Optical transmission allows massive data transfers thanks to its tremendous transport capacity. Sustained efforts have been made to address the physical constraints that limit its flexibility, but as for today, translucent architectures involving electrical treatments are still favored. Optical Burst Switching (OBS) appears as a promising alternative to Optical Circuit Switching (OCS) in an all-optical framework - provided contention resolution is efficiently solved. In this paper, we investigate further contention resolution with a recourse to the electrical domain. We will thwart the drawbacks of the resulting translucent architecture, i.e., increased delays and equipment cost, by exploiting traffic grooming. Firstly, the contending payload will be submitted to a local aggregation process instead of being stored in dedicated buffers. Secondly, we propose a new transmission scheme, called CAROBS, that exploits dynamic traffic grooming, both at the network edge and in the core network. At the network edge, aggregation queues with different destinations can contribute to build burst trains whose cars can be later optically discarded, one or several at a time, at some intermediate nodes. In the core network, local traffic can be inserted at the tail/head of the train subject to some conditions. A proper dynamic management of the burst trains and of the aggregation pools allows a reduction of the aggregation delays and of the contention rate. Combining our grooming scheme with our translucent architecture annihilates the drawback of signal conversion: Loss-less transfer is achieved without neither impacting the delay nor the equipment cost. It results in an impressive throughput increase that puts OBS close to the performance of OCS with similar features and services, allowing to take full advantage of the high responsiveness to bursty traffic that OBS enjoys.

All-optical circuit switching prevents contention by forbidding multiplexing beyond the wavelength granularity. The drawbacks of such a coarse granularity can be reduced thanks to translucent architectures with MSPPs (MultiService Provisioning Platforms). Another switching paradigm of interest is of

Loss-less transfers are commonly associated with circuit switching: Circuit switching in all-optical networks (OCS) prevent from collisions by avoiding multiplexing below the wavelength granularity. Spectral multiplexing can be combined with time multiplexing in opaque networks at the expense of add

Peer-to-peer (P2P) systems are quite attractive due to their ability to deliver large amounts of data at a reduced deployment cost. They offer an interesting paradigm for media streaming applications that can benefit from the inherent self organization and resource scalability made available by participating peers. However, most studies on P2P live video streaming deal mainly with the feasibility of such an approach with a focus on maximizing throughput and resiliency. Little has been done to investigate the impact ofthe characteristics of the streaming source and of the participating peers on the P2P system performance and more specifically, on the playback delays experienced by peers. Through the use of an integer linear programming model, this paper investigates several related trade offs we need toconsider, to fully exploit the potential of a P2P streaming system.

A major challenge in the optical networking in the metropolitan area is the design of a robust and simple access scheme for building a flexible metro network. In this regard, we consider the bus-based approach of the Passive Optical Networks (PON) with the asynchronous CSMA/CA protocol. We present an enhancement of the CSMA/CA scheme, called LOCOMOTIVE, which takes benefit from the transparent nodes and unidirectional transmission of data signal on optical channel. Idea of the LOCOMOTIVE consists in creating trains of packets, where packets are glued one after another. As illustrated in this paper on simulation based results, the LOCOMOTIVE increases substantially the access protocol performance and achieves efficient utilization of network resources by minimizing the bandwidth fragmentation phenomenon. Index Terms-- Asynchronous optical CSMA/CA, variable length packet, shared bus, bandwidth fragmentation performance.

A simple analytical model is presented to evaluate the effective multicast capacity of unidirectional multi-channel slotted ring networks where each node is equipped with one fixed or tuned transmitter and an array of fixed receivers (i.e., FT-FR w and TT-FR w systems). Furthermore, an approximate approach is developed to compare the mean access delay of a multicast packet between these networks and the TT-FR system [1]. The approach is based on the discrete Geom/Geom/1 queue [2] and on the computation of blocking probabilities. Moreover, the analysis is validated by simulations and the impact of self-similar traffic is shown. The presented methodology enables the comparison of performance of future wavelength division multiplexing (WDM) multicast-capable medium access (MAC) protocols, allowing destination stripping, in terms of effective transmission, and multicast throughput capacity in addition to the access delay.

