Given the increasing popularity of real-time video communication over enterprise wireless networks, ensuring good quality of experience is becoming critical. A common problem in such networks is that the clients typically adopt the strategy of associating to the access point with the highest signal strength, which can lead to load imbalance, unfair throughput allocation, and low overall video quality. In this paper, we propose to improve load balancing across video clients by selectively triggering directed roams of chosen clients, while explicitly accounting for the cost of roaming. We formulate the problem of identifying the directed roams to perform as an optimization problem where the objective function reflects both the overall utility of all video clients and the cost due to roaming. We subsequently apply relaxations to the original NP-Hard problem, and propose a three-stage algorithm that efficiently computes an approximate solution. Simulations show that by selectively roaming just 5% of the clients, we increase throughput relative to an initial strongest signal based association by at least 10% for over half of the clients and by at least 20% for over a quarter of the clients. We also increase aggregate throughput, improve fairness and thereby increase the overall video quality.

Multiparty videoconferences, or more generally multiparty video calls, are gaining a lot of popularity as they offer a rich communication experience. These applications have, however, large requirements in terms of both network and computational resources and have to deal with sets of heterogenous clients. The multiparty videoconferencing systems are usually either based on expensive central nodes, called Multipoint Control Units (MCU), with transcoding capabilities, or on a peer-to-peer architecture where users cooperate to distribute more efficiently the different video streams. Whereas the first class of systems requires an expensive central hardware, the second one depends completely on the redistribution capacity of the users, which sometimes might neither provide sufficient bandwidth nor be reliable enough. In this work, we propose an alternative solution where we use a central node to distribute the video streams, but at the same time we maintain the hardware complexity and the computational requirements of this node as low as possible, for example, it has no video decoding capabilities. We formulate the rate allocation problem as an optimization problem that aims at maximizing the Quality of Service (QoS) of the videoconference. We propose two different distributed algorithms for solving the optimization problem: the first algorithm is able to find an approximate solution of the problem in a one-shot execution, whereas the second algorithm, based on Lagrangian relaxation, performs iterative updates of the optimization variables in order to gradually increase the value of the objective function. The two algorithms, though being disjointed, nicely complement each other. If executed in sequence, they allow us to achieve both a quick approximate rate reallocation, in case of a sudden change of the system conditions, and a precise refinement of the variables, which avoids problems caused by possible faulty approximate solutions. We have further implemented our solution in a network simulator where we show that our rate allocation algorithm is able to properly optimize users’ QoS. We also illustrate the benefits of our solution in terms of network usage and overall utility when compared to a baseline heuristic method operating on the same system architecture.

Due to the presence of buffers in the inner network nodes, each congestion event leads to buffer queueing and thus to an increasing end-to-end delay. In the case of delay sensitive applications, a large delay might not be acceptable and a solution to properly manage congestion events while maintaining a low end-to-end delay is required. Delay-based congestion algorithms are a viable solution as they target to limit the experienced end-to-end delay. Unfortunately, they do not perform well when sharing the bandwidth with congestion control algorithms not regulated by delay constraints (e.g., loss-based algorithms). Our target is to fill this gap, proposing a novel congestion control algorithm for delay-constrained communication over best effort packet switched networks. The proposed algorithm is able to maintain a bounded queueing delay when competing with other delay-based flows, and avoid starvation when competing with loss-based flows. We adopt the well-known price-based distributed mechanism as congestion control, but: 1) we introduce a novel non-linear mapping between the experienced delay and the price function and 2) we combine both delay and loss information into a single price term based on packet interarrival measurements. We then provide a stability analysis for our novel algorithm and we show its performance in the simulation results carried out in the NS3 framework. Simulation results demonstrate that the proposed algorithm is able to: achieve good intra-protocol fairness properties, control efficiently the end-to-end delay, and finally, protect the flow from starvation when other flows cause the queuing delay to grow excessively.

Multiparty videoconferences, or more generally multiparty video calls, are gaining a lot of popularity as they offer a rich communication experience. These applications have however, large requirements in terms of both network and computational resources and have to deal with sets of heterogenous clients. The multiparty videoconferencing systems can be grouped in two classes. They are based either on expensive central nodes, called multipoint control units (MCU), with transcoding capabilities, or, on a peer-to-peer strategy where users help each other to distribute the different video streams. Whereas the first one requires an expensive central hardware, the second one depends completely on the redistribution capacity of the users, which sometimes might neither provide sufficient bandwidth nor be reliable enough. In this work we propose an alternative solution where we use a central node to distribute the video streams but at the same time we maintain the hardware complexity and the computational requirements of this node as low as possible. The proposed solution uses a distributed algorithm to allocate the users' rates in a Quality of Service (QoS) aware manner. The allocation algorithm is also extremely fast and is able to quickly reallocate the rates in case the conditions change. We have further implemented our solution in a network simulator where we show that our rate allocation algorithm is able to properly optimize users' QoS and adapt to dynamic changes in the system. We also illustrate the benefits of our solution in terms network usage and average utility when compared to a baseline heuristic method operating on the same system architecture.

