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INTRODUCTION. Extraction and separation of foreign proteins expressed in E. coli commonly employs multistep fractionation procedures combining centrifugation, mechanical disruption, volume reduction, filtration, and finally capture and recovery in order to separate cells from culture media and purify the target protein from the cell extracts[1,2]. These methods are not amenable to rapid high throughput (HT) purification of thousands of proteins. We demonstrate here a HT-compatible reagent, PopCulture™, and associated method that eliminate the necessity for multiple mechanical fractionation procedures. The entire culturing, extraction, and purification can be achieved in the original culture tube or multiwell plate. This "in-media" protein purification procedure may be adapted to high throughput robotic processing of samples for proteomics research and any application that would benefit from the increased speed and convenience it provides. METHOD. Protein is isolated by culturing cells in liquid media under conditions for optimal target protein expression, chemically lysing the cells directly in their liquid media with a concentrated mixture of specialized detergents and buffer (*PopCulture™ Reagent), reduction of the extract’s viscosity by the addition of *Benzonase® nuclease and *lysozyme, capturing the target protein by adsorption to an *affinity matrix, washing the capture matrix-target protein complex to remove spent culture media and unwanted cellular contaminants, and elution of the purified protein from the adsorptive matrix (*Novagen Inc., Madison, WI). RESULTS. We have employed the PopCulture Reagent and protocol to purify a GST Tag™/His Tag® fusion protein using immobilized glutathione (GST-Bind Resins) or immobilized metal chelation (IMAC; His-Bind® Resins). The results are summarized in Table 1 and Figs. 1 and 2.

We've known for a while that the Internet has ossified as a result of the race to optimize existing applications or enhance security. NATs, performance-enhancing-proxies,firewalls and traffic normalizers are only a few of the middleboxes that are deployed in the network and look beyond the IP header to do their job. IP itself can't be extended because "IP options are not an option". Is the same true for TCP?In this paper we develop a measurement methodology for evaluating middlebox behavior relating to TCP extensions and present the results of measurements conducted from multiple vantage points. The short answer is that we can still extend TCP, but extensions' design is very constrained as it needs to take into account prevalent middlebox behaviors. For instance, absolute sequence numbers cannot be embedded in options, as middleboxes can rewrite ISN and preserve undefined options. Sequence numbering also must be consistent for a TCP connection, because many middleboxes only allow through contiguous flows.We used these findings to analyze three proposed extensions to TCP. We find that MPTCP is likely to work correctly in the Internet or fallback to regular TCP. TcpCrypt seems ready to be deployed, however it is fragile if resegmentation does happen---for instance with hardware offload. Finally, TCP extended options in its current form is not safe to deploy.

The latest large-scale data centers offer higher aggregate bandwidth and robustness by creating multiple paths in the core of the net- work. To utilize this bandwidth requires different flows take different paths, which poses a challenge. In short, a single-path transport seems ill-suited to such networks.We propose using Multipath TCP as a replacement for TCP in such data centers, as it can effectively and seamlessly use available bandwidth, giving improved throughput and better fairness on many topologies. We investigate what causes these benefits, teasing apart the contribution of each of the mechanisms used by MPTCP.Using MPTCP lets us rethink data center networks, with a different mindset as to the relationship between transport protocols, rout- ing and topology. MPTCP enables topologies that single path TCP cannot utilize. As a proof-of-concept, we present a dual-homed variant of the FatTree topology. With MPTCP, this outperforms FatTree for a wide range of workloads, but costs the same.In existing data centers, MPTCP is readily deployable leveraging widely deployed technologies such as ECMP. We have run MPTCP on Amazon EC2 and found that it outperforms TCP by a factor of three when there is path diversity. But the biggest benefits will come when data centers are designed for multipath transports.

Proceedings of: ACM MobiArch 2011, The 6th ACM International Workshop on Mobility in the Evolving Internet Architecture, June 28, 2011, Washington, D.C. Host mobility has traditionally been solved at the network layer, but even though Mobile IP has been standardised for 15 years, it hasn’t been supported by operators. IP’s double role as a location identif er and communication endpoint identif er brings a number of functional and performance problems. We argue that the best place to handle mobility is at the transport layer. While this is not a new argument, we believe that the emerging standard of Multipath TCP (MPTCP) can be used to solve many issues related to mobility. MPTCP naturally implements make-before-break, can be incrementally deployed, is backwards compatible with standard TCP, and could even ease incremental adoption of IPv6. Using simulations and indoor experiments with WiFi and 3G, we show that MPTCP gives better throughput, achieves smoother handoffs, and can be tuned to lower energy consumption. This research was supported by Trilogy (http://www.trilogy-project.org), a research project (ICT-216372) partially funded by the European Community under its Seventh Framework Programme. European Community's Seventh Framework Program This work was partly funded by POSDRU/89/1.5/S/62557 Publicado

