Packet Loss

Reliable transport protocols such as TCP are tuned to perform well in traditional networks where packet losses occur mostly because of congestion. However, networks with wireless and other lossy links also suffer from significant losses due to bit errors and handoffs. TCP responds to all losses by invoking congestion control and avoidance algorithms, resulting in degraded end-to-end performance in wireless and lossy systems. In this paper, we compare several schemes designed to improve the performance of TCP in such networks. We classify these schemes into three broad categories: end-to-end protocols, where loss recovery is performed by the sender; link-layer protocols, that provide local reliability; and split-connection protocols, that break the end-to-end connection into two parts at the base station. We present the results of several experiments performed in both LAN and WAN environments, using throughput and goodput as the metrics for comparison.

Wireless sensor networks promise fine-grain monitoring in a wide variety of environments. Many of these environments (e.g., indoor environments or habitats) can be harsh for wireless communication. From a networking perspective, the most basic aspect of wireless communication is the packet delivery performance: the spatio-temporal characteristics of packet loss, and its environmental dependence. These factors will deeply impact the performance of data acquisition from these networks.

We discuss findings from a large-scale study of Internet packet dynamics conducted by tracing 20,000 TCP bulk transfers between 35 Internet sites. Because we traced each 100 Kbyte transfer at both the sender and the receiver, the measurements allow us to distinguish between the end-to-end behaviors due to the different directions of the Internet paths, which often exhibit asymmetries. We characterize the prevalence of unusual network events such as out-of-order delivery and packet corruption; discuss a robust receiver-based algorithm for estimating "bottleneck bandwidth" that addresses deficiencies discovered in techniques based on "packet pair"; investigate patterns of packet loss, finding that loss events are not well-modeled as independent and, furthermore, that the distribution of the duration of loss events exhibits infinite variance; and analyze variations in packet transit delays as indicators of congestion periods, finding that congestion periods also span a wide range of time scales. In x 3 we characterize unusual network behavior: out-of-order delivery, replication, and packet corruption. Then in x 4 we discuss a robust algorithm for estimating the "bottleneck" bandwidth that limits a connection's maximum rate. This estimation is crucial for subsequent analysis because knowing the bottleneck rate lets us determine when the closely-spaced TCP data packets used for our network probes are correlated with each other. (We note that the stream of ack packets returned by the TCP data receiver in general is not correlated, due to the small size and larger spacing of the acks.) Once we can determine which probes were correlated and which not, we then can turn to analysis of end-to-end Internet packet loss (x 5) and delay (x 6). In x 7 we briefly summarize our findings, a number of which challenge commonly-held assumptions about network behavior.

This paper presents a full duplex radio design using signal inversion and adaptive cancellation. Signal inversion uses a simple design based on a balanced/unbalanced (Balun) transformer. This new design, unlike prior work, supports wideband and high power systems. In theory, this new design has no limitation on bandwidth or power. In practice, using a channel sounder, we find that the signal inversion technique alone can cancel at least 45dB across a 40MHz bandwidth. Further, combining signal inversion cancellation with cancellation in the digital domain can reduce self-interference by up to 73dB for a 10MHz OFDM signal.

Multicast stream authentication and signing is an im-portant and challenging problem. Applications include the continuous authentication of radio and TV Internet broad-casts, and authenticated data distribution by satellite. The main challenges are fourfold. First, authenticity must be ...

One of the main challenges of securing broadcast communication is source authentication, or enabling receivers of broadcast data to verify that the received data really originates from the claimed source and was not modified en route. This problem is complicated by ...

This article proposes a novel approach to assess the stability of linear systems with delayed and sampled-data inputs. The paper considers both asynchronous sampling and input delay based on an extension of existing results on the stability of sampled-data systems to the case where a delay is introduced in the control loop. The proposed method provides easy tractable sufficient conditions for asymptotic stability of sampled-data systems under asynchronous sampling and transmission delays. The period and delay-dependent conditions are expressed using computable linear matrix inequalities. Several examples show the efficiency of the stability criteria.

