Notes on the use of RTP for shared workspace applications

The Real-time Transport Protocol, RTP, has become the dominant protocol for streaming audio and video in IP-based environments. A number of proposals have been made which attempt to build on this success and apply RTP for shared workspace applications. We discuss the needs of such applications and the features provided by RTP, with an aim to showing why RTP is not appropriate for such uses

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

Ultra-reliable Low-latency, Energy-efficient and Computing-centric Software Data Plane for Network Softwarization

Network softwarization plays a significantly important role in the development and deployment of the latest communication system for 5G and beyond. A more flexible and intelligent network architecture can be enabled to provide support for agile network management, rapid launch of innovative network services with much reduction in Capital Expense (CAPEX) and Operating Expense (OPEX). Despite these benefits, 5G system also raises unprecedented challenges as emerging machine-to-machine and human-to-machine communication use cases require Ultra-Reliable Low Latency Communication (URLLC). According to empirical measurements performed by the author of this dissertation on a practical testbed, State of the Art (STOA) technologies and systems are not able to achieve the one millisecond end-to-end latency requirement of the 5G standard on Commercial Off-The-Shelf (COTS) servers. This dissertation performs a comprehensive introduction to three innovative approaches that can be used to improve different aspects of the current software-driven network data plane. All three approaches are carefully designed, professionally implemented and rigorously evaluated. According to the measurement results, these novel approaches put forward the research in the design and implementation of ultra-reliable low-latency, energy-efficient and computing-first software data plane for 5G communication system and beyond

Angle-Resolved Observation of X-Ray Second Harmonic Generation in Diamond

This thesis reports angularly-resolved observation of X-ray second harmonic generation (XSHG) in diamond at several phase-matching geometries. The XSHG signal was produced by ultra-short, highly intense X-ray pulses with a photon energy of 9.831 keV generated by a free-electron laser. In some geometries for high pulse energies more than 10 second harmonic photons per pulse were generated. Different phase-matched geometries were used for XSHG to investigate the angular dependence of the efficiency of the process. Furthermore, for each phase-matching condition, the quadratic dependence for second harmonic generation at each geometry was verified and the crystal rocking curves were measured. The results for the angular dependence of the efficiency are in good agreement with the theory as shown by simulations (done by Priyanka Chakraborti). Improved interpretations of the data are enabled by a new analysis algorithm (developed within the framework of this thesis) that interprets the detected photon signal using a deeper understanding of the used 2D detector. Further an extensive analysis on established interpretation methods is presented. Adviser: Matthias Fuch

Characterization of a Single Photon Sensing and Photon Number Resolving CMOS Detector for Astrophysics

Next-generation NASA missions, such as the LUVIOR and HabEx concepts, require single photon counting large-format detectors. Charge Coupled Devices (CCDs) have typically been used for optical applications in similar flagship missions of the past. CCDs have excellent properties in most metrics but have their own challenges for single photon counting applications. First, typical CCDs have a read noise of a few electrons, although recent modifications (EMCCDs) use an on-chip gain to amplify the signal above the read noise. Secondly, the signal is carried by charge that is transferred across the detector array. While CCDs for NASA missions are carefully fabricated to minimize defects, continuous bombardment from high energy radiation in space will damage the detector over the lifetime of the mission. This will degrade the charge transfer efficiency and in turn, reduce the single photon counting ability of the CCD. CMOS devices offer a different architecture that mitigates some of these problems. In CMOS image sensors, each pixel has its own charge to voltage converter and in-pixel amplifier mitigating issues found with charge transfer efficiency. Additional circuits that are critical to operation of the sensor can be incorporated on chip allowing for a parallel readout architecture that increases frame rate and can decrease read noise. This thesis is a collection of work for the characterization of a room temperature characterization, low-noise, single photon counting and photon number resolving CMOS detector. The work performed in this thesis will provide the framework for a technology development project funded by NASA Cosmic Origins (COR) program office. At the end of the two-year project, a megapixel CMOS focal plane array will be demonstrated to satisfy the stated needs of the LUVOIR and HabEx future astrophysics space mission concepts with a launch date near the 2040s

JTP, an energy-aware transport protocol for mobile ad hoc networks (PhD thesis)

Wireless ad-hoc networks are based on a cooperative communication model, where all nodes not only generate traffic but also help to route traffic from other nodes to its final destination. In such an environment where there is no infrastructure support the lifetime of the network is tightly coupled with the lifetime of individual nodes. Most of the devices that form such networks are battery-operated, and thus it becomes important to conserve energy so as to maximize the lifetime of a node. In this thesis, we present JTP, a new energy-aware transport protocol, whose goal is to reduce power consumption without compromising delivery requirements of applications. JTP has been implemented within the JAVeLEN system. JAVeLEN [RKM+08], is a new system architecture for ad hoc networks that has been developed to elevate energy efficiency as a first-class optimization metric at all protocol layers, from physical to transport. Thus, energy gains obtained in one layer would not be offset by incompatibilities and/or inefficiencies in other layers. To meet its goal of energy efficiency, JTP (1) contains mechanisms to balance end-toend vs. local retransmissions; (2) minimizes acknowledgment traffic using receiver regulated rate-based flow control combined with selected acknowledgments and in-network caching of packets; and (3) aggressively seeks to avoid any congestion-based packet loss. Within this ultra low-power multi-hop wireless network system, simulations and experimental results demonstrate that our transport protocol meets its goal of preserving the energy efficiency of the underlying network. JTP has been implemented on the actual JAVeLEN nodes and its benefits have been demonstrated on a real system

Real-time data flow models and congestion management for wire and wireless IP networks

Includes abstract.Includes bibliographical references (leaves 103-111).In video streaming, network congestion compromises the video throughput performance and impairs its perceptual quality and may interrupt the display. Congestion control may take the form of rate adjustment through mechanisms by attempt to minimize the probability of congestion by adjusting the rate of the streaming video to match the available capacity of the network. This can be achieved either by adapting the quantization parameter of the video encoder or by varying the rate through a scalable video technique. This thesis proposes a congestion control protocol for streaming video where an interaction between the video source and the receiver is essential to monitor the network state. The protocol consists of adjusting the video transmission rate at the encoder whenever a change in the network conditions is observed and reported back to the sender