Community Question Answering (CQA) in different domains is growing at a large scale because of the availability of several platforms and huge shareable information among users. With the rapid growth of such online platforms, a massive amount of archived data makes it difficult for moderators to retrieve possible duplicates for a new question and identify and confirm existing question pairs as duplicates at the right time. This problem is even more critical in CQAs corresponding to large software systems like askubuntu where moderators need to be experts to comprehend something as a duplicate. Note that the prime challenge in such CQA platforms is that the moderators are themselves experts and are therefore usually extremely busy with their time being extraordinarily expensive. To facilitate the task of the moderators, in this work, we have tackled two significant issues for the askubuntu CQA platform: (1) retrieval of duplicate questions given a new question and (2) duplicate question confirmation time prediction. In the first task, we focus on retrieving duplicate questions from a question pool for a particular newly posted question. In the second task, we solve a regression problem to rank a pair of questions that could potentially take a long time to get confirmed as duplicates. For duplicate question retrieval, we propose a Siamese neural network based approach by exploiting both text and network-based features, which outperforms several state-of-the-art baseline techniques. Our method outperforms DupPredictor [33] and DUPE [1] by 5% and 7% respectively. For duplicate confirmation time prediction, we have used both the standard machine learning models and neural network along with the text and graph-based features. We obtain Spearman's rank correlation of 0.20 and 0.213 (statistically significant) for text and graph based features respectively. We shall place all our codes and data in the public domain upon acceptance.

We propose a novel Evolutionary Algorithm (EA) based on the Differential Evolution algorithm for solving global numerical optimization problem in real-valued continuous parameter space. The proposed MadDE algorithm leverages the power of the multiple adaptation strategy with respect to the control parameters and search mechanisms, and is tested on the benchmark functions taken from the CEC 2021 special session & competition on single-objective bound-constrained optimization. Experimental results indicate that MadDE is able to achieve superior performance on global numerical optimization problems when compared against state-of-the-art real-parameter optimizers. We also provide a hyperparameter optimization algorithm SUBHO for improving the search performance of any EA by finding an optimal set of control parameters, and demonstrate its efficacy in enhancing MadDE's performance on the same benchmark. The source code of our implementation is publicly available at https://github.com/subhodipbiswas/MadDE.

Amazon Aurora is a high-throughput cloud-native relational database offered as part of Amazon Web Services (AWS). One of the more novel differences between Aurora and other relational databases is how it pushes redo processing to a multi-tenant scale-out storage service, purpose-built for Aurora. Doing so reduces networking traffic, avoids checkpoints and crash recovery, enables failovers to replicas without loss of data, and enables fault-tolerant storage that heals without database involvement. Traditional implementations that leverage distributed storage would use distributed consensus algorithms for commits, reads, replication, and membership changes and amplify cost of underlying storage. In this paper, we describe how Aurora avoids distributed consensus under most circumstances by establishing invariants and leveraging local transient state. Doing so improves performance, reduces variability, and lowers costs.

Amazon Aurora is a relational database service for OLTP workloads offered as part of Amazon Web Services (AWS). In this paper, we describe the architecture of Aurora and the design considerations leading to that architecture. We believe the central constraint in high throughput data processing has moved from compute and storage to the network. Aurora brings a novel architecture to the relational database to address this constraint, most notably by pushing redo processing to a multi-tenant scale-out storage service, purpose-built for Aurora. We describe how doing so not only reduces network traffic, but also allows for fast crash recovery, failovers to replicas without loss of data, and fault-tolerant, self-healing storage. We then describe how Aurora achieves consensus on durable state across numerous storage nodes using an efficient asynchronous scheme, avoiding expensive and chatty recovery protocols. Finally, having operated Aurora as a production service for over 18 months, we share the lessons we have learnt from our customers on what modern cloud applications expect from databases.

Online games are a rapidly growing Internet application. In order to run a successful online game, game companies and game infrastructure providers must properly manage game workloads and content so that they can maximize player satisfaction while minimizing their own costs. Toward this end, this paper provides a comprehensive, long-term analysis of several popular online games and their players using one of the richest data sets available for online games.

