The use of virtualisation technology in the cloud enables services to migrate within and across geographically diverse data centres, e.g., to enable load balancing and fault tolerance. An important part of securing cloud services is being able to detect anomalous behaviour, caused by attacks, that is evident in network traffic. However, it is not clear whether virtual service migration adversely affects the performance of contemporary network-based anomaly detection approaches. In this paper, we explore this issue, and show that wide-area virtual service migration can adversely affect state of the art approaches to network flow-based anomaly detection techniques, potentially rendering them unusable.

High quality online video streaming, both live and on-demand, has become an essential part of consumers' every-day lives. The popularity of video streaming has placed a heavy burden on the network infrastructure that now has to transfer an enormous amount of data very quickly to the end-user. To further exacerbate the situation, the Video-on-Demand (VoD) distribution paradigm uses a unicast independent flow for each user request. This results in multiple duplicate flows carrying the same video assets many times end-to-end. We present OpenCache: a highly configurable, efficient and transparent in-network caching service that aims to improve the VoD distribution efficiency by caching video assets as close to the end-user as possible. OpenCache leverages Software Defined Networking to benefit last mile environments by improving network utilisation and increasing the Quality of Experience for the end-user. Our evaluation on a pan-European OpenFlow testbed uses adaptive video streaming and demonstrates that with the use of OpenCache, the external link utilisation is reduced by 100%. Furthermore the streaming application receives better quality video and observes higher throughput, lower latency and shorter start up and buffering times.

Multimedia streaming applications are traditionally delivered over UDP. Recent measurements show that more and more multimedia streaming data are over TCP as web-based TV, P2P streaming, video sharing websites are getting increasingly popular. To improve the quality of experience (QoE) for users and to cope with variability in TCP throughput, streaming applications typically implement buffers. Yet, for improving the QoE and the streaming quality, e.g., playback continuity and timeliness, it is critical to dimension buffers and the initial buffering delay appropriately. In this paper, we first develop a model for TCP streaming systems and an analytical framework to assess the QoE. Our emphasis is on buffer occupancy, which depends on the TCP arriving rate and the playout rate (the coding rate). We observe that TCP window “bounds”, namely congestion window sizes immediately before a triple duplicate or timeout event, allow to distinguish the minimum and maximum buffer occupancy for TCP streaming systems. As confirmed by experiments, the proposed analytical framework allows to estimate the frequency of buffer overflow or underflow events if buffer sizes and the initial buffering delays are known parameters, or conversely, to dimension the buffer and delay appropriately. We further extend our model and analysis for P2P multicast streaming systems. Simulations and experiments in real networks validate our proposed analytical framework in terms of underflow/overflow probabilities and delay.

Channel aware schedulers of modern wireless networks - such as the popular Proportional Fairness Scheduler (PFS) - improve throughput performance by exploiting channel fluctuations while maintaining fairness among the users. In order to simplify the analysis, PFS was introduced and vastly investigated in a model where frame losses do not occur, which is of course not the case in practical wireless networks. Recent studies focused on the efficiency of various implementations of PFS in a realistic model where frame losses can occur. In this work we show that the common straight forward adaptation of PFS to frame losses exposes the system to a malicious attack (which can alternatively be caused by malfunctioning user equipment) that can drastically degrade the performance of innocent users. We analyze the factors behind the vulnerability of the system and propose a modification of PFS designed for the frame loss model which is resilient to such malicious attack while maintaining the fairness properties of original PFS.

Interference is an inherent characteristic of wireless (multihop) communications. Adding interference-awareness to important control functions, e.g., routing, could significantly enhance the overall network performance. Despite some initial efforts, it is not yet clearly understood how to best capture the effects of interference in routing protocol design. Most existing proposals aim at inferring its effect by actively probing the link. However, active probe measurements impose an overhead and may often misrepresent the link quality due to their interaction with other networking functions. Therefore, in this paper we follow a different approach and: 1) propose a simple yet accurate analytical model for the effect of interference on data reception probability, based only on passive measurements and information locally available at the node; 2) use this model to design an efficient interference-aware routing protocol that performs as well as probing-based protocols, yet avoids all pitfalls related to active probe measurements. To validate our proposal, we have performed experiments in a real testbed, setup in our indoor office environment. We show that the analytical predictions of our interference model exhibit good match with both experimental results as well as more complicated analytical models proposed in related literature. Furthermore, we demonstrate that a simple probeless routing protocol based on our model performs at least as good as well-known probe-based routing protocols in a large set of experiments including both intraflow and interflow interference.

