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.

Congestion and flow control is an integral part of any Internet data transport protocol. It is widely accepted that the congestion avoidance mechanisms of TCP have been one of the key contributors to the success of the Internet. However, TCP is ill-suited to real-time multimedia streaming applications. Its bursty transmission, and abrupt and frequent wide rate fluctuations cause high delay jitters and sudden quality degradation of multimedia applications. For asymmetric networks such as wireless networks, cable modems, ADSL, and satellite networks, transmitting feedback for (almost) every packet received as it is done in TCP causes congestion in the reverse path. In this environment, TCP may severely underutilize the forward path throughput. Use of multicast further complicates the problem; TCP-like frequent feedback from each receiver to the sender in a large scale multicast session cause well-known scalability limitations (e.g. acknowledgment implosion). We have developed a...

protocols for networks with either backbone or last-hop wireless links. As our basic video transport protocol, we use UDP in conjunction with a congestion control mechanism extended with an LDA. For congestion control, we use the TCP-Friendly Rate Control (TFRC) algorithm. We extend TFRC to use an LDA when a connection uses at least one wireless link in the path between the sender and receiver. One goal of this paper is to evaluate various LDAs under different wireless network topologies and competing traffic. A second goal of this paper is to propose and evaluate a new LDA, called ZigZag, as well as a class of hybrid algorithms based upon ZigZag. We then evaluate these LDAs via simulation. Based upon our simulation results, we find that no single base algorithm performs well across all topologies and competition. However, the hybrid algorithms perform well across topologies, competition, and in some cases match or exceed the performance of the best base LDA for a given scenario.

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.

DCCP, the Datagram Congestion Control Protocol, is a new transport protocol in the TCP/UDP family that provides a congestion-controlled flow of unreliable datagrams. Delay-sensitive applications, such as streaming media and telephony, prefer timeliness to reliability. These applications have historically used UDP and implemented their own congestion control mechanisms---a difficult task---or no congestion control at all. DCCP will make it easy to deploy these applications without risking congestion collapse. It aims to add to a UDP-like foundation the minimum mechanisms necessary to support congestion control, such as possibly-reliable transmission of acknowledgement information. This minimal design should make DCCP suitable as a building block for more advanced application semantics, such as selective reliability. We introduce and motivate the protocol and discuss some of its design principles. Those principles particularly shed light on the ways TCP's reliable byte-stream semantics influence its implementation of congestion control.

A large and rapidly growing proportion of users connect to the Internet via residential broadband networks such as Digital Subscriber Lines (DSL) and cable. Residential networks are often the bottleneck in the last mile of today’s Internet. Their characteristics critically affect Internet applications, including voice-over-IP, online games, and peer-to-peer content sharing/delivery systems. However, to date, few studies have investigated commercial broadband deployments, and rigorous measurement data that characterize these networks at scale are lacking. In this paper, we present the first large-scale measurement study of major cable and DSL providers in North America and Europe. We describe and evaluate the measurement tools we developed for this purpose. Our study characterizes several properties of broadband networks, including link capacities, packet round-trip times and jitter, packet loss rates, queue lengths, and queue drop policies. Our analysis reveals important ways in which residential networks differ from how the Internet is conventionally thought to operate. We also discuss the implications of our findings for many emerging protocols and systems, including delay-based congestion control (e.g., PCP) and network coordinate systems (e.g., Vivaldi).

This paper considers AIMD-based (Additive-Increase Multiplicative-Decrease) congestion control mechanisms that are TCP-compatible (i.e., that compete reasonably fairly with TCP), but that reduce their sending rate less sharply than does TCP in response to a single packet drop. The paper then briefly compares these smoother AIMD-based congestion control mechanisms with TFRC (TCP-Friendly Rate Control), which makes use of equation-based congestion control.

Traditional approaches to receiver-driven layered multicast have advocated the benefits of cumulative layering, which can enable coarse-grained congestion control that complies with TCP-friendliness equations over large time scales. In this paper, we quantify the costs and benefits of using non-cumulative layering and present a new, scalable multicast congestion control scheme which provides a fine-grained approximation to the behavior of TCP additive increase / multiplicative decrease (AIMD). In contrast to the conventional wisdom, we demonstrate that finegrained rate adjustment can be achieved with only modest increases in the number of layers and aggregate bandwidth consumption, while using only a small constant number of control messages to perform either additive increase or multiplicative decrease.

Abstract—We provide an overview of an architecture of today’s Internet streaming media delivery networks and describe various problems that such systems pose with regard to video coding. We demonstrate that based on the distribution model (live or on-demand), the type of the network delivery mechanism (unicast versus multicast), and optimization criteria associated with particular segments of the network (e.g., minimization of distortion for a given connection rate, minimization of traffic in the dedicated delivery network, etc.), it is possible to identify several models of communication that may require different treatment from both source and channel coding perspectives. We explain how some of these problems can be addressed using a conventional framework of temporal motion-compensated, transform-based video compression algorithm, supported by appropriate channel-adaptation mechanisms in client and server components of a streaming media system. Most of these techniques have already been implemented in RealNetworks ® RealSystem ® 8 and its RealVideo ® 8 codec, which we are using throughout the paper to illustrate our results. Index Terms—Internet media delivery networks, scalable video coding, streaming media, video compression. I.

TCP performs poorly on paths that reorder packets significantly, where it misinterprets out-of-order delivery as packet loss. The sender responds with a fast retransmit though no actual loss has occurred. These repeated false fast retransmits keep the sender’s window small, and severely degrade the throughput it attains. Requiring nearly in-order delivery needlessly restricts and complicates Internet routing systems and routers. Such beneficial systems as multi-path routing and parallel packet switches are difficult to deploy in a way that preserves ordering. Toward a more reordering-tolerant Internet architecture, we present enhancements to TCP that improve the protocol’s robustness to reordered and delayed packets. We extend the sender to detect and recover from false fast retransmits using DSACK information, and to avoid false fast retransmits proactively, by adaptively varying dupthresh. Our algorithm is the first that adaptively balances increasing dupthresh, to avoid false fast retransmits, and limiting the growth of dupthresh, to avoid unnecessary timeouts. Finally, we demonstrate that TCP’s RTO estimator tolerates delayed packets poorly, and present enhancements to it that ensure it is sufficiently conservative, without using timestamps or additional TCP header bits. Our simulations show that these enhancements significantly improve TCP’s performance over paths that reorder or delay packets. 1. Introduction and