Queueing Delay Minimization in Overloaded Networks via Rate Control

We first develop a rate-distortion (R-D) model for DCT-based video coding incorporating the macroblock (MB) intra refreshing rate. For any given bit rate and intra refreshing rate, this model is capable of estimating the corresponding coding distortion even before a video frame is coded. We then present a theoretical analysis of the picture distortion caused by channel errors and the subsequent inter-frame propagation. Based on this analysis, we develop a statistical model to estimate such channel errors induced distortion for different channel conditions and encoder settings. The proposed analytic model mathematically describes the complex behavior of channel errors in a video coding and transmission system. Unlike other experimental approaches for distortion estimation reported in the literature, this analytic model has very low computational complexity and implementation cost, which are highly desirable in wireless video applications. Simulation results show that this model is able to accurately estimate the channel errors induced distortion with a minimum delay in processing. Based on the proposed source coding R-D model and the analytic channel-distortion estimation, we derive an analytic solution for adaptive intra mode selection and joint source-channel rate control under time-varying wireless channel conditions. Extensive experimental results demonstrate that this scheme significantly improves the end-to-end video quality in wireless video coding and transmission.

A novel method is proposed which minimizes the tune-in time of channel zapping, i.e. changing from one audiovisual service to another, in IP datacasting (IPDC) over Digital Video Broadcasting for Handheld terminals (DVB-H). DVB-H uses a time-sliced transmission scheme enabling a receiver to turn radio reception off for those time-slices that are not of interest to the user and thus reducing the power consumption used for radio reception. One of the significant factors in tune-in time is the time from the start of media decoding to the start of correct output from decoding, which is minimized when a time-slice starts with a random access point picture such as an independent decoding refresh (IDR) picture in H.264/AVC standard. In IPDC over DVB-H, encapsulation to time-slices is performed independently from encoding in a network element called IP encapsulator. At the time of encoding, time-slice boundaries are typically not known exactly, and it is therefore impossible to govern the location of IDR pictures relative to time-slices. It is proposed that an additional stream consisting of IDR pictures only is transmitted to the IP encapsulator, which replaces pictures in a normal bit stream with IDR pictures according to time-slice boundaries in order to achieve the minimum tune-in delay. Replacing pictures causes a mismatch in the pixel values of the reference pictures between the encoder and decoder and the mismatch error is propagated in the reconstructed video. It has to be ensured that the propagated error is subjectively negligible. Furthermore, the stream resulting from the operation of the IP encapsulator should comply with the Hypothetical Reference Decoder (HRD) specification of H.264/AVC. Error propagation caused by the proposed splicing method is analysed and a video rate control system is proposed to satisfy the HRD requirements for the spliced stream. Simulation results show that in addition to fulfilling H.264/AVC compliancy, good average quality of decoded video is achieved with minimum tune-in delay and complexity

In Wireless Sensor Networks, congestion occurs when the traffic rate is high. This happens when the an event is detected in a network. Congestion causes packet loss thus degrading the performance of the network. Hence it necessitates to develop an effective congestion control technique. This paper focuses on congestion due to concurrent transmission. We have proposed an efficient protocol to detect and control congestion in a MAC. The occurrence of congestion is detected by calculating a new metrics called congestion scale. When the congestion scale exceeds the threshold value it intimates, that, congestion has occurred. Congestion notification signal is send to all the nodes. On receiving the notification signal all nodes adjust their transmission rate to control congestion. We have implemented Hop-by-Hop Rate Control Technique(HRCT) to control congestion and to guarantee both high throughput and minimum delay. This technique is implemented successfully in NS-2 simulator. Finally, simulation results have demonstrated the effectiveness of our proposed algorithm.

Analysis of fading channels is usually made under the framework of either information theory or network theory. Information theory is usually concerned with the total throughput of the channel (Shannon capacity) while network theory considers also other performance measures such as average delay and buffer overflow probability. In a contemporary communication system operating in a fading environment, the performance measure is usually a combination of delay and throughput measures under various power constraints. Optimizing both delay and throughput is somewhat contradictory since achieving maximal throughput depends only on the fading channel while minimal average delay depends also on the arrival process of data to the transmitter. We define a performance criterion of both transmission power and average delay. Dynamic programming is employed to find the optimal power/rate control for this criterion as well as the system performance. Numerical results are presented as well as asymptotic analytic results for large buffers.