In this paper, we propose a fully distributed network model for rate control in application layer multicast based on the utility-price model to maximize the aggregate utilities, and accordingly design an original primal algorithm and a typical dual algorithm both with very small messaging overhead.

In this paper, we study the topic of distributed and optimal congestion control for scalable video streams in application-layer multicast (ALM). We propose a TCP-friendly, fully distributed synchronous algorithm based on the utility-price model which maximizes the global utilities for the streams in the application-layer multicast tree. With the help of numerical study, we show that our proposed algorithm optimizes the overall video quality for fine-grained scalable streams, while minimizing the messaging overhead in the application-layer multicast channel.

Interference is an inherent property of wireless multihop networks. Adding interference-awareness to their control functions can significantly enhance the overall network performance. In this paper we present an analytical model for the probability that a transmission destined to an arbitrary network node is successful in the presence of interference from other nodes in the network. We introduce the concept of interference areas and interference zones to express this probability as a function of the network density, node transmission probability, radio propagation environment, and network card reception sensitivity. Our derivation includes a simpleMAC model, which captures the carrier sense function of many MAC protocols. Contrary to measurementbased models, our derivation only requires information that is locally available to the nodes, avoiding all measurement-related pitfalls. The validation of our model against experiments in a real testbed, set up for this purpose in our indoor office environment, shows good match of the experimental results with the analytical predictions. Interestingly our model predictions follow closely those of more elaborate state-of-the-art analytical models. Finally, to demonstrate the real utility of our model, we have implemented on our testbed a routing metric that explicitly takes interference into account via our derivation. The throughputs of the resulting routes compare favorably with those achieved by a well-known probe-based routing metric.

In this comments paper, we revisit the network model introduced in Cui, Y., et al (2006). We discuss the inaccuracy of the model and, to correct the network model, we propose to apply directed capacity constraints for directed flows. Based on a comparison of numerical results, we show that the corrected model leads to better accuracy than the original model.

As mobile devices are getting more and more ubiquitous, the paradigm of wireless Mobile Ad hoc NETworks (MANETs) is gaining popularity. Especially distributed real-time applications are attractive for their users in this environment. To help manage these applications in MANETs, previously we developed a distributed algorithm called PBS (priority based selection). PBS divides the ad hoc network into zones and selects and maintains a set of nodes, which act as zone servers, to manage these zones. The zone server selection applies node priority comparison, which reflects the node's capabilities to act as zone server. In this paper, we propose a mechanism called NWC (node weight computation) to be applied in the node priority comparison of PBS. NWC computes the node weight, on which the priority comparison is based, as the weighted linear combination of the node parameters. The parameter weights are extracted from the so-called service profile which reflects the characteristics and requirements of the given service. To create this profile we developed a procedure using factorial design via simulations and multi-objective optimization. Our NWC mechanism, applying the created service profile, assigns the highest priorities to the best suited nodes for a given service type and thus designates the most powerful nodes to be selected as zone servers by PBS. Moreover, the weight computation of NWC can be used by other clustering algorithms in MANETs, too.

The increasing number and diversity of wireless mobile devices are driving a revolutionary change in the way users are accessing and using services and resources. Mobile ad hoc networks (MANETs), temporarily created by mobile devices, offer opportunities for sharing services and resources made available by member nodes. Especially, real-time applications have great potentials in mobile ad hoc environments as MANETs form an ideal technological basis for ad hoc real-time interaction among people. However, real-time applications require low latency connections with a minimum of jitter and packet loss rate from the underlying network. To meet these demands is a challenging task in mobile ad hoc networks due to the high level of node mobility, properties of the wireless communication channel and the lack of central administration. In this paper, via a simulation study we analyze and evaluate some Quality of Service (QoS) extensions, like priority queuing or rate control policies, to ad hoc routing in IEEE 802.11 wireless mobile ad hoc networks from the aspects of real-time applications. We show how these QoS extensions can improve the performance of AODV routing protocol and why we selected AODV as the starting point of our work.

In spite of significant work on real-time content distribution, providing consistent service quality across the Internet remains a challenge. Multicast applications further exacerbate that problem as the number of participants-clients, providers and administrative domains-and the level of heterogeneity in a single session is increased. Recently the increasing complexity of managing and operating the Internet has sparked off interest in autonomous networking. In this paper, we introduce vital enhancements to the elementary single source multicast service that facilitates incremental deployment, self-configuration, self-optimization and self-healing. We further introduce an autonomic multicast architecture where a self-monitoring network layer multicast automatically adapts the network layer forwarding service to meet the service quality requirements requested by application level services.

