RFC 6437 introduces the 20-bit flow label field in the IPv6 header and recommends it to be used alongside source and destination IP addresses for efficient flow classification. A recent paper from Google authors explains how they have begun to use this field in their IPv6 network to select an alternative path, thereby preventing congestion or outages. However, there is no prior research investigating whether connections make use of this field on the Internet. In this paper, we present a comprehensive analysis of the 9-year MAWI dataset (2016-2024), the Internet traces gathered from a large backbone link in Japan, and examine whether the IPv6 flow label is indeed employed on the Internet. Our findings reveal an increasing upward trend in the adoption of the flow label field. In particular, this trend correlates with the use of QUIC, whose specification—different from TCP—explicitly repeats the recommendation to utilize the flow label.

Network researchers, academics, and students often need to deal with complex scripting in order to design a network topology, configure network nodes, automate experimentation, and visualize network performance metrics. Current simulation or emulation tools lack the ability to provide a comprehensive understanding of the network. This paper introduces TeachNet, a lightweight and user-friendly network tool that provides a graphical user interface for designing, configuring, and automating network experiments. We explain TeachNet’s design and present results to demonstrate its effectiveness and efficiency using our TCP experimentation-focused prototype, and Mininet in the backend. TeachNet creates a repository of log files and offers a framework for future research that is easily extensible to accommodate other tools such as OMNeT++ and ns3.

RFC 6437 specifies the usage of an IPv6 3-tuple (flow label, source and destination address fields) for flow classification in routers. Is this the only flow classification approach in use with IPv6 when the flow label is not zero? If so, end hosts could benefit from knowing that multiple flows between the same host pair traverse the same network bottleneck (for example, single-path congestion control coupling mechanisms could then reduce loss and delay). This paper attempts to answer this question with an investigation of whether path consistency is achievable via the IPv6 3-tuple in the Internet. We employ a novel method that extends Paris traceroute to control the path behavior solely using IPv6 network-layer header data, and conduct a large-scale measurement study from globally stationed vantage points to examine i) the extent to which the flow label is delivered to its destination, and ii) how it influences the behavior of load-balancing algorithms. Our results suggest that the flow label always passes through the network unaltered, but the 3-tuple is not a reliable flow identifier in the current Internet.

Technological advancements have provided wireless links with very high data rate capacity for 5G/6G mobile networks and WiFi 6, which will be widely deployed by 2025. However, the capacity can have substantial fluctuations, violating the assumption at the transport layer that the capacity is (almost) steady. In this paper, we present a general and efficient, yet deployable solution to this problem through a novel design empowered with a rich theory, allowing a significantly improved experience in using new technologies, especially mobile cellular services. We employ the well-known theory of food-chain models in biology, where a bottleneck link can be modeled as prey, while flows are predators. We extend this model to a chain of predators and preys to form a multi-hop congestion controller, called LGCC. Through simulation evaluation with real-life 5G traces we show the effectiveness of LGCC, compared with the state-of-the-art ABC (Accel-Brake Control). Our results show an order of magnitude bottleneck queuing delay decrease, with only a small decrease in throughput because LGCC tries to never exceed link capacities. LGCC’s design can additionally open a new paradigm in stable multi-hop congestion control and flow aggregation.

To fight global warming, carbon emissions must urgently be reduced. In this paper, we look at opportunities to do so by diminishing the operational energy usage of an important always-on element of the Internet: server systems. Our measurements from a server host indicate that there is probably not much to be gained by making a significant software change - but, depending on the system, a simple, easily overlooked configuration update may make a difference at no performance cost. This difference is small per server (in the order of an LED light bulb). Given the multitude of permanently operational servers in the world, it may however be significant at scale.

In response to ossification and privacy concerns, post-TCP transport protocols such as QUIC are designed to be "paranoid"---opaque to meddling middleboxes by encrypting and authenticating the header and payload---making it impossible for Performance-Enhancing Proxies (PEPs) to provide the same assistance as before. We propose a research agenda towards an alternate approach to PEPs, creating a sidecar protocol that is loosely-coupled to the unchanged and opaque, underlying transport protocol. The key technical challenge to sidecar protocols is how to usefully refer to the packets of the underlying connection without ossification. We have made progress on this problem by creating a tool we call a quACK (quick ACK), a concise representation of a multiset of numbers that can be used to efficiently decode the randomly-encrypted packet contents a sidecar has received. We implement the quACK and discuss how to achieve several applications with this approach: alternate congestion control, ACK reduction, and PEP-to-PEP retransmission across a lossy subpath.

