<p>The first part of the thesis (Part I) is devoted to find methods to describe transient behaviour of traffic processes, where the main emphasis is put on the description and analysis of excess periods and excess volumes of quite general stochastic processes. By assuming that traffic changes on different time scales, the transient characteristics such as excess periods could be important measures to describe periods of congestion on a communication link and moreover, the corresponding excess volume will represent lost information during such periods. Although the results obtained are of rather general nature, they provide some rather fundamental insight into transient characteristics of traffic processes. The distributions of the length of excess periods may then be expressed it terms of some excess probabilities that are related to the minimum of the process in the time interval considered. Similar relationsfor the excess volumes are harder to obtain and require the joint probability of the arrived volume and the minimum of the process in the same time interval.</p>

The first part of the thesis (Part I) is devoted to find methods to describe transient behaviour of traffic processes, where the main emphasis is put on the description and analysis of excess periods and excess volumes of quite general stochastic processes. By assuming that traffic changes on different time scales, the transient characteristics such as excess periods could be important measures to describe periods of congestion on a communication link and moreover, the corresponding excess volume will represent lost information during such periods. Although the results obtained are of rather general nature, they provide some rather fundamental insight into transient characteristics of traffic processes. The distributions of the length of excess periods may then be expressed it terms of some excess probabilities that are related to the minimum of the process in the time interval considered. Similar relationsfor the excess volumes are harder to obtain and require the joint probability of the arrived volume and the minimum of the process in the same time interval.

Some of the applications in modern data networks are delay sensitive (e.g., video and voice). An end-to-end delay analysis is needed to estimate the required network resources of delay sensitive applications. The schedulers used in the network can impact the resulting delays to the applications. When multiple applications are multiplexed in a switch, a scheduler is used to determine the precedence of the arrivals from different applications. Computing the end-to-end delay and queue sizes in a network of schedulers is difficult and the existing solutions are limited to some special cases (e.g., specific type of traffic). The theory of Network Calculus employs the min-plus algebra to obtain performance bounds. Given an upper bound on the traffic arrival in any time interval and a lower bound on the available service (called the service curve) at a network element, upper bounds on the delay and queue size of the traffic in that network element can be obtained. An equivalent end-to-end service curve of a tandem of queues is the min-plus convolution of the service curves of all nodes along the path. A probabilistic end-to-end delay bound using network service curve scales with O(H logH) in the path length H. This improves the results of the conventional method of adding per-node delay bounds scaling with O(H^3). We have used and advanced Network Calculus for end-to-end delay analysis in a network of schedulers. We formulate a service curve description for a large class of schedulers which we call Delta-schedulers. We show that with this service curve, tight single node delay and backlog bounds can be achieved. In an end-to-end scenario, we formulate a new convolution theoii rem which considerably improves the end-to-end probabilistic delay bounds. We specify our probabilistic end-to-end delay and backlog bounds for exponentially bounded burstniess (EBB) traffic arrivals. We show that the end-to-end delay varies considerably by the type of schedulers along the path. Using these bounds, we also show that a if the number of flows increases, the queues inside a network can be analyzed in isolation and regardless of the network effect.

Some of the applications in modern data networks are delay sensitive (e.g., video and voice). An end-to-end delay analysis is needed to estimate the required network resources of delay sensitive applications. The schedulers used in the network can impact the resulting delays to the applications. When multiple applications are multiplexed in a switch, a scheduler is used to determine the precedence of the arrivals from different applications. Computing the end-to-end delay and queue sizes in a network of schedulers is difficult and the existing solutions are limited to some special cases (e.g., specific type of traffic). The theory of Network Calculus employs the min-plus algebra to obtain performance bounds. Given an upper bound on the traffic arrival in any time interval and a lower bound on the available service (called the service curve) at a network element, upper bounds on the delay and queue size of the traffic in that network element can be obtained. An equivalent end-to-end service curve of a tandem of queues is the min-plus convolution of the service curves of all nodes along the path. A probabilistic end-to-end delay bound using network service curve scales with O(H logH) in the path length H. This improves the results of the conventional method of adding per-node delay bounds scaling with O(H^3). We have used and advanced Network Calculus for end-to-end delay analysis in a network of schedulers. We formulate a service curve description for a large class of schedulers which we call Delta-schedulers. We show that with this service curve, tight single node delay and backlog bounds can be achieved. In an end-to-end scenario, we formulate a new convolution theoii rem which considerably improves the end-to-end probabilistic delay bounds. We specify our probabilistic end-to-end delay and backlog bounds for exponentially bounded burstniess (EBB) traffic arrivals. We show that the end-to-end delay varies considerably by the type of schedulers along the path. Using these bounds, we also show that a if the number of flows increases, the queues inside a network can be analyzed in isolation and regardless of the network effect.

