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Recent growth of the multimedia content delivery over the Internet and the popularity of the peer-to-peer (P2P) architecture have opened new horizons for emerging novel services over the Internet. Currently, most of multimedia services are being offered to the end users by using set-top boxes installation on the client’s premises, with integrated media storage capabilities and their adaptation. The organization of the end-clients in P2P fashion has great potential to change business models to offer new value-added multimedia services and therefore to generate substantial revenue for service providers. In this paper, we present a mechanism to organize the sender peers in hierarchical hybrid overlay networks. The objective of such organization is to facilitate the receiver peer (content consumer) to select best sender peers for the provision of better QoS (Quality of Service). To construct the hybrid overlay networks, peers offering the same video quality are placed together at the same level of overlay networks. The organization of sender peers within these overlays is subject to (1) the semantic of the video provided by the peer (base layer, or enhancement layers) and (2) the QoS offered by each peer along the end-to-end path. The proposed streaming mechanism is receiver-centric where receiver peer selects a number of sender peers from the overlay networks to receive media contents. The performance evaluation performed using ns-2 simulator shows that hybrid overlays organization mechanism is helpful to enhance the overall QoS by significant improvement in received video packets throughput, the packets drop ratio and transmission delay.

Internet videoconferencing has emerged as a viable medium for communication and entertainment. However, its widespread use is being challenged. This is because videoconference end-users frequently experience perceptual quality impairments such as video frame freezing and voice dropouts due to changes in network conditions on the Internet. These impairments cause extra and correspondingly lead to unwanted that affects user Quality of Experience (QoE) and Internet congestion. Hence, it is important to measure and subsequently minimize the extra end-user interaction effort in a videoconferencing system. In this paper, we describe a novel active measurement scheme that considers end-user interaction effort and the corresponding network bandwidth consumption to provide videoconferencing interaction QoE measurements. The scheme involves a “Multi-Activity Packet-Trains” (MAPTs) methodology to dynamically emulate a videoconference session’s participant interaction patterns and corresponding video activity levels that are affected by transient changes in network conditions. Also, we describe the implementation and validation of the tool we have developed to measure the videoconferencing interaction QoE on a network path using our proposed scheme.

Predicting future calls can be the next advanced feature of the intelligent phone as the phone service providers are looking to offer new services to their customers. Call prediction can be useful to many applications such as planning daily schedule and attending unwanted communications (e.g. voice spam). Predicting calls is a very challenging task. We believe that this is a new area of research. In this paper, we propose a Call Predictor (CP) that computes the probability of receiving calls and makes call prediction based on caller’s behavior and reciprocity. The proposed call predictor is tested with the actual call logs. The experimental results show that the call predictor performs reasonably well with false positive rate of 2.4416%, false negative rate of 2.9191%, and error rate of 5.3606%.

Fourth generation systems (4G) will provide multimedia group communication sessions to multiple mobile users with distinct requirements. This way, it is expected the control of the quality level, connectivity and ubiquitous access for multi-user multimedia sessions across heterogeneous and mobile networks with seamless capability. This paper analyses the requirements of a control architecture to provide (QoS), connectivity and seamless handover management for multi-user sessions in 4G systems, and introduces the (Q3M) proposal. In addition, simulation results present the efficiency of this approach concerning the session setup time and packet losses during handover.

IEEE802.16 (WiMAX) is a wireless metropolitan area network standard with high transmission speed and great coverage. Point to multi-point (PMP) is the traditional WiMAX transmission mode. Wireless mesh network (WMN) is an attractive and useful structure which is suggested to be adopted in WiMAX. A smart WiMAX structure with mesh is proposed in our paper, which supports direct data transmission between SSs (subscribe stations) with the control of the BS if possible. This mesh model is easy to achieve, and could improve the system performance such as throughput and propagation delay. With the help of the BS, several key problems in pure wireless mesh network are satisfactorily solved, such as coexistence, interference and billing. Simulation results are presented in this article to show the enhancement gain of the system performance by the smart model. The idea of virtual cell is put forward to benefit analysis, and the performance with AMC scheme is considered too. More concrete descriptions and explanations are given in the article.

In this paper, we show the feasibility of real-time flow monitoring with controllable accuracy in today’s IP networks. Our approach is based on Netflow and A-GAP. A-GAP is a protocol for continuous monitoring of network state variables, which are computed from device metrics using aggregation functions, such as SUM, AVERAGE and MAX. A-GAP is designed to achieve a given monitoring accuracy with minimal overhead. A-GAP is decentralized and asynchronous to achieve robustness and scalability. The protocol incrementally computes aggregation functions inside the network and, based on a stochastic model, it dynamically configures local filters that control the overhead and accuracy. We evaluate a prototype in a testbed of 16 commercial routers and provide measurements from a scenario where the protocol continuously estimates the total number of FTP flows in the network. Local flow metrics are read out from Netflow buffers and aggregated in real-time. We evaluate the prototype for the following criteria. First, the ability to effectively control the trade off between monitoring accuracy and processing overhead; second, the ability to accurately predict the distribution of the estimation error; third, the impact of a sudden change in topology on the performance of the protocol. The testbed measurements are consistent with simulation studies we performed for different topologies and network sizes, which proves the feasibility of the protocol design, and, more generally, the feasibility of effective and efficient real-time flow monitoring in large network environments.

As mobile devices and functionalities have increased and become intelligent, many related problems have occurred. Especially, the degraded quality of service caused by the shadow area has given the end-users much inconvenience. In addition, the credibility of the service providers, the network operators, and the manufacturers of the mobile devices have also decreased. In order to solve these problems, we need to monitor the radio frequency (RF) signal related information such as the received signal strength (RSS) for finding the shadow areas. So far, no appropriate method has been given. In this paper, we propose a RF signal monitoring method for the quality of service (QoS) improvement based on the Open Mobile Alliance (OMA) Device Management (DM) standard. We have defined the management objects (MOs) for finding the shadow areas and design the management operations for collecting the MOs at the central server. We have developed based on MOs and the management operations. We also present the result of the performance evaluation of our proposed management operations.

In this paper, we provide an online monitor and control approach for adaptive bandwidth allocations in a QoS-aware multimedia network system. Accurate management of bandwidth allocations with a goal of maximizing the revenues is quite complex, due to the interactions among various data flows that dynamically share the network bandwidth. So, we adopt a heuristics-aided control that iteratively adjusts the bandwidth allocation based on the observed packet loss rate and delays. In terms of ’control theory’, the bandwidth allocation and the packet loss rate constitute the system input and output respectively, with the heuristics-based bandwidth adjustment strategies incorporated in a controller along the feedback loop. A ’control-theoretic’ treatment of the QoS adaptation problem allows studying the stability and convergence properties of the QoS delivered to the applications, while maximizing the connectivity service provider’s revenues.

Due to missing copyright, this chapter was removed from the online version of this volume.

We introduce a new network planning approach for management of multicast communications in large data dissemination networks. Our approach, addressing , considers both the common user interests and geographical dependencies while evaluating multicast configurations. We first perform to partition users into virtual domains based on their characteristics within the same geographical vicinity. Second, based on the resource availability of a given network, mechanism is used to evaluate the benefits of forming a new multicast group and the cost of the management overhead. We evaluate the performance of the proposed approach using our simulation software. Preliminary results show that, using two-stage planning, the performance of our approach is at least 40% better than the existing greedy approaches.