In 2021-2022, the Computer Science and Math departments at Colorado State University replaced the traditional two-course Calculus sequence (Calculus I and II) with a single-semester course tailored to computer science needs. Unintentionally, this change also led the department to accept classic Calculus I taken by students before they transferred into Computer Science as meeting the major's Calculus requirement. We now have four years of data regarding the success of Computer Science students who only took (some version) of Calculus I, vs. those who took both Calculus I and II.

Scientific data volume is growing in size, and as a direct result, the need for faster transfers is also increasing. The scientific community has sought to leverage parallel transfer methods using multi-threaded and multi-source download models to reduce download times. In multi-source transfers, a client downloads data from multiple replicated servers in parallel. Tools such as Aria2 and BitTorrent support such multi-source transfers and have shown improved transfer times.In this work, we introduce Multi-Source Data Transfer Protocol, MDTP, which further improves multi-source transfer performance. MDTP logically divides a file request into smaller chunk requests and distributes the chunk requests across multiple servers. Chunk sizes are adapted based on each server’s performance but selected in a way that ensures each round of requests completes around the same time. We formulate this chunk-size allocation problems as a variant of the bin-packing problem, where adaptive chunking efficiently fills the available capacity "bins" corresponding to each server.Our evaluation shows that MDTP reduces transfer times by 10–22% compared to Aria2, the fastest alternative. Comparisons with other protocols, such as static chunking and BitTorrent, demonstrate even greater improvements. Additionally, we show that MDTP distributes load proportionally across all available replicas, not just the fastest ones, which improves throughput. Finally, we show MDTP maintains high throughput even when latency increases or bandwidth to the fastest server decreases.

Inspired by prior work suggesting undetected errors were becoming a problem on the Internet, we set out to create a measurement system to detect errors that the TCP checksum missed. We designed a client-server framework in which the servers sent known files to clients. We then compared the received data with the original file to identify undetected errors introduced by the network. We deployed this measurement framework on various public testbeds. Over the course of 9 months, we transferred a total of 26 petabytes of data. Scaling the measurement framework to capture a large number of errors proved to be a challenge. This paper focuses on the challenges encountered during the deployment of the measurement system. We also present the interim results, which suggest that the error problems seen in prior works may be caused by two distinct processes: (1) errors that slip past TCP and (2) file system failures. The interim results also suggest that the measurement system needs to be adjusted to collect exabytes of measurement data, rather than the petabytes that prior studies predicted.

IEEE CRC-32 is widely used in data communications to detect transmission errors. The decision to use CRC-32 was made in the 1970s, based on studies that compared the performance of various 32-bit CRCs on relatively small (2 000 bit) packets subjected to errors described by a renewal process. In today’s networks we often see far larger packets (8 000 bits or more) and experiments have shown that errors do not fit a renewal process. The world has changed. To get a sense of that change, we captured 1 900 gigabit WIFI (802.11n) transmissions errors to better understand error distributions and patterns. Our data suggest that a considerable fraction of errors could slip past IEEE CRC-32. It may soon be time to consider a new generation of error codes.

Although IP and its overlying protocols, such as TCP and UDP, are ubiquitous, they were originally designed for point-to-point connections between computers in reasonably fixed locations. They are less suited to mobile networks and broadcast communications. In this paper, we present an alternative to IP that is based on a publish-subscribe approach. The approach that we present combines an application publish-subscribe programming model with a content delivery network, which provides several advantages in certain communication environments, including quality of service based on application level needs; efficient support for reliable broadcast; support for disadvantaged, intermittent, and limited communications; and more efficient reliability and fault tolerance. The paper presents our approach, based on a streamlined Data Distribution Service and simplified Content Delivery Network, a motivating example in which the publish-subscribe based distribution and network provides advantages, and a contrast to TCP/IP in the example context.

We present a transport protocol whose goal is to reduce power consumption without compromising delivery requirements of applications. To meet its goal of energy efficiency, our transport protocol (1) contains mechanisms to balance end-to-end vs. local retransmissions; (2) minimizes acknowledgment traffic using receiver regulated rate-based flow control combined with selected acknowledgements and in-network caching of packets; and (3) aggressively seeks to avoid any congestion-based packet loss. Within a recently developed ultra low-power multi-hop wireless network system, extensive simulations and experimental results demonstrate that our transport protocol meets its goal of preserving the energy efficiency of the underlying network.