In 1992, the CEC agreed to finance a one-year project called “Piloting of Multimedia Integrated Services for European Researcher (MICE)”. This paper outlines the background, purpose and aims of the project. The technical work required to interconnect existing facilities, and issues relating to influencing factors—such as standards and infrastructure—are discussed in detail. In conclusion, we provide an outlook on the future of multimedia conferencing and highlights some of the next steps to be taken.

Recently new data center topologies have been proposed that offer higher aggregate bandwidth and location independence by creating multiple paths in the core of the network. To effectively use this bandwidth requires ensuring different flows take different paths, which poses a challenge.Plainly put, there is a mismatch between single-path transport and the multitude of available network paths. We propose a natural evolution of data center transport from TCP to multipath TCP. We show that multipath TCP can effectively and seamlessly use available bandwidth, providing improved throughput and better fairness in these new topologies when compared to single path TCP and randomized flow-level load balancing. We also show that multipath TCP outperforms laggy centralized flow scheduling without needing centralized control or additional infrastructure.

This document describes Session Initiation Protocol (SIP), an application-layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants. These sessions include Internet telephone calls, multimedia distribution, and multimedia conferences.

To search the web quickly, search engines partition the web index over many machines, and consult every partition when answering a query. To increase throughput, replicas are added for each of these machines. The key parameter of these algorithms is the trade-off between replication and partitioning: increasing the partitioning level improves query completion time since more servers handle the query, but may incur non-negligible startup costs for each sub-query. Finding the right operating point and adapting to it can significantly improve performance and reduce costs. We introduce Rendezvous On a Ring (ROAR), a novel distributed algorithm that enables on-the-fly re-configuration of the partitioning level. ROAR can add and remove servers without stopping the system, cope with server failures, and provide good load-balancing even with a heterogeneous server pool. We demonstrate these claims using a privacy-preserving search application built upon ROAR.

There are moves in the Internet architecture community to add multipath capabilities to TCP, so that end-systems will be able to shift their traffic away from congested parts of the network. We study two problems relating to the design of multipath TCP. (i) We investigate stochastic packet-level behaviour of some proposed multipath congestion control algorithms, and find that they do not behave how we might expect from fluid modeling: they tend to flap randomly between their available paths. We explain why, and propose a congestion control algorithm that does not flap. (ii) We consider how the path choice offered by the network affects the ability of end-systems to shift their traffic between a pool of resources. We define a `resource poolability' metric, which measures for each resource how easy it is for traffic to be shifted away from that resource e.g. in the event of a traffic surge or link failure.

In this paper we investigate the building of a virtual router platform that ensures isolation and fairness between concurrent virtual routers. Recent developments in commodity x86 hardware enable us to take advantage of the flexibility and wealth of resources available to a software router in order to build a virtual router platform. Using commodity x86 hardware we show that it is viable to run highly experimental and untrusted router systems along side a production router on the same hardware platform without sacrificing performance. We investigate the extent to which we can isolate a virtual router running experimental code from other virtual routers.

Since the ARPAnet, network designers have built localized mechanisms for statistical multiplexing, load balancing, and failure resilience, often without understanding the broader implications. These mechanisms are all types of resource pooling, whichmeans making a collection of resources behave like a single pooled resource. We believe that the natural evolution of the Internet is that it should achieve resource pooling by harnessing the responsiveness of multipath-capable end systems. We argue that this approach will solve the problems and limitations of the current piecemeal approaches. This research was supported by Trilogy (http://www.trilogy-project.org), a research project (ICT-216372) partially funded by the European Community under its Seventh Framework Programme. European Community's Seventh Framework Program Publicado

In this paper, we evaluate the performance of a software IP router forwarding plane inside the Xen virtual machine monitor environment with a view to identifying (some) design issues in Virtual Routers. To this end, we evaluate and compare the forwarding performance of two identical Linux software router configurations, run either above the Xen hypervisor or within vanilla Linux. Even with minimal sized packets, we show that the Xen DomO privileged domain offers near native forwarding performance at the condition that the sollicitation to unpriviledged domains stay minimal, whereas Xen unprivileged domains offer very poor performance in every cases. This shows that an important design principle for virtual router platforms must be to handle all forwarding, for all virtual routers, onto the same forwarding engine, in order to avoid much detrimental per-packet context switching.

