As an efficient distribution mechanism, Peer-to-Peer (P2P) technology has become a tremendously attractive solution to offload servers in large-scale video streaming applications. However, in providing on-demand asynchronous streaming services, P2P streaming design faces two major challenges: how to schedule efficient video sharing between peers with asynchronous playback progresses? how to provide incentives for peers to contribute their resources to achieve a high level of system-wide Quality-of-Experience (QoE)? In this paper, we present iPASS, a novel mesh-based P2P VoD system, to address these challenges. Specifically, iPASS adopts a dynamic buffering-progress-based peering strategy to achieve high peer bandwidth utilization with low system maintenance cost. To provide incentives for peer uploading, iPASS employs a differentiated prefetching design that enables peers with higher contribution prefetch content at higher speed. A distributed adaptive taxation algorithm is developed to balance the system-wide QoE and service differentiations among heterogeneous peers. To assess the performance of iPASS, we built a detailed packet-level P2P VoD simulator and conducted extensive simulations. It was demonstrated that iPASS can completely offload server when the average peer upload bandwidth is more than 1.2 times the streaming rate. Furthermore, we showed that the distributed incentive algorithm motivates peers to contribute and collaboratively achieve a high level of system wide QoE.

As an efficient distribution mechanism, peer-to-peer technology has become a tremendously attractive solution to offload servers in large scale video streaming applications. However, in providing on-demand asynchronous streaming services, P2P streaming design faces two major challenges: how to schedule efficient video sharing between peers with asynchronous playback progresses? how to provide incentives for peers to contribute their resources to achieve a high level of system-wide quality-of-experience (QoE)? In this paper, we present iPASS, a novel mesh-based P2P VoD system, to address these challenges. Specifically, iPASS adopts a dynamic buffering-progress-based peering strategy to achieve high peer bandwidth utilization with low system maintenance cost. To provide incentives for peer uploading, iPASS employs a differentiated pre-fetching design that enables peers with higher contribution pre-fetch content at higher speed. Through packet-level simulations, it was demonstrated that iPASS can effectively offload server and the proposed distributed incentive algorithm motivates peers to contribute and collectively achieve a high level of of QoE.

In this paper we study the video streaming bandwidth of peer-to-peer streaming networks where the underlying topology is a complex network. We focus on the maximal streaming rate and how it depends on the type of network. We consider networks such as small world networks, scale free networks, locally connected networks and random networks. The experimental results indicate that a more connected graph does not necessarily imply a higher streaming rate, whereas properties such as the existence of a Hamiltonian path from the source do.

We develop a framework of distributed and stateless solutions for implicitly given packing linear programs, which are solved by multiple agents operating in a cooperative but uncoordinated manner. This is motivated by multi-commodity flow problems where flows can be split along possibly exponentially many paths.

Compared with the traditional client/server streaming model, peer-assisted video streaming has been shown to provide better scalability with lower infrastructure cost. In this paper, we describe how peer-assisted video streaming can be implemented through real-time service oriented architecture. Our architecture, dubbed PACTS, is composed of 5 distinct service components: directory service, content service, peer download service, peer upload service and rate composition service. PACTS is designed to meet the specific QoS requirements of different users, entered through a simple Web interface. It also provides a distributed implementation of the content and peer directory services for effectively handling flash crowd situations, when a large number of users join during a very short period of time. We discuss the functionalities of the main PACTS services and specify the workflow of how these services work together to deliver real-time streaming services leveraging the benefits of peer to peer technology. By organizing elements of traditional video streaming and peer to peer computing into loosely-coupled composable middleware services and distributing them among participating entities, PACTS enables high-quality low- cost video streaming at a large scale and in real time.

This paper studies TCP performance in a stationary multihop wireless network using IEEE 802.11 for channel access control. We first show that, given a specific network topology and flow patterns, there exists an optimal window size W* at which TCP achieves the highest throughput via maximum spatial reuse of the shared wireless channel. However, TCP grows its window size much larger than W* leading to throughput reduction. We then explain the TCP throughput decrease using our observations and analysis of the packet loss in an overloaded multihop wireless network. We find out that the network overload is typically first signified by packet drops due to wireless link-layer contention, rather than buffer overflow-induced losses observed in the wired Internet. As the offered load increases, the probability of packet drops due to link contention also increases, and eventually saturates. Unfortunately the link-layer drop probability is insufficient to keep the TCP window size around W'*. We model and analyze the link contention behavior, based on which we propose link RED that fine-tunes the link-layer packet dropping probability to stabilize the TCP window size around W*. We further devise adaptive pacing to better coordinate channel access along the packet forwarding path. Our simulations demonstrate 5 to 30 percent improvement of TCP throughput using the proposed two techniques.