We present Trickle, an algorithm for propagating and maintaining code updates in wireless sensor networks. Borrowing techniques from the epidemic/gossip, scalable multicast, and wireless broadcast literature, Trickle uses a "polite gossip" policy, where motes periodically broadcast a code summary to local neighbors but stay quiet if they have recently heard a summary identical to theirs. When a mote hears an older summary than its own, it broadcasts an update. Instead of flooding a network with packets, the algorithm controls the send rate so each mote hears a small trickle of packets, just enough to stay up to date. We show that with this simple mechanism, Trickle can scale to thousand-fold changes in network density, propagate new code in the order of seconds, and impose a maintenance cost on the order of a few sends an hour.

TCP is a reliable transport protocol tuned to perform well in traditional networks where congestion is the primary cause of packet loss. However, networks with wireless links and mobile hosts incur significant losses due to biterrors and handoff. This environment violates many of the assumptions made by TCP, causing degraded end-to-end performance. In this paper, we describe the additions and modifications to the standard Internet protocol stack (TCP/IP) to improve end-to-end reliable transport performance in mobile environments. The protocol changes are made to network-layer software at the base station and mobile host, and preserve the end-to-end semantics of TCP. One part of the modifications, called the snoop module, caches packets at the base station and performs local retransmissions across the wireless link to alleviate the problems caused by high bit-error rates. The second part is a routing protocol that enables low-latency handoff to occur with negligible data loss. We have implemented this new protocol stack on a wireless testbed. Our experiments show that this system is significantly more robust at dealing with unreliable wireless links than normal TCP; we have achieved throughput speedups of up to 20 times over regular TCP and handoff latencies over 10 times shorter than other mobile routing protocols.

In this paper, we investigate the impact of radio irregularity on the communication performance in wireless sensor networks. Radio irregularity is a common phenomenon which arises from multiple factors, such as variance in RF sending power and different path losses depending on the direction of propagation. From our experiments, we discover that the variance in received signal strength is largely random; however, it exhibits a continuous change with incremental changes in direction. With empirical data obtained from the MICA2 platform, we establish a radio model for simulation, called the Radio Irregularity Model (RIM). This model is the first to bridge the discrepancy between spherical radio models used by simulators and the physical reality of radio signals. With this model, we are able to analyze the impact of radio irregularity on some of the well-known MAC and routing protocols. Our results show that radio irregularity has a significant impact on routing protocols, but a relatively small impact on MAC protocols. Finally, we propose six solutions to deal with radio irregularity. We evaluate two of them in detail. The results obtained from both the simulation and a running testbed demonstrate that our solutions greatly improve communication performance in the presence of radio irregularity.

We show that mutual exchange of independent information between two nodes in a wireless network can be efficiently performed by exploiting network coding and the physicallayer broadcast property offered by the wireless medium. The proposed approach improves upon conventional solutions that separate the processing of the two unicast sessions, corresponding to information transfer along one direction and the opposite direction. We propose a distributed scheme that obviates the need for synchronization and is robust to random packet loss and delay, and so on. The scheme is simple and incurs minor overhead.

Testing of network protocols and distributed applications has become increasingly complex, as the diversity of networks and underlying technologies increase, and the adaptive behavior of applications becomes more sophisticated. In this paper, we present NIST Net , a tool to facilitate testing and experimentation with network code through emulation. NIST Net enables experimenters to model and effect arbitrary performance dynamics (packet delay, jitter, bandwidth limitations, congestion, packet loss and duplication) on live IP packets passing through a commodity Linux-based PC router. We describe the emulation capabilities of NIST Net; examine its architecture; and discuss some of the implementation challenges encountered in building such a tool to operate at very high network data rates while imposing minimal processing overhead. Calibration results are provided to quantify the fidelity and performance of NIST Net over a wide range of offered loads (up to 1 Gbps), and a diverse set o...

This paper consists of a categorization of optical buffering strategies for optical packet switches, and a comparison of the performance of these strategies both with respect to packet loss/delay and bit error rate (BER) performance. Issues surrounding optical buffer implementation are discussed, and representative architectures are introduced under different categories. Conclusions are drawn about packet loss and BER performance, and about the characteristics an architecture should have to be practical. It is shown that there is a strong case for the use of optical regeneration for successful cascading of these architectures.

