Secure End-to-End Communications in Mobile Networks

2009 - 2010Cellular communication has become an important part of our daily life. Besides using cell phones for voice communication, we are now able to access the Internet, conduct monetary transactions, send voice, video and text messages and new services continue to be added. The frequencies over which voice is transmitted are public, so voice encryption is necessary to avoid interception of the signal over the air. But once the signal reaches the operators Base Station (BS), it will be transmitted to the receiver over a wired or wireless mean. In either case, no protection is de ned. This does not seem a problem, but this is not true. Along the path across operator network, voice is at risk. It will only be encrypted again, with a di erent key, from the BS to the receiver if the receiver is herself a mobile user. Moreover, voice encryption is not mandatory. The choice whether or not to accept an unprotected communication is up to the network. When adopted, the same encryption algorithm is used for sending SMS messages between mobile telephones and base stations and for encrypting of calls. Unfortunately, vulnerabilities in this encryption systems were already revealed more than 10 years ago and more continue to be discovered. Currently the most popular communication technologies are the GSM and the UMTS. The UMTS is in use as a successor to GSM. Along with mobile phone services, It provides rapid data communication. The security algo- rithms in UMTS di ers from GSM in two important ways: encryption and mutual authentication. Although security standards have been improved, the end- to-end security is not provided... [edited by Author]IX n.s

Video Quality Measurement For 3G Handsets

Internet provides many services. VOIP (Voice over IP) is one such service also known as Internet Telephony or IP Telephony. Using VOIP we can make voice telephony calls, participate in video conferences, etc over data networks (WAN'S and LAN'S) or internet. VOIP operates by first converting voice data into digital form, organizing them into packets, transmitting them through the most convenient route to their destination and finally reassembling them at the destination. Protocols like SIP/RTP, H.323, MGCP are designed which perform all the above steps. This project aims to make a video call from a 3G Mobile to an IP phone via Asterisk Gateway. Asterisk to act as bridge for video call between 3G-IP network must capture the audio/video stream from 3G mobile, convert captured stream into an IP compatible stream and send stream to an IP client and vice-versa. Asterisk needs to support AMR codec for audio and MPEG-4 codec for video and H.324M protocol stack for capturing audio/video streams from 3G Mobile. Asterisk currently supports audio codec's like GSM, G.729, A-law, and U-law. It allows H.261, H.263 video streams as pass-through. It supports VOIP protocols like SIP/RTP, MGCP, and H.323 which allows it to interface with other devices. This project aims to implement AMR codec, H.324M protocol stack, MPEG-4, bridging functions between SIP/RTP-ISDN and 3G Mobile in Asterisk which allows a 3G phone to call a SIP client via Asterisk. This thesis discusses the implementation of AMR in asterisk as well as SIP protocol and SIP soft phones

Video Quality Measurement for 3G Handset

The quality of video has become a decisive factor for the consumer of 3G video services to choose his mobile operator. It is, therefore, critical for 3G network operator, equipment provider and service provider to measure and hence maintain the video quality of video services they offer. A project has been proposed in University of Plymouth to develop a test platform to evaluate video quality for 3G handset using Asterisk PBX server. For this purpose, support for 3G-324M protocol and all the audio and video codecs (i.e. H.263 baseline level 10 and MPEG-4 simple profile @ level 0) mandated and recommended by 3G- 324M standard should be added in to Asterisk®. The purpose of this thesis is to identify the correct software implementation of H.263 baseline level 10 and MEPG-4 simple profile @ level 0 video codecs so that they can then be incorporated in to Asterisk®. This is the part of the above mentioned project. Open source FFmpeg-libavcodec is believed to support both MPEG-4 and H.263 codecs. Similarly Telenor H.263 codec is also free to use. This project tests both the capabilities and suitability of the above mentioned software packages/codecs for adding in to Asterisk to perform the required encoding and decoding. Experiments showed that FFmpeg-libavcodec can neither decode nor encode to MPEG-4 simple profile @ level 0. It seems that FFmpeg requires some major modifications in its source code to support MPEG-4 simple profile @ level 0 codec. Although FFmepg can decode and encode to H.263 baseline level 10, but it does not offer a fine control over bitrate while encoding, and reports very high muxing overhead while decoding, H.263 baseline level 10. Telenor H.263 codec can decode and encode to H.263 baseline level 10.without any problem. Telenor H.263 codec is, therefore, more suitable for incorporating in to Asterisk® than FFmpeg for decoding and encoding to H.263 baseline level 10 bitstreams. ISchool of Computing, Communication and Electronic