We propose a new objective for network research: to build a fundamentally different sort of network that can assemble itself given high level instructions, reassemble itself as requirements change, automatically discover when something goes wrong, and automatically fix a detected problem or explain why it cannot do so.We further argue that to achieve this goal, it is not sufficient to improve incrementally on the techniques and algorithms we know today. Instead, we propose a new construct, the Knowledge Plane, a pervasive system within the network that builds and maintains high-level models of what the network is supposed to do, in order to provide services and advice to other elements of the network. The knowledge plane is novel in its reliance on the tools of AI and cognitive systems. We argue that cognitive techniques, rather than traditional algorithmic approaches, are best suited to meeting the uncertainties and complexity of our objective.

The design of the IP protocol makes it difficult to reliably identify the originator of an IP packet. Even in the absence of any deliberate attempt to disguise a packet's origin, widespread packet forwarding techniques such as NAT and encapsulation may obscure the packet's true source. Techniques have been developed to determine the source of large packet flows, but, to date, no system has been presented to track individual packets in an efficient, scalable fashion. We present a hash-based technique for IP traceback that generates audit trails for traffic within the network, and can trace the origin of a single IP packet delivered by the network in the recent past. We demonstrate that the system is effective, space efficient (requiring approximately 0.5% of the link capacity per unit time in storage), and implementable in current or next-generation routing hardware. We present both analytic and simulation results showing the system's effectiveness.

Current routing protocols are monolithic, specifying the algorithm used to construct forwarding tables, the metric used by the algorithm (generally some form of hop count), and the protocol used to distribute these metrics as an integrated package. The flexible intra-AS routing environment (FIRE) is a link-state, intradomain routing protocol that decouples these components. FIRE supports run-time-programmable algorithms and metrics over a secure link-state distribution protocol. By allowing the network operator to dynamically reprogram both the properties being advertised and the routing algorithms used to construct forwarding tables, FIRE enables the development and deployment of novel routing algorithms without the need for a new protocol to distribute state. FIRE supports multiple concurrent routing algorithms and metrics, each constructing separate forwarding tables. By using operator-specified packet filters, separate classes of traffic may be routed using completely different routing algorithms, all supported by a single routing protocol. This paper presents an overview of FIRE, focusing particularly on FIRE's novel aspects with respect to traditional routing protocols. We consider deploying several current unicast and multicast routing algorithms in FIRE, and describe our Java-based implementation.

From my perspective, one of the more distressing aspects of the past fifteen years is the evolution of data communications and distributed computing into distinct and somewhat disconnected disciplines. It is getting to the point where researchers in each discipline often don't really understand what is going on in the other discipline, even though the two disciplines are logically tightly linked to one another.

Current routing protocols are monolithic, specifying the algorithm used to construct forwarding tables, the metric used by the algorithm (generally some form of hop-count), and the protocol used to distribute these metrics as an integrated package. The Flexible Intra-AS Routing Environment (FIRE) is a link-state, intra-domain routing protocol that decouples these components. FIRE supports run-time-pro- grammable algorithms and metrics over a secure link-state distribution protocol. By allowing the network operator to dynamically reprogram both the information being advertised and the routing algorithm used to construct forwarding tables in Java, FIRE enables the development and deployment of novel routing algorithms without the need for a new protocol to distribute state. FIRE supports multiple concurrent routing algorithms and metrics, each constructing separate forwarding tables. By using operator-specified packet filters, separate classes of traffic are routed using completely different routing algorithms, all supported by a single routing protocol.

Traces of Internet packets from the past two years show that between 1 packet in 1,100 and 1 packet in 32,000 fails the TCP checksum, even on links where link-level CRCs should catch all but 1 in 4 billion errors. For certain situations, the rate of checksum failures can be even higher: in one hour-long test we observed a checksum failure of 1 packet in 400. We investigate why so many errors are observed, when link-level CRCs should catch nearly all of them.

This article introduces Smart Packets and describes the smart Packets architecture, the packet formats, the language and its design goals, and security considerations. Smart Packets is an Active Networks project focusing on applying active networks technology to network management and monitoring. Messages in active networks are programs that are executed at nodes on the path to one or more target hosts. Smart Packets programs are written in a tightly encoded, safe language specifically designed to support network management and avoid dangerous constructs and accesses. Smart Packets improves the management of large complex networks by (1) moving management decision points closer to the node being managed, (2) targeting specific aspects of the node for information rather than exhaustive collection via polling, and (3) abstracting the management concepts to language constructs, allowing nimble network control.

