In tree-based multicast systems, a relatively small number of interior nodes carry the load of forwarding multicast messages. This works well when the interior nodes are dedicated infrastructure routers. But it poses a problem in cooperative application-level multicast, where participants expect to contribute resources proportional to the benefit they derive from using the system. Moreover, many participants may not have the network capacity and availability required of an interior node in high-bandwidth multicast applications. SplitStream is a high-bandwidth content distribution system based on application-level multicast. It distributes the forwarding load among all the participants, and is able to accommodate participating nodes with different bandwidth capacities. We sketch the design of SplitStream and present some preliminary performance results.

CDNs have been widely used to provide low latency, scalability, fault tolerance, and load balancing for the delivery of web content and more recently streaming media. We propose a system that improves the performance of streaming media CDNs by exploiting the path diversity provided by existing CDN infrastructure. Path diversity is provided by the different network paths that exist between a client and its nearby edge servers; and multiple description (MD) coding is coupled with this path diversity to provide resilience to losses. In our system, MD coding is used to code a media stream into multiple complementary descriptions, which are distributed across the edge servers in the CDN. When a client requests a media stream, it is directed to multiple nearby servers which host complementary descriptions. These servers simultaneously stream these complementary descriptions to the client over different network paths.

With the explosive growth of video applications over the Internet, many approaches have been proposed to stream video effectively over packet switched, best-effort networks. Many use techniques from source and channel coding, or implement transport protocols, or modify system architectures in order to deal with delay, loss, and time-varying nature of the Internet. In our previous work, we proposed a framework with a receiver driven protocol to coordinate simultaneous video streaming from multiple senders to a single receiver in order to achieve higher throughput, and to increase tolerance to packet loss and delay due to network congestion. The receiver-driven protocol employs two algorithms: rate allocation and packet partition. The rate allocation algorithm determines the sending rate for each sender; the packet partition algorithm ensures no senders send the same packets, and at the same time, minimizes the probability of late packets. In this paper, we propose a novel rate allocation scheme to be used with Forward Error Correction (FEC) in order to minimize the probability of packet loss in bursty loss environments such as those caused by network congestion. Using both simulations and actual Internet experiments, we demonstrate the effectiveness of our rate allocation scheme in reducing packet loss, and hence, achieving higher visual quality for the streamed video.

Packet loss and end-to-end delay limit delay sensitive applications over the best effort packet switched networks such as the Internet. In our previous work, we have shown that substantial reduction in packet loss can be achieved by sending packets at appropriate sending rates to a receiver from multiple senders, using disjoint paths, and by protecting packets with forward error correction. In this paper, we propose a Path Diversity with Forward error correction (PDF) system for delay sensitive applications over the Internet in which, disjoint paths from a sender to a receiver are created using a collection of relay nodes. We propose a scalable, heuristic scheme for selecting a redundant path between a sender and a receiver, and show that substantial reduction in packet loss can be achieved by dividing packets between the default path and the redundant path. NS simulations are used to verify the effectiveness of PDF system.

Abstract—Enabling video transport over ad hoc networks is more challenging than over other wireless networks. The wireless links in an ad hoc network are highly error prone and can go down frequently because of node mobility, interference, channel fading, and the lack of infrastructure. However, the mesh topology of ad hoc networks implies that it is possible to establish multiple paths between a source and a destination. Indeed, multipath transport provides an extra degree of freedom in designing error resilient video coding and transport schemes. In this paper, we propose to combine multistream coding with multipath transport, to show that, in addition to traditional error control techniques, path diversity provides an effective means to combat transmission error in ad hoc networks. The schemes that we have examined are: 1) feedback based reference picture selection; 2) layered coding with selective automatic repeat request; and 3) multiple description motion compensation coding. All these techniques are based on the motion compensated prediction technique found in modern video coding standards. We studied the performance of these three schemes via extensive simulations using both Markov channel models and OPNET Modeler. To further validate the viability and performance advantages of these schemes, we implemented an ad hoc multiple path video streaming testbed using notebook computers and IEEE 802.11b cards. The results show that great improvement in video quality can be achieved over the standard schemes with limited additional cost. Each of these three video coding/transport techniques is best suited for a particular environment, depending on the availability of a feedback channel, the end-to-end delay constraint, and the error characteristics of the paths. Index Terms—Ad hoc networks, error resilience, IEEE 802.11, multipath transport, video transport, wireless networks.