Scheutzow et al. have presented an analytical model to evaluate the capacity of the unidirectional and bidirectional TT-FR packet-switched ring network under multicast traffic (i.e., a ring where each node is equipped with one tuned transmitter, TT, and one fixed receiver, FR). In this paper, we investigate the impact of multicast on the access delay of the bidirectional TT-FR, by means of an analytical model based on the computation of blocking probability and on the discrete Geom/Geom/1 queue. Moreover, the capacity and the delay of the TT-FRW system (i.e., each ring node has as many fixed receivers as wavelength channels W) are calculated and compared to those of TT-FR. The obtained results are validated by using extensive simulations, and the self-similarity influence is also shown. This work offers an important and simple analytical tool to compare the impact of multicast on access delays and multicast capacity in future slotted ring networks

In this paper, we focus on the capacity of a multicell UMTS system. We first determine an upper bound on the other cell interference and obtain novel expressions for the SIR and powers for both the uplink and the downlink. The former is an asynchronous CDMA system, often using one of three types of receivers : matched filter, minimum mean-square error (MMSE) and decorrelator. In the latter, the SIR depends on the distance between the user and the base station. We show that the inter-cell interference alters the capacity of the system for all kinds of receivers, though the effective bandwidth expressions remain identical to the ones obtained in the single-cell case. As an application, we show how to use this analytical model to develop a new efficient connection admission control (CAC) algorithm.

In this paper, we develop an efficient call admission control (CAC) algorithm for UMTS systems. We first introduce the expressions that we developed for signal-to-interference (SIR) for both uplink and downlink, to obtain a novel CAC algorithm that takes into account, in addition to SIR constraints, the effects of mobility, coverage as well as the wired capacity in the LMTS terrestrial radio access network (UTRAN). for the uplink, and the maximal transmission power of the base station, for the downlink. As of its implementation, we investigate the measurement-based approach as a means to predict future, both handoff and new, call arrivals and thus manage different priority levels. Compared to classical CAC algorithms, our CAC mechanism achieves better performance in terms of outage probability and QoS management.

The need to guarantee quality-of-service (QoS) to multimedia applications leads to a tight integration between the routing and forwarding functions in the Internet. multiprotocol label switching tries to provide a global solution for this integration. In this context, multipoint-to-point (m2p) networks appear as a key architecture since they provide a cheaper way to connect edge nodes than point-to-point connections. M2p networks have been mainly studied for their load balancing ability. In this paper, we go a step further: we propose and evaluate a traffic management scheme that provides deterministic QoS guarantees for multimedia sources in an m2p network. We first derive an accurate upper bound on the end-to-end delay in an m2p architecture based on the concept of additivity. Broadly speaking, an m2p network is additive if the maximum end-to-end delay is equal to the sum of local maximum delays. We then introduce two admission control algorithms for additive networks: a centralized algorithm and a distributed algorithm and discuss their complexity and their scalability.

For original paper see Chlamtac et al. (IEEE/ACM Trans. Networking, vol.6, no.4, p.422-31, 1998 August). We prove that the buffer bound in the above paper, can be improved by using a modification of the proofs in the original paper together with so-called network calculus bounds. We also show that the delay bound in the above paper, is the sum of worst-case queueing delays at all nodes along the path of a connection.

Quality of service (QoS) is a major issue in the recent and current developments of telecommunication networks, notably in the ATM-based B-ISDN or the emerging QoS-capable IP networks. As in most cases where IP runs over ATM, translation of QoS between them, through the translation of their respective performance parameters (PPs), is central to the overall end-to-end performance and is the focus of this framework. We specifically set the end-to-end path which encompasses the major entities of the system, identify the layers of concern and their mechanisms, define inter-protocolar, inter-layer PPs and discuss relevant issues related to their translation between the protocols as well as between the layers, and relate them to the overall end-to-end QoS in IP and ATM.

Various space priority mechanisms and their detailed performance evaluation are described. A comparative performance study is given, indicating the excellent performance characteristics of a simple buffer management scheme called partial buffer sharing. The introduction of a second bearer capability provides a 10/sup -6/ cell loss rate instead of 10/sup -10/.>

The author examines some of the dimensioning problems encountered when using an asynchronous time-division network, to carry synchronous traffic. The basic switch is built around a memory, buffering the frames to be processed. The incoming flows are to be emitted on the output trunks in some order depending on external constraints, so the buffers play in fact the role of a reordering stage; the authors calculates their size, so that the rejection probability meets the grade-of-service criteria. He compares briefly two memory organizations for synchronous time-division switches, namely, common memory and full partitioning.>