Deploying a new network architecture in the Internet requires changing some, but not necessarily all elements between communicating applications. One way to achieve gradual deployment is a proxy or gateway which "translates" between the new architecture and TCP/IP. We present such a proxy, called "Performance Enhancing Proxy for Deploying Network Architectures (PEP-DNA)", which allows TCP/IP applications to benefit from advanced features of a new network architecture without having to be redeveloped. Our proxy is a kernel-based Linux implementation which can be installed wherever a translation needs to occur between a new architecture and TCP/IP domains. We discuss the proxy operation in detail and evaluate its efficiency and performance in a local testbed, demonstrating that it achieves high throughput with low additional latency overhead. In our experiments, we use the Recursive InterNetwork Architecture (RINA) and Information-Centric Networking (ICN) as examples, but our proxy is modular and flexible, and hence enables realistic gradual deployment of any new "clean-slate" approaches.

Most Internet hosts today support multiple access technologies and network interfaces. Multipath transport protocols, like MPTCP, are being deployed (e.g., in smartphones), allowing transparent simultaneous use of multiple links. Besides providing increased resilience to link failures, multipath transports may better exploit available (aggregate) capacity across all interfaces. The safest way to ensure fairness is to assume that any subflows of a multipath end-to-end connection may share bottleneck links, but knowledge of non-shared bottlenecks could allow multipath senders to exploit more capacity without being unfair to other flows. The problem of reliably detecting the existence of (non)-shared bottlenecks is not trivial and is compounded by the fact that bottlenecks may change due to traffic dynamics. In this paper we focus on practical methods to reliably group flows that share, possibly dynamic, bottlenecks online and in a passive manner (i.e., without injecting measurement traffic). We introduce a novel dynamic clustering algorithm that we apply to update our previous shared bottleneck flow grouping (SBFG) method standardized by the IETF, based on delay statistics. We also adapt an offline SBFG method based on wavelet filters to enable it for online operation. These SBFG methods are evaluated by a simple testbed, rigorous simulation and real-world Internet experiments in a testbed comprised of multihomed hosts. Our results suggest that there is no clear winner, and selection of the “best” SBFG method will have to consider tradeoffs regarding accuracy, lag, and application requirements.

DiffServ was designed to implement service provider quality of service (QoS) policies, where routers change and react upon the DiffServ Code Point (DSCP) in the IP header. However, nowadays, applications are beginning to directly set the DSCP themselves, in the hope that this will yield a more appropriate service for their respective video, audio and data streams. WebRTC is a prime example of such an application. We present measurements, for both IPv4 and IPv6, of what happens to DSCP values along Internet paths after an end system has set them without any prior agreement between a customer and a service provider. We find that the DSCP is often changed or zeroed along the path, but detrimental effects from using the DSCP are extremely rare; moreover, DSCP values sometimes remain intact (potentially having an effect on traffic) for several AS hops. This positive result motivates an analysis of the potential latency impact from such DSCP usage, for which we present the first measurement results. We find that routers at approximately 3% of more than 100,000 links differentiate between the WebRTC DSCP values (EF, AF42 and CS1) and consistently reduce delay in comparison with probes carrying a zero value (CS0) under congestion. In contrast, routers at around 2% of these links increase the delay by a comparable amount under congestion, uniformly for EF, AF42 and CS1.

XTRA (XFSM for Transport) aims at providing a first attempt towards a “code-once-port-everywhere” platform-agnostic programming abstraction tailored to the deployment of transport layer functions. XTRA's programming abstraction not only fits SW platforms, but is specifically designed to harness, with no re-coding effort, the offloading opportunities offered by CPU-less HW boards or smart NICs. We demonstrate the viability of XTRA with three completely different implementations of the underlying execution engine (HW proof-of-concept on a NetFPGA board, User-space SW over Linux' Open Data Plane, and NS3 emulator). Flexibility is shown via a number of example applications, ranging from a variety of congestion control algorithms, to a middlebox-type TCP proxy functionality, up to a customized “Timer-Based” (TB) TCP which leverages the native reliance of XTRA on timers, so as to produce a loss recovery operation which, despite being formalized only via a handful of code lines, performs almost comparable with the highly optimized Linux and FreeBSD implementations.