The continuous introduction of converged services such as VoIP and Video-On-Demand has created many operational challenges for service providers. In this paper, we describe how to use location-aware technologies, not only to integrate disparate management applications, but also to transform the underlying process to use geographical views as the focal point of management operations. Based on an engagement with a large Cable provider, we have designed and implemented 3i - Integrated Infrastructure Intelligence - to address key issues in the service assurance process. 3i is highly componentized and provides an intuitive way for creating role-based views through dynamic scoping, event aggregation and status projection, and location-driven active probing. We analyze the current service assurance process and compare it with the improved process after introducing 3i. Overall, the re-engineered process offers base execution improvements in alarm collection, problem drill-down and reporting, as well as complexity improvements throughout. 3i is a fully implemented tool and has demonstrated capabilities beyond its original intended scope as a decision support tool in planning and marketing functions.

In this paper, we design and evaluate the prototype of a network planning service utilizing the coordinate based embedding of network hosts. The kernel of our prototype consists of a scalable network distance embedding method, a core set of services built on top of this embedding, and a generic set of APIs exposed to the applications for utilizing these services. The implemented service core consists of four generic services that we argue to be common to a wide range of applications requiring management of services and monitoring of network distance among Internet hosts. The proposed service does not require any support from the end-hosts. We evaluate the implemented service core using real Internet data and demonstrate its efficacy. Our experience provides several key insights for the design and management of a suitable embedding scheme.

Viruses spreading over the Internet can cause significant damage and the loss of network security. On the other hand, the anti-virus process also plays an important part affecting the dynamics of the virus spreading. The spreading dynamics of most viruses depend on the underlying network topology. While much research attention has been paid in developing the anti-virus software/tools, the dynamics and propagating model of the virus and anti-virus spreading in the topology-aware networks is neither well understood, nor thoroughly studied. To remedy this deficiency, we model and analyze the spreading characteristics of viruses as coexisting with the anti-virus spreading process in the two-layer small-world topology. Applying the fluid analysis, we derive the analytical solutions to the two-layer model. The simulations experiments confirm the validity of our fluid analyses in characterizing both virus and anti-virus spreading dynamics.

VoIP applications require the ability to identify speakers in real time. This paper presents compressed speaker recognition (CSR), an innovative approach to perform speaker recognition directly from the compressed voice packets. CSR performs online speaker recognition from live packet streams of compressed voice packets by performing fast clustering over a defined subset of the features available in each compressed voice packet. Our experimental results show that CSR is highly scalable and accurate across a broad range of speakers

Cheating in on-line games is a prevalent problem for both game makers and players. The popular real-time strategy game genre is especially vulnerable to cheats, as it is frequently hosted as a peer-to-peer game. As the genre has moved towards a distributed simulation approach to gameplay, the number of cheats has been reduced to bug exploits and “maphacks”: a form of information exposure that reveals the opponent’s units and positions when they should be hidden. This paper proposes a technique for detecting maphacking based on bit commitment and explores the tradeoffs in network traffic and information exposure inherent in reducing information exposure in peer-to-peer games.

Viruses propagating over the Internet can result in significant damage and the loss of network security. On the other hand, the anti-virus process is already a part of the virus spreading process and plays an important role affecting the dynamics of virus spreading. The spreading dynamics of most viruses depend upon the underlying network topology. While much research effort has been made in the development of anti-virus software/tools, the dynamics and propagating model of virus and anti-virus spreading in topology-aware networks is neither well understood, nor thoroughly investigated. To remedy this deficiency, we propose a two-layer small-world network topology to model the spreading characteristics of viruses as coexisting with the anti-virus spreading process. Simulation experiments verify the analytical results in characterizing both the virus and anti-virus spreading dynamics.

While deploying the next generation of optical networks with a mesh topology, telecommunications carriers are being confronted with a choice between wavelength switches that can switch traffic at SONET STS-48 (2.5 Gbps) granularity and subwavelength grooming capable switches that can switch at STS-1 (51 Mbps) granularity. The former consumes high fragmented/unused capacity to support low capacity end-to-end circuits using high capacity STS-48 channels (given current subwavelength traffic levels) while the latter may require relatively complicated hardware design that decreases switch scalability. Two-tier network architectures combine the benefits of STS-1 and STS-48 switches by using an upper tier of STS-48 switches for routing and restoration and a lower tier of STS-1 switches for grooming efficiency. A partial two-tier architecture, where STS-1 switches are restricted to a subset of the network nodes, has been shown in to closely match the grooming benefits of a full lower STS-1 tier. We furnish a detailed upper hound analysis of how the fragmented/unused capacity in STS-48 channels (fragmentation loss) varies with the grooming capability of a network for arbitrary traffic scenarios. We show that the upper bounds derived in this paper are in agreement with results obtained using efficient routing and grooming algorithms discussed. Because the bounds obtained do not make any assumptions about traffic and are easy to compute, they are suited for incorporation into a network engineering tool for deciding strategic placement of STS-1 switches in partial two-tier networks. Our work is not biased towards any particular network architecture but aims to analyze the grooming efficiency of two-tier networks.