We describe our experiences in contributing to the implementation of a new protocol, DCCP, in the Linux kernel. Being the first implementation in a main-stream operating system, we are the first ones to explore the implications and the unexpected issues that could arise from developing this protocol. We will focus on how the DCCP implementation differs from that of TCP and the performance issues that we have encountered.

Recent advances in the defense of networked computers use instrumented binaries to track tainted data and can detect attempted break-ins automatically. These techniques identify how the transfer of execution to the attacker takes place, allowing the automatic generation of defenses. However, as with many technologies, these same techniques can also be used by the attackers: the information provided by detectors is accurate enough to allow an attacker to create a new worm using the same vulnerability, hijacking the exploit. Hijacking changes the threat landscape by pushing attacks to extremes (targeting selectively or creating a rapidly spreading worm), and increasing the requirements for automatic worm containment mechanisms. In this paper, we show that hijacking is feasible for two categories of attackers: those running detectors and those using Self-Certifying Alerts, a novel mechanism proposed by Costa et al. for end-to-end worm containment. We provide a discussion of the effects of hijacking on the threat landscape and list a series of possible countermeasures.

Fast-growing Internet applications like streaming media and telephony prefer timeliness to reliability, making TCP a poor fit. Unfortunately, UDP, the natural alternative, lacks congestion control. High-bandwidth UDP applications must implement congestion control themselves-a difficult task-or risk rendering congested networks unusable. We set out to ease the safe deployment of these applications by designing a congestion-controlled unreliable transport protocol . The outcome, the Datagram Congestion Control Protocol or DCCP, adds to a UDP-like foundation the minimum mechanisms necessary to support congestion control. We thought those mechanisms would resemble TCP's, but without reliability and, especially, cumulative acknowledgements, we had to reconsider almost every aspect of TCP's design. The resulting protocol sheds light on how congestion control interacts with unreliable transport, how modern network constraints impact protocol design, and how TCP's reliable bytestream semantics intertwine with its other mechanisms, including congestion control.

The 802.11 encryption standard Wired Equivalent Privacy (WEP) is still widely used today despite the numerous discussions on its insecurity. In this paper, we present a novel vulnerability which allows an attacker to send arbitrary data on a WEP network after having eavesdropped a single data packet. Furthermore, we present techniques for real-time decryption of data packets, which may be used under common circumstances. Vendor produced mitigation techniques which cause frequent WEP re-keying prevent traditional attacks, whereas our attack remains effective even in such scenarios. We implemented a fully automatic version of this attack which demonstrates its practicality and feasibility in real networks. As even rapidly re-keyed networks can be quickly compromised, we believe WEP must now be abandoned rather than patched yet again.

Publish-subscribe is a powerful paradigm for distributed communication based on decoupled producers and consumers of information. Its event-driven nature makes it very appealing for large-scale data dissemination infrastructures. Various architectures were proposed in recent years that provide very diverse features. However, there are few well-defined metrics in the publish-subscribe area that would allow their evaluation and comparison. In this paper, we provide a broad overview of relevant quality-of-service metrics and describe their specific meaning in the context of distributed and decentralized publish-subscribe systems. Our goal is to provide a common base for future evaluations of emerging systems and for the design of qualityof- service aware publish-subscribe infrastructures.

Defending against DoS attacks is extremely difficult; effective solutions probably require significant changes to the Internet architecture. We present a series of architectural changes aimed at preventing most flooding DoS attacks, and making the remaining attacks easier to defend against. The goal is to stimulate a debate on trade-offs between the flexibility needed for future Internet evolution and the need to be robust to attack.

A number of large-scale distributed Internet applications could potentially benefit from some level of knowledge about the relative proximity between its participating host nodes. For example, the performance of large overlay networks could be improved if the application-level connectivity between the nodes in these networks is congruent with the underlying IP-level topology. Similarly, in the case of replicated Web content, client nodes could use topological information in selecting one of multiple available servers. For such applications, one need not find the optimal solution in order to achieve significant practical benefits. Thus, these applications, and presumably others like them, do not require exact topological information and can instead use sufficiently informative hints about the relative positions of Internet hosts. In this paper, we present a binning scheme whereby nodes partition themselves into bins such that nodes that fall within a given bin are relatively close to one another in terms of network latency. Our binning strategy is simple (requiring minimal support from any measurement infrastructure), scalable (requiring no form of global knowledge, each node only needs knowledge of a small number of well-known landmark nodes) and completely distributed (requiring no communication or cooperation between the nodes being binned). We apply this binning strategy to the two applications mentioned above: overlay network construction and server selection. We test our binning strategy and its application using simulation and Internet measurement traces. Our results indicate that the performance of these applications can be significantly improved by even the rather coarse-grained knowledge of topology offered by our binning scheme.

