12 research outputs found
A Survey on Modeling of Human States in Communication Behavior
The Technical Committee on Communication BehaviorEngineering addresses the research question How do we construct a com-munication network system that includes users?. The growth in highlyfunctional networks and terminals has brought about greater diversity inusers\u27 lifestyles and freed people from the restrictions of time and place.Under this situation, the similarities of human behavior cause traffic aggre-gation and generate new problems in terms of the stabilization of networkservice quality. This paper summarizes previous studies relevant to com-munication behavior from a multidisciplinary perspective and discusses theresearch approach adopted by the Technical Committee on CommunicationBehavior Engineering
QoE-Based Low-Delay Live Streaming Using Throughput Predictions
Recently, HTTP-based adaptive streaming has become the de facto standard for
video streaming over the Internet. It allows clients to dynamically adapt media
characteristics to network conditions in order to ensure a high quality of
experience, that is, minimize playback interruptions, while maximizing video
quality at a reasonable level of quality changes. In the case of live
streaming, this task becomes particularly challenging due to the latency
constraints. The challenge further increases if a client uses a wireless
network, where the throughput is subject to considerable fluctuations.
Consequently, live streams often exhibit latencies of up to 30 seconds. In the
present work, we introduce an adaptation algorithm for HTTP-based live
streaming called LOLYPOP (Low-Latency Prediction-Based Adaptation) that is
designed to operate with a transport latency of few seconds. To reach this
goal, LOLYPOP leverages TCP throughput predictions on multiple time scales,
from 1 to 10 seconds, along with an estimate of the prediction error
distribution. In addition to satisfying the latency constraint, the algorithm
heuristically maximizes the quality of experience by maximizing the average
video quality as a function of the number of skipped segments and quality
transitions. In order to select an efficient prediction method, we studied the
performance of several time series prediction methods in IEEE 802.11 wireless
access networks. We evaluated LOLYPOP under a large set of experimental
conditions limiting the transport latency to 3 seconds, against a
state-of-the-art adaptation algorithm from the literature, called FESTIVE. We
observed that the average video quality is by up to a factor of 3 higher than
with FESTIVE. We also observed that LOLYPOP is able to reach a broader region
in the quality of experience space, and thus it is better adjustable to the
user profile or service provider requirements.Comment: Technical Report TKN-16-001, Telecommunication Networks Group,
Technische Universitaet Berlin. This TR updated TR TKN-15-00
A layered multicast packet video system
Due to the character of the original source materials and the nature of batch digitization, quality control issues may be present in this document. Please report any quality issues you encounter to [email protected], referencing the URI of the item.Includes bibliographical references.Issued also on microfiche from Lange Micrographics.Software based desktop videoconferencing tools are developed to demonstrate techniques necessary for video delivery in heterogeneous packet networks. Using the current network infrastructure and no network resource reservation, a one-to-many implementation is designed around a two-layer pyramidal video coder. During periods of congestion, the network routers give priority to the base layer, which by itself allows reconstruction of reasonable quality video. Receiver feedback is used to lower the output rate of the encoder's low priority pyramidal layer when all receivers are suffering high packet loss. Each of the two layers is transmitted on a separate multicast channel. Under persistent congestion, an individual receiver will discard the low priority pyramidal layer, which allows the network to prune the multicast tree 'd congestion. A new scheme is examined where if the other receivers back and avoi are agreeable, the source will respond to a receiver pruning its pyramidal layer by lowering its rate and allowing the receiver to quickly rejoin the pyramidal layer at a quality level higher than what the high priority base layer can provide by itself. Another new scheme is described where an agent on the receiver's local router provides spare capacity information to assist the receiver in its decision to rejoin the pyramidal layer
Rate-adaptive H.264 for TCP/IP networks
While there has always been a tremendous demand for streaming video over
TCP/IP networks, the nature of the application still presents some challenging issues.
These applications that transmit multimedia data over best-effort networks like the
Internet must cope with the changing network behavior; specifically, the source encoder
rate should be controlled based on feedback from a channel estimator that probes the
network periodically. First, one such Multimedia Streaming TCP-Friendly Protocol
(MSTFP) is considered, which iteratively integrates forward estimation of network status
with feedback control to closely track the varying network characteristics. Second, a
network-adaptive embedded bit stream is generated using a r-domain rate controller.
The conceptual elegance of this r-domain framework stems from the fact that the
coding bit rate ) (R is approximately linear in the percentage of zeros among the
quantized spatial transform coefficients ) ( r , as opposed to the more traditional, complex
and highly nonlinear ) ( Q R characterization. Though the r-model has been
successfully implemented on a few other video codecs, its application to the emerging
video coding standard H.264 is considered. The extensive experimental results show thatrobust rate control, similar or improved Peak Signal to Noise Ratio (PSNR), and a faster
implementation
Wavelet-based adaptive video coding for packet-switching networks
Thesis (M.S.)--Massachusetts Institute of Technology, Dept. of Electrical Engineering and Computer Science, 1995.Includes bibliographical references (leaves [92]-97).by Ye Gu.M.S
Real-time compressed video transmission across the common data link
The advances in high speed computer networks and digital communication techniques have enabled the rapid and extensive dissemination of information throughout the modern defense infrastructure. One of the challenges in networking today is real-time dissemination of information. This thesis proposes a solution for a specific aspect of this challenge, namely, the transmission of real-time compressed video data across the Common Data Link (CDL), a high-speed military data link designed to operate in high error environments. Current research is primarily focused on the transmission of real- time data across low-error links. This thesis proposes, simulates, and analyzes a mechanism which guarantees that (1) delay bounds are met for real-time flows despite network overload and (2) a minimum acceptable image quality is maintained despite the presence of highly correlated errors. These highly correlated errors are characteristic of the type of electromagnetic jamming likely to be encountered by the CDL. This mechanism consists of four fundamental requirements: (1) a hierarchical image compression scheme, (2) rate control at the source, (3) bandwidth allocation within all encountered network nodes, and (4) dynamic forward error correction. The proposed solution is modeled in the OPNET simulation environment, and the validity and feasibility of the mechanism are verified. In addition, the simulation is interfaced with a compression/decompression algorithm running in MATLAB to enable the subjective analysis of actual images before and after transmission in various jamming scenarios. The results demonstrate the effectiveness of the proposed solution in meeting delay guarantees and maintaining image quality.http://archive.org/details/realtimecompress109457522U.S. Navy (U.S.N.) author