Wireless multimedia services are increasingly becoming popular boosting the need for better quality-of-experience (QoE) with minimal costs. The standard codecs employed by these systems remove spatio-temporal redundancies to minimize the bandwidth required. However, this increases the exposure of the system to transmission errors, thus presenting a significant degradation in perceptual quality of the reconstructed video sequences. A number of mechanisms were investigated in the past to make these codecs more robust against transmission errors. Nevertheless, these techniques achieved little success, forcing the transmission to be held at lower bit-error rates (BERs) to guarantee acceptable quality. This paper presents a novel solution to this problem based on the error detection capabilities of the transport protocols to identify potentially corrupted group-of-blocks (GOBs). The algorithm uses a support vector machine (SVM) at its core to localize visually impaired macroblocks (MBs) that require concealment within these GOBs. Hence, this method drastically reduces the region to be concealed compared to state-of-the-art error resilient strategies which assume a packet loss scenario. Testing on a standard H.263++ codec confirms that a significant gain in quality is achieved with error detection rates of 97.8% and peak signal-to-noise ratio (PSNR) gains of up to 5.33 dB. Moreover, most of the undetected errors provide minimal visual artifacts and are thus of little influence to the perceived quality of the reconstructed sequences.

Easy to deploy, robust, and highly resilient to failures, epidemic algorithms are a potentially effective mechanism for propagating information in large peer-to-peer systems deployed on Internet or ad hoc networks. It is possible to adjust the parameters of epidemic algorithm to achieve high reliability despite process crashes and disconnections, packet losses, and a dynamic network topology. Although researchers have used epidemic algorithms in applications such as failure detection, data aggregation, resource discovery and monitoring, and database replication, their general applicability to practical, Internet-wide systems remains open to question. We describe four key problems: membership maintenance, network awareness, buffer management, and message filtering, and suggest some preliminary approaches to address them.

The Adhoc Multicast Routing Protocol AMRoute presents a novel approach for robust IP Multicast in mobile adhoc networks by exploiting user-multicast trees and dynamic logical cores. It creates a bidirectional, shared tree for data distribution using only group senders and receivers as tree nodes. Unicast tunnels are used as tree links to connect neighbors on the user-multicast tree. T h us, AMRoute does not need to be supported by n e t work nodes that are not interested capable of multicast, and group state cost is incurred only by group senders and receivers. Also, the use of tunnels as tree links implies that tree structure does not need to change even in case of a dynamic network topology, which reduces the signaling tra c and packet loss. Thus AMRoute does not need to track network dynamics; the underlying unicast protocol is solely responsible for this function. AMRoute does not require a speci c unicast routing protocol; therefore, it can operate seamlessly over separate domains with di erent unicast protocols. Certain tree nodes are designated by AMRoute as logical cores, and are responsible for initiating and managing the signaling component of AMRoute, such as detection of group members and tree setup. Logical cores di er signi cantly from those in CBT and PIM-SM, since they are not a central point f o r data distribution and can migrate dynamically among member nodes. Simulation results demonstrate that AMRoute signaling tra c and join latency remain at relatively low l e v els for typical group sizes. The results also indicate that group members receive a high proportion of data multicast by a sender, even in the case of a dynamic network.

In this paper, optimal control of linear time-invariant (LTI) systems over unreliable communication links is studied. The motivation of the problem comes from growing applications that demand remote control of objects over Internet-type or wireless networks where links are prone to failure. Depending on the availability of acknowledgment (ACK) signals, two different types of networking protocols are considered. Under a TCP structure, existence of ACK signals is assumed, unlike the UDP structure where no ACK packets are present. The objective here is to meansquare (m.s.) stabilize the system while minimizing a quadratic performance criterion when the information flow between the controller and the plant is disrupted due to link failures, or packet losses. Sufficient conditions for the existence of stabilizing optimal controllers are derived. ᭧

We present the unequal loss protection (ULP) framework in which unequal amounts of forward error correction are applied to progressive data to provide graceful degradation of image quality as packet losses increase. We develop a simple algorithm that can find a good assignment within the ULP framework. We use the Set Partitioning in Hierarchical Trees coder in this work, but our algorithm can protect any progressive compression scheme. In addition, we promote the use of a PMF of expected channel conditions so that our system can work with almost any model or estimate of packet losses. We find that when optimizing for an exponential packet loss model with a mean loss rate of 20% and using a total rate of 0.2 bits per pixel on the Lenna image, good image quality can be obtained even when 40% of transmitted packets are lost.