It is a widely held belief that packet reordering in the Internet is a pathological behavior, or more precisely, that it is an uncommon behavior caused by incorrect or malfunctioning network components. Some studies of Internet traffic have reported seeing occasional packet reordering events and ascribed these events to "route fluttering", router "pauses" or simply to broken equipment. We have found, however, that parallelism in Internet components and links is causing packet reordering under normal operation and that the incidence of packet reordering appears to be substantially higher than previously reported. More importantly, we observe that in the presence of massive packet reordering transmission control protocol (TCP) performance can be profoundly effected. Perhaps the most disturbing observation about TCP's behavior is that large scale and largely random reordering on the part of the network can lead to self-reinforcingly poor performance from TCP.

Aggressive research on gigabit-per-second networks has led to dramatic improvements in network transmission speeds. One result of these improvements has been to put pressure on router technology to keep pace. This paper describes a router, nearly completed, which is more than fast enough to keep up with the latest transmission technologies. The router has a backplane speed of 50 Gb/s and can forward tens of millions of packets per second.

Checksum and cyclic redundancy check (CRC) algorithms have historically been studied under the assumption that the data fed to the algorithms was uniformly distributed. This paper examines the behavior of checksums and CRCs over real data from various UNIX file systems. We show that, when given real data in small to modest pieces (e.g., 48 bytes), all the checksum algorithms have skewed distributions. These results have implications for CRCs and checksums when applied to real data. They also can cause a spectacular failure rate for both the TCP and ones-complement Fletcher (1983) checksums when trying to detect certain types of packet splices. When measured over several large file systems, the 16 bit TCP checksum performed about as well as a 10-bit CRC. We show that for fragmentation-and-reassembly error models, the checksum contribution of each fragment are, in effect, colored by the fragment's offset in the splice. This coloring explains the performance of Fletcher's sum on nonuniform data, and shows that placing checksum fields in a packet trailer is theoretically no worse than a header checksum field. In practice, the TCP trailer sums outperform even Fletcher header sums.

Checksum and CRC algorithms have historically been studied under the assumption that the data fed to the algorithms was entirely random. This paper examines the behavior of checksums and CRCs over real data from various UNIX® file systems. We show that, when given real data in small to modest pieces (e.g., 48 bytes), all the checksum algorithms have skewed distributions. In one dramatic case, 0.01% of the check values appeared nearly 19% of the time. These results have implications for CRCs and checksums when applied to real data. They also cause a spectacular failure rate for the both TCP and Fletcher's checksums when trying to detect certain types of packet splices.

Presents an adaptive flow synchronization protocol that permits synchronized delivery of data to and from geographically distributed sites. Applications include inter-stream synchronization, synchronized delivery of information in a multisite conference, and synchronization for concurrency control in distributed computations. The contributions of this protocol in the area of flow synchronization are the ability to synchronize over arbitrary topologies, the introduction of an adaptive synchronization delay, the flexibility to maintain multiple synchronization groups, and the use of a modular architecture that permits the application to tailor synchronization calculations to its service requirements. The authors take advantage of network protocols capable of maintaining network clock synchronization in the millisecond range.>

As an experiment in protocol optimizations, the authors undertook to improve the performance of a stateless protocol, namely the user datagram protocol (UDP) in the 4.3 BSD Unix kernel. The authors describe the successful optimizations that were done, along with measurements that shows a UDP performance improvement of between 25-35% on CISC and RISC systems, and overall kernal improvement of between 12% and 18%.>

article Free AccessImproving round-trip time estimates in reliable transport protocols Authors: Phil Karn Bell Communications Research, Morristown, NJ Bell Communications Research, Morristown, NJView Profile , Craig Partridge Bolt Beranek and Newman, Inc., Cambridge, MA Bolt Beranek and Newman, Inc., Cambridge, MAView Profile Authors Info & Claims ACM Transactions on Computer SystemsVolume 9Issue 4Nov. 1991 pp 364–373https://doi.org/10.1145/118544.118549Published:01 November 1991Publication History 94citation1,450DownloadsMetricsTotal Citations94Total Downloads1,450Last 12 Months87Last 6 weeks4 Get Citation AlertsNew Citation Alert added!This alert has been successfully added and will be sent to:You will be notified whenever a record that you have chosen has been cited.To manage your alert preferences, click on the button below.Manage my AlertsNew Citation Alert!Please log in to your account Save to BinderSave to BinderCreate a New BinderNameCancelCreateExport CitationPublisher SiteeReaderPDF

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Key areas of agreement and disagreement in the network management community are presented using the evolving open systems interconnection network management architecture as an informal guide. A key observation is that the open-networking community has apparently settled on a management model that places management information base (MIB) on each network node and manages these MIBs remotely with application level protocols. Various network systems based on this model are examined, along with some systems that use alternative management architectures.>