Micro protocol engineering for unstructured carriers: On the embedding of steganographic control protocols into audio transmissions

Network steganography conceals the transfer of sensitive information within unobtrusive data in computer networks. So-called micro protocols are communication protocols placed within the payload of a network steganographic transfer. They enrich this transfer with features such as reliability, dynamic overlay routing, or performance optimization --- just to mention a few. We present different design approaches for the embedding of hidden channels with micro protocols in digitized audio signals under consideration of different requirements. On the basis of experimental results, our design approaches are compared, and introduced into a protocol engineering approach for micro protocols.Comment: 20 pages, 7 figures, 4 table

Advanced Compression and Latency Reduction Techniques Over Data Networks

Applications and services operating over Internet protocol (IP) networks often suffer from high latency and packet loss rates. These problems are attributed to data congestion resulting from the lack of network resources available to support the demand. The usage of IP networks is not only increasing, but very dynamic as well. In order to alleviate the above-mentioned problems and to maintain a reasonable Quality of Service (QoS) for the end users, two novel adaptive compression techniques are proposed to reduce packets’ payload size. The proposed schemes exploit lossless compression algorithms to perform the compression process on the packets’ payloads and thus decrease the overall net- work congestion. The first adaptive compression scheme utilizes two key network performance indicators as design metrics. These metrics include the varying round-trip time (RTT) and the number of dropped packets. The second compression scheme uses other network information such as the incoming packet rate, intermediate nodes processing rate, average packet waiting time within a queue of an intermediate node, and time required to perform the compression process. The performances of the proposed algorithms are evaluated through Network Simulator 3 (NS3). The simulation results show an improvement in network conditions, such as the number of dropped packets, network latency, and throughput

TCP performance enhancement in wireless networks via adaptive congestion control and active queue management

The transmission control protocol (TCP) exhibits poor performance when used in error-prone wireless networks. Remedy to this problem has been an active research area. However, a widely accepted and adopted solution is yet to emerge. Difficulties of an acceptable solution lie in the areas of compatibility, scalability, computational complexity and the involvement of intermediate routers and switches. This dissertation rexriews the current start-of-the-art solutions to TCP performance enhancement, and pursues an end-to-end solution framework to the problem. The most noticeable cause of the performance degradation of TCP in wireless networks is the higher packet loss rate as compared to that in traditional wired networks. Packet loss type differentiation has been the focus of many proposed TCP performance enhancement schemes. Studies conduced by this dissertation research suggest that besides the standard TCP\u27s inability of discriminating congestion packet losses from losses related to wireless link errors, the standard TCP\u27s additive increase and multiplicative decrease (AIMD) congestion control algorithm itself needs to be redesigned to achieve better performance in wireless, and particularly, high-speed wireless networks. This dissertation proposes a simple, efficient, and effective end-to-end solution framework that enhances TCP\u27s performance through techniques of adaptive congestion control and active queue management. By end-to-end, it means a solution with no requirement of routers being wireless-aware or wireless-specific . TCP-Jersey has been introduced as an implementation of the proposed solution framework, and its performance metrics have been evaluated through extensive simulations. TCP-Jersey consists of an adaptive congestion control algorithm at the source by means of the source\u27s achievable rate estimation (ARE) —an adaptive filter of packet inter-arrival times, a congestion indication algorithm at the links (i.e., AQM) by means of packet marking, and a effective loss differentiation algorithm at the source by careful examination of the congestion marks carried by the duplicate acknowledgment packets (DUPACK). Several improvements to the proposed TCP-Jersey have been investigated, including a more robust ARE algorithm, a less computationally intensive threshold marking algorithm as the AQM link algorithm, a more stable congestion indication function based on virtual capacity at the link, and performance results have been presented and analyzed via extensive simulations of various network configurations. Stability analysis of the proposed ARE-based additive increase and adaptive decrease (AJAD) congestion control algorithm has been conducted and the analytical results have been verified by simulations. Performance of TCP-Jersey has been compared to that of a perfect , but not practical, TCP scheme, and encouraging results have been observed. Finally the framework of the TCP-Jersey\u27s source algorithm has been extended and generalized for rate-based congestion control, as opposed to TCP\u27s window-based congestion control, to provide a design platform for applications, such as real-time multimedia, that do not use TCP as transport protocol yet do need to control network congestion as well as combat packet losses in wireless networks. In conclusion, the framework architecture presented in this dissertation that combines the adaptive congestion control and active queue management in solving the TCP performance degradation problem in wireless networks has been shown as a promising answer to the problem due to its simplistic design philosophy complete compatibility with the current TCP/IP and AQM practice, end-to-end architecture for scalability, and the high effectiveness and low computational overhead. The proposed implementation of the solution framework, namely TCP-Jersey is a modification of the standard TCP protocol rather than a completely new design of the transport protocol. It is an end-to-end approach to address the performance degradation problem since it does not require split mode connection establishment and maintenance using special wireless-aware software agents at the routers. The proposed solution also differs from other solutions that rely on the link layer error notifications for packet loss differentiation. The proposed solution is also unique among other proposed end-to-end solutions in that it differentiates packet losses attributed to wireless link errors from congestion induced packet losses directly from the explicit congestion indication marks in the DUPACK packets, rather than inferring the loss type based on packet delay or delay jitter as in many other proposed solutions; nor by undergoing a computationally expensive off-line training of a classification model (e.g., HMM), or a Bayesian estimation/detection process that requires estimations of a priori loss probability distributions of different loss types. The proposed solution is also scalable and fully compatible to the current practice in Internet congestion control and queue management, but with an additional function of loss type differentiation that effectively enhances TCP\u27s performance over error-prone wireless networks. Limitations of the proposed solution architecture and areas for future researches are also addressed