We deployed 72 sensors of 10 modalities in 15 wireless and wired networked sensor systems in the environment, in objects, and on the body to create a sensor-rich environment for the machine recognition of human activities. We acquired data from 12 subjects performing morning activities, yielding over 25 hours of sensor data. We report the number of activity occurrences observed during post-processing, and estimate that over 13000 and 14000 object and environment interactions occurred. We describe the networked sensor setup and the methodology for data acquisition, synchronization and curation. We report on the challenges and outline lessons learned and best practice for similar large scale deployments of heterogeneous networked sensor systems. We evaluate data acquisition quality for on-body and object integrated wireless sensors; there is less than 2.5% packet loss after tuning. We outline our use of the dataset to develop new sensor network self-organization principles and machine learning techniques for activity recognition in opportunistic sensor configurations. Eventually this dataset will be made public.

An (m; n; b; r)-erasure-resilient coding scheme consists of an encoding algorithm and a decoding algorithm with the following properties. The encoding algorithm produces a set of n packets each containing b bits from a message of m packets containing b bits. The decoding algorithm is able to recover the message from any set of r packets. Erasure-resilient codes have been used to protect real-time tra c sent through packet based networks against packet losses. In this paper we describe a erasure-resilient coding scheme that is based on a version of Reed-Solomon codes and which has the property that r = m: Both the encoding and decoding algorithms run in quadratic time and have been customized to give the rst real-time implementations of Priority Encoding Transmission (PET) 2], 1] for medium quality video transmission on Sun

In this paper, we present Chainsaw, a p2p overlay multicast system that completely eliminates trees. Peers are notified of new packets by their neighbors and must explicitly request a packet from a neighbor in order to receive it. This way, duplicate data can be eliminated and a peer can ensure it receives all packets. We show with simulations that Chainsaw has a short startup time, good resilience to catastrophic failure and essentially no packet loss. We support this argument with real-world experiments on Planetlab and compare Chainsaw to Bullet and Splitstream using MACEDON.

We explore the use of end-to-end multicast traffic as measurement probes to infer network-internal characteristics. We have developed in an earlier paper [2] a Maximum Likelihood Estimator for packet loss rates on individual links based on losses observed by multicast receivers. This technique exploits the inherent correlation between such observations to infer the performance of paths between branch points in the multicast tree spanning the probe source and its receivers. We evaluate through analysis and simulation the accuracy of our estimator under a variety of network conditions. In particular, we report on the error between inferred loss rates and actual loss rates as we vary the network topology, propagation delay, packet drop policy, background traffic mix, and probe traffic type. In all but one case, estimated losses and probe losses agree to within 2 percent on average. We feel this accuracy is enough to reliably identify congested links in a wide-area internetwork.

In this paper, we investigate the impact of radio irregularity on wireless sensor networks. Radio irregularity is a common phenomenon which arises from multiple factors, such as variance in RF sending power and different path losses depending on the direction of propagation. From our experiments, we discover that the variance in received signal strength is largely random; however, it exhibits a continuous change with incremental changes in direction. With empirical data obtained from the MICA2 and MICAZ platforms, we establish a radio model for simulation, called the Radio Irregularity Model (RIM). This model is the first to bridge the discrepancy between spherical radio models used by simulators and the physical reality of radio signals. With this model, we investigate the impact of radio irregularity on several upper layer protocols, including MAC, routing, localization and topology control. Our results show that radio irregularity has a relatively larger impact on the routing layer than the MAC layer. It also shows that radio irregularity leads to larger localization errors and makes it harder to maintain communication connectivity in topology control. To deal with these issues, we present eight solutions to deal with radio irregularity. We evaluate three of them in detail. The results obtained from both the simulations and a running testbed demonstrate that our solutions greatly improve system performance in the presence of radio irregularity.

