A New covert channel over RTP

In this thesis, we designed and implemented a new covert channel over the RTP protocol. The covert channel modifies the timestamp value in the RTP header to send its secret messages. The high frequency of RTP packets allows for a high bitrate covert channel, theoretically up to 350 bps. The broad use of RTP for multimedia applications, including VoIP, provides plentiful opportunities to use this channel. By using the RTP header, many of the challenges present for covert channels using the RTP payload are avoided. Using the reference implementation of this covert channel, bitrates of up to 325 bps were observed. Speed decreases on less reliable networks, though message delivery was flawless with up to 1% RTP packet loss. The channel is very difficult to detect due to expected variations in the timestamp field and the flexible nature of RTP

Prediction of RoHCv1 and RoHCv2 compressor utilities for VoIP

Modern cellular networks utilising the long–term evolution (LTE) and the coming 5G set of standards face an ever–increasing demand for low–latency mobile data from connected devices. Header compression is employed to minimise the overhead for IP–based cellular network traffic, thereby decreasing the overall bandwidth usage and, subsequently, transmission delays. Here, we employ machine learning approaches for the prediction of Robust Header Compression version 1’s and version 2’s compression utility for VoIP transmissions, which allows the compression to dynamically adapt to varying channel conditions. We evaluate various regression models employing r2 and mean square error scores next to complexity (number of coefficients) based on an RTP specific training data set and separately captured live VoIP audio calls. We find that the proposed weighted Ridge regression model explains about at least 50 % of the observed results and the accuracy score may be as high as 94 % for some of the VoIP transmissions

Adaptive header compression techniques for mobile multimedia networks

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Study, Analysis and Modeling of IP Multimedia systems on next generation networks providing mobile and fixed multimedia services

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Practical free-space quantum key distribution

Within the last two decades, the world has seen an exponential increase in the quantity of data traffic exchanged electronically. Currently, the widespread use of classical encryption technology provides tolerable levels of security for data in day to day life. However, with one somewhat impractical exception these technologies are based on mathematical complexity and have never been proven to be secure. Significant advances in mathematics or new computer architectures could render these technologies obsolete in a very short timescale. By contrast, Quantum Key Distribution (or Quantum Cryptography as it is sometimes called) offers a theoretically secure method of cryptographic key generation and exchange which is guaranteed by physical laws. Moreover, the technique is capable of eavesdropper detection during the key exchange process. Much research and development work has been undertaken but most of this work has concentrated on the use of optical fibres as the transmission medium for the quantum channel. This thesis discusses the requirements, theoretical basis and practical development of a compact, free-space transmission quantum key distribution system from inception to system tests. Experiments conducted over several distances are outlined which verify the feasibility of quantum key distribution operating continuously over ranges from metres to intercity distances and finally to global reach via the use of satellites

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