This document specifies TCP-Friendly Rate Control (TFRC). TFRC is a congestion control mechanism for unicast flows operating in a best-effort Internet environment. It is reasonably fair when competing for bandwidth with TCP flows, but has a much lower variation of throughput over time compared with TCP, which makes it more suitable for applications such as telephony or streaming media where a relatively smooth sending rate is of importance.

Lack of QoS support in the Internet has not prevented the rapid growth of streaming applications. However many of these applications do not perform congestion control effectively. Thus there is significant concern about the effects on co-existing well-behaved traffic and the potential for congestion collapse. In addition, the majority of such applications are unable to perform quality adaptation on-the-fly as the available bandwidth changes during a session. This paper provides some architectural insights on the design of video playback applications in the Internet. We present fundamental design principles for Internet applications and identify end-to-end congestion control, quality adaptation and error control as the three major building blocks for Internet video playback applications. We discuss the design space for each of these components, and within that space, present an end-to-end architecture suited for playback of layered-encoded stored video streams. Our architecture reconciles congestion control and quality adaptation which occur on different timescales. It exhibits a TCP-friendly behavior by adopting the RAP protocol for end-to-end congestion control. Additionally it uses a layered framework for quality adaptation with selective retransmission to maximize the quality of the delivered stream as the available bandwidth changes. We argue that the architecture can be generalized by replacing the suggested mechanism for each component by another from the same design space as long as all components remain compatible.

It has been claimed that multicast state cannot be aggregated. In this paper, we will debunk this myth and present a simple technique that can be used to aggregate multicast forwarding state. In particular, we present an interface-centric data structure model which allows aggregation of ranges of multicast addresses in the forwarding table. Understanding the limits of possible aggregation is critical to our knowledge of how IP multicast will scale as it becomes widely deployed. We show through analysis and simulation that some aggregation is possible, even under purely random address allocation and purely random group membership distribution. We further show how other methods of allocation can significantly improve the ability to aggregate, and how non-random distributions of membership can affect aggregation both positively and negatively.

Most currently proposed solutions to application-level multicast organise the group members into an application-level mesh over which a Distance-Vector routing protocol, or a similar algorithm, is used to construct source-rooted distribution trees. The use of a global routing protocol limits the scalability of these systems. Other proposed solutions that scale to larger numbers of receivers do so by restricting the multicast service model to be single-sourced. In this paper, we propose an application-level multicast scheme capable of scaling to large group sizes without restricting the service model to a single source. Our scheme builds on recent work on Content-Addressable Networks (CANs). Extending the CAN framework to support multicast comes at trivial additional cost and, because of the structured nature of CAN topologies, obviates the need for a multicast routingalg orithm. Given the deployment of a distributed infrastructure such as a CAN, we believe our CAN-based multicast scheme offers the dual advantages of simplicity and scalability.

Multi-core CPUs, along with recent advances in memory and buses, render commodity hardware a strong candidate for software router virtualization. In this context, we present the design of a new platform for virtual routers on modern PC hardware. We further discuss our design choices in order to achieve both high performance and flexibility for packet processing.

This paper proposes a mechanism for equation-based congestion control for unicast traffic. Most best-effort traffic in the current Internet is well-served by the dominant transport protocol, TCP. However, traffic such as best-effort unicast streaming multimedia could find use for a TCP-friendly congestion control mechanism that refrains from reducing the sending rate in half in response to a single packet drop. With our mechanism, the sender explicitly adjusts its sending rate as a function of the measured rate of loss events, where a loss event consists of one or more packets dropped within a single round-trip time. We use both simulations and experiments over the Internet to explore performance.

