

We propose a solution to mitigate the performance degradation and corresponding Quality of Experience (QoE) reduction caused by packet reordering for multimedia applications which utilise unreliable transport protocols like the Datagram Congestion Control Protocol (DCCP). We analytically derive the optimum buffer size based on the applications data rate and the maximum delay tolerated by the multimedia application. We propose a dynamically adjustable buffer in the transport protocol receiver which uses this optimum buffer size. We demonstrate, via simulation results, that our solution reduces the packet loss rate, increases the perceived bandwidth and does not increase jitter in the received applications packets while still being within the application's delay limits, therefore resulting in an increased QoE of multimedia applications.

In networks with very large delay like satellite IP-based networks, standard TCP is unable to correctly grab the available resource. To overcome this problem, Performance Enhancing Proxies (PEPs), which break the end-to-end connection and simulate a receiver close enough to the sender, can be placed before the links with large delay. Although splitting PEPs does not modify the transport protocol at the end nodes, they prevent the use of security protocols such as IPsec. In this paper, we propose solutions to replace the use of PEPs named SatERN. This proposal, based on Explicit Rate Notification (ERN) protocols over IP, does not split connections and is compliant with IP-in-IP tunneling solutions. Finally, we show that the SatERN solution achieves high satellite link utilization and fairness of the satellite traffic.

We present Xstream-x264: a real-time cross-layer video streaming technique implemented within a well known open-source H.264 video encoder tool x264 for on-line video quality evaluation experiments. Xstream-x264 uses the transport protocol provided indication of the available data rate for corresponding adjustments in the video encoder. We discuss the design, implementation and the quality evaluation methodology utilized with our tool. We demonstrate via experimental results that the streaming video quality greatly improves with the presented cross-layer approach both in terms of lost frame count and the objective video quality metrics Peak Signal to Noise Ratio (PSNR).

This paper introduces a robust point-to-point transmission scheme: Tetrys, that relies on a novel on-the-fly erasure coding concept which reduces the delay for recovering lost data at the receiver side. In current erasure coding schemes, the packets that are not rebuilt at the receiver side are either lost or delayed by at least one RTT before transmission to the application. The present contribution aims at demonstrating that Tetrys coding scheme can fill the gap between real-time applications requirements and full reliability. Indeed, we show that in several cases, Tetrys can recover lost packets below one RTT over lossy and best-effort networks. We also show that Tetrys allows to enable full reliability without delay compromise and as a result: significantly improves the performance of time constrained applications. For instance, our evaluations present that video-conferencing applications obtain a PSNR gain up to 7 dB compared to classic block-based erasure codes.

This paper presents STAMP: a tool to analyse SMTP servers overlay topology. STAMP builds a weighted and oriented graph from an email database, an email log or just a simple email header allowing a post-analysis of the SMTP overlay structure and the identification of the paths used by an email. The objective of this tool is twofold. STAMP allows to automatically perform an analysis of the SMTP topology for debugging (e.g, message delay, emails loop, ...) and for metrology purposes. While several traceroute-like measurements projects try to map the Internet, to the best of our knowledge, no tool allows to drive an analysis of the SMTP overlay network. Thus, the goal of the resulting graph is to develop methods (from graph theory, statistical analysis, ...) to identify relaying problems. We aim to explore the impact of IP network problems over emails delivery (and respectively: emails' traffic over IP networks) in conjunction with IP measurements driven synchronously. In the present paper, we introduce the design and the measurement methodology of the STAMP software and as second contribution, bring out to the networking community the tool and some measurements databases.

Delay Tolerant Networks (DTN) do not provide any end to end connectivity guarantee. Thus, transporting data over such networks is a tough challenge as most of Internet applications assume a form of persistent end to end connection. While research in DTN has mainly addressed the problem of routing in various mobility contexts with the aim to improve bundle delay delivery and data delivery ratio, little attention has been paid to applications. This paper investigates the support of streaming-like applications over DTN. We identify how DTN characteristics impact on the overall performances of these applications and present Tetrys, a transport layer mechanism, which enables robust streaming over DTN. Tetrys is based on an on the fly coding mechanism able to ensure full reliability without retransmission and fast in-order bundle delivery in comparison to classical erasure coding schemes. We evaluate our Tetrys prototype on real DTN connectivity traces captured from the Rollerblading tour in Paris. Simulations show that on average, Tetrys clearly outperforms all other reliability schemes in terms of bundles delivery service.

We present experimental results for the performance of selected voice codecs using the Datagram Congestion Control Protocol (DCCP) with TCP-Friendly Rate Control (TFRC) congestion control mechanism over a satellite link. We evaluate the performance of both constant and variable data rate speech codecs (G.729, G.711 and Speex) for a number of simultaneous calls, using the ITU E-model and identify problem areas and potential for improvement. Our experiments are done on a commercial satellite service using a data stream generated by a VoIP application, configured with selected voice codecs and using the DCCP/CCID4 Linux implementation. We analyse the sources of packet losses which are a main contributor to reduced voice quality when using CCID4 and additionally analyse the effect of jitter which is one of the crucial parameters contributing to VoIP quality and has, to the best of our knowledge, not been considered previously in the published DCCP performance results. We propose modifications to the CCID4 algorithm and demonstrate how these improve the VoIP performance, without the need for additional link information other than what is already monitored by CCID4 (which is the case for Quick-Start). We also demonstrate the fairness of the proposed modifications to other flows. We identify the additional benefit of DCCP when used in VoIP admission control mechanisms and draw conclusions about the advantages and disadvantages of the proposed DCCP/CCID4 congestion control mechanism for use with VoIP applications.