The challenge addressed in this paper consists in offloading packet-based pacing to a hardware Network Interface Card, while retaining the flexibility of software timers. In this direction, we propose, design, implement, and evaluate a hardware calendar, which can be programmed via a simple yet very flexible programming interface leveraging stateful (adaptive) per-packet timers. We show, for both specific examples (exponential, linearly increasing, etc) as well as for the general case, how to derive such a per-packet timer setting from a high-level desired rate envelope. Further, we describe and evaluate an FPGA implementation which relies on a novel insertion strategy for solving collisions in the calendar's hash table.

Communication between two Internet hosts using parallel connections may result in unwanted interference between the connections. In this dissertation, we propose a sender-side solution to address this problem by letting the congestion controllers of the different connections collaborate, correctly taking congestion control logic into account. Real-life experiments and simulations show that our solution works for a wide variety of congestion control mechanisms, provides great flexibility when allocating application traffic to the connections, and results in lower queuing delay and less packet loss.

One of the main goals of multipath TCP (MPTCP) is to achieve higher throughput than regular TCP by utilizing multiple paths simultaneously. When these paths share a common bottleneck, MPTCP tries not to be more aggressive than a regular TCP flow. This is achieved by MPTCP's coupled congestion control mechanism that couples the increase factor of MPTCP's subflows in congestion avoidance. However, slow-start remains unchanged and behaves uncoupled for each subflow, affecting MPTCP and concurrent traffic at the bottleneck. We propose LISA, a simple algorithm for coupling MPTCP subflows in slow-start, and investigate the trade-off that this coupling entails. Our evaluations show that coupling in slow-start not only provides gains for MPTCP but also for a concurrent TCP at the bottleneck.

RINA, the Recursive InterNetwork Architecture, is a novel “back to basics” type approach to networking. The recursive nature of RINA calls for radically different approaches to how networking is performed. It shows great potential in many aspects, e.g. by simplifying management and providing better security. However, RINA has not been explored for congestion control yet. In this paper, we take first steps to investigate how congestion control can be performed in RINA, and demonstrate that it can be very efficient because it is applied close to where the problem happens, and through its recursive architecture, interesting effects can be achieved. We also show how easily congestion control can be combined with routing, enabling a straightforward implementation of in-network resource pooling.

Multipath TCP (MPTCP) enables the simultaneous use of multiple links for bandwidth aggregation, better resource utilization and improved reliability. Its coupled congestion control intends to reap the increased bandwidth of multiple links, while avoiding being more aggressive than regular TCP flows on every used link. We argue that this leads to a very conservative behavior when paths do not share a bottleneck. Therefore, in this paper, we first quantify the penalty of the coupled congestion control for links that do not share a bottleneck. Then, in order to overcome this penalty, we design and implement a practical shared bottleneck detection (SBD) algorithm for MPTCP, namely MPTCP-SBD. Through extensive emulations, we show that MPTCP-SBD outperforms all currently deployed MPTCP coupled congestion controls by accurately detecting bottlenecks. For the non-shared bottleneck scenario, we observe throughput gains of up to 40% with two subflows and the gains increase significantly as the number of subflows increase, reaching more than 100% for five subflows. Furthermore, for the shared bottleneck scenario, we show that MPTCP-SBD remains fair to TCP. We complement the emulation results with real-network experiments justifying its safeness for deployment.

Explicit Congestion Notification (ECN) lets a bottleneck's Active Queue Management (AQM) mechanism inform an endpoint about congestion without having to drop a packet. A recently proposed sender-side modification called Alternative Backoff with ECN (ABE) enables reduced latency while maintaining good utilization with ECN. However, under certain circumstances ABE can produce a degree of unfair behavior between ABE-enabled TCP senders and conventional TCP senders. We propose specific guidance for configuring bottleneck AQMs to assist in fairness between ABE-enabled and conventional TCP flows. We evaluate our proposal using RED, then describe how it can be applied to other AQM mechanisms and incrementally introduced into the Internet.

