Network management, whether for malfunction analysis, failure prediction, performance monitoring and improvement, generally involves large amounts of data from different sources. To effectively integrate and manage these sources, automatically finding semantic matches among their schemas or ontologies is crucial. Existing approaches on database matching mainly fall into two categories. One focuses on the schema-level matching based on schema properties such as field names, data types, constraints and schema structures. Network management databases contain massive tables (e.g., network products, incidents, security alert and logs) from different departments and groups with nonuniform field names and schema characteristics. It is not reliable to match them by those schema properties. The other category is based on the instance-level matching using general string similarity techniques, which are not applicable for the matching of large network management databases. In this article, we develop a matching technique for large NEtwork MAnagement databases (NEMA) deploying instance-level matching for effective data integration and connection. We design matching metrics and scores for both numerical and non-numerical fields and propose algorithms for matching these fields. The effectiveness and efficiency of NEMA are evaluated by conducting experiments based on ground truth field pairs in large network management databases. Our measurement on large databases with 1,458 fields, each of which contains over 10 million records, reveals that NEMA can achieve accuracy of 95%. We further compare with several other existing algorithms, and show that NEMA outperforms them by 7%-15% in numerical matching and achieves the best trade-off for non-numerical matching.

DaPIPE is a programming environment that supports a developer in writing P4 code to be integrated into the pre-existing data plane program of a network device, without affecting the pre-installed network operating system. This paper discusses the challenges related to such a task and how daPIPE addresses them. In presenting daPIPE we show how it is successfully used to add a novel feature required by a specific use case to a Tofino-based Cisco switch of the Nexus 3400 family.

Software defined network (SDN) has enabled consistent and programmable management in computer networks. However, the explosion of cloud services and content delivery networks (CDNs)-coupled with the momentum of encryption-challenges the simple per-flow management and calls for a more comprehensive approach for managing Web traffic. We propose a new approach based on a “per service” management concept, which allows to identify and prioritize all traffic of important Web services, while segregating others, even if they are running on the same cloud platform, or served by the same CDN. We design and evaluate AWESoME, automatic Web service manager, a novel SDN application to address the above problem. On the one hand, it leverages big data algorithms to automatically build models describing the traffic of thousands of Web services. On the other hand, it uses the models to install rules in SDN switches to steer all flows related to the originating services. Using traffic traces from volunteers and operational networks, we provide extensive experimental results to show that AWESoME associates flows to the corresponding Web service in real-time and with high accuracy. AWESoME introduces a negligible load on the SDN controller and installs a limited number of rules on switches, hence scaling well in realistic deployments. Finally, for easy reproducibility, we release ground truth traces and scripts implementing AWESoME core components.

HTTP(S) has become the main means to access the Internet. The web is a tangle, with (i) multiple services and applications co-located on the same infrastructure and (ii) several websites, services and applications embedding objects from CDN, ads and tracking platforms. Traditional solutions for traffic classification and metering fall short in providing visibility in users' activities. Service providers and corporate network administrators are left with huge amounts of measurements, which cannot immediately reveal the real impact of each web service on the network. Such visibility is key to dimension the network, charge users and policy traffic. This paper introduces the Web Helper Accounting Tool (WHAT), a system to uncover the overall traffic produced by specific web services. WHAT combines big data and machine learning approaches to process large volumes of network flow measurements and learn how to group traffic due to pre-defined services of interest. Our evaluation demonstrates WHAT effectiveness in enabling accurate accounting of the traffic associated to each service. WHAT illustrates the power of machine learning when applied to large datasets of network measurements, and allows network administrators to regain the lost visibility on network usage.

Thanks to the widespread use of smart devices, recent years have witnessed the proliferation of mobile apps available on online stores such as Apple iTunes and Google Play. As the number of new mobile apps continues to grow at a rapid pace, automatic classification of the apps has become an increasingly important problem to facilitate browsing, searching, and recommending them. This paper presents a framework that automatically labels apps with a richer and more detailed categorization and uses the labeled apps to study the app market. Leveraging a fine-grained, hierarchical ontology as a guide, we developed a framework not only to label the apps with fine-grained categorical information but also to induce a customized class hierarchy optimized for mobile app classification. With the classification accuracy of 93%, large-scale categorization conducted with our framework on 168,000 Google Play apps discovers novel inter-class relationships among categories of Google Play market.

