We present the novel concept of a physically assured unclonable networking protocol based on physical unclonable functions (PUF) in which peer protocol machines of a networking protocol, together with the protocol data units (PDU) that they exchange, are continuously physically authenticated jointly over time. A method is presented to realize these physical authentications for networking protocols in general. The method is based on extending the hitherto developed PUF-based ‘CRP Ratchet’ protocol - for creating unclonable links - to additionally entangle protocol PDUs with PUF input challenges so as to realize a physically unclonable protocol (PUP). Architectures are presented to employ PUP either as a protocol layer shim in the data path of PDUs, or as an adjunct process running concurrently in a protocol layer.

A continuous-time Markov chain (CTMC) model is developed for the probabilistic transient analysis of topology-based worm propagation in networks with arbitrary topologies, background traffic, link errors, and error recovery protocols. The CTMC model is analyzed using the numerical Markov chain transient analysis uniformization technique. Our specific use-case deals with radio frequency (RF) links and future worm propagation in that area. Time distribution and related measures are provided for a worm infecting a given node. An accompanying state space compression technique is developed, significantly reducing the space and time requirements of the worm propagation analysis as a function of the topology and initial infection. Examples are presented to demonstrate computational space and time requirements and expose encountered practical challenges. The developed transient CTMC worm propagation model provides a general, flexible framework for the detailed probabilistic analysis of complex computer worm propagations.

We describe a new vision of joint computation and communication resource management that goes beyond the end-to-end and client-server model of the current Internet. Enabled by a growing trend toward embedding processing and networking capabilities into "smart" network nodes and devices, for example, smartphones, watches, appliances, and automobiles, Dispersed Computing describes a new resource-centric architecture that leverages the diversity of networked computation points within the network, and the heterogeneity of network links and protocol stacks that connect them. We describe this new, resource-centric architecture as an evolution of both fog/edge networking and active networks. We illustrate the fundamental principles and the advantages over the current Internet architecture, and highlight several commercial use cases enabled by a dispersed computing architecture.

A dynamic overlay system is presented for supporting transport service needs of dispersed computing applications for moving data and/or code between network computation points and end-users in IoT or IoBT. The Network Backhaul Layered Architecture (Nebula) system combines network discovery and QoS monitoring, dynamic path optimization, online learning, and per-hop tunnel transport protocol optimization and synthesis over paths, to carry application traffic flows transparently over overlay tunnels. An overview is provided of Nebula's overlay system, software architecture, API, and implementation in the NRL CORE network emulator. Experimental emulation results demonstrate the performance benefits that Nebula provides under challenging networking conditions.

A cloud-based web service is presented for users to machine-generate network simulation programs for the opensource ns-3 network simulator. The service takes as input a userprovided network system model and delivers as output an ns-3 'main' program. An accompanying user-based modeling tool provides a graphical user interface for constructing and editing models of four layer Internet systems in terms of backbone nodes and links, different types of wireless networks (LTE, Wifi infrastructure, Wifi adhoc, Wimax, or CSMA), end-to-end application traffic sources, mobility models, routing protocols, and trace sources. Methods are also developed to make the cloudbased program generation service secure in terms of not exposing network model information in the cloud during the entirety of the program generation process. A masking technique is presented for hiding network information such as node or link names, link end-point names, IP and MAC addresses, model option choices, parameter values, and application-layer traffic end-point names. A network model decomposition technique is presented for obscuring network topology information such as the mix and graphical interconnection of nodes, links, wireless networks, wireless network elements, and application-layer traffic flows. The ns-3 program generation service facilitates the production and editing of ns-3 programs for large or complex network systems. It can also assist new users or non-experts with generating ns-3 programs.

A fast simulation technique based on importance sampling is developed for the analysis of path service availability in mesh networks with dynamic path restoration. The method combines the simulation of the path rerouting algorithm with a “dynamic path failure importance sampling” (DPFS) scheme to estimate path availabilities efficiently. In DPFS, the failure rates of network elements are biased at increased rates until path failures are observed under rerouting. The simulated model uses “failure equivalence groups,” with finite/infinite sources of failure events and finite/infinite pools of repair personnel, to facilitate the modeling of bidirectional link failures, multiple in-series link cuts, optical amplifier failures along links, node failures, and more general geographically distributed failure scenarios. The analysis of a large mesh network example demonstrates the practicality of the technique.