For many on-line games, user experience is impacted significantly by network latency. As on-line games and on-line game servers proliferate, the ability to discover and connect to nearby servers is essential for maintaining user satisfaction. In this paper, we present a redirection service for on-line games based on the geographic location of players relative to servers. As our results show, the service better meets client demand, saving each client and the Internet as a whole, thousands of miles of networking inefficiency.

We discuss enhancements to the OSPF (open shortest path first) protocol for routing and topology discovery in optical mesh networks. OSPF's opaque LSA (link state advertisement) mechanism is used to extend OSPF to disseminate optical resource related information through optical LSAs. Standard link-state database flooding mechanisms are used for distribution of optical LSAs. Each optical LSA carries optical resource information pertaining to a single optical link bundle between two adjacent OXCs (optical cross connects), allowing for fine granularity changes in topology to be incorporated in path computation algorithms. OSPF packets are carried over a single IP control channel between adjacent OXCs. We analyze the performance of OSPF with optical extensions. Specifically, we compute control channel bandwidth used due to LSA updates. We also estimate the amount of memory required to store the LSA database. Finally, we study CPU usage for computing primary and backup lightpaths. Our analysis shows that the control channel bandwidth usage, memory requirement, and CPU usage are small enough to not be limiting factors for designing optical networks with single OSPF areas consisting of a large number (more than 500) of OXCs.

In order to stem the increasing packet loss rates caused by an exponential increase in network traffic, the IETF has been considering the deployment of active queue management techniques such as RED (random early detection) (see Floyd, S. and Jacobson, V., IEEE/ACM Trans. Networking, vol.1, p.397-413, 1993). While active queue management can potentially reduce packet loss rates in the Internet, we show that current techniques are ineffective in preventing high loss rates. The inherent problem with these algorithms is that they use queue lengths as the indicator of the severity of congestion. In light of this observation, a fundamentally different active queue management algorithm, called BLUE, is proposed, implemented and evaluated. BLUE uses packet loss and link idle events to manage congestion. Using both simulation and controlled experiments, BLUE is shown to perform significantly better than RED, both in terms of packet loss rates and buffer size requirements in the network. As an extension to BLUE, a novel technique based on Bloom filters (see Bloom, B., Commun. ACM, vol.13, no.7, p.422-6, 1970) is described for enforcing fairness among a large number of flows. In particular, we propose and evaluate stochastic fair BLUE (SFB), a queue management algorithm which can identify and rate-limit nonresponsive flows using a very small amount of state information.

We propose a flow-control scheme for multicast ABR services in ATM networks. At the heart of the proposed scheme is an optimal second-order rate control algorithm, called the /spl alpha/-control, designed to deal with the variation in RM-cell round-trip time (RTT) resulting from dynamic drift of the bottleneck in a multicast tree. Applying two-dimensional rate control, the proposed scheme makes the rate process converge to the available bandwidth of the connection's most congested link sensed by the traffic source. It also confines the buffer occupancy to a target regime bounded by a finite buffer capacity as the system enters the equilibrium state. It works well irrespective of the topology of the multicast tree. Using the fluid analysis, we model the proposed scheme and analyze the system dynamics for multicast ABR traffic. We study the convergence properties and derive the optimal-control conditions for the /spl alpha/-control. The analytical results show that the scheme is stable and efficient in the sense that both the source rate and bottleneck queue length rapidly converge to a small neighborhood of the designated operating point. We present simulation results which verify the analytical observations. The simulation experiments also demonstrate the superiority of the proposed scheme to the other schemes in dealing with RM-cell RTT and link-bandwidth variations, achieving fairness in both buffer and bandwidth occupancies, and enhancing average throughput.