We describe a new carrier-sense multiple access (CSMA) protocol for multihop wireless networks, sometimes also called ad hoc networks. The CSMA protocol divides the available bandwidth into several channels and selects an idle channel randomly for packet transmission. It also employs a notion of “soft” channel reservation as it gives preference to the channel that was used for the last successful transmission. We show via simulations that this multichannel CSMA protocol provides a higher throughput compared to its single channel counterpart by reducing the packet loss due to collisions. We also show that the use of channel reservation provides better performance than multichannel CSMA with purely random idle channel selection

In order to stem the increasing packet loss rates caused by an exponential increase in network traffic, the IETF is considering the deployment of active queue management techniques such as RED . While active queue management can potentially reduce packet loss rates in the Internet, this paper shows that current techniques are ineffective in preventing high loss rates. The inherent problem with these queue management algorithms is that they all use queue lengths as the indicator of the severity of congestion. In light of this observation, a fundamentally different active queue management algorithm called BLUE is proposed. BLUE uses packet loss and link idle events to manage congestion. Using simulation and controlled experiments, BLUE is shown to perform significantly better than RED both in terms of packet loss rates and buffer size requirements in the network. As an extension to BLUE, a novel technique for enforcing fairness among a large number of flows is described. In particular, this paper proposes and evaluates Stochastic Fair BLUE (SFB), a queue management algorithm which can identify and rate-limit non-responsive flows using a very small amount of state information.

This paper investigates differentiated services in wireless packet networks using a fully distributed approach that supports service differentiation, radio monitoring, and admission control. While our proposal is generally applicable to distributed wireless access schemes, we design, implement, and evaluate our framework within the context of existing wireless technology. Service differentiation is based on the IEEE 802.11 Distributed Coordination Function (DCF) originally designed to support best-effort data services. We analyze the delay experienced by a mobile host implementing the IEEE 802.11 DCF and derive a closed-form formula.We then extend the DCF to provide service differentiation for delay-sensitive and best-effort traffic based on the results from the analysis. Two distributed estimation algorithms are proposed. These algorithms are evaluated using simulation, analysis, and experimentation. A Virtual MAC (VMAC) algorithm passively monitors the radio channel and estimates locally achievable service levels. The VMAC estimates key MAC level statistics related to service quality such as delay, delay variation, packet collision, and packet loss. We show the efficiency of the VMAC algorithm through simulation and consider significantly overlapping cells and highly bursty traffic mixes. In addition, we implement and evaluate the VMAC in an experimental differentiated services wireless testbed. A Virtual Source (VS) algorithm utilizes the VMAC to estimate application-level service quality. The VS allows application parameters to be tuned in response to dynamic channel conditions based on "virtual delay curves." We demonstrate through simulation that when these distributed virtual algorithms are applied to the admission control of the radio channel then a globally stable state can be maintained without the need for complex centralized radio resource management.

Wireless sensor networks involve very large numbers of small, low-power, wireless devices. Given their unattended nature, and their potential applications in harsh environments, we need a monitoring infrastructure that indicates system failures and resource depletion. In this paper, we briefly describe an architecture for sensor network monitoring, then focus on one aspect of this architecture: continuously computing aggregates (sum, average, count) of network properties (loss rates, energy-levels etc., packet counts). Our contributions are two-fold. First, we propose a novel tree construction algorithm that enables energy-efficient computation of some classes of aggregates. Second, we show through actual implementation and experiments that wireless communication artifacts in even relatively benign environments can significantly impact the computation of these aggregate properties. In some cases, without careful attention to detail, the relative error in the computed aggregates can be as much as 50%. However, by carefully discarding links with heavy packet loss and asymmetry, we can improve accuracy by an order of magnitude.

Congestion in wireless sensor networks not only causes packet loss, but also leads to excessive energy consumption. Therefore congestion in WSNs needs to be controlled in order to prolong system lifetime. In addition, this is also necessary to improve fairness and provide better quality of service (QoS), which is required by multimedia applications in wireless multimedia sensor networks. In this paper, we propose a novel upstream congestion control protocol for WSNs, called priority-based congestion control protocol (PCCP). Unlike existing work, PCCP innovatively measures congestion degree as the ratio of packet inter-arrival time along over packet service time. PCCP still introduced node priority index to reflect the importance of each sensor node. Based on the introduced congestion degree and node priority index, PCCP utilizes a cross-layer optimization and imposes a hop-by-hop approach to control congestion. We have demonstrated that PCCP achieves efficient congestion control and flexible weighted fairness for both single-path and multi-path routing, as a result this leads to higher energy efficiency and better QoS in terms of both packet loss rate and delay.

This paper describes current problems found with audio applications over the MBONE (Multicast Backbone), and investigates possible solutions to the most common one -packet loss. The principles of packet speech systems are discussed, and how the structure allows the use of redundancy to design viable solutions to the problem. The paper proposes the use of synthetic speech coding algorithms (vocoders) to provide redundancy, since the algorithms produce a very low bit-rate stream, which only adds a small overhead to a packet. Preliminary experiments show that normal speech repaired with synthetic quality speech is intelligible, even at very high loss rates.