Practical reliability models are developed to support the engineering of carrier-class networks that provide VoIP service to enterprise customers. The focus is on end-to-end measures of service reliability. The paper first describes the typical architecture of large-scale carrier-class VoIP networks, the message flows that exist therein, and use case variations that arise in practice. Reliability measures of interest are then defined and models are developed in terms of the call flows and element reliabilities. The failover and repair mechanisms that may be employed in the various protocol layers are also surveyed. An example analysis illustrates the modeling of a particular use case. The developed models provide a convenient practical means to estimate end-to-end VoIP service reliability prior to field trials or network deployments. Such proactive modeling is an important step in the planning and engineering of new VoIP networks.

We present a novel architecture and methods to enable a user to make IP-to-PSTN voice calls without using a VoIP service provider or their gateways. The architecture is based on a 'personal' IP-to-PSTN gateway (PIPG) deployed at the residence or business of a user, where both PSTN and Internet service are assumed to exist. The PIPG enables the bridging of a voice call between an Internet endpoint, such as a softphone, and a user's PSTN line. The architecture includes an external server for registration purposes. We also extend the architecture to support the concept of a PSTN-line sharing (PLS) application in which users' PSTN lines can be shared by members of a group, analogous to popular peer-to-peer file sharing applications. In PLS, the external server supports a directory to available PIPG as well as intelligence to assign PIPG to incoming calls of members. The blocking performance of PLS is also analyzed

Methods are developed to estimate the source–destination traffic distribution matrix of a packet network using only aggregate link and source/sink traffic measurements. The methods are useful for network planning and monitoring of large packet networks, where it is impractical to gather measurement data on every end-to-end traffic flow. The first method assumes that the distribution matrix is time-invariant. This method is of theoretical interest but provides the foundation for developing a method for the realistic case of a time-varying matrix. The second method assumes that the matrix is time-varying. It uses linear programming (LP) to find a distribution matrix that optimally fits the measurement data. A practical problem with the first two methods is that the computational requirements increase as the square of the number of network nodes. The third method is a fast exact decomposition procedure for the time-invariant case that scales with the network size. The maximum number of unknowns that needs to be solved simultaneously is equal to the number of network nodes. The final method is a fast decomposition procedure for the time-varying case. This procedure scales with the network size. It uses LP to find an approximate distribution matrix that optimally fits the measurement data. The methods are applied to simulated example networks to illustrate the accuracy and speed.

A simulation-based methodology is developed for analyzing the subjective quality of voice-over-IP calls as a function of network QoS parameters and choices in implementation and configuration. The proposed method combines the use of existing objective voice-quality measurement algorithms, such as the ITU-T P.861 PSQM, and artificial voice reference signals, such as the ITU-T P.50, with the discrete-event simulation of a network QoS model. A significant advantage of the method is that it does not involve the use of human subjects in evaluating subjective voice quality levels. This enables one to entirely automate the process of quantifying call quality as a function of network QoS and implementation choices such as packet size and codec type. Such automation enables one to realize significant time and cost savings in obtaining experimental results. A tool implementation is described that includes basic network packet loss and delay jitter models. Example numerical results are presented for the G.711 codec. The extension of the method to the subjective quality evaluation of audio, video, and multimedia signals is also described. The paper also introduces the concepts of 'subjective teletraffic engineering' and 'subjective-equivalent bandwidths'.

A method is developed to estimate call grade of service (GoS) and offered traffic for Voice over IP (VoIP) calls at a PSTN-IP network gateway. Since offered calls from the PSTN circuit-switched side may never reach a gateway due to blocking at gateway PSTN interfaces, it is not possible to measure directly at the gateway the GoS or the offered traffic. The proposed method is based on polling the gateway dial control MIB (RFC 2128) to derive the carried traffic and solving a fixed-point problem involving the Erlang-B formula to obtain the GoS and offered traffic. A monitoring system utilizing this method has been developed for a large IP telecom network. The system includes a web interface for generating graphical monitoring reports.