Abstract — This paper presents a new framework for multimedia streaming that integrates the application and network layer functionalities to meet such stringent application requirements as delay and loss. The coordination between these two layers provides more robust media transmission even under severe network conditions. In this framework, a multiple description source coder is used to produce multiple independently-decodable streams that are routed over partially link-disjoint (non-shared) paths to combat bursty packet losses. We model multi-path streaming and propose a multi-path selection method that chooses a set of paths maximizing the overall quality at the client. Overlay infrastructure is then used to achieve multi-path routing over these selected paths. The simulation results show that the average peak signal-to-noise ratio (PSNR) improves by up to 8.1 dB, if the same source video is routed over intelligently selected multiple paths instead of the shortest path or maximally link-disjoint paths. In addition to PSNR improvement in quality, the enduser experiences a more continual streaming quality. I.

The quality of real-time voice communication over best-effort networks is mainly determined by the delay and loss characteristics observed along the network path. Excessive playout buffering at the receiver is prohibitive and significantly delayed packets have to be discarded and considered as late loss. We propose to improve the tradeoff among delay, late loss rate, and speech quality using multi-stream transmission of real-time voice over the Internet, where multiple redundant descriptions of the voice stream are sent over independent network paths. Scheduling the playout of the received voice packets is based on a novel multi-stream adaptive playout scheduling technique that uses a Lagrangian cost function to trade delay versus loss. Experiments over the Internet suggest largely uncorrelated packet erasure and delay jitter characteristics for different network paths which leads to a noticeable path diversity gain. We observe significant reductions in mean end-to-end latency and loss rates as well as improved speech quality when compared to FEC protected single-path transmission at the same data rate. In addition to our Internet measurements, we analyze the performance of the proposed multi-path voice communication scheme using the ns network simulator for different network topologies, including shared network links.

Video communication is often afflicted by various forms of losses, such as packet loss over the Internet. This paper examines the question of whether the packet loss pattern, and in particular the burst length, is important for accurately estimating the expected mean-squared error distortion. Specifically, we (1) verify that the loss pattern does have a significant effect on the resulting distortion, (2) explain why a loss pattern, for example a burst loss, generally produces a larger distortion than an equal number of isolated losses, and (3) propose a model that accurately estimates the expected distortion by explicitly accounting for the loss pattern, inter-frame error propagation, and the correlation between error frames. The accuracy of the proposed model is validated with JVT/H. 26L coded video and previous frame concealment, where for most sequences the total distortion is predicted to within -4-0.25 dB for burst loss of length two packets, as compared to prior models which underestimate the distortion by about 1.5 dB. Furthermore, as the burst length increases, our prediction is within -4-0.7 dB, while prior models degrade and underestimate the distortion by over 3 dB.

Multiple description (MD) coders provide important error resilience properties. Specifically, MD coders are designed to provide good performance when the loss is limited to a single description, but it is not known in advance which description. In [1], we combined MD video coding with a path diversity transmission system for packet networks such as the Internet, where different descriptions are explicitly transmitted through different network paths, to improve the effectiveness of MD coding over a packet network by increasing the likelihood that the loss probabilities for each description are independent. The available bandwidth in each path may be similar or different, resulting in the requirement of balanced or unbalanced operation, where the bit rate of each description may differ based on the available bandwidth along its path. We design a MD video communication system that is effective in both balanced and unbalanced operation. Specifically, unbalanced MD streams are created by carefully adjusting the frame rate of each description, thereby achieving unbalanced rates of almost 2:1 while preserving MD's effectiveness and error recovery capability.

large-scale distributed network of servers, content distribution, scaleable and reliable algorithm, wide-area testbed, simulation In this work, we consider a large-scale distributed network of servers and a problem of content distribution across it. We propose a novel algorithm, called FastReplica, for an efficient and reliable replication of large files in the Internet environment. There are a few basic ideas exploited in FastReplica. In order to replicate a large file among n nodes (n is in the range of 10-30 nodes), the original file is partitioned into n subfiles of equal size and each subfile is transferred to a different node in the group. After that, each node propagates its subfile to the remaining nodes in the group. Thus instead of the typical replication of an entire file to n nodes by using n Internet paths, connecting the original node to the replication group, FastReplica exploits n x n Internet paths within the replication group where each path is used for transferring 1/n-th of the file. We design a scalable and reliable FastReplica algorithm which can be used for replication of large files to a large group of nodes. The new method is simple and inexpensive. It does not require any changes or modifications to the existing Internet infrastructure, and at the same time, it significantly reduces the file replication time as we demonstrate through experiments on a prototype implementation of FastReplica in a wide-area testbed.