Datacenter transport has attracted much recent interest, however, most proposed improvements require changing the datacenter fabric, which hinders their applicability and deployability over commodity hardware. In this paper, we present a novel congestion controller, Logistic Growth Control (LGC), for datacenters which does not require changes to the datacenter fabric. LGC uses a similar ECN marking as in DCTCP, but adapts to congestion using the logistic growth function. This function has been proven to have nice characteristics including stability, convergence, fairness, and scalability, which are very appealing for congestion control. As a result, our LGC mechanism operates in the datacenter network in a more stable and fair manner, leading to less queuing and latency. LGC also behaves better than DCTCP, and it converges to the fair share of the bottleneck link capacity irrespective of the Round-Trip-Time (RTT). We discuss the stability and fairness of LGC using a fluid model, and show its performance improvement with simulations.

When multiple congestion controlled flows traverse the same network path, their resulting rate is usually an outcome of their competition at the bottleneck. The WebRTC / RTCWeb suite of standards for inter-browser communication is required to allow prioritization. This is addressed by our previously presented mechanism for coupled congestion control, called the Flow State Exchange (FSE). Here, we present our first simulation results using two mechanisms that have been proposed for IETF standardization: Google Congestion Control (GCC) and Network-Assisted Dynamic Adaptation (NADA). These two mechanisms exhibit aspects that allow us to use a simpler “passive” algorithm in our FSE. Passive coupling allows a less time-constrained request-response style of signaling between congestion control mechanisms and the FSE, which enables the FSE to run as a stand-alone management tool.

Practical shared bottleneck detection has proved to be a difficult problem. We present a novel passive approach using efficient estimates of time and frequency domain summary statistics. The approach is not CPU nor network intensive, and has numerous potential applications in the Internet. Simulations and tests over the Internet and 3G cellular network show its efficacy in grouping flows correctly.

HTTP is a successful Internet technology on top of which a lot of the web resides. However, limitations with its current specification have encouraged some to look for the next generation of HTTP. In SPDY, Google has come up with such a proposal that has growing community acceptance, especially after being adopted by the IETF HTTPbis-WG as the basis for HTTP/2.0. SPDY has the potential to greatly improve web experience with little deployment overhead, but we still lack an understanding of its true potential in different environments. This paper offers a comprehensive evaluation of SPDY's performance using extensive experiments. We identify the impact of network characteristics and website infrastructure on SPDY's potential page loading benefits, finding that these factors are decisive for an optimal SPDY deployment strategy. Through exploring such key aspects that affect SPDY, and accordingly HTTP/2.0, we feed into the wider debate regarding the impact of future protocols.

Underlay-unawareness in P2P systems can result in sub-optimal peer selection for overlay routing and hence poor performance. The majority of underlay aware proposals for peer selection focus on finding the shortest overlay routes by selecting the nearest peers according to proximity. However, in case of multiple and parallel downloads, if the underlay paths between a downloader and its selected nearest peers share a bottleneck, this can cause congestion, leading to performance deterioration instead of improvement. This effect was neglected in previous work because, in today's Internet, the bottleneck is usually not shared as it is the end user's access link. This is no longer the case in more modern scenarios, e.g. with FTTH or with upcoming in-network caching techniques such as DECADE. We propose an improved peer selection approach for P2P applications called Fewest Common Hops (FCH) that ensures proximity based node selection having maximum path disjointness. It is a client based, infrastructure independent heuristic to optimize download time for multiple and parallel downloads in P2P content distribution applications. Simulations show that, even when FCH is implemented in the simplest possible fashion (using only traceroute), it can significantly decrease the download time.

is no longer too young to be usable: the first RFCs were published in 2006, and a stable and quite complete Linux implementation exists. DCCP over UDP has also recently been specified to address network traversal problems. But how good is the service provided to applications by this protocol? This paper identifies some deficiencies of the current implementation—the lack of transparency in the API with regard to packet loss, the coarse granularity of the lookup table used to calculate the TFRC equation, and the lack of history discounting in CCID-3—and demonstrates that they can significantly impair the performance of typical DCCP use cases such as live video streaming. Solutions are proposed to tackle all these problems, and it is shown that they considerably improve the performance and the flexibility of applications. I.