Virtual Network Functions (VNFs) are often realized using virtual machines (VMs) because they provide an isolated environment compatible with classical cloud computing technologies. However, VMs are demanding in terms of required resources (CPU and memory) and therefore not suitable for low-cost devices like residential gateways. Such equipment often runs a Linux-based operating system that includes by default a (large) number of common network functions, which can provide some of the services otherwise offered by simple VNFs, but with reduced overhead. In this paper those native software components are made available through a Network Function Virtualization (NFV) platform, thus making their use transparent from the VNF developer point of view.

There are network management, traffic engineering, and security practices adopted in today's networking that rely on the knowledge about what applications' traffic is passing through the networks. These practices might fail with mobile apps whose identity remains hidden in generic HTTP traffic. The main reason is that unlike traditional applications, most mobile apps do not use specific protocols or IP ports with distinctive features. Many enterprises and service providers are in a great need of regaining control over their networks that increasingly carry mobile traffic. In this paper we propose FLOWR, a system that automatically identifies mobile apps by continually learning the apps' distinguishing features via traffic analysis. FLOWR focuses solely on key-value pairs in HTTP headers and intelligently identifies the pairs suitable for app signatures. Our system employs a custom supervised learning approach that leverages a very limited knowledge of app-signature seeds and autonomously grows its capacity for app identification. The approach is motivated by a simple but effective hypothesis that unknown app-identifying features should co-occur with the known signatures. Our experimental results show a significant growth in flow identification coverage provided by FLOWR. Specifically, we show that FLOWR can achieve identification of 86-95% of flows related to their generating apps.

Fingerprinting the Operating System (OS) running on a device based on its traffic has several applications, such as NAT detection, policy enforcement in enterprise networks, and billing for shared access in mobile networks. In this paper, we propose to utilize several features in TCP/IP headers for OS identification, and use real traffic traces to evaluate the accuracy of fingerprinting. Our trace-driven study shows that several techniques that successfully fingerprint desktop OSes are not effective for fingerprinting mobile devices. Therefore, we propose new features for fingerprinting OSes on mobile devices. We also consider NAT/tethering detection, an important application of OS fingerprinting. We use the presence of multiple OSes from the same IP address along with TCP timestamp, clock frequency, and boot time to detect tethering. Evaluation shows that our approach effectively detects tethering and outperforms existing schemes.

A summary of a user's Internet activities, such as web visits, can closely reflect their interests and preferences. However, automating the summarization process is not trivial as it should strike a good balance between generality and specificity, while there is no gold standard for doing so. In our approach to summarizing user information, we introduce two scoring mechanisms that cooperatively optimize for polarizing criteria. Having mapped user activity information onto a category tree, the scoring mechanisms highlight the most representative tree node; the node provides an aggregated view, i.e., a summary, of the activities most representative of the user. We evaluate our approach by summarizing web activity on the network of a large cellular service provider to devise interests of individual users as well as user groups.

No abstract available.

Twitter has attracted millions of users that generate a humongous flow of information at constant pace. The research community has thus started proposing tools to extract meaningful information from tweets. In this paper, we take a different angle from the mainstream of previous works: we explicitly target the analysis of the timeline of tweets from "single users". We define a framework - named TUCAN - to compare information offered by the target users over time, and to pinpoint recurrent topics or topics of interest. First, tweets belonging to the same time window are aggregated into "bird songs". Several filtering procedures can be selected to remove stop-words and reduce noise. Then, each pair of bird songs is compared using a similarity score to automatically highlight the most common terms, thus highlighting recurrent or persistent topics. TUCAN can be naturally applied to compare bird song pairs generated from timelines of different users.