Wireless channels are vulnerable to interception. In some applications an end-to-end secure data transfer is required. However the use of cryptographic functions in communication over a wireless channel increases sensitivity to channel errors. As a result, the connection characteristics in terms of delay, throughput, and transmission energy worsen. Transmission energy is a key issue in some secure end-to-end wireless applications especially if they are running on mobile handheld devices with a limited source of energy such as batteries. That is why in most secure end-to-end wireless connections, the connection is dropped in poor channel conditions. In this thesis, models are proposed by which the performance is improved and transmission energy is lowered. A combination of a cross-layer controller, K Best Likelihood (K-BL) channel decoder, and a keyed error detection algorithm in the novel model supports the authorized receivers by a higher throughput, lower delay mean, and less transmission energy in a certain range of the Signal to Noise Ratio (SNR). This is done at the expense of additional computation at the receiving end. Ttradeoffs are examined and the simulation results of the new model are compared with those of conventional wireless communication systems. Another model is devised to mitigate the energy consumption of the Turbo Code channel decoder. The overall decoding energy consumption for each packet can be lowered by reducing the average number of iterations in the Turbo Code channel decoder. The proposed models achieve better energy consumption by reducing the number of iterations in a channel decoder that uses the Turbo decoder and by reducing the number of retransmissions in a trellis channel decoder. Furthermore, the security enhancement of the novel models is assessed in terms of the extent to which the enhancement is fully achieved.

The paradigm of separate source-channel coding is inspired by Shannon's separation result, which implies the asymptotic optimality of designing source and channel coding independently from each other. The result exploits the fact that channel error probabilities can be made arbitrarily small, as long as the block length of the channel code can be made arbitrarily large. However, this is not possible in practice, where the block length is either fixed or restricted to a range of finite values. As a result, the optimality of source and channel coding separation becomes unknown, leading researchers to consider joint source-channel coding (JSCC) to further improve the performance of practical systems that must operate in the finite block length regime. With this motivation, this thesis investigates the application of JSCC principles for multimedia communications over point-to-point, broadcast, and relay channels. All analyses are conducted from the perspective of end-to-end distortion (EED) for results that are applicable to channel codes with finite block lengths in pursuing insights into practical design. The thesis first revisits the fundamental open problem of the separation of source and channel coding in the finite block length regime. Derived formulations and numerical analyses for a source-channel coding system reveal many scenarios where the EED reduction is positive when pairing the channel-optimized source quantizer (COSQ) with an optimal channel code, hence establishing the invalidity of the separation theorem in the finite block length regime. With this, further improvements to JSCC systems are considered by augmenting error detection codes with the COSQ. Closed-form EED expressions for such system are derived, from which necessary optimality conditions are identified and used in proposed algorithms for system design. Results for both the point-to-point and broadcast channels demonstrate significant reductions to the EED without sacrificing bandwidth when considering a tradeoff between quantization and error detection coding rates. Lastly, the JSCC system is considered under relay channels, for which a computable measure of the EED is derived for any relay channel conditions with nonzero channel error probabilities. To emphasize the importance of analyzing JSCC systems under finite block lengths, the large sub-optimality in performance is demonstrated when solving the power allocation configuration problem according to capacity-based formulations that disregard channel errors, as opposed to those based on the EED. Although this thesis only considers one JSCC setup of many, it is concluded that consideration of JSCC systems from a non-asymptotic perspective not only is more meaningful, but also reveals more relevant insight into practical system design. This thesis accomplishes such by maintaining the EED as a measure of system performance in each of the considered point-to-point, broadcast, and relay cases.