This paper describes and evaluates stochastic fair blue (SFB), a novel technique for enforcing fairness among a large number of rows. SFB scalably detects and rate-limits non-responsive flows through the use of a marking probability derived from the blue queue management algorithm and a Bloom (1970) filter. Using analysis and simulation, SFB is shown to effectively handle non-responsive flows using an extremely small amount of state information.

This paper exposes an inherent weakness in current active queue manage ment techniques such asRedin that they rely on queue lengths to indicate the severity of congestion. In light of this observation, a fundamentally different active queue management algorithm calledBlueis proposed.Blueuses packet loss and link utilization to manage congestion. Using simulation and controlled experiments,Blueis shown to significantly outperformRedin providing lower packet loss rates and smaller queuing delays to networked applications such as interactive audio and video.

In this paper, we share our experience in designing and building a content-based switch which we call L5. In addition to the layer 2-3-4 information available in the packet, a content-based switch uses application level information to route traffic in the network. Making routing decisions based on information contained in the payload is not a new idea. In fact application level proxies which are functionally equivalent to a content-based switch, have been around for years. Our contribution is in combining the functionalities of an application level proxy with the data handling capabilities of a switch into a single system. In this paper, we describe the architecture of the L5 system along with the details of how application level information can be efficiently processed in the switch hardware. We cover two specific application examples that we believe are ideal candidates for content-based switching: one is routing HTTP sessions based on uniform resource locators (URL) and the other is session-aware dispatching of secure socket layer (SSL) connections.

The last couple of years has seen a growing momentum towards using the Internet for conducting business. One of the key enablers for business applications is the ability to setup secure channels across the Internet. The Secure Sockets Layer (SSL) protocol provides this capability and it is the most widely used transport layer security protocol. In this paper we investigate the performance of SSL both from a latency as well as a throughput point of view. Since SSL is primarily used to secure Web transactions, we use the SPECWeb96 benchmark suitably modified for use with the SSL protocol. We benchmark two of the more popular Web servers that are in use today and find that they are a couple of orders of magnitude slower when it comes to serving secure Web pages. We investigate the reason for this deficiency by instrumenting the SSL protocol stack with a detailed profiling of the protocol processing components. Based on our findings we suggest two modifications to the protocol that reduce the latency as well as increase the throughput at the server.

The Internet provides no support for privacy or authentication of multicast packets. However, an increasing number of applications require secure multicast services in order to restrict group membership and enforce accountability of group members. A major problem associated with the deployment of secure multicast delivery services is the scalability of the key distribution protocol. This is particularly true with regard to the handling of group membership changes, such as member departures and/or expulsions, which necessitate the distribution of a new session key to all the remaining group members. As the frequency of group membership changes increases, it becomes necessary to reduce the cost of key distribution operations. This paper explores the use of batching of group membership changes to reduce the frequency, and hence the cost, of key re-distribution operations. It focuses explicitly on the problem of cumulative member removal and presents an algorithm that minimizes the number of messages required to distribute new keys to the remaining group members. The algorithm is used in conjunction with a new multicast key management scheme which uses a set of auxiliary keys in order to improve scalability. In contrast to previous schemes which generate a fixed hierarchy of keys, the proposed scheme dynamically generates the most suitable key hierarchy by composing different keys. Our cumulative member removal algorithm uses Boolean function minimization techniques, and outperforms all other schemes known to us in terms of message complexity.

The congestion control mechanisms used in TCP have been the focus of numerous studies and have undergone a number of enhancements. However, even with these enhancements, TCP connections still experience alarmingly high loss rates, especially during times of congestion. To alleviate this problem, the IETF is considering active queue management mechanisms, such as random early detection (RED), for deployment in the network. In this paper, we first show that the effectiveness of RED depends, to a large extent, on the appropriate parameterization of the RED queue. We then show that there is no single set of RED parameters that work well under different congestion scenarios. In light of this observation, we propose and experiment with more adaptive RED gateways which self-parameterize themselves based on the traffic mix. The results show that traffic cognizant parameterization of RED gateways can effectively reduce packet loss, while maintaining high link utilizations under a range of network loads.