Low complexity solutions to provide deterministic quality over packet switched networks while achieving high resource utilization have been an open research issue for many years. Service differentiation combined with resource overprovisioning has been considered an acceptable compromise and widely deployed given that the amount of traffic requiring quality guarantees has been limited. This approach is not viable, though, as new bandwidth hungry applications, such as video on demand, telepresence, and virtual reality, populate networks invalidating the rationale that made it acceptable so far. Time-driven priority represents a potentially interesting solution. However, the fact that the network operation is based on a time reference shared by all nodes raises concerns on the complexity of the nodes, from the point of view of both their hardware and software architecture. This work analyzes the implications that the timing requirements of time-driven priority have on network nodes and shows how proper operation can be ensured even when system components introduce timing uncertainties. Experimental results on a time-driven priority router implementation based on a personal computer both validate the analysis and demonstrate the feasibility of the technology even on an architecture that is not designed for operating under timing constraints.

The deployment of wireless technologies in industrial networks is very promising mainly due to their inherent flexibility. However, current wireless solutions lack the capability to provide the deterministic, low delay service required by many industrial applications. Moreover, the high level of interference generated by industrial equipment limits the coverage that ensures acceptable performance. Multihop solutions, when combining frame forwarding with higher node density, have the potential to provide the needed coverage while keeping radio communication range short. However, in multihop solutions, the medium access time at each of the nodes traversed additively contributes to the end-to-end delay and the forwarding delay (i.e., the time required for packets to be processed, switched, and queued) at each node is to be added as well. This paper describes time-driven access and forwarding, a solution for guaranteeing deterministic delay, at both the access and forwarding level, in wireless multihop networks, analyzes its properties, and assesses its performance in industrial scenarios.

Previous works showed that the quality-of-service (QoS) requirements of multimedia applications can be optimally satisfied by pipeline forwarding (PF) by providing end-to-end delay guarantees as well as high network resource utilization. However, the unavoidable mismatch between reserved resources and the unpredictable traffic profile of a video stream has an impact on the resulting application layer quality. Therefore, a new low-complexity H.264 video encoding and packetization scheme based on a distortion-optimized macroblock grouping technique is designed here to maximize the performance of video transmission on PF networks. The scheme considers the perceptual importance of the different parts of the video data to group the most important information in few packets that are the natural candidates to receive the deterministic service provided by PF. Results show peak signal-to-noise ratio (PSNR) gains up to 2.5 dB over traditional video encoding and packetization schemes, as well as more graceful degradation in case of high network load.

Preserving integrity of applications being executed in remote machines is an open problem. Integrity requires that application code is not tampered with, prior to or during execution, by a rogue user or a malicious software agent. This paper presents a methodology to enforce runtime integrity of application code by means of an integrity- preserving software component that is combined with the application. The software component is a trusted logic that can be replaced continuously from a remote location during run-time. For added assurance, the software component produces continuous sequence of proofs of its proper operation that are verified remotely.

The widespread diffusion of private networks in SOHO scenarios is fostering an increased deployment of network address translators (NATs). The presence of NATs seriously limits end-to-end connectivity and prevents protocols like the session initiation protocol (SIP) from working properly. This document shows how the address list extension (ALEX), which was originally developed to provide dual-stack and multi-homing support to SIP, can be used, with minor modifications, to ensure end-to-end connectivity for both media and signaling flows, without relying on intermediate relay nodes whenever it is possible.

Despite its efficiency, the general approach of hardcoding protocol format descriptions in packet processing applications suffers from many limitations. Among the others, the lack of flexibility when needing to extend the software for supporting new protocols, and the proliferation of modules with similar functionality between different applications, resulting in decreased maintainability. The NetPDL language was defined for overcoming such limitations, allowing decoupling applications from the knowledge of the format of protocol headers. The main criticism to NetPDL relates to its supposed performance penalties; this paper demonstrates that this language can be effectively used for the dynamic generation of optimized, i.e. efficient and fast, packet-processing code, and presents the architecture of a compiler implemented for such purpose.

With the ever increasing amount of traffic, scalability is probably the most important factor that differentiates several existing approaches to traffic classification. This paper focuses on payload-based classification and compares the results obtained through a "lightweight" traffic classification approach with the ones obtained with a "completely stateful" approach, demonstrating that the first approach, albeit less precise, is still appropriate for a large class of applications.