Article Free Access Share on A performance monitoring system for VoIP gateways Author: Adrian E. Conway Infolibria, Inc., 411 Waverley Oaks Road, Waltham, MA Infolibria, Inc., 411 Waverley Oaks Road, Waltham, MAView Profile Authors Info & Claims WOSP '00: Proceedings of the 2nd international workshop on Software and performanceSeptember 2000 Pages 38–43https://doi.org/10.1145/350391.350401Online:01 September 2000Publication History 0citation1,594DownloadsMetricsTotal Citations0Total Downloads1,594Last 12 Months6Last 6 weeks1 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 Alerts New Citation Alert!Please log in to your account Save to BinderSave to BinderCreate a New BinderNameCancelCreateExport CitationPublisher SiteeReaderPDF

A major challenge in the development of integrated service packet networks is to devise mechanisms that can support services with different QoS requirements. Recently, a new mechanism was developed, called weighted fair blocking (WFB), that can be placed on top of almost any discrete-time multiplexing system for controlling packet loss. The WFB mechanism has a number of attractive features. It can offer sources a range of QoS loss levels or be used as a balanced fair blocking (BFB) mechanism to equalize blocking probabilities in the interest of fairness. The BFB mechanism also reduces multiplexer buffer requirements significantly. The mechanism was derived for multiple Bernoulli sources as well as for two bursty MMBP (Markov modulated Bernoulli processes) sources. In this paper, we extend the BFB mechanism to the more realistic case of multiple MMBPs. The extension is found to be theoretically tractable but significant practical complications arise. To circumvent these difficulties, a simple heuristic BFB (H-BFB) mechanism is proposed. The performance of H-BFB is found to be remarkably good in approaching that of theoretically exact BFB from the point of view of buffer gains and fairness.

The UNIX STREAMS jhcilities offer a set of kernel mechanisms for implementing layered network protocol &ware in a modular jhshion in terms of drivers, modules, and queues. The facilities simpll$ the development and integration of network protocol software in a data networking device. In the development of a data networking product, performance modeling and analysis .s/udies are required at an early stage to optimize the .s.v.stem design and minimize risk. In this paper, we develop a queueing model for the perjbrmance analysis of layered networking protocol software based on STREAMS. The formulated model is a state-dependent single-server FIFO waiting-line with feedback. The modeling methodology is applied to Racal’s FastFrameTM 600, a multi-protocol~jirame relay network access device.

A major challenge in integrated services packet networks is the design of packet multiplexing systems to support different QoS requirements. We focus on the problem of satisfying different loss requirements. In time-slotted multiplexing systems, one may have batch arrivals (i.e., simultaneous arrivals in a time slot), and it is necessary to reject some of the packets in a batch when there is an insufficient number of available buffers. We propose a simple and intelligent probabilistic mechanism for rejecting packets in order to meet the loss requirements. It may be used as a selective discard mechanism to provide different QoS levels in terms of loss (weighted fair blocking). It may also be used to achieve fairness (balanced fair blocking). It offers important advantages and gains relative to existing approaches while being very simple. It allows decoupling of buffer dimensioning from the parameterization of a discard mechanism. It is universal in the sense that it may be used on top of many discrete-time multiplexing schemes. It also reduces the buffer space needed to meet loss requirements compared to natural rejection methods such as random selection. The mechanism is derived explicitly for two Markov modulated Bernoulli sources. The derivation for an arbitrary number of Bernoulli sources is formulated as a linear programming problem.

A new mean-value type of algorithm is developed for analyzing multi-facility blocking models with state-dependent arrival rates. It can be applied to a broad class of blocking systems with simultaneous resource possession including, for example, circuit-switched networks. The underlying recursion is cast in terms of blocking probabilities and marginal state probabilities. The developments made here generalize previous results that were restricted to the case of constant arrival rates.

In this paper, we present simple recursive algorithms for computing call and time congestion in the classical Engset model with M sources and N servers. The first recursion has the complexity of O(MN) and gives the blocking probabilities for all intermediate values of M and N. The second recursion assumes a particular value of M and has the complexity of O(N). It gives the blocking probabilities for all intermediate values of N. Both recursions are similar to the well-known recurrence for computing the Erlang loss function.

Single-hop and multi-hop wavelength division multiplex (WDM) access systems have been proposed to take advantage of the large bandwidth available in lightwave mediums. In such systems, there are one or several transmitters and receivers at each node in the network. The receivers and transmitters may be tunable or set at particular wavelengths. The lightwave medium makes available a large number of channels at different wavelengths. Recently, the stochastic analysis of single-hop and multi-hop systems has been studied. The authors present a unified `multi-facility blocking model' approach to the stochastic modeling and analysis of WDM networks of both the single-hop and multi-hop type. The developed model is amenable to exact analysis by an efficient recursive algorithm. The modeling and analysis technique enables one to study the performance of arbitrary access configurations under general traffic conditions including the case of uniform traffic and the case where there are `hot spots.'