Atomic broadcast is a useful abstraction for implementing fault-tolerant distributed applications such as state-machine replication. Although a number of algorithms solving atomic broadcast have been published, the problem of bounding the memory used by these algorithms has not been given the attention it deserves. It is indeed impossible to solve repeated atomic broadcast with bounded memory in a system (non-synchronous or not equipped with a perfect failure detector) in which consensus is solvable with bounded memory. The intuition behind this impossibility is the inability to safely garbage-collect unacknowledged messages, since a sender process cannot tell whether the destination process has crashed or is just slow.The usual technique to cope with this problem is to introduce a membership service, allowing the exclusion of a slow or silent process from the group and safely discarding unacknowledged messages sent to this process. In this paper,we present a novel solution that does not rely on a membership service. We relax the specification of atomic broadcast so that it can be implemented with bounded memory, while being strong enough to still be useful for applications that use atomic broadcast, e.g., state-machine replication.

Modularity is a desirable property of complex software systems, since it simplifies code reuse, verification, maintenance, etc. However, the use of loosely coupled modules introduces a performance overhead. This overhead is often considered negligible, but this is not always the case. This paper aims at casting some light on the cost, in terms of performance, that is incurred when designing a relevant group communication protocol with modularity in mind: atomic broadcast. We conduct our experiments using two versions of atomic broadcast: a modular version and a monolithic one. We then measure the performance of both implementations under different system loads. Our results show that the overhead introduced by modularity is strongly related to the level of stress to which the system is subjected, and in the worst cases, reaches approximately 50%.

Designing and correctly implementing group communication systems (GCSs) is notoriously difficult. Assuming that processes fail only by crashing provides a powerful means to simplify the theoretical development of these systems. When making this assumption, however, one should not forget that clean crash failures provide only a coarse approximation of the effects that errors can have in distributed systems. Ignoring such a discrepancy can lead to complex GCS-based applications that pay a large price in terms of performance overhead yet fail to deliver the promised level of dependability. This paper provides a thorough study of error effects in real systems by demonstrating an error-injection-driven design methodology, where error injection is integrated in the core steps of the design process of a robust fault-tolerant system. The methodology is demonstrated for the Fortika toolkit, a Java-based GCS. Error injection enables us to uncover subtle reliability bottlenecks both in the design of Fortika and in the implementation of Java. Based on the obtained insights, we enhance Fortika's design to reduce the identified bottlenecks. Finally, a comparison of the results obtained for Fortika with the results obtained for the OCAML-based Ensemble system in a previous work, allows us to investigate the reliability implications that the choice of the development platform (Java versus OCAML) can have

Protocol composition frameworks provide off-the-shelf composable protocols to simplify the development of custom protocol stacks. All recent protocol frameworks use a general-purpose event-driven model to manage the interactions between protocols. In complex compositions, where protocols offer their service to more than one other protocol, the one-to-many interaction scheme of the eventdriven model introduces composition problems by mixing up the targets to which data (list of headers) should be delivered. To solve these problems, we propose to shift the driving force behind interactions from the events to the headers they carry. We show that the resulting domain-specific header-driven model solves the composition problems, provides statically typed header handling and enhances protocol readability.

Atomic broadcast in particular, and group communication in general, have mainly been specified and implemented in a system model where processes do not recover after a crash. The model is called crash-stop. The drawback of this model is its inability to express algorithms that tolerate the crash of a majority of processes. This has led to extend the crash-stop model to the so-called crash-recovery model, in which processes have access to stable storage, to log their state periodically. This allows them to recover a previous state after a crash. However, the existing specifications of atomic broadcast in the crash-recovery model are not satisfactory, and the paper explains why. The paper also proposes a new specification of atomic broadcast in the crash-recovery model that addresses these issues. Specifically, our new specification allows to distinguish between a uniform and a non-uniform version of atomic broadcast. The non-uniform version logs less information, and is thus more efficient. The uniform and non-uniform atomic broadcast have been implemented and compared with a published atomic broadcast algorithm. Performance results are presented.

We propose a novel approach to the composition of group communication protocols. In this approach, components are modelled as finite state machines communicating via signals. We introduce two building blocks, called adaptor and adaplexor, that ease the development and the composition of group communication protocol stacks, and we discuss how isolation can be achieved in this setting. To validate our architectural concepts, we have implemented the proposed group communication architecture in SDL.

The paper presents and compares Appia and Cactus, two frameworks for protocol composition. The comparison is based on the experience gained in implementing a fault-tolerant atomic broadcast service. The paper also gives preliminary performance results, and concludes with a discussion of the most interesting features of the two frameworks, and suggestions for an improved framework.