Wireless IEEE 802.11 networks are typically deployed as a convenient and flexible way to access wired backbones. However, due to the inherently shared nature and physical properties of their communication medium, performance limits are easily reached. Some standard medium access control (MAC) solutions trade simplicity for performance yielding an overall throughput possibly much lower than the actual transmission rate. Moreover, given that many applications with real-time requirements, such as VoIP phones, IPTV, videoconferencing, distributed gaming, are getting increasingly deployed and important to users, it is mandatory for the wireless access to support them. However, the delay and jitter introduced by current IEEE 802.11 standard MAC solutions in some critical, although not uncommon, scenarios can be unacceptable for real-time applications. This work proposes a novel MAC solution for wireless networks that can offer deterministic quality of service for priority traffic while fully supporting statistical multiplexing and possibly increasing the aggregated network throughput. Being based on IEEE 802.11 physical layers and integrating some of the IEEE 802.11 MAC mechanisms, the implementation can leverage on existing components to take advantage of their economy of scale. The throughput and delay achieved with the proposed solution are compared with the ones of standard IEEE 802.11 protocols in various simulation scenarios.

This work addresses the high-rate session scheduling problem in fractional lambda switching (FlambdaS) networks. With its global phase synchronization and pipeline forwarding (PF) operation, FlambdaS offers promising network performance and scalability over its competitors, e.g., time division multiplexing (such as SONET/SDH) and wavelength division multiplexing (WDM). Yet, Non-Immediate Forwarding (NIF) brings challenging complexity to session scheduling, where other known scheduling methods (e.g. RWTA) are not applicable. A forwarding graph is used to wholly examine the huge schedule space for an end-to-end high-rate NIF session. An efficient scheduling algorithm, eSSM, is proposed to explore all possibilities on the graph and present the optimized non-blocking schedule. Complexity bounds are then devised analytically and experimentally verified under specific circumstances. A low-complexity heuristic is proposed to avoid the complexity of eSSM in low-load networks.

Although the SIP protocol claims a complete dual-stack support, some aspects, such as interoperability between different address realms and support for multi-homed hosts, are not taken into consideration. This leads to an extensive usage of proxies as gateways, e.g., between different address realms. ALEX ("address list extension") is a simple extension to the SIP header that addresses these limitations, providing additional scalability for SIP proxies and allowing the establishment of direct channels between peers, while still guaranteeing backward compatibility with traditional SIP implementations.

Efficient scheduling for heterogeneous fractional lambda switching (FlambdaS) networks is required but challenging. A heterogeneous network implies bandwidth mismatch between links of varied bit rates. Moreover, when non-immediate forwarding (NIF) is used in FlambdaS, it increases the scheduling complexity exponentially, while decreasing the blocking probability. Thus, NIF scheduling presents a serious challenge for an algorithm to be used in a large heterogeneous FlambdaS network. In this paper, an efficient scheduling algorithm that is combined with a flexible forwarding scheme is presented. The algorithm provides a full scheduling solution for an end-to-end request in heterogeneous FlambdaS networks. Furthermore, the algorithm has linear complexity in single-channel networks and quadratic complexity in multiple-channel WDM networks.

In order to achieve ultra scalable IP packet switching it is essential to minimize "stopping" of the serial bit streams. In our recent experimental work we demonstrated how this can be achieved with an ultra-scalable switching architecture reaching multi-terabits per second (10-100 Tb/s) in a single chassis. The implemented testbed uses only off-the-shelf optical and electronic components. The scalability of this architecture is the direct outcome of how global time (i.e., UTC - coordinated universal time) and pipeline forwarding are utilized. The paper presents the design of a prototype switch and experimental activity with it.

Future Internet traffic will be dominated by on-demand streaming media flows, such as IPTV, 3D/HD video, gaming, virtual reality, and many more. Consequently, future network architectures will need to implementscalable IP packet switchingcapable of offeringpredictable performancesto such applications. Our recent experimental work demonstrated how an IP network can be implemented without "stopping" the serial bit streams. The deployed switch is very simple, scalable to 10-100 terabits per second in a single chassis, and suitable for all optical implementation. The implemented testbed uses only off-the-shelf optical and electronic components and was completed in 9-month.

After an overview of the distinctive features of interactive multimedia applications, their requirements on the network are discussed: limited delay and delay variation (Jitter) are identified as the most demanding requirements. The most common techniques for controlling service quality are presented and the two major frameworks for their deployment within IP networks, i.e., Integrated Services (IntServ) and Differentiated Services (DiffServ), outlined. The tutorial also studies the implications of the architecture of packet switches on delay bounds. Finally, the tutorial delves into more detail by analyzing the end-to-end delay components of a videoconferencing system step-by-step.

