5719748
Leopard: A Locality Aware Peer-to-Peer System with No Hot Spot
Recent research [7,12,2] has shown that Internet hosts can be efficiently (i.e., without excessive measurements) mapped to a virtual (Euclidean) coordinate system, where the geometric distance between any two nodes in this virtual space approximates their real IP network distance (latency). Based on this result, in this paper, we propose an alternative approach that inherently incorporates a virtual coordinate system into a P2P network. In our system, called Leopard, a node is assigned a coordinate in the so-called node geo space as it joins the network, and obtains neighbor relationships that reflects network proximity from the beginning. The object id space and the node geo space are then “weaved” together via a novel technique called geographically-scoped hashing. Through analysis and simulation, we show three major desirable properties of Leopard to exemplify the power of this paradigm shift: i) a constant routing stretch, i.e., IP level network latency of object look-up is proportional to the distance between a requesting node and the target object; ii) always locates a near-by copy when multiple copies exist; and iii) effectively handles “flash crowd” traffic with near optimal load balancing.
14053219
Distributed Algorithm for Service Replication in Service Overlay Network
The service overlay network (SON) is an effective mean to deploy end-to-end QoS guaranteed content delivery services on the current Internet. We model the content delivery service on a SON as a service delivery tree (SDT). Previous studies have addressed the optimal distribution tree formulation issues. In this paper, we focus on the problem of maximizing the total operation profit of the SON. In [1], authors introduced a cost model for optimal bandwidth provisioning in SON. In this paper, we extended this concept and propose an alternative approach to maximize the total effective throughput of SDT as well as to minimize the QoS violation penalty of the SON by service replication. We present both centralized and distributed algorithms for the placement of replicated servers on the SDT. Experiments are carried to quantify the merit, effectiveness and the scalability of the proposed service replication algorithm. In particular, the performance gain is very close to the exhaustive search. The algorithm performs well when we scale up the service overlay networks. Finally, we show that one only needs to perform a small number of replications to attain the optimal gain.
5876943
Adaptive packet sampling for flow volume measurement
Traffic measurement and monitoring are an important component of network management and traffic engineering. With high-speed Internet backbone links, efficient and effective packet sampling techniques for traffic measurement and monitoring are not only desirable, but also increasingly becoming a necessity. Since the utility of sampling depends on the accuracy and economy of measurement, it is important to control sampling error. In this paper we propose and analyze an adaptive, stratified random packet sampling technique for flow-level traffic measurement. In particular, we address the theoretical and practical issues involved. Through theoretical studies and experiments, we demonstrate that the proposed sampling technique provides unbiased estimation of flow size with controllable error bound, in terms of both packet and byte counts for elephant flows, while avoiding excessive oversampling.
33893889
Supporting stored video: reducing rate variability and end-to-end resource requirements through optimal smoothing
Variable-bit-rate (VBR) compressed video can exhibit significant multiple-time-scale bit-rate variability. In this paper we consider the transmission of stored video from a server to a client across a network, and explore how the client buffer space can be used most effectively toward reducing the variability of the transmitted bit rate. Two basic results are presented. First, we show how to achieve the greatest possible reduction in rate variability when sending stored video to a client with given buffer size. We formally establish the optimality of our approach and illustrate its performance over a set of long MPEG-1 encoded video traces. Second, we evaluate the impact of optimal smoothing on the network resources needed for video transport, under two network service models: deterministic guaranteed service (Chang 1994; Wrege et al. 1996) and renegotiated constant-bit-rate (RCBR) service (Grossglauser et al. 1997). Under both models, the impact of optimal smoothing is dramatic.
14141879
Improving VoIP quality through path switching
The current best-effort Internet cannot readily provide the service guarantees that VoIP applications often require. Path switching can potentially address this problem without requiring new network mechanisms, simply by leveraging the robustness to performance variations available from connectivity options such as multi-homing and overlays. In this paper, we evaluate the effectiveness and benefits of path switching in improving the quality of VoIP applications, and demonstrate its feasibility through the design and implementation of a prototype gateway. We argue for an application-driven path switching system that accounts for both network path characteristics and application-specific factors (e.g., codec algorithms, playout buffering schemes). We also develop an application path quality estimator based on the ITU-T E-model for voice quality assessment, and an application-driven path switching algorithm that dynamically adapts the time scales over which path switching decisions are made to maximize voice quality. Through network emulation and experiments over a wide-area multi-homed test bed, we show that, with sufficient path diversity, path switching can yield meaningful improvements in voice quality. Hence by exploiting the inherent path diversity of the Internet, application-driven path switching is a viable option in providing quality-of-service to applications.
