On aggregate available bandwidth in many-to-one data transfer.

Hui Shui Cheung.Thesis submitted in: August 2005.Thesis (M.Phil.)--Chinese University of Hong Kong, 2006.Includes bibliographical references (leaves 37-38).Abstracts in English and Chinese.Acknowledgement --- p.iAbstract --- p.ii摘要 --- p.iiiChapter Chapter 1 --- Introduction --- p.1Chapter Chapter 2 --- Related Work --- p.4Chapter Chapter 3 --- Single-Source Bandwidth Availability --- p.6Chapter 3.1 --- Measurement Methodology --- p.6Chapter 3.2 --- Measurement Results --- p.7Chapter Chapter 4 --- Multi-Souce Bandwidth Availability --- p.9Chapter 4.1 --- Correlation Among Senders --- p.9Chapter 4.2 --- Aggregate Bandwidth --- p.10Chapter 4.3 --- Sensitivity Analysis --- p.11Chapter Chapter 5 --- The Measurement System --- p.15Chapter 5.1 --- Overview of PlanetLab --- p.15Chapter 5.2 --- Measurement Tool --- p.16Chapter 5.3 --- Process Control --- p.17Chapter Chapter 6 --- Hybrid-Download Streaming --- p.21Chapter 6.1 --- Introduction --- p.21Chapter 6.2 --- Streaming Algorithm --- p.22Chapter 6.3 --- Performance Evaluation --- p.23Chapter Chapter 7 --- Playback-Adaptive Streaming --- p.26Chapter 7.1 --- Introduction --- p.26Chapter 7.2 --- Streaming Algorithm --- p.27Chapter 7.3 --- Adaptive Rebuffering Algorithm --- p.30Chapter 7.4 --- Performance Evaluation --- p.31Chapter Chapter 8 --- Conclusion --- p.36Bibliography --- p.3

Measurement and application of many-to-one data flows.

Ho, Po Yee.Thesis (M.Phil.)--Chinese University of Hong Kong, 2007.Includes bibliographical references (leaves 77-81).Abstracts in English and Chinese.Acknowledgements --- p.iAbstract --- p.ii摘要 --- p.iiiChapter Chapter 1 --- Introduction --- p.1Chapter Chapter 2 --- Background and Related Work --- p.4Chapter 2.1 --- Link/Path Capacity --- p.4Chapter 2.2 --- Unutilized Bandwidth --- p.5Chapter 2.3 --- Achievable Bandwidth --- p.5Chapter Chapter 3 --- Measurement Methodology --- p.7Chapter 3.1 --- PlanetLab Measurement --- p.8Chapter 3.2 --- FTP Measurement --- p.10Chapter Chapter 4 --- Analysis of Measurement Data --- p.12Chapter 4.1 --- Per-Flow Achievable Bandwidth --- p.13Chapter 4.2 --- Inter-Flow Correlation --- p.14Chapter 4.3 --- Intra-Flow Temporal Correlation --- p.16Chapter 4.4 --- Intra-Flow Bandwidth Variation --- p.18Chapter 4.5 --- Predictability of Bandwidth Properties --- p.22Chapter 4.6 --- Long-term Flow Properties --- p.26Chapter Chapter 5 --- A Mathematical Framework --- p.28Chapter 5.1 --- Bandwidth Variations --- p.28Chapter 5.2 --- Bandwidth Predictability --- p.31Chapter 5.3 --- Sensitivity Analysis --- p.34Chapter Chapter 6 --- Predictive Buffering Algorithm --- p.41Chapter 6.1 --- Related Work --- p.43Chapter 6.2 --- System Model --- p.44Chapter 6.3 --- Prediction Algorithm for Constant Bit-Rate Videos --- p.45Chapter 6.4 --- Prediction Algorithm for Variable Bit-Rate Videos --- p.46Chapter 6.5 --- Parameter Estimation --- p.47Chapter Chapter 7 --- Performance Evaluation --- p.49Chapter 7.1 --- Trace-Driven Simulation Setup --- p.49Chapter 7.2 --- Performance over CBR Videos --- p.50Chapter 7.2.1 --- Video Playback Performance --- p.51Chapter 7.2.2 --- Buffering Time --- p.57Chapter 7.3 --- Performance over VBR Videos --- p.61Chapter 7.3.1 --- Video Playback Performance --- p.62Chapter 7.3.2 --- Buffering Time --- p.66Chapter Chapter 8 --- Future Work --- p.69Chapter 8.1 --- Playback Rate Adaptation --- p.70Chapter 8.2 --- Sender Selection Algorithm --- p.71Chapter 8.3 --- Dynamic Flow Allocation --- p.72Chapter 8.4 --- Predictive Flow Allocation --- p.73Chapter 8.5 --- Challenge in P2P Applications --- p.74Chapter Chapter 9 --- Conclusion --- p.76Bibliograph

Adaptive Playout Scheduling Using Time-Scale Modification In Packet Voice Communications

A new receiver-based playout scheduling scheme is proposed, which estimates the network delay from past statistics and adaptively adjusts the playout time of the voice packets. In contrast to previous work, the adjustment is not only performed in between talkspurts, but also within the talkspurts in a highly dynamic way. Proper reconstruction of continuous output speech is achieved by scaling individual voice packets using a time-scale modification technique which modifies the rate of playout while preserving voice pitch. Subjective listening tests show that this operation does not impair audio quality. Simulation results based on Internet measurement indicate that buffering delay and loss rate can be significantly reduced by adaptive scheduling