The goal of Rate Adaptation (RA) mechanisms in 802.11 WLANs is to provide optimum system throughput under varying channel conditions (e.g. in presence of noise) by carrying out run-time prediction and selection of the most appropriate bit-rate. The cross-layer interaction of TCP, as the major transport protocol in the Internet, with different RA mechanisms and DCF is yet to be thoroughly investigated. Previously reported efforts A) have never included real-life measurements of uplink TCP traffic; B) lack a practical view because they do not consider the RA mechanisms commonly deployed in today's off-the-shelf 802.11 devices; C) miss the study of RA mechanisms in low-noise environments. This paper covers all the above shortages, by conducting real-life measurements in two different test-beds (NDlab and Emulab) alongside with simulations, to study the performance of TCP coupled with different commonly deployed RA mechanisms. Our measurements reveal that 1) most conventional RA mechanisms are unable to distinguish frame errors due to collisions from channel noise/interference, and will respond to them negatively to some extent; 2) different than downlink TCP, uplink TCP can be adversely affected by collision-triggered rate downshifts that some RA schemes exhibit even under perfect channel conditions or in low-noise environments; 3) the relatively recent Minstrel RA mechanism can counter this negative uplink behavior well, yielding almost equal performance as in the downlink case.

SUMMARY Grid applications move large amounts of data between distributed resources, and the efficiency of a Grid depends on their timely delivery within given bounds (deadlines). In most cases, the data volume and deadline are known in advance, allowing for both network planning and connection admission control (textrmCAC). We formally define the problem and, based on this formalization, describe the operation of a feasible procedure for network reservations of deadline‐constrained bulk data transfer requests. The procedure guarantees a minimum bandwidth to meet the deadlines and allows for opportunistic utilization of residual network capacity. We propose a novel analytical model based on the solution of anM/M(nc)/1/k(s)−RPSqueue. The analytical model is validated againstns−2 simulations taking into account network level details (IP and TCP protocols), showing remarkably good coherence even under heavy loads. The model is orders of magnitude faster than simulation, which enables its application to plan the capacity of Grid networks, and to enforce CAC under the hypothesis of a dominating bottleneck on the transfer route. Copyright © 2011 John Wiley & Sons, Ltd.

This work proposes a quality-oriented, real-time capable prioritization technique for media units of H.264/AVC video streams. The derivation of estimates is based on the analysis of the macroblock partitioning, the spatial extents of temporal dependencies, and the length and strength of prediction chains existing among macroblocks, thus incorporating the expected impact of error propagation. It is demonstrated how the prioritization scheme can be beneficially integrated into live streaming systems which are characterized by tight timing constraints, with the focus on content-aware selective automatic repeat request mechanisms. Additionally, it is shown how potentially limited feedback can be used to adapt the estimation process to leverage prediction preciseness. The approach is compared against existing techniques in terms of practicability and efficiency, and tested under independent and bursty loss conditions in a wired and a wireless test setup. Moreover, the performance is examined when low-latency and constant bitrate video settings are enforced by using x264's novel encoding feature periodic-intra-refresh. Results of both experiments and simulations indicate that the proposed technique outperforms all reference techniques in nearly all test cases, and that the video quality can be further improved by incorporating receiver feedback.

With the standardization of SVC, the scalable extension of H.264/AVC, layered peer-to-peer streaming has attracted more and more attention as it offers adaptability to network fluctuations and heterogeneous end users. Although overlay construction is important for system performance, not much effort has been spent on unstructured overlay construction for layered peer-to-peer streaming. Related work concentrates on layered streaming algorithms, and assumes that a list of peers for data exchange, called neighbors, is provided by traditional membership management protocols, e.g., SCAMP. Our previous studies have demonstrated that a random overlay is not good enough for layered peer-to-peer streaming. In this paper, we propose a new membership management protocol, based on peer sampling services. The protocol is quality-aware as it constructs the overlay so that (1) high capacity peers will be located at good positions in the overlay, e.g., close to the server, and (2) peers having similar capacity are likely to connect to each other. Both features are necessary to maximize bandwidth utilization of peers and to mitigate layer bottlenecks. With implementation of the protocol in PeerSim, we evaluate important graph properties of the overlay, constructed by the proposed protocol, to understand how it evolves during the streaming session with peer churn. Evaluation results show that the overlay is (1) scalable: it is stable with different sizes, from hundreds up to 10000 peers; and (2) robust: the good features are maintained or recovered fast under a high peer churn rate, and it only becomes disconnected when more than 86% of the peers are removed from the network.