We propose a flow control scheme for multicast ABR services in ATM networks. At the heart of the proposed scheme is an optimal second-order rate control algorithm, called the /spl alpha/-control, designed to deal with the variation in RM-cell round-trip time (RTT) resulting from dynamic "drift" of the bottleneck in a multicast tree. Applying two-dimensional rate control, the proposed scheme makes the rate process converge to the available bandwidth of the connection's most congested link. It also confines the buffer occupancy to a target regime bounded by a finite buffer capacity. It works well irrespective of the topology of the multicast tree. Using the fluid approximation, we model the proposed scheme and analyze the system dynamics for multicast ABR traffic. We study the convergence properties and derive the optimal control conditions for the /spl alpha/-control. The analytical results show that the scheme is stable and efficient in the sense that both the source rate and bottleneck queue length rapidly converge to a small neighborhood of the designated operating point. We present simulation results which verify the analytical observations. The simulation results also demonstrate the effectiveness of the proposed scheme in dealing with RM-cell RTT and link-bandwidth variations, and in achieving fairness in both buffer and bandwidth occupancies.

A large number of Internet applications are sensitive to overload conditions in the network. While these applications have been designed to adapt somewhat to the varying conditions in the Internet, they can benefit greatly from an increased level of predictability in network services. We propose minor extensions to the packet queueing and discard mechanisms used in routers, coupled with simple control mechanisms at the source that enable the network to guarantee minimal levels of throughput to different sessions while sharing the residual network capacity in a cooperative manner. The service realized by the proposed mechanisms is an interpretation of the controlled-load service being standardized by the Internet Engineering Task Force. Although controlled-load service can be used in conjunction with any transport protocol, our focus in this paper is on understanding its interaction with Transmission Control Protocol (TCP). Specifically, we study the dynamics of TCP traffic in an integrated services network that simultaneously supports both best-effort and controlled-load sessions. In light of this study, we propose and experiment with modifications to TCP's congestion control mechanisms in order to improve its performance in networks where a minimum transmission rate is guaranteed. We then investigate the effect of network transients, such as changes in traffic load and in service levels, on the performance of controlled-load as well as best-effort connections. To capture the evolution of integrated services in the Internet, we also consider situations where only a selective set of routers are capable of providing service differentiation between best-effort and controlled-load traffic. Finally, we show how the service mechanisms proposed here can be embedded within other packet and link scheduling frameworks in a fully evolved integrated services Internet.

This paper examines the use of adaptable priority marking for providing soft bandwidth guarantees to individual connections or connection groups over the Internet. In contrast to other proposals for service differentiation which focus on providing firm performance guarantees, the proposed scheme does not require resource reservation for individual connections and can be supported with minimal changes to the network infrastructure. It uses modest support from the network in the form of priority handling for appropriately marked packets and relies on intelligent transmission control mechanisms at the edges of the network to achieve the desired throughput levels. The paper describes the control mechanisms and evaluates their behaviour in various network environments. These mechanisms are shown to have several salient features which make them suitable for deployment in an evolving Internet.

This paper explores the performance impact of supporting QoS guarantees on communication in TCP/IP protocol stacks at Unix-like end hosts. We first demonstrate the efficacy of our RSVP-based QoS architecture in providing the desired QoS to individual connections via application-level experiments using UDP sessions and TCP connections on an ATM network. We then identify and measure, via detailed profiling, the overheads imposed by the individual components of the QoS architecture, such as traffic policing, traffic shaping, and buffer management. Our measurements reveal that traffic policing overheads are largely offset by savings due to per-session buffer pre-allocation, and, for ATM networks, a faster path through the network interface layer. In the latter case the data path latency for compliant packets can even be slightly smaller than the default best-effort data path latency. Traffic shaping presents more challenges, primarily because of interactions with the operating system CPU scheduler. We discuss the performance implications of traffic shaping and suggest techniques to reduce or mask some of the associated overheads.

The Internet Engineering Task Force (IETF) is currently in the process of overhauling the architecture of the Internet to meet new challenges and support new applications. One of the most important components of that venture is the enhancement of the Internet service model from a classless best effort service architecture to an integrated services architecture supporting a multitude of classes and types of services. This paper presents the design, implementation, and experiences with a protocol architecture for the integrated services Internet. It is based on the emerging standards for resource reservation in the Internet, namely, the RSVP protocol and the associated service specifications defined by the IETF. Our architecture represents a major functional enhancement to the traditional TCP/IP protocol stack. It is scalable in terms of performance and number of network sessions, and supports a wide variety of network interfaces ranging from legacy LAN interfaces, such as Token Ring and Ethernet, to high-speed ATM interfaces. The paper also describes the implementation of this architecture on the IBM AIX platform and our experiences with the system. We then present a performance analysis of the system which quantifies the overheads imposed by all components of the QoS support, such as traffic policing, traffic shaping, and buffer management.

