ARQ protocol for joint source and channel coding and its applications

Shannon\u27s separation theorem states that for transmission over noisy channels, approaching channel capacity is possible with the separation of source and channel coding. Practically, the situation is different. Infinite size blocks are needed to achieve this theoretical limit. Also, time-varying channels require a different approach. This leads to many approaches for source and channel coding. This dissertation will address a joint source and channel coding that suits Automatic Repeat Request (ARQ) application and applies it to packet switching networks. Following aspects of the proposed joint source and channel coding approach will be presented: The design of the proposed joint source and channel coding scheme. The approach is based on a variable length coding scheme which adapts the arithmetic coding process for joint source and channel coding. The protocol using this joint source and channel coding scheme in communication systems. The error recovery technique of the proposed scheme is presented. The application of the scheme and protocol. The design is applied to wireless TCP network and real-time video transmissions. The coding scheme embeds the redundancy needed for error detection in source coding stage. The self-synchronization property of lossless compression is utilized by decoder to detect channel errors. With this approach, error detection may be delayed. The delay in detection is referred to as error propagation distance. This work analyzes the distribution of error propagation distance. The error recovery technique of this joint source and channel coding for ARQ (JARQ) protocol is analyzed. Throughput is studied using signal flow graph for both independent channel and nonindependent channels. A packet combining technique is presented which utilizes the non-uniform distribution of error propagation distance to increase the throughput. The proposed scheme may be applied to many areas. In particular, two applications are discussed. A TCP/JARQ protocol stack is introduced and the coordination between TCP and JARQ layers is discussed to maximize system performance. By limiting the number of retransmission, the proposed scheme is applied to real-time transmission to meet timing requirement

Differential Radio Link Protocol: An Improvement To Tcp Over Wireless Networks

New generations of wireless cellular networks, including 3G and 4G technologies, are envisaged to support more mobile users and a variety of wireless multimedia services. With an increasing demand for wireless multimedia services, the performance of TCP becomes a bottleneck as it cannot differentiate between the losses due to the nature of air as a medium and high data load on the network that leads to congestion. This misinterpretation by TCP leads to a reduction in the congestion window size thereby resulting in reduced throughput of the system. To overcome this scenario Radio Link Protocols are used at a lower layer which hides from TCP the channel related losses and effectively increases the throughput. This thesis proposes enhancements to the radio link protocol that works underneath TCP by identifying decisive frames and categorizing them as {\em crucial} and {\em non-crucial}. The fact that initial frames from the same upper layer segment can afford a few trials of retransmissions and the later frames cannot, motivates this work. The frames are treated differentially with respect to FEC coding and ARQ schemes. Specific cases of FEC and ARQ strategies are then considered and it is shown qualitatively as how the differential treatment of frames can improve the performance of the RLP and in effect that of TCP over wireless networks

A General Framework for Analyzing, Characterizing, and Implementing Spectrally Modulated, Spectrally Encoded Signals

Fourth generation (4G) communications will support many capabilities while providing universal, high speed access. One potential enabler for these capabilities is software defined radio (SDR). When controlled by cognitive radio (CR) principles, the required waveform diversity is achieved via a synergistic union called CR-based SDR. Research is rapidly progressing in SDR hardware and software venues, but current CR-based SDR research lacks the theoretical foundation and analytic framework to permit efficient implementation. This limitation is addressed here by introducing a general framework for analyzing, characterizing, and implementing spectrally modulated, spectrally encoded (SMSE) signals within CR-based SDR architectures. Given orthogonal frequency division multiplexing (OFDM) is a 4G candidate signal, OFDM-based signals are collectively classified as SMSE since modulation and encoding are spectrally applied. The proposed framework provides analytic commonality and unification of SMSE signals. Applicability is first shown for candidate 4G signals, and resultant analytic expressions agree with published results. Implementability is then demonstrated in multiple coexistence scenarios via modeling and simulation to reinforce practical utility

Quality of service and channel-aware packet bundling for capacity improvement in cellular networks

Title from PDF of title page, viewed on May 26, 2011VitaIncludes bibliographical references (p. 76-84)Thesis (Ph.D.)--School of Computing and Engineering. University of Missouri--Kansas City, 2011We study the problem of multiple packet bundling to improve spectral efficiency in cellular networks. The packet size of real-time data, such as VoIP, is often very small. However, the common use of time division multiplexing limits the number of VoIP users supported, because a packet has to wait until it receives a time slot, and if only one small VoIP packet is placed in a time slot, capacity is wasted. Packet bundling can alleviate such a problem by sharing a time slot among multiple users. A recent revision of cdma2000 1xEV-DO introduced the concept of the multi-user packet (MUP) in the downlink to overcome limitations on the number of time slots. However, the efficacy of packet bundling is not well understood, particularly in the presence of time varying channels. We propose a novel QoS and channel-aware packet bundling algorithm that takes advantage of adaptive modulation and coding. We show that optimal algorithms are NP complete and recommend heuristic approaches. We also show that channel utilization can be significantly increased by slightly delaying some real-time packets within their QoS requirements while bundling those packets with like channel conditions. We validate our study through extensive OPNET simulations with a complete EV-DO implementation.Introduction -- Related work -- Background on wireless systems -- Multiple packet bundling -- Evaluation -- Conclusion