Software defined radio testbed of television white space for video transmission

Recently, television white space (TVWS) has grabbed a lot of attention from researchers in the Cognitive Radio (CR) area. This underutilized spectrum is one of the possible solutions for spectrum scarcity problem in wireless communication. Thus, many research works have been carried out in order to find a suitable method to utilize this spectrum in an efficient manner. Nevertheless, the actual hardware implementation on utilizing this spectrum is still lacking. Therefore, in this research, an Orthogonal Frequency Division Multiplexing (OFDM) real-time video transmission is proposed using software defined radio (SDR) platform. Two modulation schemes are used namely Phase-shift keying (PSK) with its Binary-PSK (BPSK) and Quadrature-PSK (QPSK) and Quadrature amplitude modulation (QAM) with 16QAM and 64QAM modes. The free channel used in this work is selected under ultra high frequency (UHF) band based on the energy detection, which is either on channel 54 or channel 56. The proposed system is developed with the physical (PHY) layer design of the transmitter and receiver in GNU Radio and integration of medium access control (MAC) layer functionality. Video capture and display programs are designed based on OpenCV modules. The performance of this design is evaluated based on two types of environment, indoor and outdoor, with packet delivery ratio (PDR) and end-to-end delay (EED) as the performance metrics. Three types of video motion are used in the experimentation which are fast (mobile), medium (foreman) and slow (akiyo). Under allocated bandwidth of 1.0 MHz, optimal performances of PDR and EED for both scenarios are shown. In the indoor scenario, QPSK½ exhibits the best performance with 0.92 of PDR and 24.7 seconds of EED for akiyo. Meanwhile for foreman and mobile, BPSK¾ achieves the best performance with PDR of 0.96 and 0.95 and EED of 33.2 seconds and 35.0 seconds, respectively. In the outdoor scenario, the best performance of PDR is achieved by 16QAM½ with 0.9 and 23.5 seconds of EED for akiyo. For foreman and mobile, QPSK½ exhibits the best performance with 0.94 and 0.9 of PDR and 31.2 seconds and 32.5 seconds of EED, respectively. In conclusion, the proposed design exhibits promising solutions for the OFDM real-time video transmission over TVWS

Reliable multimedia transmission over wireless sensor network

Nowadays, video streaming application is widely used in wired as well as wireless environment. Extending this application into Wireless Sensor Networks (WSN) for IEEE 802.15.4 network has attracted lots of attention in the research community. Reliable data transmission is one of the most important requirements in WSN especially for multimedia application. Moreover, multimedia application requires high bandwidth and consumes large memory size in order to send video data that requires small end-to-end (ETE) delay. To overcome this problem, rate control serves as an important technique to control the bit rate of encoded video for transmission over a channel of limited bandwidth and low data rate which is 250kbps with small Maximum Transmission Unit (MTU) size of 127 bytes. Therefore, a rate control model called enhanced Video Motion Classification based (e-ViMoC) model using an optimal combination of parameter setting is proposed in this thesis. Another challenging task to maintain the video quality is the design of an enhanced transport protocol. Standard transport protocols cannot be directly applied in WSN specifically, but some modifications are required. Therefore, to achieve high reliability of video transmission, the advantages of User Datagram Protocol (UDP) features are applied to the proposed transport protocol called Lightweight Reliable Transport Protocol (LRTP) to tailor to the low data rate requirement of WSN. Besides, priority queue scheme is adopted to reduce the end-to-end delay while maintaining the reliability and energy efficiency. Evalvid simulation tool and exhaustive search method are used to determine optimal combination of quantization scale (q), frame rate (r) and Group of Picture (GOP) size (l) values to control the bit rate at the video encoder. The model of e-ViMoC is verified both with simulation and experimental work. The proposed transport protocol has been successfully studied and verified through simulation using Network Simulator 2 (NS-2). From the simulation results, the proposed e-ViMoC encoded video enhances the Packet Delivery Ratio (PDR) by 5.14%, reduces the energy consumed by 16.37%, improves the Peak Signal to Noise Ratio (PSNR) by 4.37% and reduces the ETE delay by 23.69% in average, compared to non-optimized encoded video. The tested experiment experiences slightly different result where the PDR is 6% lower than simulation results. Meanwhile, the combination of e-ViMoC and LRTP outperforms the standard transport protocol by average improvement of 142.9% for PDR, average reduction of 8.87% for energy consumption, average improvement of 4.1% for PSNR, and average reduction of 19.38% for ETE delay. Thus, the simulation results show that the combination of proposed e-ViMoC and LRTP provides better reliability performance in terms of the PDR while simultaneously improves the energy efficiency, the video quality and ETE delay