We present a novel mechanism for estimating the perceptual importance of network packets constituting a video stream. We explain how this information can be used to extend existing rateless coding schemes and present first results demonstrating the expected increase in perceptual quality.

The Internet-wide deployment of new transport protocols such as the Stream Control Transmission Protocol (SCTP) is a difficult matter. While SCTP could be beneficial in many cases, it is still a major challenge to enable applications to use the new protocol. We believe that its deployment could be significantly accelerated by introducing an intermediate step: transparent usage below TCP, such that TCP-based applications automatically obtain a benefit. We have implemented a Linux based TCP-SCTP mapping tool that exploits SCTP's multi-streaming feature, and carried out measurements which show that such an approach can significantly improve the performance of TCP applications. However, and perhaps more importantly, we also encountered difficulties, which lead us to make some concrete recommendations regarding future research and the implementation and specification of SCTP itself.

Congestion control in transport protocols can be performed implicit when packets are lost, or explicit with feedback from the network. In this paper we compare some aspects of implicit and explicit congestion control using a variant of CUBIC with Quick-Start, called CUBIQ. We compare CUBIQ with regular TCP, XCP and CUBIC, and find that CUBIQ performs as well as XCP in most cases, and much better than XCP when XCP is hampered by small buffers in large BDP-networks. These results indicate that the feedback provided by Quick-Start may be enough for a sender when it needs to find the right transmission rate, also in the initial phase of a communication.

Consumer broadband services are increasingly a mix of TCP-based and UDP-based applications, often with quite distinct requirements for interactivity and network performance. Consumers can experience degraded service when application traffic collides at a congestion point between home LANs, service provider edge networks and fractional-Mbit/sec `broadband' links. We illustrate two key issues that arise from the impact of TCP-based data transfers on real-time traffic (such as VoIP or online games) sharing a broadband link. First, well-intentioned modifications to traditional TCP congestion control can noticeably increase the latencies experienced by VoIP or online games. Second, superficially-similar packet dropping rules in broadband gateways can induce distinctly different packet loss rates in VoIP and online game traffic. Our observations provide cautionary guidance to researchers who model such traffic mixes, and to vendors implementing equipment at either end of consumer links.

TCP is quite a heavyweight protocol when serving very small web pages. We introduce a server-side kernel modification which enables a web server to perform HTTP over a UDP socket while the kernel provides a regular TCP interface 'on the wire' to remote clients. We show that our 'stateless' TCP modification can greatly reduce a server's CPU usage (>20%) and TCP related memory requirements(>90%), potentially enabling it to serve small web pages even under extreme overload conditions.

Over the past years, several schemes have been proposed to estimate the perceptual distortion in video quality caused by packet loss. Some of them are intuitively designed and do not correlate well with actual distortion values in most real-world scenarios. Others simulate decoders' error concealment measures and use the quality degradation of reconstructed regions as basis for calculating importance estimates. However, such techniques do not take error propagation in the temporal domain into account and are computationally expensive, especially when estimates cannot be derived in parallel to, or as side-product of the video encoding process. We therefore propose a novel approach which targets all previously mentioned problems and inspects dependencies between media units at the level of macroblocks. We show how our scheme can be applied to live video streaming systems using rateless codes for error protection and identify pitfalls which have to be considered. Test results indicate that the proposed unequal error protection scheme considerably outperforms previous approaches, independent of the resolution of the test sequences.

