ADAPTIVE SPEECH QUALITY IN VOICE-OVER-IP COMMUNICATIONS

The quality of VoIP communication relies significantly on the network that transports the voice packets because this network does not usually guarantee the available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage the voice-over-IP stream dynamically, changing parameters as needed to assure quality. The main objective of this dissertation is to develop an adaptive speech encoding system that can be applied to conventional (telephony-grade) and wideband voice communications. This comprehensive study includes the investigation and development of three key components of the system. First, to manage VoIP quality dynamically, a tool is needed to measure real-time changes in quality. The E-model, which exists for narrowband communication, is extended to a single computational technique that measures speech quality for narrowband and wideband VoIP codecs. This part of the dissertation also develops important theoretical work in the area of wideband telephony. The second system component is a variable speech-encoding algorithm. Although VoIP performance is affected by multiple codecs and network-based factors, only three factors can be managed dynamically: voice payload size, speech compression and jitter buffer management. Using an existing adaptive jitter-buffer algorithm, voice packet-size and compression variation are studied as they affect speech quality under different network conditions. This study explains the relationships among multiple parameters as they affect speech transmission and its resulting quality. Then, based on these two components, the third system component is a novel adaptive-rate control algorithm that establishes the interaction between a VoIP sender and receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average voice quality than traditional VoIP

Priority-Based Resource Allocation for Downlink OFDMA Systems Supporting RT and NRT Traffics

Efficient radio resource management is essential in Quality-of-Service (QoS) provisioning for wireless communication networks. In this paper, we propose a novel priority-based packet scheduling algorithm for downlink OFDMA systems. The proposed algorithm is designed to support heterogeneous applications consisting of both real-time (RT) and non-real-time (NRT) traffics with the objective to increase the spectrum efficiency while satisfying diverse QoS requirements. It tightly couples the subchannel allocation and packet scheduling together through an integrated cross-layer approach in which each packet is assigned a priority value based on both the instantaneous channel conditions as well as the QoS constraints. An efficient suboptimal heuristic algorithm is proposed to reduce the computational complexity with marginal performance degradation compared to the optimal solution. Simulation results show that the proposed algorithm can significantly improve the system performance in terms of high spectral efficiency and low outage probability compared to conventional packet scheduling algorithms, thus is very suitable for the downlink of current OFDMA systems

Cross-Layer Capacity Optimization In OFDMA Systems: WiMAX And LTE

Given the broad range of applications supported, high data rate required and low latency promised; dynamic radio resource management is becoming vital for newly emerging air interface technologies such as wireless interoperability for microwave access (Wimax) and long term evolution (lte) adopted by international standards. This thesis considers orthogonal frequency division multiple access (ofdma) system, which has been implemented in both Wimax and lte technologies as their air interface multiple access mechanism. A framework for optimized resource allocation with quality of service (qos) support that aims to balance between service provider\u27s revenue and subscriber\u27s satisfaction is proposed. A cross-layer optimization design for subchannel, for Wimax, and physical resource block (prb), for lte, and power allocations with the objective of maximizing the capacity (in bits/symbol/hz) subject to fairness parameters and qos requirements as constraints is presented. An adaptive modulation and coding (amc)-based cross-layer scheme has also been proposed in this thesis by adopting an amc scheme together with the cross-layer scheme to realize a more practical and viable resource allocation. The optimization does not only consider users channel conditions but also queue status of each user as well as different qos requirements. In the proposed framework, the problem of power allocation is solved analytically while the subchannel/prb allocation is solved using integer programming exhaustive search. The simulation and numerical results obtained in this thesis have shown improved system performance as compared to other optimization schemes known in literature

Networking Mechanisms for Delay-Sensitive Applications

The diversity of applications served by the explosively growing Internet is increasing. In particular, applications that are sensitive to end-to-end packet delays become more common and include telephony, video conferencing, and networked games. While the single best-effort service of the current Internet favors throughput-greedy traffic by equipping congested links with large buffers, long queuing at the congested links hurts the delay-sensitive applications. Furthermore, while numerous alternative architectures have been proposed to offer diverse network services, the innovative alternatives failed to gain widespread end-to-end deployment. This dissertation explores different networking mechanisms for supporting low queueing delay required by delay-sensitive applications. In particular, it considers two different approaches. The first one assumes employing congestion control protocols for the traffic generated by the considered class of applications. The second approach relies on the router operation only and does not require support from end hosts