Wireless channels are vulnerable to interception. In some applications an end-to-end secure data transfer is required. However the use of cryptographic functions in communication over a wireless channel increases sensitivity to channel errors. As a result, the connection characteristics in terms of delay, throughput, and transmission energy worsen. Transmission energy is a key issue in some secure end-to-end wireless applications especially if they are running on mobile handheld devices with a limited source of energy such as batteries. That is why in most secure end-to-end wireless connections, the connection is dropped in poor channel conditions. In this thesis, models are proposed by which the performance is improved and transmission energy is lowered. A combination of a cross-layer controller, K Best Likelihood (K-BL) channel decoder, and a keyed error detection algorithm in the novel model supports the authorized receivers by a higher throughput, lower delay mean, and less transmission energy in a certain range of the Signal to Noise Ratio (SNR). This is done at the expense of additional computation at the receiving end. Ttradeoffs are examined and the simulation results of the new model are compared with those of conventional wireless communication systems. Another model is devised to mitigate the energy consumption of the Turbo Code channel decoder. The overall decoding energy consumption for each packet can be lowered by reducing the average number of iterations in the Turbo Code channel decoder. The proposed models achieve better energy consumption by reducing the number of iterations in a channel decoder that uses the Turbo decoder and by reducing the number of retransmissions in a trellis channel decoder. Furthermore, the security enhancement of the novel models is assessed in terms of the extent to which the enhancement is fully achieved.

IEEE 802.11 has become the standard in wireless LAN. Originally, 802.11 is designed for the best-effort services only. To support the increasing demand of delay-sensitive applications, IEEE 802.11 Task Group E is developing a QoS-aware MAC protocol, EDCF, for differentiated services. However, when the network becomes congested, there exists unexpected packet delay due to collisions and retransmissions. This thesis proposes a dynamic contention control (DCC) scheme to reduce packet delay and increase the percentage of packets arriving in time. DCC estimates per-hop delay, denoted as Mrtt, and end-to-end delay, denoted as Sigma_t, based on either the received MAC-layer ACK or the control packets of a reactive routing protocol. Then, Mrtt and Sigma_t are used to dynamically adjust the associated contention window for each priority. Besides, when a frame is retransmitted, the backoff time is determined according to the remaining end-to-end delay instead of a uniformly distributed random number. For the propose of evaluation, we perform simulations on the well-known network simulator, NS-2. DCC is compared with the EDCF and one previously proposed scheme, AEDCF. The simulation results demonstrate the effectiveness and superiority of DCC.

The increasing use of electronic components in today&rsquo / s automobiles demands more powerful in-vehicle network communication protocols. FlexRay protocol, which is expected to be the de-facto standard in the near future, is a deterministic, fault tolerant and fast protocol designed for in vehicle communication. The current de-facto in-vehicle communication standard, CAN, and the future in-vehicle communication standard FlexRay will exist together in future cars. Data exchange between these two standards will be performed via Gateway units. In this thesis, end-to-end performance of a FlexRay-CAN network connected by a Gateway is evaluated as well as Gateway functionality and processing delay. The results of the experiments, which are performed for a realistic message set with various scheduling schemes, are presented and discussed.

Quality of Service (QoS) in real-time media applications can be defined as the ability to guarantee the delivery of packets from source to destination over best-effort networks within some constraints. These constraints defined as the QoS metrics are end-to-end packet delay, delay jitter, throughtput, and packet losses. Transporting real-time media applications over best-effort networks, e.g. the Internet, is an area of current research. Both the Transmission Control Protocol (TCP) and the User Datagram Protocol (UDP) have failed to provide the desired QoS. This research aims at developing application-level end-to-end QoS controls to improve the user-perceived quality of real-time media applications over best-effort networks, such as, the public Internet. In this research an end-to-end packet based approach is developed. The end-to- end packet based approach consists of source buffer, network simulator ns-2, destina- tion buffer, and controller. Unconstrained model predictive control (MPC) methods are implemented by the controller at the application layer. The end-to-end packet based approach uses end-to-end network measurements and predictions as feedback signals. Effectiveness of the developed control methods are examined using Matlab and ns-2. The results demonstrate that sender-based control schemes utilizing UDP at transport layer are effective in providing QoS for real-time media applications transported over best-effort networks. Significant improvements in providing QoS are visible by the reduction of packet losses and the elimination of disruptions during the playback of real-time media. This is accompanied by either a decrease or increase in the playback start-time.