Although ironically it does not offer any real-time guarantee, Internet is a popular solution to support multimedia time-constrained applications (e.g. VoIP, Video Conferencing, ...).

This paper studies and analyses the benefits of favoring the transfer or packets of a TCP flow over a best-effort network. Specifically, we aim at studying whether we could improve the pace of short data request, such as HTTP request, by giving a high priority to that are not previously enqueued inside a core router. Following the idea that long-lived TCP flows greatly increase the routing queue delay, the motivation of this work is to minimise the impact in terms of delay, introduced by long-lived TCP flows over shot TCP flows. Thus, this forwarding scheme avoids to delay packets that do not belong to a flow already enqueued inside a router in order to avoid delay penalty to short flow. We define metrics to study the behaviour of such forwarding scheme and run several experiments over a complex and realistic topology. The results obtained present interesting and unexpected property of this forwarding scheme where not only short TCP flows take benefit of such routing mechanism.

The behaviour of the TCP AIMD algorithm is known to cause queue length oscillations when congestion occurs at a router output link. Indeed, due to these queueing variations, end-to-end applications experience large delay jitter. Many studies have proposed efficient Active Queue Management (AQM) mechanisms in order to reduce queue oscillations and stabilize the queue length. These AQM are mostly improvements of the Random Early Detection (RED) model. Unfortunately, these enhancements do not react in a similar manner for various network conditions and are strongly sensitive to their initial setting parameters. Although this paper proposes a solution to overcome the difficulties of setting these parameters by using a Kohonen neural network model, another goal of this study is to investigate whether cognitive intelligence could be placed in the core network to solve such stability problem. In our context, we use results from the neural network area to demonstrate that our proposal, named Kohonen-RED (KRED), enables a stable queue length without complex parameters setting and passive measurements.

TCP-friendly rate control (TFRC) is a congestion control technique that trade-offs responsiveness to the network conditions for a smoother throughput variation. We take advantage of this trade-off by calculating the rate gap between the theoretical TCP throughput and the smoothed TFRC throughput. Any rate gain from this rate gap is then opportunistically used for video coding. We define a frame complexity measure to determine the additional rate to be used from the rate gap and then perform a rate negotiation to determine the target rate for the encoder and the final sending rate. Results show that although this method has a more aggressive sending rate compared to TFRC, it is still TCP-friendly, does not contribute too much to network congestion and achieves a reasonable video quality gain over the conventional method.

Congestion control is a fundamental mechanism for the stability of the Internet and is a central mechanism for TCP. However, this congestion control mechanism focuses mainly on the core network state and is blind to the characteristics of wireless and mobile access networks. Moreover, TCP window bas

Based on an identification and evaluation of the subtle counterproductive interactions between the WLANs MAC layer and the transport layer, this paper shows a new approach towards congestion control for WLANs. We introduce a specialization of TFRC (MTFRC: mobile TFRC), which is adapted to wireless access networks. This TFRC specialization requires only slight changes to the standard TFRC protocol. Simulation results show substantial improvements for applications over TFRC in scenarios where the bottleneck situates on the MAC layer of the mobile nodes.

The features of mobility, which enormously impact on how communication is evolving into the future, represent a particular challenge in todaypsilas wireless networking research. After an identification and evaluation of the gap between the discontinuities of the communication service inherent to the physical layer of mobile networks and the continuity requirements issue from the stream centric multimedia applications, we propose a novel middleware 3MOI (Middleware for the Mobility Management Over the Internet) which can support efficient and context-aware mobility management and satisfy new mobility requirements such as dynamical location management, fast handover, and continuous connection support.

Pervasive communications are increasingly sent over mobile devices and personal digital assistants. This trend has been observed during the last football world cup where cellular phones service providers have measured a significant increase in multimedia traffic. To better carry multimedia traffic, the IETF standardized a new TCP Friendly Rate Control (TFRC) protocol. However, the current receiver-based TFRC design is not well suited to resource limited end systems. We propose a scheme to shift resource allocation and computation to the sender. This sender based approach led us to develop a new algorithm for loss notification and loss rate computation. We demonstrate the gain obtained in terms of memory requirements and CPU processing compared to the current design. Moreover this shifting solves security issues raised by classical TFRC implementations. We have implemented this new sender-based TFRC, named TFRCught, and conducted measurements under real world conditions.

In the context of a reconfigurable transport protocol, this paper introduces two protocol instances based on the composition and specialisation of the TFRC congestion control and Selective Acknowledgment mechanisms. The two resulting transport architectures lead respectively to the QTPAF protocol, specifically designed to operate over QoS-enabled networks and the QTPlight protocol, specifically designed for resource-limited end systems connected to powerful servers. QTPAF combines QoS-aware TFRC congestion control with full reliability to provide a transport service similar to TCP but additionally taking into account network-level bandwidth reservations. QTPlight proposes a modification of TFRC that shifts from the receiver to the sender the complexity of the loss rate estimation mechanism. This modification allows to alleviate the processing and communication load of "light" resource limited mobile receivers. We present the concept of these protocols and their adaptation in the EuQoS European project framework.

No abstract available.

Research reported here deals with a communication architecture with guaranteed end-to-end quality of service (QoS) in an IPv6 environment providing differentiated services within a single DiffServ domain. The article successively presents the design principles and services of the proposed architecture, their implementation over a national platform, and experimental measurements evaluating the QoS provided at the user level.