Adaptive network abstraction layer packetization for low bit rate H.264/AVC video transmission over wireless mobile networks under cross layer optimization

Master'sMASTER OF ENGINEERIN

Cooperative systems based signal processing techniques with applications to three-dimensional video transmission

Three-dimensional (3-D) video has recently emerged to offer an immersive multimedia experience that can not be offered by two-dimensional (2-D) video applications. Currently, both industry and academia are focused on delivering 3-D video services to wireless communication systems. Modern video communication systems currently adopt cooperative communication and orthogonal frequency division multiplexing (OFDM) as they are an attractive solution to combat fading in wireless communication systems and achieve high data-rates. However, this strong motivation to transmit the video signals over wireless systems faces many challenges. These are mainly channel bandwidth limitations, variations of signal-to-noise ratio (SNR) in wireless channels, and the impairments in the physical layer such as time varying phase noise (PHN), and carrier frequency offset (CFO). In response to these challenges, this thesis seeks to develop efficient 3-D video transmission methods and signal processing algorithms that can overcome the effects of error-prone wireless channels and impairments in the physical layer. In the first part of the thesis, an efficient unequal error protection (UEP) scheme, called video packet partitioning, and a new 3-D video transceiver structure are proposed. The proposed video transceiver uses switching operations between various UEP schemes based on the packet partitioning to achieve a trade- off between system complexity and performance. Experimental results show that the proposed system achieves significantly high video quality at different SNRs with the lowest possible bandwidth and system complexity compared to direct transmission schemes. The second part of the thesis proposes a new approach to joint source-channel coding (JSCC) that simultaneously assigns source code rates, the number of high and low priority packets, and channel code rates for the application, network, and physical layers, respectively. The proposed JSCC algorithm takes into account the rate budget constraint and the available instantaneous SNR of the best relay selection in cooperative systems. Experimental results show that the proposed JSCC algorithm outperforms existing algorithms in terms of peak signal-to-noise ratio (PSNR). In the third part of the thesis, a computationally efficient training based approach for joint channel, CFO, and PHN estimation in OFDM systems is pro- posed. The proposed estimator is based on an expectation conditional maximization (ECM) algorithm. To compare the estimation accuracy of the proposed estimator, the hybrid Cram´er-Rao lower bound (HCRB) of hybrid parameters of interest is derived. Next, to detect the signal in the presence of PHN, an iterative receiver based on the extended Kalman filter (EKF) for joint data detection and PHN mitigation is proposed. It is demonstrated by numerical simulations that, compared to existing algorithms, the performance of the proposed ECM-based estimator in terms of the mean square error (MSE) is closer to the derived HCRB and outperforms the existing estimation algorithms at moderate-to-high SNRs. Finally, this study extends the research on joint channel, PHN, and CFO estimation one step forward from OFDM systems to cooperative OFDM systems. An iterative algorithm based on the ECM in cooperative OFDM networks in the presence of unknown channel gains, PHNs and CFOs is applied. Moreover, the HCRB for the joint estimation problem in both decode-and-forward (DF) and amplify-and-forward (AF) relay systems is presented. An iterative algorithm based on the EKF for data detection and tracking the unknown time-varying PHN throughout the OFDM data packet is also used. For more efficient 3-D video transmission, the estimation algorithms and UEP schemes based packet portioning were combined to achieve a more robust video bit stream in the presence of PHNs. Applying this combination, simulation results demonstrate that promising bit-error-rate (BER) and PSNR performance can be achieved at the destination at different SNRs and PHN variance. The proposed schemes and algorithms offer solutions for existing problems in the techniques for applications to 3-D video transmission

Session initiation protocol for wireless channels

The Session Initiation Protocol (SIP) was designed for wire line networks. It was developed to initiate, modify and terminate sessions between two hosts on a network. When the Internet expanded to include wireless hosts, SIP did not scale well for these wireless hosts because of the nature of the wireless channel. Also, there were issues with mobility and real time communication. This thesis proposes improvements to some of the extensions to SIP, for better performance over wireless channels. We investigate the call setup time for various transport mechanisms viz. TCP and UDP, and study the performance of a dynamic Session Timers compared to the current standard of a periodic refresh mechanism, where the frequency of UPDATEs vary with the condition of the wireless channel. We also propose a handoff algorithm that reduces the handover time with decreased packet losses