This paper studies TCP performance over multihop wireless networks that use the IEEE 802.11 protocol as the access method. Our analysis and simulations show that, given a specific network topology and flow patterns, there exists a TCP window size W*, at which TCP achieves best throughput via improved spatial channel reuse. However, TCP does not operate around W*, and typically grows its average window size much larger; this leads to decreased throughput and increased packet loss. The TCP throughput reduction can be explained by its loss behavior. Our results show that network overload is mainly signified by wireless link contention in multihop wireless networks. As long as the buffer size at each node is reasonably large (say, larger than 10 packets), buffer overflow-induced packet loss is rare and packet drops due to link-layer contention dominate. Link-layer drops offer the first sign for network overload. We further show that multihop wireless links collectively exhibit graceful drop behavior: as the offered load increases, the link contention drop probability also increases, but saturates eventually. In general, the link drop probability is insufficient to stabilize the average TCP window size around W*. Consequently, TCP suffers from reduced throughput due to reduced spatial reuse. We further propose two techniques, link RED and adaptive pacing, through which we are able to improve TCP throughput by 5% to 30% in various simulated topologies. Some simulation results are also validated by real hardware experiments.

Transport protocol design for supporting multimedia streaming in mobile ad hoc networks is challenging because of unique issues, including mobility-induced disconnection, reconnection, and high out-of-order delivery ratios; channel errors and network congestion. In this paper, we describe the design and implementation of a transmission control protocol (TCP)-friendly transport protocol for ad hoc networks. Our key design novelty is to perform multimetric joint identification for packet and connection behaviors based on end-to-end measurements. Our NS-2 simulations show significant performance improvement over wired TCP friendly congestion control and TCP with explicit-link-failure-notification support in ad hoc networks.

Transport protocol design for mobile ad hoc networks is challenging because of unique issues, including mobility-induced disconnection, reconnection, and high out-of-order delivery ratios; channel errors; and network congestion. We describe the design and implementation of a TCP-friendly transport protocol for ad hoc networks. Our key design novelty is to perform multi-metric joint identification for packet and connection behaviors based on end-to-end measurements. Our testbed measurements and ns-2 simulations show a significant performance improvement over standard TCP in ad hoc networks.

Several recent studies have indicated that TCP performance degrades significantly in mobile ad hoc networks. This paper examines how badly TCP may perform in such networks and provides a quantitative characterization of this performance gap. Previous approaches typically made comparisons by ignoring the inherent dynamics such as mobility, channel error and shared-channel contention. Our work provides a realistic, achievable TCP throughput upper bound, and may serve as a benchmark for future TCP modifications in ad hoc networks. Our simulation findings indicate that node mobility, especially mobility-induced network disconnection and reconnection events, has the most significant impact on TCP performance. TCP NewReno merely achieves about 10% of a reference TCPs throughput in such cases. As mobility increases, the relative throughput drop ranges from almost 0% in the static case to 1000% in a highly mobile scenario (mobility speed is 20 m/sec). In contrast, congestion and mild channel error (say, 1%) have less visible effect on TCP (with less than 10% performance drop compared with the reference TCP).

Cost-effectively collecting distributed state information is a challenging problem. There is perhaps no single perfect solution since different distributed application environments pose different requirements from the information collection process. Knowledge of the environment, in terms of traffic conditions, load models etc., plays a key role in determining tradeoffs between accuracy (needed to ensure quality-of-service requirements) and cost-effectiveness. We develop an adaptive information collection algorithm that utilizes network traffic knowledge characterized using a time series model. The algorithm utilizes an information collection architecture consisting of a directory service integrated into the middleware layer with monitoring modules distributed across the network. The cost-effectiveness of the proposed information collection algorithm proposed is verified in simulations over diverse network traffic patterns, i.e. Internet WAN (TCP), MPEG (multimedia) and Web access traffic traces. Our results show that the proposed adaptive information collection algorithm compensates for inaccuracies in network traffic predictions in a cost-effective manner.

In this paper, we present and evaluate algorithms to address combined path and server selection (CPSS) problems in highly dynamic multimedia environments. Our goal is to ensure effective utilization of network and server resources while tolerating imprecision in system state information. Components within the framework implement the optimized scheduling policies as well as collect/update the network and server parameters using a directory service. We present and analyze multiple policies to solve the combined path and server selection (CPSS) problem. In addition, we study multiple techniques for updating the directory service with system state information. We further evaluate the performance of the CPSS policies under different update mechanisms and study the implications of the CPSS policies on directory service management.

The evolving global information infrastructure consists of a widearea networking backbone that provides connectivity among service providers and clients requesting multimedia services via different applications. As this infrastructure scales, service providers replicate data and resources on the network to serve more concurrent clients. Adaptive and intelligent scheduling techniques are required to increase the utilization of their resources and handle an increasing number of requests. Scheduling for multimedia applications must guarantee desired Quality-of-Service (QoS) from both the network path and the server. Furthermore, with the increasing amount of mobile clients and highly dynamic network topologies, optimizing resource utilization becomes complicated. In a highly dynamic and ad-hoc environment where clients are mobile, load sensitive routing and scheduling techniques must be able to tolerate some information