Quality of service differentiation for multimedia delivery in wireless LANs

Delivering multimedia content to heterogeneous devices over a variable networking environment while maintaining high quality levels involves many technical challenges. The research reported in this thesis presents a solution for Quality of Service (QoS)-based service differentiation when delivering multimedia content over the wireless LANs. This thesis has three major contributions outlined below: 1. A Model-based Bandwidth Estimation algorithm (MBE), which estimates the available bandwidth based on novel TCP and UDP throughput models over IEEE 802.11 WLANs. MBE has been modelled, implemented, and tested through simulations and real life testing. In comparison with other bandwidth estimation techniques, MBE shows better performance in terms of error rate, overhead, and loss. 2. An intelligent Prioritized Adaptive Scheme (iPAS), which provides QoS service differentiation for multimedia delivery in wireless networks. iPAS assigns dynamic priorities to various streams and determines their bandwidth share by employing a probabilistic approach-which makes use of stereotypes. The total bandwidth to be allocated is estimated using MBE. The priority level of individual stream is variable and dependent on stream-related characteristics and delivery QoS parameters. iPAS can be deployed seamlessly over the original IEEE 802.11 protocols and can be included in the IEEE 802.21 framework in order to optimize the control signal communication. iPAS has been modelled, implemented, and evaluated via simulations. The results demonstrate that iPAS achieves better performance than the equal channel access mechanism over IEEE 802.11 DCF and a service differentiation scheme on top of IEEE 802.11e EDCA, in terms of fairness, throughput, delay, loss, and estimated PSNR. Additionally, both objective and subjective video quality assessment have been performed using a prototype system. 3. A QoS-based Downlink/Uplink Fairness Scheme, which uses the stereotypes-based structure to balance the QoS parameters (i.e. throughput, delay, and loss) between downlink and uplink VoIP traffic. The proposed scheme has been modelled and tested through simulations. The results show that, in comparison with other downlink/uplink fairness-oriented solutions, the proposed scheme performs better in terms of VoIP capacity and fairness level between downlink and uplink traffic

Reducing Latency in Internet Access Links with Mechanisms in Endpoints and within the Network

Excessive and unpredictable end-to-end latency is a major problem for today’s Internet performance, affecting a range of applications from real-time multimedia to web traffic. This is mainly attributed to the interaction between the TCP congestion control mechanism and the unmanaged large buffers deployed across the Internet. This dissertation investigates transport and link layer solutions to solve the Internet’s latency problem on the access links. These solutions operate on the sender side, within the network or use signaling between the sender and the network based on Explicit Congestion Notification (ECN). By changing the sender’s reaction to ECN, a method proposed in this dissertation reduces latency without harming link utilization. Real-life experiments and simulations show that this goal is achieved while maintaining backward compatibility and being gradually deployable on the Internet. This mechanism’s fairness to legacy traffic is further improved by a novel use of ECN within the network

Dual-Mode Congestion Control Mechanism for Video Services

Recent studies have shown that video services represent over half of Internet traffic, with a growing trend. Therefore, video traffic plays a major role in network congestion. Currently on the Internet, congestion control is mainly implemented through overprovisioning and TCP congestion control. Although some video services use TCP to implement their transport services in a manner that actually works, TCP is not an ideal protocol for use by all video applications. For example, UDP is often considered to be more suitable for use by real-time video applications. Unfortunately, UDP does not implement congestion control. Therefore, these UDP-based video services operate without any kind of congestion control support unless congestion control is implemented on the application layer. There are also arguments against massive overprovisioning. Due to these factors, there is still a need to equip video services with proper congestion control.Most of the congestion control mechanisms developed for the use of video services can only offer either low priority or TCP-friendly real-time services. There is no single congestion control mechanism currently that is suitable and can be widely used for all kinds of video services. This thesis provides a study in which a new dual-mode congestion control mechanism is proposed. This mechanism can offer congestion control services for both service types. The mechanism includes two modes, a backward-loading mode and a real-time mode. The backward-loading mode works like a low-priority service where the bandwidth is given away to other connections once the load level of a network is high enough. In contrast, the real-time mode always demands its fair share of the bandwidth.The behavior of the new mechanism and its friendliness toward itself, and the TCP protocol, have been investigated by means of simulations and real network tests. It was found that this kind of congestion control approach could be suitable for video services. The new mechanism worked acceptably. In particular, the mechanism behaved toward itself in a very friendly way in most cases. The averaged TCP fairness was at a good level. In the worst cases, the faster connections received about 1.6 times as much bandwidth as the slower connections

