System Support for Bandwidth Management and Content Adaptation in Internet Applications

This paper describes the implementation and evaluation of an operating system module, the Congestion Manager (CM), which provides integrated network flow management and exports a convenient programming interface that allows applications to be notified of, and adapt to, changing network conditions. We describe the API by which applications interface with the CM, and the architectural considerations that factored into the design. To evaluate the architecture and API, we describe our implementations of TCP; a streaming layered audio/video application; and an interactive audio application using the CM, and show that they achieve adaptive behavior without incurring much end-system overhead. All flows including TCP benefit from the sharing of congestion information, and applications are able to incorporate new functionality such as congestion control and adaptive behavior.Comment: 14 pages, appeared in OSDI 200

Unicast UDP Usage Guidelines for Application Designers

Publisher PD

Holographic and 3D teleconferencing and visualization: implications for terabit networked applications

Abstract not available

SSthreshless Start: A Sender-Side TCP Intelligence for Long Fat Network

Measurement shows that 85% of TCP flows in the internet are short-lived flows that stay most of their operation in the TCP startup phase. However, many previous studies indicate that the traditional TCP Slow Start algorithm does not perform well, especially in long fat networks. Two obvious problems are known to impact the Slow Start performance, which are the blind initial setting of the Slow Start threshold and the aggressive increase of the probing rate during the startup phase regardless of the buffer sizes along the path. Current efforts focusing on tuning the Slow Start threshold and/or probing rate during the startup phase have not been considered very effective, which has prompted an investigation with a different approach. In this paper, we present a novel TCP startup method, called threshold-less slow start or SSthreshless Start, which does not need the Slow Start threshold to operate. Instead, SSthreshless Start uses the backlog status at bottleneck buffer to adaptively adjust probing rate which allows better seizing of the available bandwidth. Comparing to the traditional and other major modified startup methods, our simulation results show that SSthreshless Start achieves significant performance improvement during the startup phase. Moreover, SSthreshless Start scales well with a wide range of buffer size, propagation delay and network bandwidth. Besides, it shows excellent friendliness when operating simultaneously with the currently popular TCP NewReno connections.Comment: 25 pages, 10 figures, 7 table

Unified congestion control for unreliable transport protocols

Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1999.Includes bibliographical references (p. 55-58).by Hariharan Shankar Rahul.S.M

Adaptive delivery of real-time streaming video

Thesis (M.Eng.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 2001.Includes bibliographical references (p. 87-92).While there is an increasing demand for streaming video applications on the Internet, various network characteristics make the deployment of these applications more challenging than traditional Internet applications like email and the Web. The applications that transmit data over the Internet must cope with the time-varying bandwidth and delay characteristics of the Internet and must be resilient to packet loss. This thesis examines these challenges and presents a system design and implementation that ameliorates some of the important problems with video streaming over the Internet. Video sequences are typically compressed in a format such as MPEG-4 to achieve bandwidth efficiency. Video compression exploits redundancy between frames to achieve higher compression. However, packet loss can be detrimental to compressed video with interdependent frames because errors potentially propagate across many frames. While the need for low latency prevents the retransmission of all lost data, we leverage the characteristics of MPEG-4 to selectively retransmit only the most important data in order to limit the propagation of errors. We quantify the effects of packet loss on the quality of MPEG-4 video, develop an analytical model to explain these effects, and present an RTP-compatible protocol-which we call SR-RTP--to adaptively deliver higher quality video in the face of packet loss. The Internet's variable bandwidth and delay make it difficult to achieve high utilization, Tcp friendliness, and a high-quality constant playout rate; a video streaming system should adapt to these changing conditions and tailor the quality of the transmitted bitstream to available bandwidth. Traditional congestion avoidance schemes such as TCP's additive-increase/multiplicative/decrease (AIMD) cause large oscillations in transmission rates that degrade the perceptual quality of the video stream. To combat bandwidth variation, we design a scheme for performing quality adaptation of layered video for a general family of congestion control algorithms called binomial congestion control and show that a combination of smooth congestion control and clever receiver-buffered quality adaptation can reduce oscillations, increase interactivity, and deliver higher quality video for a given amount of buffering. We have integrated this selective reliability and quality adaptation into a publicly available software library. Using this system as a testbed, we show that the use of selective reliability can greatly increase the quality of received video, and that the use of binomial congestion control and receiver quality adaptation allow for increased user interactivity and better video quality.by Nicholas G. Feamster.M.Eng

Best effort measurement based congestion control

Abstract available: p.