Multimedia in mobile networks: Streaming techniques, optimization and User Experience

1.UMTS overview and User Experience 2.Streaming Service & Streaming Platform 3.Quality of Service 4.Mpeg-4 5.Test Methodology & testing architecture 6.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

Performance evaluation of information and communications technology infrastructure for smart distribution network applications

This thesis was submitted for the degree of Master of Philosophy and awarded by Brunel University.Current electrical networks require secure, scalable and cost-effective Information and Communications Technology (ICT) solutions to facilitate the novel functionalities required by Smart Grids. Countries around the globe are investigating alternative energy sources to mitigate the current energy crisis and environmental issues experienced by many countries due to global warming, rapid growth of population, inefficient energy management, dwindling fossil fuel resources, etc. Therefore, alternative or renewable energy sources, such as wind, solar, hydro, combined heat and power, etc., are required to mitigate such a crisis and such sources will also need to be integrated in to the power grid in a distributed manner. Such distributed energy sources are mainly connected to the distribution networks and introduce huge challenges to the distribution network operator (DNO). Many of these challenges cannot be dealt with effectively using existing network operation mechanisms therefore the research and development of novel ICT solutions to support smart distribution network operation is required. This research investigated suitable ICT solutions to enable the Smart Grid to tackle these challenges and proposes ICT infrastructure models that can be used for simulation studies in order to investigate cost-effective, scalable and secure solutions for the DNOs. Initially, a Quality of Service (QoS) monitoring test-bed was proposed to evaluate the performance of bandwidth intensive applications, such as smart meter data transmission. Simulation studies for different communication technologies, cellular and Power Line Communication (PLC), were also carried out and the simulation models were verified using experimental test results. Finally, the modelling and analysis of smart metering infrastructure was carried out using simulation and extensive studies were performed to evaluate the data transmission rate performance for different configurations of smart meters and concentrators

Evaluating the effectiveness of mobile telecommunication services in Durban and Lagos.

Thesis (M.Com.)-University of KwaZulu-Natal, Westville, 2011.Telecommunication includes voice, video and internet communication services. Thus, mobile telecommunication services involve voice communication, video streaming, graphics and television services at high speed. Technology development in mobile telecommunication has enabled users to exchange data using cell phones, laptops and other telecommunication devices. More so, understanding the concept of user experience is very important in the context of provision of mobile telecommunication services. This research will evaluate the effectiveness of mobile telecommunication services in Durban, South Africa and Lagos, Nigeria amongst first-year IT students of the University of KwaZulu-Natal and Lagos State University. The research is focused on the actual experience and perceptions of first-year IT students. The study will examine the factors that influence first-year IT students' judgment of the quality of mobile telecommunication services. It will also access the impact of quality of mobile telecommunication services on the loyalty of first-year IT students towards their network operator. However, Technology Acceptance Model (TAM) is the theory adopted for this research, which explains how attitude of users determine the intention to use technology and the intention eventually influences the overall use of such technology. The objectives of this research highlight opportunities associated with understanding first-year IT students' experiences and perceptions of mobile telecommunication services in UKZN, Durban and LASU, Lagos. Other opportunities include giving an insight into the operations of network providers, determine the quality of mobile telecommunication services and understanding the impact of mobile telecommunication services on students in UKZN and LASU. Another significance of this study allows network providers to understand students' behaviour and to respond to their preferences