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

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

Adaptive scheduling in cellular access, wireless mesh and IP networks

Networking scenarios in the future will be complex and will include fixed networks and hybrid Fourth Generation (4G) networks, consisting of both infrastructure-based and infrastructureless, wireless parts. In such scenarios, adaptive provisioning and management of network resources becomes of critical importance. Adaptive mechanisms are desirable since they enable a self-configurable network that is able to adjust itself to varying traffic and channel conditions. The operation of adaptive mechanisms is heavily based on measurements. The aim of this thesis is to investigate how measurement based, adaptive packet scheduling algorithms can be utilized in different networking environments. The first part of this thesis is a proposal for a new delay-based scheduling algorithm, known as Delay-Bounded Hybrid Proportional Delay (DBHPD), for delay adaptive provisioning in DiffServ-based fixed IP networks. This DBHPD algorithm is thoroughly evaluated by ns2-simulations and measurements in a FreeBSD prototype router network. It is shown that DBHPD results in considerably more controllable differentiation than basic static bandwidth sharing algorithms. The prototype router measurements also prove that a DBHPD algorithm can be easily implemented in practice, causing less processing overheads than a well known CBQ algorithm. The second part of this thesis discusses specific scheduling requirements set by hybrid 4G networking scenarios. Firstly, methods for joint scheduling and transmit beamforming in 3.9G or 4G networks are described and quantitatively analyzed using statistical methods. The analysis reveals that the combined gain of channel-adaptive scheduling and transmit beamforming is substantial and that an On-off strategy can achieve the performance of an ideal Max SNR strategy if the feedback threshold is optimized. Finally, a novel cross-layer energy-adaptive scheduling and queue management framework EAED (Energy Aware Early Detection), for preserving delay bounds and minimizing energy consumption in WLAN mesh networks, is proposed and evaluated with simulations. The simulations show that our scheme can save considerable amounts of transmission energy without violating application level QoS requirements when traffic load and distances are reasonable

Improving network routing performance in dynamic environments

In this dissertation, we study methods for improving the routing performance of computer communication networks in dynamic environments. The dynamic environments we considered in this work include both network topology changes and traffic demand changes. In the first part, We propose a novel fast rerouting scheme for link state routing protocols. Link state routing protocols are widely used by todayâÂÂs ISPs on their backbone networks. The global update based rerouting of link state protocols usually takes seconds to complete which affects real time applications like Voice over IP. In our scheme, usually, only routers directly connected to failed links are involved in rerouting. For other cases, only a small number of neighboring routers are also involved. Since our scheme calculates rerouting paths in advance, rerouting can be done faster than previous reactive approaches. The computation complexity of our scheme is less than previous proactive approaches. In the second part, we study Multihoming Route Control (MRC) that is a technology used by multihomed stub networks recently. By selecting ISPs with better quality, MRC can improve routing performance of stub networks significantly. We first study the stability issue of distributed MRC and propose two methods to avoid possible oscillations of traditional MRC. The first MRC method is based on âÂÂoptimal routingâÂÂ. The idea is to let the stub networks belonging to a same organization coordinate their MRC and thus avoid oscillations. The second method is based on âÂÂuser-optimal routingâÂÂ. The idea is to allow MRC devices to use multiple paths for traffic to one destination network and switch traffic between paths smoothly when path quality or the traffic matrix changes. A third MRC method we propose is for MRC of traffic consisting of TCP flows of different sizes on paths with bottlenecks of limited capacity. Based on analysis of quality characteristics of bottleneck links, we propose a greedy MRC approach that works in small timescales. Simulation results show that the proposed MRC method can greatly improve routing performance for the MRC sites as well as the overall routing performance of all sites in the network

TCP flow aware adaptive path switching in diffserv enabled MPLS networks

Cataloged from PDF version of article.We propose an adaptive flow-level multi-path routing-based traffic engineering solution for an IP backbone network carrying TCP/IP traffic. Incoming TCP flows are switched between two explicitly routed paths, namely the primary and secondary paths (PP and SP), for resilience and potential goodput improvement at the TCP layer. In the proposed architecture, PPs receive a preferential treatment over SPs using differentiated services mechanisms. The reason for this choice is not for service differentiation but for coping with the detrimental knock-on effect stemming from the use of longer SP that is well known for conventional network load balancing algorithms. Moreover, both paths are congestion-controlled using Explicit Congestion Notification marking at the core and Additive Increase Multiplicative Decrease rate adjustment at the ingress nodes. The delay difference between PP and SP is estimated using two per-egress rate-controlling buffers maintained at the ingress nodes for each path, and this delay difference is used to determine the path over which a new TCP flow will be routed. We perform extensive simulations using ns-2 in order to demonstrate the viability of the proposed distributed adaptive multi-path routing method in terms of per-flow TCP goodput. The proposed solution consistently outperforms the single-path routing policy and provides substantial per-flow goodput gains under poor PP conditions. Moreover, highest goodput improvements under the proposed scheme are achieved by flows that receive the lowest goodputs with single-path routing, while the performance of the flows with high goodputs with single-path routing does not deteriorate with the proposed path switching technique. Copyright # 2011 John Wiley & Sons, Ltd