Quality-adaptive media streaming

Thesis (Master)--Izmir Institute of Technology, Computer Engineering, Izmir, 2012Includes bibliographical references (leaves: 37-40)Text in English; Abstract: Turkish and Englishxi, 40 leavesIn this study, an adaptive method for maximizing network bandwidth utilization for real-time media streaming applications is presented. The proposed method implements a rate control approach over the transport protocol RTP. RTP is coupled with an existing multimedia codec, H.264. A controller that keeps the RTP packet loss fraction at a predefined reference point is implemented. During the course of the stream transmission, the information about the network state is generated by the RTP/RTCP and sent to the server by the clients. Packet loss fraction parameter is fed into the controller. Controlling the multimedia codec bitrate directly affects the packet transmission rate, therefore RTP packet transmission rate is also controlled. Two control approaches are proposed. Firstly, a PID controller is introduced. This PID controller is designed without any self adaptation and manually tuned to maximize all of the available bandwidth. Secondly, a model reference adaptive controller (MRAC) is proposed. This MRAC controller constantly adjusts its parameters according to a reference model. The output of the TCP Friendly Rate Control Algorithm (TFRC) is used as the model to keep the MRAC controller friendly towards other flows flows at a level that the application requires. Simulations are provided to demonstrate the operation of the proposed methods. In the simulations, a content streaming scenario is run against background traffic for the available bandwidth in a bottleneck network configuration

Quality Of Service Enhancement In Ip Based Networks Using Diffserv

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2003Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2003Bu çalışmada, Diffserv mimarisi ile internet üzerinde servis kalitesi sağlama tartışılmıştır. Son on yılda IP tabanlı internette servis kalitesi sağlanamıyordu. İnternette bugün sağlanabilen tek servis ‘Best Effort (BE)’ adı verilen servistir. Yani, ağ, herhangi bir paketi hedefine ulaştırırken mümkün olan en verimli yolu kullanır ancak herhangi bir garanti ya da kaynak ayırımı yapmaz. Başka bir deyişle, trafik mümkün olduğu kadar hızlı ve herhangi bir zaman sınırı ya da miktar açısından garanti verilmeden işlenir. İnternetin ticari bir yapıya bürünmesiyle ‘Servis Kalitesi’ sağlanma ihtiyacı daha da artmıştır. Bu ihtiyaç farklı mimarilerin doğmasına yol açmıştır: IntServ ve DiffServ olmak üzere.Tez çalışmasında DiffServ mimarisi problem ve avantajlarıyla ele alınmıştır. Son bölümde ise NS2 ağ simulasyon yazılımı kullanılarak DiffServ implementasyonları yapılmış sonuçları karşılaştırmalı olarak verilmiştir.In this study improving Quality of Service (QoS) on the Internet with DiffServ architecture is discussed. Within the past decade, it is certainly not support for Quality of Service (QoS) over the IP-based ubiquitous Internet. The Internet as it stands today only support one service class called -Best-Effort (BE) Service. The network would make an earnest attempt to deliver packets to their destinations but with no guarantees and/or special resources allocated for any of the packets. With another words, traffic is processed as quickly as possible but there is no guarantee as to timeliness or actual delivery or even how much can be delivered (i.e. throughput). With the rapid transformation of the Internet into a commercial infrastructure, demands for Quality of Service (QoS) have rapidly developed. This need was resulted to different architectures: IntServ and DiffServ. In the study some DiffServ implementations are discussed with their problems and gains. At the last section with using NS2 simulation programming language some DiffServ implementations are given with the compared results.Yüksek LisansM.Sc

Call admission control for adaptive bit-rate VoIP over 802.11 WLAN.