Time-driven Switching (TDS) networks with non-immediate forwarding (NIF) provides scheduling flexibility and consequently, reduces the blocking probability (blocking is defined to take place when transmission capacity is available, but without a feasible schedule). However, it has been shown that with NIF scheduling complexity may grow exponentially. Efficiently finding a schedule from an exponential set of potential schedules is the focus of this paper. The work first presents the mathematical formulation of the NIF scheduling problem, under a wide variety of networking requirements, then introduces an efficient (i.e., having at most polynomial complexity) search algorithm that guarantees to find at least one schedule whenever such a schedule exists. The novel algorithm uses 'trellis' representations and the well-known survivor-based searching principle.

Given a client/server application, how can the server entrust the integrity of the remote client, albeit the latter is running on an un-trusted machine? To address this research problem, we propose a novel approach based on the client-side generation of an execution signature, which is remotely checked by the server, wherein signature generation is locked to the entrusted software by means of code integrity checking. Our approach exploits the features of dynamic aspect-oriented programming (AOP) to extend the power of code integrity checkers in several ways. This paper both presents our approach and describes a prototype implementation for a messaging application.

This paper proposes a novel adaptive AQM (advanced queue management) approach called Time-Driven Early Discard (TED). The basic underlying idea is to set a deadline on packet service time in routers, beyond which packets are discarded. TED is shown to improve fairness among TCP connections sharing congested links when the deadline is chosen proportional to their round trip time (RTT). TED is adaptive in limiting the throughput of only those connections that traverse congested links. In fact, as demonstrated by the presented results, TCP connections traversing parts of the network with enough available capacity can achieve the maximum throughput enabled by their transmission window as corresponding packets do not reach their deadline . Finally, the paper shows how TED can be instrumental in enabling TCP to deploy shorter retransmission timeouts, which results in prompter reactivity to loss, hence improved performance overall in terms of achieved goodput.

This paper reports on the implementation of Time-Driven Priority (TDP) scheduling on a FreeBSD platform. This work is part of a TDP prototyping and demonstration project aimed at showing the implications of TDP deployment in packet-switched networks, especially benefits for real-time applications. This paper focuses on practical aspects related to the implementation of the technology on a Personal Computer (PC)-based router and presents the experimental results obtained on a testbed network. The basic building blocks of a TDP router are described and implementation choices are discussed. The relevant results achieved and here presented can be categorized into two types: qualitative results, including the successful integration of all needed blocks and the insight obtained on the complexity related to the implementation of a TDP router, and quantitative ones, including measures of achievable network utilization and of jitter experienced on a fully-loaded TDP network. The outcome demonstrates the effectiveness of the presented implementation while confirming TDP points of strength.

As traffic on the Internet continues to grow exponentially, there is a real need to solve transmission and switching scalability. Moreover, future Internet traffic will be dominated by streaming media flows, such as video-telephony, video-conferencing, 3D video, virtual reality, and many more. Consequently, network solutions will need to offer quality of service and traffic engineering together with the above mentioned scalability - i.e., over-provisioning is not likely be a viable solution to accommodate streaming media traffic. This paper describes the architecture of a ultra-scalable IP switch and the first experiments with a prototypal implementation. The switch scalability is a consequence of it operating pipeline forwarding of packets, which also results in quality of service guarantees for UDP-based streaming applications, while preserving elastic TCP-based traffic as is, i.e., without affecting any existing applications based on "best- effort" services. Moreover, the prototype demonstrates the low complexity of pipeline forwarding implementation as the deployed network gear was realized from off-the-shelf components in only nine months through the design, implementation, and testing efforts of the authors.

This paper tackles the issue of optimizing the transport of video over packet networks with respect to both resource utilization and user perceived quality. Previous work showed that the quality of service requirements of multimedia applications can be satisfied by pipeline forwarding of packets. However, the current Internet is not based on such technology and its incremental introduction raises questions on how to handle video packets generated by pipeline forwarding unaware sources at the interface between a subnetwork deploying conventional packet scheduling techniques and one implementing pipeline forwarding. This work proposes to use the perceptual importance of the carried video samples to determine which packets shall be transferred with pipeline forwarding - thus receiving deterministic service - and which with a traditional, e.g., best effort or differentiated, service. Simulation results with the first implemented variants of this solution are presented.