This paper investigates differentiated services in wireless packet networks using a fully distributed approach that supports service differentiation, radio monitoring and admission control. Service differentiation is based on the IEEE 802.11 Distributed Coordination Function (DCF) originally designed to support best-effort data services. We extend the Distributed Coordination Function to provide service differentiation for delay sensitive and best-effort traffic. Two distributed estimation algorithms are proposed and analyzed. A Virtual MAC (VMAC) algorithm passively monitors the radio channel and estimates locally achievable service levels. The Virtual MAC estimates key MAC level statistics related to service quality such as delay, delay variation, packet collision and packet loss. We show the efficiency of the Virtual MAC algorithm and consider significantly overlapping cells and highly bursty traffic mixes. A Virtual Source (VS) algorithm utilizes the Virtual MAC to estimate application level service quality. The Virtual Source allows application parameters to be tuned in response to dynamic channel conditions based on "virtual delay curves". We demonstrate through simulation that when these distributed virtual algorithms are applied to the admission control of the radio channel then a globally stable state can be maintained without the need for complex centralized radio resource management. Finally, we discuss a distributed service level management scheme that builds on the proposed algorithms to offer continuous service with handoff.

In this paper, we present a model for TCP/IP ow c o n trol mechanism. The rate at which data is transmitted increases linearly in time until a packet loss is detected. At t h a t p o i n t, the transmission rate is divided by a constant factor. Losses are generated by some exogenous random process which i s only assumed to be stationary. This allows us to account for any correlation and any distribution of inter-loss times. We obtain an explicit expression for the throughput of a TCP connection and bounds on the throughput when there is a limit on the congestion window size. In addition, we study the e ect of the TimeOut mechanism on the throughput. A set of experiments is conducted over the real Internet and a comparison is provided with other models which make simple assumptions on the inter-loss time process.

Packet switching of variable-bit-rate real-time video sources is a means for efficient sharing of communication resources while maintaining uniform picture quality. Performance analysis for the statistical multiplexing of such video sources is required as a first step towards assessing the feasibility of packet video. This paper extends our earlier work in modeling video sources which have been coded using interframe coding schemes and in carrying out buffer queueing analysis for the multiplexing of several such sources. Our previous models and analysis were suitable for relatively uniform activity scenes. Here, we consider models for scenes with multiple activity levels, which lead to sudden changes in the coder output bit rates. We present correlated Markov models for the corresponding sources, and using a flowequivalent queueing analysis, obtain common buffer queue distributions and probabilities of packet loss. Our results demonstrate the efficiency of packet video on a single link, due to the smoothing effect of multiplexing several variable-bit-rate video sources.

There is a critical need for new thinking regarding overload traffic management in sensor networks. It has now become clear that experimental sensor networks (e.g., mote networks) and their applications commonly experience periods of persistent congestion and high packet loss, and in some cases even congestion collapse. This significantly impacts application fidelity measured at the physical sinks, even under light to moderate traffic loads, and is a direct product of the funneling effect; that is, the many-to-one multi-hop traffic pattern that characterizes sensor network communications. Existing congestion control schemes are * This work is supported by the Army Research Office (ARO) under Award W911NF-04-1-0311 on resilient sensor networks. Andrew T. Campbell also thanks the UK Engineering and Physical Sciences Research Council (EPSRC) for its support during his sabbatical year (2003-2004) at the Computer Lab., Cambridge University, as an EPSRC Visiting Fellow.

The loss behavior of wireless networks has become the focus of many recent research efforts. Although it is generally agreed that wireless communications experience higher error rates than wireline, the nature of these lossy links is not fully understood. This paper describes an effort to characterize the loss behavior of the AT&T WaveLAN, a popular in-building wireless interface. Using a trace-based approach, packet loss information is recorded, analyzed, and validated. Our results indicate that WaveLAN experiences an average packet error rate of 2 to 3 percent. Further analysis reveals that these errors are not independent, making it hard to model them with a simple two-state Markov chain. We derive another model based on the distributions of the error and error-free length of the packet streams. For validation, we modulate both the error models and the traces in a simulator. Simulations based on traces yield an average TCP throughput of about 5 percent less than simulations based our best error model.