Existing anycast routing protocols solely route packets to the closest group member. In this paper, we introduce density-based anycast routing, a new anycast routing strategy particularly suitable for unstable networks. Instead of routing packets merely on proximity information to the closest member, density-based anycast routing considers the number of available anycast group members for its routing decision. To evaluate the benefits of densitybased routing, we present a unified model to analyze pure proximitybased, pure density-based, as well as combined routing strategies. With an extensive simulation study, we then evaluate these strategies in multiple mobile scenarios. The two main results are that (i) density-based routing increases the probability of successful packet delivery when the network is unstable; and (ii) for particular mobile scenarios, density-based routing finds even shorter routes compared to traditional proximity-based routing. Finally, we discuss implementation issues and propose a solution to dynamically adapt the protocol’s parameter settings. I.

End-to-end transport protocols perform poorly in mobile environments, primarily due to the frequent route breaks induced by node mobility. We present a framework for hop-by-hop transport protocols and propose a new reliable transport protocol: SaFT (store-and-forward transport). We evaluate its performance in mobile ad hoc networks vis-a-vis TCP. Overall, SaFT achieves up to 3 times faster delivery of messages in such networks. We conclude that store-and-forward protocols are suited well to provide reliable communication in mobile ad hoc networks

As mobile devices are getting more ubiquitous, the paradigm of wireless mobile ad hoc networks (MANETs) is gaining popularity. However, MANETs impose new challenges because of their self-organizing, mobile and error-prone nature. Mobility prediction can mitigate the problems emerging from node mobility.In this paper, we propose an approach called XCoPred to predict link quality variations based on pattern matching which can be exploited for mobility prediction. XCoPred doesn't require the use of any external hardware or reference point. Each MANET node monitors the Signal to Noise Ratio (SNR) of its links to obtain a time series of SNR measurements. When a prediction is required, the node tries to detect patterns similar to the current situation in the history of the SNR values of its links by applying the normalized cross-correlation function. The found matches are then used as the base of the prediction. Simulations have shown that fairly accurate predictions around $2~dB$ of absolute average prediction error can be achieved with XCoPred in case of appropriate parameter settings and scenarios showing clear node mobility patterns.

Using distributed real-time applications in mobile environments, e.g., multiplayer games or collaborative working tools, is getting popular as mobile devices and wireless networks are becoming ubiquitous. Especially mobile networked gaming, in regard to the current trends, is considered by game developers, mobile device manufacturers and service providers to be a very attractive source of future revenue. Furthermore, the appearance and evolution of new communication paradigms like mobile ad hoc networking offer new ways and unique features for real-time mobile applications and even for mobile gaming. However, ad hoc networks reserve special challenges mainly due to their self-organized behavior and the resource constraints of the participating mobile devices. One of these challenges is how we can manage applications and support their smooth running in this dynamic and error prone environment. In this paper, we present our algorithm called PBS (Priority Based Selection) which is based on graph theory using Dominating Sets to give support to distributed services in mobile ad hoc networks. PBS computes an appropriate Dominating Set of the network graph in a fully distributed manner and it is the first approach in contrast to the existing algorithms that offers continuous maintenance of this set even in dynamically changing network topologies. To get an appreciation about PBS we discuss and analyze our algorithm, evaluate it via simulations and show how the Dominating Set created and maintained by applying PBS can be used in managing real-time multiplayer games in mobile ad hoc networks.

This paper presents a media- and TCP-friendly rate-based congestion control algorithm (MTFRCC) for scalable video streaming in the Internet. The algorithm integrates two new techniques: i) a utility-based model using the rate-distortion function as the application utility measure for optimizing the overall video quality; and ii) a two-timescale approach of rate averages (long-term and short-term) to satisfy both media and TCP-friendliness. We evaluate our algorithm through simulation and compare the results against the TCP-friendly rate control (TFRC) algorithm. For assessment, we consider five criteria: TCP fairness, responsiveness, aggressiveness, overall video quality, and smoothness of the resulting bit rate. Our simulation results manifest that MTFRCC performs better than TFRC for various congestion levels, including an improvement of the overall video quality.