XML is a technology that has been widely adopted for data exchange, particularly in Web and e-commerce applications. This paper proposes the use of XML also for network packet processing. It presents some XML-based languages for data exchange and it identifies some examples in which XML can enable a new, modular design of network applications while maintaining the required high processing efficiency. These technologies have been implemented in the NetBee library, which provides an excellent way to give an insight of the performance obtainable with the proposed approach.

This paper describes a novel approach to traffic analysis in high speed networks based on data mining techniques. Data mining techniques are here applied as a means to effectively process the significant amount of captured data. The paper provides a first evaluation of the proposed approach in terms of its ability of extracting relevant information and its computational requirements. Such evaluation is based on experiments run on a prototypal implementation of the proposed approach.

Fractional Lambda Switching (FλS) is a novel proposal for the management of all-optical networks with sub-wavelength provisioning capability. The unique characteristic of FλS is the utilization of the UTC (coordinated universal time) for alignment and switching. Several central research issues are still open in FλS and need to be formally defined and analyzed. Within the scope of this paper, we introduce three novel switch architecture designs that are based on the use of tunable lasers. As an important goal, we introduce the notion “scheduling feasibility” that measure the number of possible different scheduling between an input and output time-frames.

A large set of tools for network monitoring and accounting, security, traffic analysis and prediction - more broadly, for network operation and management - require direct and efficient real-time access to data traveling on the network. Software tools are often preferred because of their low cost and high versatility. However, these tools are often considered to suffer from performance problems on high-speed networks. We demonstrate that, despite the common belief, the performance limits for software real-time network analysis tools are still far from being reached and it can even be improved with limited hardware support. We analyze the performance of a widely used library for network analysis, WinPcap, highlight its bottlenecks, and propose some solutions that almost double the overall speed, thus enabling the deployment of software-based tools on high speed networks.

Traffic monitoring and analysis based on general purpose systems with high speed interfaces, such as Gigabit Ethernet and 10 Gigabit Ethernet, requires carefully designed software in order to achieve the needed performance. One approach to attain such a performance relies on deploying multiple processors. This work analyses some general issues in multiprocessor systems that are particularly critical in the context of packet capture and network monitoring applications. More important, a new algorithm is proposed to coordinate multiple producers concurrently accessing a shared buffer, which is instrumental in packet capture on symmetrical multiprocessor machines.

In general topology networks, routing from one node to another over a tree embedded in the network is intuitively a good strategy, since it typically results in a route length of O(logn) links, n being the number of nodes in the network. Routing from one node to another over a ring embedded in the network results in route length of O(n) links. However, in group (many-to-many) multicast, the overall number of links traversed by each packet, i.e., the networks elements on which resources must possibly be reserved, is typically O(N) for both tree and ring embedding, where N is the size of the group. The paper focuses on tree versus ring embedding for real-time group multicast in which all packets should reach all the nodes in the group with a bounded end-to-end delay. Real-time properties are guaranteed by the deployment of time-driven priority in network nodes. In order to have a better understanding of the nontrivial problem of ring versus tree embedding, we consider static, dynamic and adaptive group multicast scenarios. Tree and ring embedding are compared using different metrics. The results are interesting and counterintuitive, showing that embedding a tree is not always the best strategy. In particular, dynamic and adaptive multicast on a tree require a protocol for updating state information during operation of the group. Such a protocol is not required on the ring where the circular topology and implicit token passing mechanisms are sufficient. Moreover, the bandwidth allocation on the ring for the three multicast scenarios is O(N), while on a general tree it is O(N) for the static multicast scenario and O(N/sup 2/) for the dynamic and adaptive multicast scenarios.