We discuss here the implementation aspects of recently developed tools of computational intelligence for tackling the issue of joint trajectory generation of a class of multi-joint cooperative robotic systems. This is closely related to the inverse kinematics problem which usually represents a heavy computational burden on the processing power of any complex robotic structure. High nonlinearities, heavy coupling between the degrees of freedom, and time variant configuration of the robot structure heavily contribute to these difficulties. Soft computing techniques have surged in recent years as effective computational tools for emulating the human capabilities when dealing with complex systems. Some of them are used here to synthesize approaches capable of substantially improving solving the inverse kinematics problem for a class of robotic systems and help generating the joint trajectories in a faster way

The ability to quickly deploy new protocols, and provide flexibility to both applications and network administrators has been an oft-cited reason for active networks to be seriously examined. This paper describes an architecture and implementation that provides dynamic installation of network services and flexibility to active code. An example of weighted fair queuing being installed dynamically into a running network is examined.

Raptor codes have been standardized as application layer forward error correction (FEC) codes for Multimedia Broadcast and Multicast Services (MBMS) and Digital Video Broadcast (DVB) due to their extraordinary advanced FEC protection and performance. Raptor codes are known to have characteristics very close to ideal and provide a wide range of operating flexibility and efficiency unmatched by other codes, and in particular they are an excellent implementation of fountain codes. Until now, investigations of the application of Raptor codes to UMTS and EPGRS have used an overall system model that does not accurately model the physical channel and user mobility, simplistically assuming independent random packet losses at the application layer. We investigate MBMS in UMTS in a much more realistic and complete simulation environment by considering advanced and complete channel models that simulate the physical channel and user mobility in a cellular network. We use this realisitic simulation environment to determine optimal system parameters under different mobility models, with different bearer parameters, and without and with selective combining. More specifically, we investigate joint settings of the Raptor code rate, the Turbo code rate, transmission power, etc., to find settings which provide reliable download delivery of files using minimal transmission energy. One of our main results is that optimal system-wide operating points use low transmission power and a modest amount of Turbo coding that results in relatively large radio packet loss rates that is compensated for by using a substantial amount of Raptor coding. These optimal operating points use far less transmission energy for download delivery of files than possible operating points without Raptor.

TCP performs poorly on paths that reorder packets significantly, where it misinterprets out-of-order delivery as packet loss. The sender responds with a fast retransmit though no actual loss has occurred. These repeated false fast retransmits keep the sender's window small, and severely degrade the throughput it attains. Requiring nearly in-order delivery needlessly restricts and complicates Internet routing systems and routers. Such beneficial systems as multi-path routing and parallel packet switches are difficult to deploy in a way that preserves ordering. Toward a more reordering-tolerant Internet architecture, we present enhancements to TCP that improve the protocol's robustness to reordered and delayed packets. We extend the sender to detect and recover from false fast retransmits using DSACK information, and to avoid false fast retransmits proactively, by adaptively varying dupthresh. Our algorithm is the first that adaptively balances increasing dupthresh, to avoid false fast retransmits, and limiting the growth of dupthresh, to avoid unnecessary timeouts. Finally, we demonstrate that TCP's RTO estimator tolerates delayed packets poorly, and present enhancements to it that ensure it is sufficiently conservative, without using timestamps or additional TCP header bits. Our simulations show that these enhancements significantly improve TCP's performance over paths that reorder or delay packets.

We describe a new carrier-sense multiple access (CSMA) protocol for multihop wireless networks, using multiple channels and a distributed channel selection scheme. The proposed protocol divides the available bandwidth into AE channels where the transmitting station selects an appropriate channel for packet transmission. The selection criterion is based on the interference power measurements on the channels. We show via simulations that this multichannel CSMA protocol provides a higher throughput compared to its single channel counterpart by reducing the packet loss due to collisions.