Evaluación de algoritmos de control de retardo en voz sobre internet

Este proyecto de tesis de maestría trata acerca de los algoritmos de control de retardo en VoIP. Las aplicaciones interactivas sobre Internet se utilizan extensamente en nuestros días. Aplicaciones P2P como Skype, VoIPBuster han incorporado exitosamente la VoIP. Un algoritmo de control de playout implementa un buffer en el lado del receptor, así guarda los paquetes recibidos. Entonces, el algoritmo calcula un tiempo límite para cada paquete. Si el paquete se perdió en la red o llegó después de su tiempo límite esperado, el paquete se considera perdido en el receptor. Un algoritmo de playout considera el compro-miso entre las pérdidas y el retardo de tal forma que optimice la interactividad de la sesión de VoIP. Nos enfocamos en la clase de algoritmos que actualizan el retardo de playout al ini-cio de cada frase. Estudiamos un algoritmo NLMS originálmente propuesto por DeLeon y posteriormente modificado por Shallwani para probarlos extensivamente bajo las mismas condiciones de trabajo. Los resultados que obtenemos indican por un lado que el algoritmo de Shallwani puede tener errores de depuración, y por el otro lado, que la detección de picos de retardo puede mejorarse. Así, decidimos mejorar la detección de picos de retardo que propone Shallwani y comparar el desempeño de nuestro algoritmo con los algoritmos de DeLeon y de Shallwani. Encontramos, que para la mayoría de los casos, usando trazas reales de audio, que han sido muy utilizadas en otros trabajos, nuestro algoritmo se desempeña mejor

Excitação multi-taxa usando quantização vetorial estruturada em árvore para o codificador CS-ACELP com aplicação em VoIP

Dissertação (mestrado) - Universidade Federal de Santa Catarina, Centro Tecnológico. Programa de Pós-Graduação em Engenharia Elétrica.Este trabalho apresenta um estudo sobre codificação multi-taxa estruturada sobre o algoritmo CS-ACELP (Conjugate-Structure Algebraic-Code-Excited Linear-Prediction) e a especificação G.729, cujo objetivo é propor um codificador com taxa variável, através da busca da melhor excitação fixa usando codebook estruturado em árvore, para aplicações VoIP (Voice-over-IP). A mudança progressiva do transporte de voz das redes de circuito para as redes IP (Internet Protocol), apesar dos diversos aspectos positivos, tem exposto algumas deficiências intrínsecas destas, mais apropriadas ao tráfego de #melhor esforço# do que ao tráfego com requisitos de tempo. Esta proposta está inserida no conjunto das iniciativas, no âmbito do transmissor, que procuram minimizar os efeitos danosos da rede sobre a qualidade da voz reconstruída. O codebook proposto tem estrutura em árvore binária, concebida a partir de uma heurística onde os vetores CS-ACELP são ordenados por valor de forma decrescente. Uma estratégia particular de armazenamento dos nós, envolvendo simplificação nos centróides, codificação diferencial e geração automática dos dois últimos níveis da árvore, permite reduzir o espaço de armazenamento de 640 para apenas 7 kwords. Através deste modelo chega-se a 13 taxas de codificação, de 5,6 a 8,0 kbit/s, com passo de 0,2 kbit/s. A relação sinal ruído fica em 1,5 dB abaixo da mesma medida na especificação G.729 para a taxa de 5,6 kbit/s, e apenas 0,6 dB abaixo quando na taxa 8,0 kbit/s. Testes subjetivos mostraram uma qualidade bastante aceitável para a taxa mínima e praticamente indistinguível do codec original na taxa máxima. Além disso, a busca da melhor excitação é 2,4 vezes mais rápida em comparação ao codec G.729 e pode ser totalmente compatível com este se a taxa for fixa em 8,0 kbit/s. This work presents a study about multi-rate coding structured over CS-ACELP (Conjugate-Structure Algebraic-Code-Excited Linear-Prediction) algorithm and G.729 standard, whose purpose is to come up with a variable rate codec by means of best fixed excitation search using a tree structured codebook, for VoIP (Voice-over-IP) applications. The progressive change of voice transmission from circuit switched to IP (Internet orks, besides its many positive aspects, has exposed some natural deficiencies of the latter, better suited to best effort traffics than traffics with time requirements. This proposition can be inserted in the bunch of efforts, related to the sender, that seek to reduce the network impairments over the quality of reconstructed voice. The suggested codebook has a binary tree structure heuristically conceived where algebraic CSACELP vectors are disposed by value in a decreasing order. Additionally, a particular approach to store the tree nodes are considered, which involves centroid implification, differential coding and automatic generation of the last two layers of the tree, squeezing the storing space from 640 down to 7 kwords. Through this model we reach 13 coding rates, ranging from 5.6 to 8.0 kbit/s, with 0.2 kbit/s step. The signal-to-noise ratio is 1.5 dB below the same measure for G.729 standard at the rate 5.6 kbit/s, and just 0.6 dB lower at 8.0 kbit/s. Subjective tests pointed to an acceptable quality at minimum rate and virtually indistinguishable quality from the original codec at the maximum one. Also, searching for the best fixed excitation is 2.4 times faster than G.729 and can be truly compatible with it if the rate is fixed in 8 kbit/s