Abstract
XCP-b proposes a modification to the XCP router algorithm that computes the spare bandwidth. The modification removes the need for an XCP router to know the exact capacity of the channel, making it possible to use the XCP-b variant in transmission media where the capacity is hard to measure. An example of this kind of medium is the IEEE 802.11. Previous work shows that XCP-b behaves well in single-hop wireless networks and that it Outperforms TCP in terms of fairness, queuing delay, stability and efficiency when the bandwidth delay product of the network grows. In this paper we extend the validation and evaluation of XCP-b to the case of multi-hop wireless networks, both stand-alone and as access networks to other wired networks. The results show that XCP-b maintains its fundamental characteristics in wireless multi-hop scenarios, such as stable throughput and low standing queues, while distributing the bandwidth fairly and using the available capacity efficiently. The simulations also show that XCP-b produces congestion window values that are closer than TCP to the theoretical upper-bound which maximizes spatial reuse.

Abstract
Emerging access networks will use heterogeneous wireless technologies such as 802.11, 802.16 or UMTS, to offer users the best access to the Internet. Layer 2 access networks will consist of wireless bridges (access points) that isolate, concatenated, or in mesh provide access to mobile nodes. The transport of real time traffic over these networks may demand new QoS signalling, used to reserve resources. Besides the reservation, the new signalling needs to address the dynamics of the wireless links, the mobility of the terminals, and the multicast traffic. In this paper a new protocol is proposed aimed at solving this problem-the QoS Abstraction Layer (QoSAL). Existing only at the control plane, the QoSAL is located above the layer 2 and hides from layer 3 the details of each technology with respect to the QoS and to the network topology. The QoSAL has been designed, simulated, and tested. The results obtained demonstrate its usefulness in 4G networks.

Abstract

Abstract
The IP Multimedia Subsystem (IMS), standardized by 3GPP, can be seen as a way of offering Internet services, such as the Web access, electronic mail or instant messaging, through any access technology, in terms of terminals and networks. This paper presents a proposal to solve the problem of the dynamic adaptation of multimedia services provided in the Web context. The proposed solution is based on the real-time generation of user interfaces conditioned by the user context. The solution is mainly characterized by the approach used for resolving the existing dependencies among user interface variables and by the mechanism for acquiring the user context information, which uses the Parlay middleware.

Abstract
The integration of multiple network access technologies in the terminal is one of the most important trends in 4G networks. This integration places telecommunications service providers, and in particular operators, under the significant challenge of being able to transform their services in order to adapt them to a great variety of delivery contexts. This paper describes a generic methodology for the adaptation of telecommunications services provided in the context of an operator. A number of scenarios are examined which show how Parlay middleware can be used by trusted service providers to easily access network resources and context information for enhancing location-based and adaptable multimedia telecommunications services.

Abstract
This paper presents the results of a CMOS-VLSI implementation of a realistic computational model proposed by Walter Freeman for the olfactory system. This model, in later years, has been studied for engineering applications such as auto-association and classification. The analogue nature of the model motivates analogue VLSI implementations. However, the dimension and complexity of such system poses many obstacles to an analogue electronic implementation; one such is the massive interconnectivity which size increases with the square of the number of inputs (channels). We suggest a multiplexing procedure that puts the burden of interconnectivity over a digital system that is simpler to design and makes the analogue system more treatable. The procedure naturally samples the signals. To avoid smoothing filters, a discrete-time solution was also employed. Although with such approach the time resolution is reduced, the advantages overcome the detriments. Previous work has shown that the model can be efficiently discretized using DSP techniques, resulting on a system that is able to predict, on sample-by-sample basis, the behaviour of the VLSI circuit, allowing for a simple and flexible way to adjust the circuit parameters. We present the measured circuit results that are further confronted with the digital implementation.