In wireless sensor networks, many protocols assume that if node A is able to interfere with node B's packet reception, node B is within node A's communication range. It is also assumed that if node B is within node A's communication range, node A is able to interfere with node B's packet reception from any transmitter. While these assumptions may be useful in protocol design, they are not valid, according to the real experiments we conducted in MICA2 platform. For a strong link that has a high packet delivery ratio, the interference range is observed smaller than the communication range, while for a weak link that has a low packet delivery ratio, the interference range is larger than the communication range. So using communication range information alone is not enough to design real collisionfree media access control protocols. This paper presents a radio interference detection protocol (RID) and its variation (RID-B) to detect run-time radio interference relations among nodes. The interference detection results are used to design real collision-free TDMA protocols. With extensive simulations in GlomoSim, and with sensor network application scenarios, we observe that the TDMA which uses the interference detection results has 100% packet delivery ratio, while the traditional TDMA has packet loss up to 60%, in heavy load. In addition to the scheduling-based TDMA protocols, we also explore the application of interference detection on contention-based MAC protocols.

The recently developed notion of TCP-compatibility has led to a number of proposals for alternative congestion control algorithms whose long-term throughput as a function of a steady-state loss rate is similar to that of TCP. Motivated by the needs of some streaming and multicast applications, these algorithms seem poised to take the current TCP-dominated Internet to an Internet where many congestion control algorithms co-exist. An important characteristic of these alternative algorithms is that they are slowly-responsive, refraining from reacting as drastically as TCP to a single packet loss.

We develop an anycast mechanism at the link layer for wireless ad hoc networks. The goal is to exploit path diversity in the link layer by choosing the best next hop to forward packets when multiple next hop choices are available. Such choices can come from a multipath routing protocol, for example. This technique can reduce transmission retries and packet drop probabilities in the face of channel fading. We develop an anycast extension of the IEEE 802.11 MAC layer based on this idea. We implement the protocol in an experimental proof-of-concept testbed using the Berkeley motes platform and S-MAC protocol stack. We also implement it in the popular ns-2 simulator and experiment with the AOMDV multipath routing protocol and Ricean fading channels. We show that anycast performs significantly better than 802.11 in terms of packet delivery, particularly when the path length or effect of fading is large. Further we experiment with anycast in networks that use multiple channels and those that use directional antennas for transmission. In these networks, deafness and hidden terminal problems are the main source of packet loss. We implemented anycast as extension of 802.11 like protocols that were proposed for these special networks. We are able to show that anycast is capable of enhancing the performance of these protocols by simply making use of the path diversity whenever it is available. ¢ Note that while physical layer techniques can mask effect of fading and interference, this work does not target physical layer techniques. Here, the interest is working on beyond physical layer capabilities, by exploring alternative paths.

We present a comprehensive set of measurements of a 2.4 GHz DSSS wireless LAN and analyze its behavior. We examine issues such as host and interface heterogeneity, bidirectional (TCP) traffic and error modeling, that have not been previously analyzed. We uncover multiple problems with TCP and UDP performance in this system. We investigate the causes of these problems (radio hardware, device drivers, network protocols) and discuss the effectiveness of proposed improvements.

Variable bit rate (VBR) compressed video is expected to become one of the major loading factors in high-speed packet networks such a s A TM-based B-ISDN. However, recent measurements based on long empirical traces (complete movies) revealed that VBR video trac possesses self-similar (or fractal) c haracteristics, meaning that the dependence in the trac stream lasts much longer than traditional models can capture. In this paper, we present a unied approach which, in addition to accurately modeling the marginal distribution of empirical video records, also models directly both the short and the long-term empirical autocorrelation structures. We also present simulation results using synthetic data and compare with results based on empirical video traces. Furthermore, we extend the application of ecient estimation techniques based on importance sampling that we had used before only for simple fractal processes. We use importance sampling techniques to eciently estimate low probabilities of packet losses that occur when a multiplexer is fed with synthetic trac from our self-similar VBR video model.

Recently, optical packet switch architectures, composed of devices such as optical switches, fiber delay lines, and passive couplers, have been proposed to overcome the electromagnetic interference (EMI), pinout and interconnection problems that would be encountered in future large electronic switch cores. However, attaining the buffer size (buffer depth) in optical packet switches required in practice is a major problem; in this paper, a new solution is presented. An architectural concept is discussed and justified mathematically that relies on cascading many small switches to form a bigger switch with a larger buffer depth. The number of cascaded switches is proportional to the logarithm of the buffer depth, providing an economical and feasible hardware solution. Packet loss performance, control and buffer dimensioning are considered. The optical performance is also modeled, demonstrating the feasibility of buffer depths of several thousand, as required for bursty traffic.