Cui, Yuanyuan.Thesis (M.Phil.)--Chinese University of Hong Kong, 2009.Includes bibliographical references (p. 64-68).Abstract also in Chinese.Chapter Chapter 1 --- Introduction --- p.1Chapter 1 .1 --- Motivations and Contributions --- p.1Chapter 1.2 --- Related Works --- p.3Chapter 1.3 --- Organization of the Thesis --- p.4Chapter Chapter 2 --- Background --- p.5Chapter 2.1 --- IEEE 802.11 --- p.5Chapter 2.1.1 --- IEEE 802.11 Topologies --- p.5Chapter 2.1.2 --- IEEE 802.11 MAC --- p.8Chapter 2.2 --- Voice over Internet Protocol (VoIP) --- p.11Chapter 2.2.1 --- A VoIP system --- p.11Chapter 2.2.2 --- QoS requirements for VoIP --- p.11Chapter 2.2.3 --- VoIP speech codecs --- p.12Chapter 2.3 --- VoIP over WLAN --- p.13Chapter 2.3.1 --- System Architecture of VoIP over WLAN --- p.14Chapter 2.3.2 --- VoIP Capacity over WLAN --- p.15Chapter 2.4 --- Skype --- p.16Chapter Chapter 3 --- Skype Rate Adaptation Mechanism --- p.17Chapter 3.1 --- Experimental Setting --- p.17Chapter 3.2 --- Overview --- p.19Chapter 3.3 --- Flow Rate Region --- p.20Chapter 3.4 --- Feedback: Receiver Report (RR) --- p.21Chapter 3.5 --- Bandwidth Usage Target (BM) --- p.24Chapter 3.6 --- Summary of Skype Rate Adaptation Mechanism --- p.28Chapter 3.7 --- Skype-emulating Traffic Generator --- p.28Chapter Chapter 4 --- "Call Admission, Fairness and Stability Control" --- p.32Chapter 4.1 --- Unfair and Instability problems for AVoIP --- p.32Chapter 4.1.1 --- Analysis --- p.32Chapter 4.1.2 --- Simulation Evaluation --- p.34Chapter 4.2 --- CFSC scheme --- p.37Chapter 4.2.1 --- Pre-admission Bandwidth-reallocation Call Admission Control (PBCAC) --- p.39Chapter 4.2.2 --- Fairness Control --- p.42Chapter 4.2.3 --- Stability Control --- p.43Chapter Chapter 5 --- Performance Evaluation of CFSC --- p.44Chapter 5.1 --- Evaluation of Fairness Control --- p.44Chapter 5.2 --- Evaluation of Stability Control --- p.46Chapter 5.3 --- Evaluation of PBCAC --- p.46Chapter 5.4 --- Evaluation of complete CFSC --- p.49Chapter Chapter 6 --- Conclusion --- p.51Appendices --- p.53References --- p.6

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic

Enhancement of perceived quality of service for voice over internet protocol systems

Voice over Internet Protocol (WIP) applications are becoming more and more popular in the telecommunication market. Packet switched V61P systems have many technical advantages over conventional Public Switched Telephone Network (PSTN), including its efficient and flexible use of the bandwidth, lower cost and enhanced security. However, due to the IP network's "Best Effort" nature, voice quality are not naturally guaranteed in the VoIP services. In fact, most current Vol]P services can not provide as good a voice quality as PSTN. IP Network impairments such as packet loss, delay and jitter affect perceived speech quality as do application layer impairment factors, such as codec rate and audio features. Current perceived Quality of Service (QoS) methods are mainly designed to be used in a PSTN/TDM environment and their performance in V6IP environment is unknown. It is a challenge to measure perceived speech quality correctly in V61P system and to enhance user perceived speech quality for VoIP system. The main goal of this project is to evaluate the accuracy of the existing ITU-T speech quality measurement method (Perceptual Evaluation of Speech Quality - PESQ) in mobile wireless systems in the context of V61P, and to develop novel and efficient methods to enhance the user perceived speech quality for emerging V61P services especially in mobile V61P environment. The main contributions of the thesis are threefold: (1) A new discovery of PESQ errors in mobile VoIP environment. A detailed investigation of PESQ performance in mobile VoIP environment was undertaken and included setting up a PESQ performance evaluation platform and testing over 1800 mobile-to-mobile and mobileto- PSTN calls over a period of three months. The accuracy issues of PESQ algorithm was investigated and main problems causing inaccurate PESQ score (improper time-alignment in the PESQ algorithm) were discovered . Calibration issues for a safe and proper PESQ testing in mobile environment were also discussed in the thesis. (2) A new, simple-to-use, V611Pjit ter buffer algorithm. This was developed and implemented in a commercial mobile handset. The algorithm, called "Play Late Algorithm", adaptively alters the playout delay inside a speech talkspurt without introducing unnecessary extra end-to-end delay. It can be used as a front-end to conventional static or adaptive jitter buffer algorithms to provide improved performance. Results show that the proposed algorithm can increase user perceived quality without consuming too much processing power when tested in live wireless VbIP networks. (3) A new QoS enhancement scheme. The new scheme combines the strengths of adaptive codec bit rate (i. e. AMR 8-modes bit rate) and speech priority marking (i. e. giving high priority for the beginning of a voiced segment). The results gathered on a simulation and emulation test platform shows that the combined method provides a better user perceived speech quality than separate adaptive sender bit rate or packet priority marking methods