Content distribution is becoming increasingly important with the decreasing cost of broadband access and the growing number of Internet users. Many of today's content distribution applications require a group communication infrastructure. Source-Specific Multicast (SSM) constitutes a promising alternative to the slow takeoff of multicast as a ubiquitous service, whether at the network- or application layer. Still, implementing any-source functionality on top of SSM remains an interesting research topic.In this paper we present how application semantics can be applied to multi-source distribution over source-specific trees. Specifically, we present two novel tree management methods that incorporate application semantics and evaluate them through simulations against the state of the art. The evaluation results manifest that our methods significantly improve perceived quality of service in the multi-source sessions we consider.

In mobile ad hoc networks, we envision a network where mobile users obtain services from close-by instances. The architecture of today's Internet was designed for fixed users that obtain services from stationary servers and is not well suited for such scenarios. The reason is that (i) the architecture combines identity and location in an IP address and thus forces mobile elements to change their identity when moving over subnet boundaries; and that (ii) the layered architecture implies a separation of service discovery/selection and routing, which is inflexible and also leads to protocol overhead. In this paper, we revise the existing Internet architecture and propose a novel architecture that is better suited for mobile ad hoc networks. There, clients bind to location-independent service identifiers and send packets that are routed to any instance of the desired service in proximity. The routing mechanism is based on the concept of (electrical) fields with which packets are forwarded towards a region with a high density of service nodes. As a result, this architecture increases the probability of successful packet delivery and leads to a robust routing substrate even in very unstable network conditions

Service discovery in mobile ad hoc networks is challenging because of the absence of any central intelligence in the network. Traditional solutions, as used in the Internet, are hence not well suited for mobile ad hoc networks. We present a novel decentralized service discovery mechanism for ad hoc networks. The basic idea is to distribute information about available services to the network neighborhood. We achieve this by using the analogy of an electrostatic field. A service is modeled by a (positive) point charge, and service request packets are seen as (negative) test charges which are attracted by the service instances. We map the physical model to a mobile ad hoc network in a way where each network element calculates a potential value and routes service requests towards the neighbor with the highest potential, hence towards a service instance. Our approach allows for differentiation of service instances based on their capacity. We define the required protocols and methods which we implemented in a network simulator. Using extensive simulations, we evaluate the performance and robustness of the mechanisms. The results indicate good performance and convergence, even in highly mobile environments. We believe that this technique can, and should, be further exploited, e.g., as a routing protocol in mobile networks.

In the context of realizing service deployment on high performance active nodes, we address the problem of installing and configuring software components in complex, heterogeneous node environments. Service deployment is difficult on high-performance nodes due to their complex architectures. They are often based on multiprocessors and run service components in multiple concurrent execution environments. Many types of execution environments have been developed each of which is usually optimized for a certain type of tasks. Advanced active nodes provide more than one to support the whole spectrum of services. Also, different types of active nodes usually support different sets of execution environments. All this motivates us to develop a service deployment scheme that can cope with heterogeneous active nodes. This paper presents our approach to this problem, called Chameleon. It has two important aspects. First, the service model we propose is based on components with two types of interfaces - a data flow interface for programming packet flows and a control interface for controlling and managing the service components. A service is structured as an arbitrary tree of such components. Second, the service specification is independent of any particular node architecture. During the service deployment phase, the service specification is resolved recursively on each node offering the service and is driven by node-specific parameters. The result of this resolution is a tree of service components, which can differ among different types of nodes. Our solution allows a service to take full advantage of specific node features, such as those related to performance or security. This paper advocates a service model that is specialized for active networking. We base the model on two basic abstractions, namely composable containers and connectors.

Finding the longest matching prefix from a database of keywords is an old problem with a number of applications, ranging from dictionary searches to advanced memory management to computational geometry. But perhaps today's most frequent best matching prefix lookups occur in the Internet, when forwarding packets from router to router. Internet traffic volume and link speeds are rapidly increasing; at the same time, a growing user population is increasing the size of routing tables against which packets must be matched. Both factors make router prefix matching extremely performance critical.In this paper, we introduce a taxonomy for prefix matching technologies, which we use as a basis for describing, categorizing, and comparing existing approaches. We then present in detail a fast scheme using binary search over hash tables, which is especially suited for matching long addresses, such as the 128 bit addresses proposed for use in the next generation Internet Protocol, IPv6. We also present optimizations that exploit the structure of existing databases to further improve access time and reduce storage space.