3065647
A core stateless bandwidth broker architecture for scalable support of guaranteed services
We present a novel bandwidth broker architecture for scalable support of guaranteed services that decouples the QoS control plane from the packet forwarding plane. More specifically, under this architecture, core routers do not maintain any QoS reservation states, whether per-flow or aggregate. Instead, the QoS reservation states are stored at and managed by a bandwidth broker. There are several advantages of such a bandwidth broker architecture. Among others, it avoids the problem of inconsistent QoS states faced by the conventional hop-by-hop, distributed admission control approach. Furthermore, it allows us to design efficient admission control algorithms without incurring any overhead at core routers. The proposed bandwidth broker architecture is designed based on a core stateless virtual time reference system developed recently. This virtual time reference system provides a unifying framework to characterize, in terms of their abilities to support delay guarantees, both the per-hop behaviors of core routers and the end-to-end properties of their concatenation. We focus on the design of efficient admission control algorithms under the proposed bandwidth broker architecture. We consider both per-flow end-to-end guaranteed delay services and class-based guaranteed delay services with flow aggregation. Using our bandwidth broker architecture, we demonstrate how admission control can be done on a per domain basis instead of on a "hop-by-hop" basis. Such an approach may significantly reduce the complexity of the admission control algorithms. In designing class-based admission control algorithms, we investigate the problem of dynamic flow aggregation in providing guaranteed delay services and devise a new apparatus to effectively circumvent this problem. We conduct detailed analyses to provide theoretical underpinning for our schemes as well as to establish their correctness. Simulations are also performed to demonstrate the efficacy of our schemes.
14900758
Distributed Source Coding for Satellite Communications
Inspired by mobile satellite communications systems, we consider a source coding system which consists of multiple sources, multiple encoders, and multiple decoders. Each encoder has access to a certain subset of the sources, each decoder has access to certain subset of the encoders, and each decoder reconstructs a certain subset of the sources almost perfectly. The connectivity between the sources and the encoders, the connectivity between the encoders and the decoders, and the reconstruction requirements for the decoders are all arbitrary. Our goal is to characterize the admissible coding rate region. Despite the generality of the problem, we have developed an approach which enables us to study all cases on the same footing. We obtain inner and outer bounds of the admissible coding rate region in terms of /spl Gamma//sub N/* and /spl Gamma/~/sub N/*, respectively, which are fundamental regions in the entropy space defined by Yeung (1991). So far, there has not been a full characterization of /spl Gamma//sub N/*, so these bounds cannot be evaluated explicitly except for some special cases. Nevertheless, we obtain an alternative outer bound which can be evaluated explicitly. We show that this bound is tight for all the special cases for which the admissible coding rate region is known. The model we study in this paper is more general than all previously reported models on multilevel diversity coding, and the tools we use are new in multiuser information theory.
6801587
Performance analysis of k-fold multicast networks
Multicast involves transmitting information from a single source to multiple destinations, and is an important operation in high-performance networks. A k-fold multicast network was recently proposed as a cost-effective solution to providing better quality-of-service functions in supporting real-world multicast applications. To give a quantitative basis for network designers to determine the suitable value of system parameter k under different traffic loads, in this paper, we propose an analytical model for the performance of k-fold multicast networks under Poisson traffic. We first give the stationary distribution of network states, and then derive the throughput and blocking probability of the network. We also conduct extensive simulations to validate the analytical model, and the results show that the analytical model is very accurate under the assumptions made. The analytical and simulation results reveal that by increasing the fold of the network, network throughput increases very fast when the fanouts of multicast connections are relatively small, compared with the network size.
2128570
Peer-to-peer based multimedia distribution service
Recently, there are many research interests in providing efficient and scalable multimedia distribution service. However, stringent quality-of-service (QoS) requirements for media distribution, as well as dynamically changing and heterogeneous network capacity in today's best effort Internet, bring many challenges. In this paper, we introduce a novel framework for multimedia distribution service based on peer-to-peer (P2P) networks. A topology-aware overlay is proposed in which hosts self-organize into groups. End hosts within the same group have similar network conditions and can easily collaborate with each other to achieve QoS awareness. In order to improve media delivery quality and provide high service availability, we further propose two distributed heuristic replication strategies, intergroup replication and intragroup replication, based on this topology-aware overlay. Specifically, intergroup replication is aimed to improve the efficiency of media content delivery between the group where a request is issued and the group where the content is stored. Also, intragroup replication is targeted at improving the availability of the content. Extensive simulation results show that the latency in our proposed architecture is 20% less than that of the FreeNet and 50% less than that of the randomly replication system. Simulation results also show that the video quality in our system is much better than that in the other two systems. Our P2P-based approach is also distributed, scalable, cost effective, and aware of the performance.
43800370
LYM-type inequalities for tEC/AUED codes
Some LYM-type inequalities are introduced to obtain a relationship between the parameters of t-error correcting/all unidirectional error-detecting codes of length n. As an application of these inequalities, new lower bounds on the redundancies of systematic tEC/AUED codes are derived and an extensive table is given. >
6522316
Visual echo cancellation in a projector-camera-whiteboard system
We propose to incorporate a whiteboard into a projector-camera system. The whiteboard serves as the writing surface (input) as well as the projecting surface (output). The ability to write and draw on top of computer-projected content opens up many new opportunities for real-time collaborations between people located on-site and remotely. Such applications inevitably require extracting handwritings from video images that contain both handwritings and the projected content. By analogy with echo cancellation in audio conferencing, we call this problem visual echo cancellation. This paper presents one approach to accomplish the task. Our visual echo cancellation algorithm estimates the incident light and derives the surface albedo based on both incident light and reflection. By estimating the albedo, we can extract the writings and recover their colors. Our approach includes two basic components of projector-camera systems: geometric calibration and color calibration. The first one solves the mapping between the position in the camera view and the position in the projector screen, while the second one solves the mapping between the actual color of the projected content and that seen by the camera.