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

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic

Sync & Sense Enabled Adaptive Packetization VoIP

The quality and reliability problem of VoIP comes from the fact that VoIP relies on the network to transport the voice packets. The inherent problem of VoIP is that there is a mismatch between VoIP and the network. Namely, VoIP has a strict requirement of bandwidth, delay, and loss, but the network (particularly best-effort service networks) cannot guarantee such a requirement. A solution to deal with this problem is to enhance VoIP with an adaptive-rate control, called adaptive-rate VoIP. Adaptive-rate VoIP has the ability to detect the state of the network and adjust the transmission accordingly. Therefore, it gives VoIP the intelligence to optimize its performance, and making it resilient and robust to the service offered by the network. The objective of this dissertation is to develop an adaptive-rate VoIP system. We take a comprehensive approach in the study and development. Adaptive-rate VoIP is generally composed of three components: rate adaptation, network state detection, and adaptive-rate control. In the rate adaptation component, we study optimizing packetization, which can be used as an alternative means for rate adaptation. An advantage is that rate adaptation is independent of the speech coder. With this method, an adaptive-rate VoIP can be based on any constant bitrate speech coder. The study shows that the VoIP performance is primarily affected by three factors: packetization, network load, and significance of VoIP traffic; and, optimizing packetization allows us to ensure the highest possible performance. In the network state detection component, we propose a novel measurement methodology called Sync & Sense of periodic stream. Sync & Sense is unique in that it can virtually synchronize the transmission and reception timing of the VoIP session without requiring a synchronized clock. The simulation result shows that Sync & Sense can accurately measure one-way network delay. Other benefits of Sync & Sense include the ability to estimate the available network bandwidth and the full spectrum of the delays of the VoIP session. In the adaptive-rate control component, we consider the design choices and develop an adaptive-rate control that makes use of the first two components. The integration of the three components is a novel and unique adaptive-rate VoIP called Sync & Sense Enabled Adaptive Packetization VoIP. The simulation result shows that our adaptive VoIP can optimize the performance under any given network condition, and deliver a better performance than traditional VoIP. The simulation result also demonstrates that our adaptive VoIP possesses the desirable properties, which include fast response to network condition, aggressiveness to compete for the needed share of bandwidth, TCP-friendliness, and fair bandwidth allocation

A simulation study of adaptive voice communications on IP networks

This paper presents simulation results outlining the behavior of rate-adaptive voice communications over IP networks. In the considered architecture, voice coders adapt their rate to the current state of the network so as to generate only the bandwidth that the network is capable of carrying. An algorithm is proposed for driving the transmission rate of voice sources on the basis of estimations of the network conditions, measured in terms of packet delays and losses. The effectiveness of the proposed solution is then investigated in various scenarios which comprise: (i) a dedicated network in which the available bandwidth is exclusively shared between adaptive voice connections; (ii) a scenario in which adaptive voice sources compete with other TCP-like sources; and (iii) an uncontrolled network environment. We have compared the performance of the rate-adaptive against the non-adaptive (i.e. fixed-rate) approach for the transport of voice over IP. Using a rate-adaptive approach, more voice communications can be carried while maintaining a good quality of service, even on non-segregated networks