Video distribution over the Internet poses many challenges. Due to the best-effort nature of today's public data networks, end system applications cannot rely on either bandwidth or delay guarantees. We designed and implemented a prototype of a multicast video distribution architecture involving knowledgeable active routers, a scalable video codec based on the wavelet transformation, and a high-performance video scaling algorithm implemented as a router plug-in. The plug-in scales the video with an average overhead of only 22 /spl mu/s per video datagram and is installed on-the-fly on the routers after the sender starts transmitting video for the first time. Through experiments on our test network, we show that we can dramatically improve the video quality on the receivers (up to 15 dB PSNR) by scaling the video on the routers to almost any target bandwidth. The target bandwidth is evaluated by the router solely based on monitoring of the load situation of the router's downstream links and can be adjusted within 50 ms.

Present-day Internet protocol routers typically employ monolithic operating systems that are not easily upgradable and extensible. With the rapid rate of protocol development it is becoming increasingly important to dynamically upgrade router software in an incremental fashion. We have designed and implemented a high-performance, modular, extended services router software architecture in the Net BSD operating system kernel. This architecture allows code modules, called plugins, to be dynamically added and configured at run time. One of the novel features of our design is the ability to bind different plugins to individual flows; this allows for distinct plugin implementations to seamlessly coexist in the same runtime environment. We achieve high performance through a carefully designed modular architecture, an innovative packet classification algorithm that is highly efficient, and by caching that exploits the flow-like characteristics of Internet traffic. Compared to a monolithic best effort kernel, our implementation requires an average increase in packet processing overhead of only 8%, or 600 cycles per packet when running on an Intel Pentium Pro at 233 MHz. By shortcutting the forward loop based on the per-flow state we establish, we can forward packets up to three times faster than the best effort kernel.

Electronic commerce has attracted a great deal of attention recently. Among the different types of electronic commerce, business-to-business is the one most widespread in terms of turnover. An important aspect in business-to-business e-commerce scenarios is how to meet the response time and throughput requirements of applications in spite of execution taking place across corporate boundaries and, in the future, via the Internet instead of using leased lines. Given the unpredictable variations of available bandwidth in today's Internet, providing application quality-of-service guarantees for these requirements is a complex task. Due to the current trend to run multimedia applications over the Internet, we assume that the future Internet will not only support best-effort service but also guaranteed service, e.g. via network resource reservation. This allows us to also take advantage of this service for e-commerce applications. The problem remains non-trivial due to the wide variations among application scenarios. As a first step towards implementing a generic solution, this paper proposes a scheme which enables virtual business processes to meet response time and throughput requirements through network resource reservation. The paper describes the application scenarios we have in mind, the WISE (Workflow-based Internet SErvices) system where these ideas are being implemented, and the mechanisms involved in providing quality-of-service guarantees at the application level.

Middleware supporting secure applications in a distributed environment faces several challenges. Scalable security in the context of multicasting or broadcasting is especially hard when privacy and authenticity is to be assured to highly dynamic groups where the application allows participants to join and leave at any time. Unicast security is well-known and has widely advanced into production state. But proposals for multicast security solutions that have been published so far are complex, often require trust in network components, or are inefficient. In this paper, we propose a framework of new approaches for achieving scalable security in IP multicasting. Our solutions assure that newly joining members are not able to understand past group traffic and that leaving members may not follow future communication. For versatility, our framework supports a range of closely related schemes for key management, ranging from tightly centralized to fully distributed, and even allows switching between these schemes on-the-fly with low overhead. Operations have low complexity [O(log N) for joins or leaves], thus granting scalability even for very large groups. We also present a novel concurrency-enabling scheme, which was devised for fully distributed key management. In this paper, we discuss the requirements for secure multicasting, present our flexible system, and evaluate its properties based on the existing prototype implementation.

Active networking allows the network infrastructure to be programmable. Previous research focused on two commonly separated approaches: "capsules" and "programmable switches". Capsules are typically small programs in packets which flow through the network and are executed in-band on nodes receiving them. Programmable switches are network devices which offer a back-door to inject code by a network administrator out-of-band in order to enhance the device's capabilities. By combining these two approaches, this paper proposes a novel system architecture which allows both application specific data processing in network nodes as well as rapid deployment of new network protocol implementations. Instead of carrying code, data packets carry pointers to digitally signed active modules initially loaded on-the-fly, inband from trusted code servers on the network. Packet processing runs at high speed, may access and modify the whole network subsystem and no potentially slow virtual machines are needed.

No abstract available.