Design and implementation of a secret communication system based on TCP retransmission steganography

This paper presents the design, implementation and experimental results of a communication system based on a network steganography method known as retransmission steganography (RSTEG). This technique, proposed by Mazurczyk et al. in 2009, is aimed at many network protocols that use a retransmission mechanism. Essentially, RSTEG works by not acknowledging a well-received package in order to invoke a retransmission, whose payload data will be exchanged by a steganogram. A client/server architecture has been built with a simple TCP implementation based on retransmission time-outs (RTO) that includes the RSTEG functionality. The application is able to communicate using an HTTP implementation built on top of it. The performed experiments evaluate the steganographic bandwidth and detectability of the implemented method. An average bandwidth of 6.4 kB/s was achieved for a 5% of retransmission probability on a 132 kB/s TCP throughput.Aquest article presenta el disseny, implementació i resultats experimentals d'un sistema de comunicació basat en el mètode d'esteganografia de xarxa conegut com a esteganografia de les retransmissions (RSTEG). Aquesta tècnica, proposada l'any 2009 per Mazurczyk et al., està enfocada a protocols de xarxa que utilitzen retransmissions. Concretament, RSTEG consisteix a no confirmar un paquet rebut per tal de forçar la seva retransmissió. Aquest paquet retransmès haurà substituït les dades d'usuari per un esteganograma. S'ha construït una arquitectura client/servidor mitjançant una implementació senzilla de TCP que inclou aquesta funcionalitat. L'aplicació es comunica mitjançant una implementació d'HTTP construïda a sobre. S'han realitzat experiments per tal d'avaluar la capacitat i la detectabilitat del mètode implementat, amb una banda ampla de 6,4 kB/s de mitjana per un 5% de probabilitat de retransmissió sobre una connexió TCP a 132 kB/s.Este artículo presenta el diseño, implementación y resultados experimentales de un sistema de comunicación basado en el método de esteganografía de red conocido como esteganografía de las retransmisiones (RSTEG). Esta técnica, propuesta en 2009 por Mazurczyk et al., está enfocada a protocolos de red que utilizan retransmisiones. Concretamente, RSTEG consiste en no confirmar un paquete recibido para forzar su retransmisión. Este paquete retransmitido habrá sustituido sus datos de usuario por un esteganograma. Se ha construido una arquitectura cliente/servidor mediante una implementación sencilla de TCP que incluye dicha funcionalidad. La aplicación se comunica mediante una implementación de HTTP construida por encima. Se han realizado experimentos para evaluar la capacidad y la detectabilidad del método implementado, obteniendo una banda ancha de 6,4 kB/s de media para un 5% de probabilidad de retransmisión sobre una conexión TCP a 132 kB/s

Evaluating and improving the performance of video content distribution in lossy networks