We propose a simple mechanism named carry-over round robin (CORR) for scheduling cells in asynchronous transfer mode networks. We quantify the operational complexity of CORR scheduling and show that it is comparable to that of a simple round-robin scheduler. We then show that, albeit its simplicity, CORR is very competitive with much more sophisticated and significantly more complex scheduling disciplines in terms of performance. We evaluate the performance of CORR using both analysis and simulation, We derive analytical bounds on the worst case end-to-end delay achieved by a CORR scheduler for different traffic arrival patterns. Using traffic traces from MPEG video streams, we compare the delay performance of CORR with that of packet-by-packet generalized processor sharing (PGPS) and stop-and-go (SG). Our results show that, in terms of delay performance, CORR compares favorably with both PGPS and SG. We also analyze the fairness properties of CORR and show that it achieves near perfect fairness.

An important challenge for the future growth of the Internet is to design routers that can forward the exponentially increasing volume of traffic, and at the same time provide the service differentiation needed by new applications. In this paper, we describe the architecture, implementation, and initial experiences with a system designed to meet this challenge. This system, which we call a QoS capable switch-router (QSR), combines the salient features of switching and routing technologies to provide high throughput and support the different classes of service being defined by the IETF. It consists of a core (ATM) switch fabric connecting intelligent adapters, each capable of both routing and switching pockets. A control engine is responsible for routing, RSVP signalling, and resource management. We have built a prototype network of 3 systems connected to several UNIX hosts, and have conducted preliminary performance measurements on this network.

The paper presents the design and implementation of a quality of service architecture for the Internet. The architecture is based on the emerging standards for resource reservation in the Internet, namely the RSVP protocol and the associated service specifications defined by the Internet Engineering Task Force. Our architecture represents a major functional enhancement to the traditional sockets based communication subsystem, while preserving application programming interface and binary compatibility with existing applications. It is scalable and supports a variety of network interfaces ranging from legacy LAN interfaces, such as token ring and Ethernet, to high speed ATM interfaces. We also describe our initial experiences with the implementation of this architecture on the IBM AIX platform.

One of the major attractions of asynchronous transfer mode (ATM) networks for transporting bursty video traffic is its ability to exploit the multiplexing gains of packet switching while providing quality of service guarantees. Unfortunately, most of the multiplexing mechanisms proposed in the literature fail to exploit the multiplexing gains of ATM. We propose a multirate service mechanism that allows a session to be served at different rates at different times. Applications generating bursty data, such as variable bit-rate (VBR) video, can take advantage of multirate service by requesting a high rate of service for brief periods of bursty arrivals and a much lower rate of service for all other times. Consequently, the applications can improve their delay performance without reserving a high bandwidth for the entire duration of the sessions. Furthermore, the scheduler can multiplex the peaks and the lulls in service rates of different sessions and improve the utilization of the system. Using MPEG video traces from a number of applications, we show that multirate servers outperform single-rate PGPS (packet-by-packet generalized processor sharing) servers and CBR (constant bit-rate) servers in terms of number of connections admitted, while providing the same level of service guarantees. We also investigate the performance of multirate service when service quality need not be guaranteed. We refer to this as predictive service. We propose a measurement-based admission control procedure for predictive service, and show that it helps increase the size of the admissible region even further.

We propose a work-conserving scheduling mechanism for providing deterministic performance guarantees in ATM networks. The most attractive feature of the proposed mechanism, which we call carry-over round robin (CORR), is its simplicity. It is an extension of weighted round robin scheduling. We have derived closed form bounds for worst case end-to-end delay when CORR is used in conjunction with the composite leaky bucket, and moving window regulators. Our results show that albeit its simplicity, CORR is very competitive with some of the more complex scheduling disciplines such as packet-by-packet generalised processor sharing and stop-and-go queueing.