Fractional Lambda (/spl lambda/) Switching (F/spl lambda/S/spl trade/) adds the necessary efficiency to wavelength division multiplexing (WDM) while preserving the simplicity of whole /spl lambda/ switching. Due to its provisioning capability, from a fraction of STS-1 to a full optical channel capacity, F/spl lambda/S/spl trade/ will extend the reach of optical networking all the way to the network edges in the metro and enterprise. Finally, F/spl lambda/S/spl trade/ uniquely enables the implementation of dynamic all-optical switches since its operation does not require (1) optical processing (e.g., packet header processing) and (2) optical buffering.

Videoconferencing is an important global application-it enables people around the globe to interact when distance separates them. In order for the participants in a videoconference call to interact naturally, the end-to-end delay should be below human perception; even though an objective and unique figure cannot be set, 100 ms is widely recognized as the desired one-way delay requirement for interaction. Since the global propagation delay can be about 100 ms, the actual end-to-end delay budget available to the system designer (excluding propagation delay) can be no more than 10 ms. We identify the components of the end-to-end delay in various configurations with the objective of understanding how it can be kept below the desired 10-ms bound. We analyze these components step-by-step through six system configurations obtained by combining three generic network architectures with two video encoding schemes. We study the transmission of raw video and variable bit rate (VBR) MPEG video encoding over (1) circuit switching; (2) synchronous packet switching; and (3) asynchronous packet switching. In addition, we show that constant bit rate (CBR) MPEG encoding delivers unacceptable delay-on the order of the group of pictures (GOP) time interval-when maximizing the quality for static scenes. This study aims at showing that having a global common time reference, together with time-driven priority (TDP) and VBR MPEG video encoding, provides adequate end-to-end delay, which is (1) below 10 ms; (2) independent of the network instant load; and (3) independent of the connection rate. The resulting end-to-end delay (excluding propagation delay) can be smaller than the video frame period, which is better than what can be obtained with circuit switching.

This paper shows how to provide an adaptive real-time group multicast (many-to-many) communication service. Adaptive means that the number of nodes that transmit to the multicast group is continuously changing. In order to meet deterministic quality-of-service (QoS) requirements of a real-time group multicast, some communication resources are reserved. We show (1) how bandwidth is reserved for each multicast group and (2) how an active source can dynamically share the bandwidth allocated to this multicast group with other active group members. Quality-of-service support for a real-time multicast group is based on time-driven priority. In this scheme the time is divided into time frames of fixed duration, and all the time frames are aligned by using a common global time reference, which can be obtained from the Global Positioning System. Bandwidth is allocated to a multicast group as a whole, rather than individually to each user. The allocation is done by reserving time intervals within time frames in a periodic fashion. This type of allocation raises two problems that are studied in this paper: (1) scheduling: how time intervals are reserved to each multicast group and (2) adaptive sharing: how the active (transmitting) participants can dynamically share the time intervals that have been reserved for their multicast group. The proposed approach is based on the embedding of multiple virtual rings, one for each multicast group. By using the virtual rings, it is simple to route messages to all the participants while minimizing the bound on the buffer sizes and queueing delays. The final part of this paper introduces a scalable growth of the multicast group by adding multiple subtrees to the virtual ring.

In general topology networks, routing from one node to another over a tree embedded in the network is intuitively a good strategy, since it typically results in a route length of O(log n) links, being n the number of nodes in the network. Routing from one node to another over a ring embedded in the network would result in route length of O(n) links. However, in group (many-to-many) multicast, the overall number of links traversed by each packet, i.e., the networks elements on which resources must be possibly reserved, is typically O(N) for both tree and ring embedding, where N is the size of the group. This paper focuses on the tree versus ring embedding for real-time group multicast in which all packets should reach all other nodes in the group with a bounded end-to-end delay. In this work, real-time properties are guaranteed by the deployment of time driven priority in network nodes. In order to have a better understanding of the non-trivial problem of ring versus tree embedding, we consider the following group multicast scenarios: (i) static-fixed subset of active nodes, (ii) dynamic-fixed number of active nodes (i.e., the identity of active nodes is changing over time, but its size remains constant), and (iii) adaptive the number and identity of active nodes change over time. Tree and ring embedding are compared using the following metrics: (i) end-to-end delay bound, (ii) overall bandwidth allocated to the multicast group, and (iii) signaling overhead for sharing of the resources allocated to the group. The results are interesting and counter-intuitive, since, as shown, embedding a tree is not always the best strategy. In particular, dynamic and adaptive multicast on a tree requires a protocol for updating state information and coordinates the operation of the group. Such a protocol is not required on the ring where the circular topology, and implicit token passing mechanisms are sufficient. Moreover, the bandwidth allocation on the ring for the three multicast scenarios is O(N); while on a general tree it is O(N) for the static multicast scenario and O(N/sup 2/) for the dynamic and adaptive multicast scenarios.