In the first part of this paper, we present a simple extension of the well-known TCP steady-state throughput equation that can be used to calculate the throughput of several flows that share an end-to-end path. The value of this extension, which we show to work well with simulations as well as real-life measurements, is its practical applicability. Thus, in the second part of this paper, we present its application in MulTFRC, a TCP-friendly rate control (TFRC)-based congestion control mechanism that is fair to a number of parallel TCP flows while maintaining a smoother sending rate than multiple real TFRC flows do. MulTFRC enables its users to prioritize transfers by controlling the fairness among them in an almost arbitrary fashion.

Grid applications need to move large amounts of data between distributed resources within deterministic timeframes. In most cases it is possible to specify the volume and the deadline in advance. We formally define and analyze a mechanism of network reservations of bulk data transfer requests having opportunistic utilization of residual network capacity and analyze it using an M/M/1/N-RPS queue. We compare the analytical results of our mechanism with the analytical results obtained from an M/M/N/N queue, for the case in which there is no opportunistic sharing of residual capacity. We validate the analytical results through simulations in C++. The analytical model is also validated against ns-2 simulations taking into account network level details (IP and TCP protocols), showing remarkably good coherence even under heavy loads. The model is orders of magnitude faster than simulation, which enables its application for capacity planning of Grid networks (also known as network dimensioning), and to enforce connection admission control (CAC) under the practical hypothesis of a dominating bottleneck on the transfer route.

WiFi, i.e., IEEE 802.11 is one of the most widely used technologies to implement internet access in today's networks. Typical frame error rates in IEEE 802.11 are much higher than in wired links. This negatively impacts the behavior of TCP, which assumes that packet loss is due to congestion. This paper presents LN-MAC, a cross-layer solution based on explicit loss notification (ELN), where the IEEE 802.11 MAC is enhanced to provide loss notifications to the TCP layer. Extensive simulation results demonstrate the benefits of LN-MAC. Numerical results demonstrate that under any condition TCP throughput is always higher when using LN-MAC than when using conventional IEEE 802.11.

A sudden delay spike or reordering in the network can cause TCP to experience a loss event. Since loss is interpreted as a sign of congestion in TCP, this causes the protocol to reduce its sending rate. Several mechanisms for detecting and reacting to such spurious loss events have been proposed; each of them has some advantages and disadvantages. We extend this space with a new detection mechanism which complements the existing ones well. Furthermore, the mechanism is easy to implement because it only needs the sender to intelligently interpret feedback from ECN nonce.

The two transport protocols DCCP and UDP-Lite can make use of data that are known to be erroneous, provided that the link layer hands over such data. A similar functionality has been suggested for TCP. In order to investigate the potential of these mechanisms in WiFi networks, we carried out a measurement study where we examined how often information about corrupt data reaches the transport layer when the corruption control at the link layer is disabled. Our results suggest that this may be a rare occurrence in certain scenarios.

Due to the growth of unresponsive UDP traffic in the Internet, it becomes increasingly important for ISPs to amply shape the traffic that leaves their network. Ideally, flows should be forced to be TCP-friendly; to this end, knowledge about certain end-to-end path characteristics is needed. We present a suitable mechanism (the Burst-PiggyBack (BPB) technique) that obtains the necessary information at a device that is located close to the sender without requiring any changes at the communicating peers or in routers. The method’s hypothesis is that an injected probe packet at the end of a burst is treated similar to the burst. The results acquired from simulations showed strong correlations between the loss rate and the RTT of bursts and probe packets.

Properly reacting to congestion is known to be a requirement for maintaining the stability of the Internet; in addition, UDP based applications should do so for their own sake if they are delay- or loss-sensitive. We measured the responsiveness of the three popular streaming media applications RealPlayer, Windows MediaPIayer and Quicktime with a varying amount of cross traffic and present a comparison of the results.

Since the IETF "TCPSAT" Working Group was chartered in 1997, numerous issues with TCP over satellite have been addressed. Although the difficulties encountered are the same, it seems as though adaptive multimedia communication over satellite has been neglected. We identify the common problems, discuss their impact and introduce a method to deal with some of them-a protocol that queries routers for QoS related information.

We describe a simple protocol that enhances the communication between end nodes and "the network". Other than the majority of QoS signaling systems, it achieves scalability by avoiding per-flow state. We also show how it can be used to decrease an adaptive multimedia application's packet loss ratio.