The contributions in this research are split in to three distinct, but related, areas. The focus of the work is based on improving the efficiency of video content distribution in the networks that are liable to packet loss, such as the Internet. Initially, the benefits and limitations of content distribution using Forward Error Correction (FEC) in conjunction with the Transmission Control Protocol (TCP) is presented. Since added FEC can be used to reduce the number of retransmissions, the requirement for TCP to deal with any losses is greatly reduced. When real-time applications are needed, delay must be kept to a minimum, and retransmissions not desirable. A balance, therefore, between additional bandwidth and delays due to retransmissions must be struck. This is followed by the proposal of a hybrid transport, specifically for H.264 encoded video, as a compromise between the delay-prone TCP and the loss-prone UDP. It is argued that the playback quality at the receiver often need not be 100% perfect, providing a certain level is assured. Reliable TCP is used to transmit and guarantee delivery of the most important packets. The delay associated with the proposal is measured, and the potential for use as an alternative to the conventional methods of transporting video by either TCP or UDP alone is demonstrated. Finally, a new objective measurement is investigated for assessing the playback quality of video transported using TCP. A new metric is defined to characterise the quality of playback in terms of its continuity. Using packet traces generated from real TCP connections in a lossy environment, simulating the playback of a video is possible, whilst monitoring buffer behaviour to calculate pause intensity values. Subjective tests are conducted to verify the effectiveness of the metric introduced and show that the results of objective and subjective scores made are closely correlated

Adaptive filtering of MPEG system streams in IP networks

Congestion and large differences in available link bandwidth create challenges for the design of applications that want to deliver high quality video over the Internet. We present an efficient adaptive filter for MPEG System streams that can be placed in the network (e.g., as an active service). This filter adjusts the bandwidth demands of an MPEG System stream to the available bandwidth without transcoding while maintaining synchronization between the streams embedded in the MPEG System. The filter is network-friendly: it is fair with respect to other (TCP) competing streams and it avoids generating bursty traffic. This paper presents the system architecture and an evaluation of our implementation in three different operating environments: a networking testbed in a laboratory environment, a home-user scenario (DSL line with 640Kbit/s), and a wide area network covering the Atlantic (server in Europe, client in the US). Moreover we examine the network-friendliness of the adaptation protocol and the relationship between the quality of the received continuous media and the protocol's aggressiveness. Our architecture is based on efficient MPEG System filtering to achieve high-quality video over best-effort network

Techniques for End-to-End Tcp Performance Enhancement Over Wireless Networks

Today’s wireless network complexity and the new applications from various user devices call for an in-depth understanding of the mutual performance impact of networks and applications. It includes understanding of the application traffic and network layer protocols to enable end-to-end application performance enhancements over wireless networks. Although Transport Control Protocol (TCP) behavior over wireless networks is well known, it remains as one of the main drivers which may significantly impact the user experience through application performance as well as the network resource utilization, since more than 90% of the internet traffic uses TCP in both wireless and wire-line networks. In this dissertation, we employ application traffic measurement and packet analysis over a commercial Long Term Evolution (LTE) network combined with an in-depth LTE protocol simulation to identify three critical problems that may negatively affect the application performance and wireless network resource utilization: (i) impact of the wireless MAC protocol on the TCP throughput performance, (ii) impact of applications on network resource utilization, and (iii) impact of TCP on throughput performance over wireless networks. We further propose four novel mechanisms to improve the end-to-end application and wireless system performance: (i) an enhanced LTE uplink resource allocation mechanism to reduce network delay and help prevent a TCP timeout, (ii) a new TCP snooping mechanism, which according to our experiments, can save about 20% of system resources by preventing unnecessary video packet transmission through the air interface, and (iii) two Split-TCP protocols: an Enhanced Split-TCP (ES-TCP) and an Advanced Split-TCP (AS-TCP), which significantly improve the application throughput without breaking the end-to-end TCP semantics. Experimental results show that the proposed ES-TCP and AS-TCP protocols can boost the TCP throughput by more than 60% in average, when exercised over a 4G LTE network. Furthermore, the TCP throughput performance improvement may be even superior to 200%, depending on network and usage conditions. We expect that these proposed Split-TCP protocol enhancements, together with the new uplink resource allocation enhancement and the new TCP snooping mechanism may provide even greater performance gains when more advanced radio technologies, such as 5G, are deployed. Thanks to their superior resource utilization efficiency, such advanced radio technologies will put to greater use the techniques and protocol enhancements disclosed through this dissertation