The question of whether technologies supporting mobile code are bringing significant benefits to the design and implementation of distributed applications is still an open one. Even more difficult is to identify precisely under which conditions a design exploiting mobile code is preferable over a traditional one. In this work, we present an in-depth evaluation of several mobile code design paradigms against the traditional client-server architecture, within the application domain of network management. The evaluation is centered around a quantitative model, which is used to determine precisely the conditions for the selection of a design paradigm minimizing the network traffic related to management.

Videoconferencing is an important global application -it enables people around the globe to interact when they are far from one another. In order for the participants in a video-conference call to interact naturally, the end-to-end delay should be below human perception-about 100 ms. Since the global propagation delay can be about 100 ms, the actual end-to-end delay budget available to the system designer (excluding propagation delay) can be no more than 10 ms. We identify the components of the end-to-end delay in various configurations with the objective of understanding how it can be kept below the desired 10 ms bound. We analyze these components going step-by-step through six system configurations obtained by combining three generic network architectures with two video encoding schemes. We study the transmission of raw video and variable bit rate (VBR) MPEG video encoding over (i) circuit switching, (ii) synchronous packet switching, and (iii) asynchronous packet switching. In addition, we show that constant bit rate (CBR) MPEG encoding delivers unacceptable delay, which is on the order of the group of pictures (GOP) time interval. This study shows that having a global common time reference, together with time-driven priority (TDP) and VBR MPEG video encoding, provides adequate end-to-end delay, which is (i) below 10 ms, (ii) independent of the network instant load, and (iii) independent of the connection rate. The resulting end-to-end delay (excluding propagation delay) can be smaller than the video frame period, which is better than what can be obtained with circuit switching.

This paper tackles the problem of analyzing the correctness and performance of a computer network protocol. Given the complexity of the problem, no currently used technique is able to achieve good results: formal techniques can discover some bugs but can be applied to over-simplified models, only; on the other hand, statistical techniques relying on simulation often fail to find some critical cases for the protocol. Our proposed approach relies on coupling a genetic algorithm with a simulator of the system under verification. Genetic algorithms recently proved themselves excellent tools for giving good, yet approximate, solution to hard-to-solve problems. To prove the effectiveness of our approach, we applied it to the quantitative verification of a network protocol: the complexity of this problem prevents the application of exact techniques, while experimental results show that the verification results we obtained are better than one can achieve with traditional statistical methods. As an example, the approach is applied to the verification of the TCP protocol operating on a given network. A genetic algorithm is able to find a configuration of the traffic over the network that sensitizes a critical problem in the TCP protocol.

This paper shows how to provide an adaptive real-time group multicast (many-to-many) communication service. Such a service can be used by applications, like audio/video tele-conferencing, that require low loss, and bounded delay and jitter. In order to meet deterministic quality of service (QoS) requirements of a real-time group multicast, some communication resources are reserved. In this work we show (i) how bandwidth is reserved for each group, and (ii) how an active user in a multicast group can dynamically share, in an efficient and fair manner, the bandwidth allocated to its group. Quality of service support for a real-time multicast group is based on time driven priority. In this scheme the time is divided into time frames of fixed duration and all the time frames are aligned by using a global time reference which can be obtained from GPS (global positioning system). Bandwidth is allocated to a multicast group as a whole, rather than individually to each user. The allocation is done by reserving time intervals within time frames in some periodic fashion. This sort of allocation raises two problems that are studied in this paper: (1) scheduling: how time intervals are reserved to each multicast group, and (2) adaptive sharing: how the participants dynamically share the time intervals that have been reserved for their multicast group. The proposed approach is based on embedding multiple virtual rings, one for each multicast group. By using the virtual rings it is simple to route messages to all the participants, while minimizing the bound on the buffer sizes and queuing delays.