We propose a multirate service mechanism that allows a network session to be served at different rates at different times. Applications generating bursty data, such as VBR video, can take advantage of multirate service by requesting a high rate of service for brief periods of bursty arrivals and a lower rate of service at other times. Consequently, an application can improve its delay performance without reserving high bandwidth for the entire duration of a session. Using MPEG video traces from a number of applications, we show that a multirate server outperforms single rate PGPS (packet-by-packet generalized processor sharing) servers in terms of number of connections admitted, while providing the same level of service guarantees. We also investigate the performance of multirate service when service quality need not be guaranteed. We refer to this as predictive service. We show that multirate servers are superior to single rate servers in providing predictive services.

At the data-link layer, ATM offers a number of features, such as high-bandwidth and per-session quality-of-service (QoS) guarantees, making it particularly attractive to multimedia applications. Unfortunately, many of these features are not visible to applications because of the inadequacies of existing higher-level protocol architectures. Although there is considerable effort underway to tune these protocols for ATM networks, we believe that a new ATM-specific protocol stack is essential to effectively exploit all the benefits of ATM. We describe the semantics of such a protocol stack, and discuss its advantages over traditional protocol architectures from the perspective of multimedia applications. The performance impact of the new protocol architecture is experimentally demonstrated on a video conferencing testbed built around IBM RS/6000's equipped with prototype hardware for video/audio processing, and connected via ATM links.

Management of replicated data has received considerable attention in the last few years. Several replica control schemes have been proposed which work in the presence of both node and communication link failures. However, this resiliency to failure inflicts a performance penalty in terms of the communication overhead incurred. Though the issue of performance of these schemes from the standpoint of availability of the system has been well addressed, the issue of message overhead has been limited to the analysis of worst case and best case message bounds. In this paper we derive expressions for computing the average message overhead of several well known replica control protocols and provide a comparative study of the different protocols with respect to both average message overhead and system availabilities.

This paper proposes a traffic control scheme for integrated services ATM networks. The control strategy comprises of two components: a shaping mechanism at the network entry point and a frame based service discipline at the switches. The shaper enforces a short term peak rate, and a long term average rate. The multiplexing scheme at a switch allocates a guaranteed bandwidth to a connection. A connection may get more than the guaranteed amount, up to a connection specific maximum, if slack bandwidth is available. By imposing an upper bound on the allocated bandwidth, we secure a better handle on the delay jitter. Unlike most frame-based schemes, our scheme allows allocation of bandwidth at any arbitrary granularity. We suggest a simple admission control policy and derive deterministic bounds on end-to-end delay and jitter. An outline of a hardware realization of the scheme is also presented.>

This paper addresses the problem of guaranteed delivery of messages with hard deadlines in a DQDB network. The authors present a cyclic reservation scheme capable of allocating bandwidth with any arbitrary granularity and provide deterministic delay guarantees. They propose two implementations of the allocation scheme within the framework of DQDB medium access control protocol. The proposed implementations are very simple, incur minimal overhead and require only minor changes in the adopted standard.>

Management of replicated data has received considerable attention in the last few years. Several replica control schemes have been proposed which work in the presence of both node and communication link failures. However, this resiliency to failure inflicts a performance penalty in terms of the communication overhead incurred. Though the issue of performance of these schemes, from the standpoint of availability of the system, has been well addressed, the issue of message overhead has been limited to the analysis of worst-case and best-case message bounds. In this paper, we compare several well-known replica management protocols and control schemes in terms of their average-case message overhead. We also consider the tradeoff between the message overhead and availability, and we define the system model considered. Analytical expressions are derived for five well-known replica control protocols. The results are discussed with numerical examples.>

We study the problem of transmitting time constrained synchronous messages in a slotted ring based local area network, carrying synchronous and asynchronous traffic. A bandwidth allocation scheme for synchronous messages is developed on top of a media access control protocol that assigns preemptive priority to synchronous traffic over asynchronous traffic. We derive sufficient conditions for schedulability of time critical synchronous messages and show that the scheme achieves high levels of schedulable utilization. A slot access protocol is proposed for synchronous streams that implements the allocation scheme with minimal additional overhead and loss of schedulable utilization. The protocol is distributed in the sense that any node can locally determine if it can use a slot, without exchanging any explicit messages with other nodes.>