Control of transport dynamics in overlay networks

Transport control is an important factor in the performance of Internet protocols, particularly in the next generation network applications involving computational steering, interactive visualization, instrument control, and transfer of large data sets. The widely deployed Transport Control Protocol is inadequate for these tasks due to its performance drawbacks. The purpose of this dissertation is to conduct a rigorous analytical study on the design and performance of transport protocols, and systematically develop a new class of protocols to overcome the limitations of current methods. Various sources of randomness exist in network performance measurements due to the stochastic nature of network traffic. We propose a new class of transport protocols that explicitly accounts for the randomness based on dynamic stochastic approximation methods. These protocols use congestion window and idle time to dynamically control the source rate to achieve transport objectives. We conduct statistical analyses to determine the main effects of these two control parameters and their interaction effects. The application of stochastic approximation methods enables us to show the analytical stability of the transport protocols and avoid pre-selecting the flow and congestion control parameters. These new protocols are successfully applied to transport control for both goodput stabilization and maximization. The experimental results show the superior performance compared to current methods particularly for Internet applications. To effectively deploy these protocols over the Internet, we develop an overlay network, which resides at the application level to provide data transmission service using User Datagram Protocol. The overlay network, together with the new protocols based on User Datagram Protocol, provides an effective environment for implementing transport control using application-level modules. We also study problems in overlay networks such as path bandwidth estimation and multiple quickest path computation. In wireless networks, most packet losses are caused by physical signal losses and do not necessarily indicate network congestion. Furthermore, the physical link connectivity in ad-hoc networks deployed in unstructured areas is unpredictable. We develop the Connectivity-Through-Time protocols that exploit the node movements to deliver data under dynamic connectivity. We integrate this protocol into overlay networks and present experimental results using network to support a team of mobile robots

Adaptive multimedia streaming control algorithm in wireless LANs and 4G networks

E-learning has become an important service offered over the Internet. Lately many users are accessing learning content via wireless networks and using mobile devices. Most content is rich media-based and often puts significant pressure on the existing wireless networks in order to support high quality of delivery. In this context, offering a solution for improving user quality of experience when multimedia content is delivered over wireless networks is already a challenging task. Additionally, to support this for mobile e-learning over wireless LANs becomes even more difficult. If we want to increase the end-used perceived quality, we have to take into account the users’ individual set of characteristics. The fact that users have subjective opinions on the quality of a multimedia application can be used to increase their QoE by setting a minimum quality threshold below which the connection is considered to be undesired. Like this, the use of precious radio resources can be optimized in order to simultaneously satisfy an increased number of users. In this thesis a new user-oriented adaptive algorithm based on QOAS was designed and developed in order to address the user satisfaction problem. Simulations have been carried out with different adaptation schemes to compare the performances and benefits of the DQOAS mechanism. The simulation results are showing that using a dynamic stream granularity with a minimum threshold for the transmission rate, improves the overall quality of the multimedia delivery process, increasing the total number of satisfied users and the link utilization The good results obtained by the algorithm in IEEE 802.11 wireless environment, motivated the research about the utility of the newly proposed algorithm in another wireless environment, LTE. The study shows that DQOAS algorithm can obtain good results in terms of application perceived quality, when the considered application generates multiple streams. These results can be improved by using a new QoS parameters mapping scheme able to modify the streams’ priority and thus allowing the algorithms decisions to not be overridden by the systems’ scheduler

Adaptive Bitrate Streaming in Cloud Gaming

Cloud gaming streams games as video from a server to a client device making it susceptible to network congestion. Adaptive bitrate streaming estimates network capacity and sets encoding parameters to avoid exceeding the bandwidth of the connection. BBR is a congestion control algorithm as an alternative to current loss-based congestion control. We designed and implemented a bitrate adaptation heuristic based on BBR into GamingAnywhere, an open source cloud gaming platform. We conducted a user study and did objective analysis comparing our modified version to the original. Through our results, we found that our adaptive system was less challenging for players and improved retention rates and that there was no statistically significant difference in visual quality from objective testing

Best effort measurement based congestion control

Abstract available: p.