Current multicast routing protocols can be classified into three types according to how the multicast tree is established: broadcast and prune (e.g., DVMRP, PIM-DM), membership advertisement (e.g., MO-SPF), and rendezvous-based (e.g., CBT, PIM-SM). Rendezvous-based protocols associate with each logical multicast group address, a physical unicast address, referred to as the 'core' or 'rendezvous point' (RP). Members first join a multicast tree rooted at this rendezvous point in order to receive data packets sent to the group. Rendezvous mechanisms are well suited to large wide-area networks because they distribute group-specific data and membership information only to those routers that are on the multicast distribution tree. However, rendezvous protocols require a bootstrap mechanism to map each logical multicast address to its current physical rendezvous point address. The bootstrap mechanism must adapt to network and router failures but should minimize unnecessary changes in the group-to-RP mapping. In addition, the bootstrap mechanism should be transparent to the hosts. This paper describes and analyzes the bootstrap mechanism developed for PIM-SM. The mechanism employs an algorithmic mapping of multicast group to rendezvous point address, based on a set of available RPs distributed throughout a multicast domain. The primary evaluation measures are convergence time, message distribution overhead, balanced assignment of groups to RPs, and host impact. The mechanism as a whole, and the design lessons in particular, are applicable to other rendezvous-based multicast routing protocols as well.

Multicast routing enables efficient data distribution to multiple recipients. However, existing work has concentrated on extending single-domain techniques to wide-area networks, rather than providing mechanisms to realize inter-domain multicast on a global scale in the Internet.We describe an architecture for inter-domain multicast routing that consists of two complementary protocols. The Multicast Address-Set Claim (MASC) protocol forms the basis for a hierarchical address allocation architecture. It dynamically allocates to domains multicast address ranges from which groups initiated in the domain get their multicast addresses. The Border-Gateway Multicast Protocol (BGMP), run by the border routers of a domain, constructs inter-domain bidirectional shared trees, while allowing any existing multicast routing protocol to be used within individual domains. The resulting shared tree for a group is rooted at the domain whose address range covers the group's address; this domain is typically the group initiator's domain. We demonstrate the feasibility and performance of these complementary protocols through simulation.This architecture, together with existing protocols operating within each domain, is intended as a framework in which to solve the problems facing the current multicast addressing and routing infrastructure.

IP Multicast, Lightweight Sessions and Application Level Framing provide guidelines by which multimedia conferencing tools can be designed, but they do not provide specific solutions. In this paper, we use these design principles to guide the design of a multicast based shared editor, and examine the consequences of taking a loose consistency approach to achieve good performance in the face of network failures and losses.

Once the concept of Computer Supported Cooperative Work (CSCW) tools was established, many software developers responded enthusiastically, and quite a number of groupware applications were built and released in the space of a few years. Few of these applications, however, were taken up with the same enthusiasm by their intended users. Grudin (1990) studied several of the failed projects and detected a high level of resistance among users against adopting these applications. He attributed this resistance to inadequacies in the design of groupware applications, resulting from the developers’ failure to appreciate that technology which aims to support collaborative work groups has to address issues and concerns that are fundamentally different from those encountered in the development of other application software. In Grudin’s view, approaches developed in Human-Computer Interaction (HCI) and Human Factors (HF) were ill equipped to tackle these problems, since they were based on experiences and observations of individual users interacting with stand-alone systems. Consequently, they focused on optimizing screen designs and interaction styles. Developing effective and successful groupware applications, however, required approaches that provided an in-depth understanding of the collaborative activities that these applications were trying to support.

This study examines the accuracy of parents' reports about their children's sleep behavior and their response to a behavioral treatment. Twenty-eight sleep-disordered and thirty control children aged 12-36 months were filmed during three nights using an infrared camera and their sleep behavior compared with parental reports. There were significant differences among the groups with parents of good sleepers being less accurate in reporting on their children's sleep behavior. Poor sleepers also had more behavior problems, a more difficult temperament and more adverse early medical histories. The good sleepers woke up as frequently as the poor sleepers. However, they managed to soothe themselves back to sleep without disturbing anyone. Virtually all poor sleepers showed significant improvement following treatment.

MESSIE is a collaborative authoring environment to support the production of large-scale document by teams of geographically distributed groups of authors working with heterogenous systems. The environment allows authors to submit text at various stages of gestation (e.g. list of topics, first draft) to a shared filestore via email. All authors collaborating on a document can read each other's contributions, and add suggestions, comments and additional material directly to the document. The system integrates automatically answered electronic mail, shared file store administration, and a version control tool in a UNIX environment. The paper describes design and implementation strategy, and reports observations and a number of changes which were made during a 4-month trial period with three collaborative authoring teams.

In this paper we provide an overview of multimedia conferencing on the Internet. The protocols mentioned are all specified elsewhere as Internet-Drafts or RFCs. Each RFC gives details of the protocol itself, how it works and what it does. This document attempts to provide the reader with an overview of how the components fit together and of some of the assumptions made, as well as some statement of direction for those components still in a nascent stage.