Present day routers typically employ monolithic operating systems which are not easily upgradable and extensible. With the rapid rate of protocol development it is becoming increasingly important to dynamically upgrade router software in an incremental fashion. We have designed and implemented a high performance, modular, extended integrated services router software architecture in the NetBSD operating system kernel. This architecture allows code modules, called plugins, to be dynamically added and configured at run time. One of the novel features of our design is the ability to bind different plugins to individual flows; this allows for distinct plugin implementations to seamlessly coexist in the same runtime environment. High performance is achieved through a carefully designed modular architecture; an innovative packet classification algorithm that is both powerful and highly efficient; and by caching that exploits the flow-like characteristics of Internet traffic. Compared to a monolithic best-effort kernel, our implementation requires an average increase in packet processing overhead of only 8%, or 500 cycles/2.1ms per packet when running on a P6/233.

Internet address lookup is a challenging problem because of increasing routing table sizes, increased traffic, higher speed links, and the migration to 128 bit IPv6 addresses. IP routing lookup requires computing the best matching prefix, for which standard solutions like hashing were believed to be inapplicable. The best existing solution we know of, BSD radix tries, scales badly as IP moves to 128 bit addresses. Our paper describes a new algorithm for best matching prefix using binary search on hash tables organized by prefix lengths. Our scheme scales very well as address and routing table sizes increase: independent of the table size, it requires a worst case time of log2(address bits) hash lookups. Thus only 5 hash lookups are needed for IPv4 and 7 for IPv6. We also introduce Mutating Binary Search and other optimizations that, for a typical IPv4 backbone router with over 33,000 entries, considerably reduce the average number of hashes to less than 2, of which one hash can be simplified to an indexed array access. We expect similar average case behavior for IPv6.

In contrast to traditional one-to-one applications, distributed multimedia applications are based on communication scenarios that involve huge but often limited groups of participants. In general, tele-teaching and virtual reality scenarios require multicasting capabilities that have to deal with reliable data, e.g., simulation data, in addition to isochronous data, e.g., audio or video, that allow for losses depending an the compression technique or color tables. In particular, reliable control messages are needed for multicast signalling purposes. Existing networks have to be enhanced by integrating reliable multicast data transmission protocols to overcome this deficiency. The presented approach provides a network independent error-control scheme for a multicast protocol, offering a high stability in the local area, and an adaptable solution for metropolitan and wide area networks.

Shared window systems allow collaboration transparent, single-user applications interacted with on multiple users' workstations, enabling the members of a cooperative ensemble to simultaneously share and revise information. The paper presents a system capable of sharing applications running under the X Window System. In contrast to previously implemented systems, the present shared window system addresses issues that are crucial for general-purpose use. The shared window system is policy-free, i.e., there are no preferred policies for handling issues such as admission and floor control. Instead, it offers a set of essential mechanisms on top of which various policies and user paradigms may be realized. Further, the system distributes the sharing functionality among all sites involved in a cooperative activity. Measurements have shown a positive impact of this on the overall performance of the system and thus justified the viability of the design decisions taken.>

Da CaPo provides an environment for the dynamic configuration of protocols. Configuration is done with respect to application requirements, properties of the offered network services, and the available resources in the end systems. The goal of the configuration is to provide a service with minimal necessary functionality for each request, i.e., to diminish protocol complexity and so to increase protocol performance. Modules serve as basic building blocks for the protocol configuration. Common software engineering principles like encapsulation and information hiding as well as a unified module interface allow the unconstrained configuration of modules to protocols. The Da CaPo runtime environment links modules to protocols in one UNIX process and realizes an efficient data transport inside the end systems because performance reducing operations like data copying or process switches are minimized.>

Today's programming methodology emphasizes the study of static aspects of programs. In practice, however, monitoring a program in execution, i.e., monitoring a process, is routinely done by any programmer whose task it is to produce a reliable piece of software. There are two reasons why one might want to examine the dynamic aspects of a program: first, to evaluate the performance of a program, and hence to assess its overall behavior; and second, to demonstrate the presence of programming errors, isolate erroneous program code, and correct it. This latter task is commonly called ``debugging a program'' and requires a detailed insight into the innards of a program being executed. Today, many computer systems are being used to measure and control real-world processes. The pace of execution of these systems and their control programs is therefore bound to timing constraints imposed by the real-world process. As a step towards solving the problems associated with execution monitoring of real-time programs, we develop a set of appropriate concepts and define the basic requirements for a real-time monitoring facility. As a test case for the theoretical treatment of the topic, we design hardware and software for an experimental real-time monitoring system and describe its implementation.