Modeling and Evaluating Feedback-Based Error Control for Video Transfer

Packet loss can be detrimental to real-time interactive video over lossy networks because one lost video packet can propagate errors to many subsequent video frames due to the encoding dependency between frames. Feedback-based error control techniques use feedback information from the decoder to adjust coding parameters at the encoder or retransmit lost packets to reduce the error propagation due to data loss. Feedback-based error control techniques have been shown to be more effective than trying to conceal the error at the encoder or decoder alone since they allow the encoder and decoder to cooperate in the error control process. However, there has been no systematic exploration of the impact of video content and network conditions on the performance of feedback-based error control techniques. In particular, the impact of packet loss, round-trip delay, network capacity constraint, video motion and reference distance on the quality of videos using feedback-based error control techniques have not been systematically studied. This thesis presents analytical models for the major feedback-based error control techniques: Retransmission, Reference Picture Selection (both NACK and ACK modes) and Intra Update. These feedback-based error control techniques have been included in H.263/H.264 and MPEG4, the state of the art video in compression standards. Given a round-trip time, packet loss rate, network capacity constraint, our models can predict the quality for a streaming video with retransmission, Intra Update and RPS over a lossy network. In order to exploit our analytical models, a series of studies has been conducted to explore the effect of reference distance, capacity constraint and Intra coding on video quality. The accuracy of our analytical models in predicting the video quality under different network conditions is validated through simulations. These models are used to examine the behavior of feedback-based error control schemes under a variety of network conditions and video content through a series of analytic experiments. Analysis shows that the performance of feedback-based error control techniques is affected by a variety of factors including round-trip time, loss rate, video content and the Group of Pictures (GOP) length. In particular: 1) RPS NACK achieves the best performance when loss rate is low while RPS ACK outperforms other repair techniques when loss rate is high. However RPS ACK performs the worst when loss rate is low. Retransmission performs the worst when the loss rate is high; 2) for a given round-trip time, the loss rate where RPS NACK performs worse than RPS ACK is higher for low motion videos than it is for high motion videos; 3) Videos with RPS NACK always perform the same or better than videos without repair. However, when small GOP sizes are used, videos without repair perform better than videos with RPS ACK; 4) RPS NACK outperform Intra Update for low-motion videos. However, the performance gap between RPS NACK and Intra Update drops when the round-trip time or the intensity of video motion increases. 5) Although the above trends hold for both VQM and PSNR, when VQM is the video quality metric the performance results are much more sensitive to network loss. 6) Retransmission is effective only when the round-trip time is low. When the round-trip time is high, Partial Retransmission achieves almost the same performance as Full Retransmission. These insights derived from our models can help determine appropriate choices for feedback-based error control techniques under various network conditions and video content

Evaluation and optimisation of Less-than-Best-Effort TCP congestion control mechanisms

Increasing use of online software installation, updates, and backup services, as well as the popularity of user-generated content, has increased the demand for band-width in recent years. Traffic generated by these applications — when receiving a ‘fair-share’ of the available bandwidth — can impact the responsiveness of delay-sensitive applications. Less-than-Best-Effort TCP congestion control mechanisms aim to allow lower-priority applications to utilise excess bandwidth with minimum impact to regular TCP carrying delay-sensitive traffic. However, no previous study has evaluated the performance of a large number of this class of congestion con-trol mechanisms. This thesis quantifies the performance of existing Less-than-Best-Effort TCP congestion control mechanisms, and proposes a new mechanism to im-prove the performance of these mechanisms with high path delay. This study first evaluated the performance of seven Less-than-Best-Effort conges-tion control mechanisms in realistic scenarios under a range of network conditions in a Linux testbed incorporating wired Ethernet and 802.11n wireless links. The seven mechanisms evaluated were: Apple LEDBAT, CAIA Delay-Gradient (CDG), RFC6817 LEDBAT, Low Priority, Nice, Westwood-LP, and Vegas. Of these mecha-nisms, only four had existing implementations for modern operating systems. The remaining three mechanisms — Apple LEDBAT, Nice, and Westwood-LP — were implemented based on published descriptions and available code fragments to fa-cilitate this evaluation. The results of the evaluation suggest that Less-than-Best-Effort congestion control mechanisms can be divided into two categories: regular TCP-like mechanisms, and low-impact mechanisms. Of the low-impact mechanisms, two mechanisms were identified as having desirable performance characteristics: Nice and CDG. Nice pro-vides background throughput comparable to regular TCP while maintaining low queuing delay in low path delay settings. CDG has the least impact on regular TCP traffic, at the expense of reduced throughput. In high path-delay settings, these reductions to throughput experienced by CDG are exacerbated, while Nice has a greater impact on regular TCP traffic. To address the very low throughput of existing Less-than-Best-Effort congestion control mechanisms in high path-delay settings, a new Less-than-Best-Effort TCP congestion control algorithm was developed and implemented: Yield TCP. Yield utilises elements of a Proportional-Integral controller to better interpret and re-spond to changes in queuing delay to achieve this goal while also reducing the impact on regular TCP traffic over TCP-like mechanisms. Source code for the im-plementation of Yield developed for this research has also been made available. The results of evaluating Yield indicate that it successfully addresses the low through-put of low-impact Less-than-Best-Effort mechanisms in high delay settings, while also reducing the impact on foreground traffic compared to regular TCP-like con-gestion control mechanisms. Yield also performs similarly to Nice in low delay settings, while also achieving greater intra-protocol fairness than Nice across all settings. These results indicate that Yield addresses the weaknesses of Nice and CDG, and is a promising alternative to existing Less-than-Best-Effort congestion control algorithms