Using Transcoding for Hidden Communication in IP Telephony

The paper presents a new steganographic method for IP telephony called TranSteg (Transcoding Steganography). Typically, in steganographic communication it is advised for covert data to be compressed in order to limit its size. In TranSteg it is the overt data that is compressed to make space for the steganogram. The main innovation of TranSteg is to, for a chosen voice stream, find a codec that will result in a similar voice quality but smaller voice payload size than the originally selected. Then, the voice stream is transcoded. At this step the original voice payload size is intentionally unaltered and the change of the codec is not indicated. Instead, after placing the transcoded voice payload, the remaining free space is filled with hidden data. TranSteg proof of concept implementation was designed and developed. The obtained experimental results are enclosed in this paper. They prove that the proposed method is feasible and offers a high steganographic bandwidth. TranSteg detection is difficult to perform when performing inspection in a single network localisation.Comment: 17 pages, 16 figures, 4 table

AN ANALYSIS OF VOICE OVER INTERNET PROTOCOL (VOIP) AND ITS SECURITY IMPLEMENTATION

Voice over Internet Protocol (VoIP) has been in existence for a number of years but only quite recently has it developed into mass adoption. As VoIP technology penetrates worldwide telecommunications markets, the advancements achieved in performance, cost reduction, and feature supportmake VoIP a convincingproposition for service providers, equipment manufacturers, and end users. Since the introduction of mass-market VoIP services over broadband Internet in 2004, security and safeguarding are becoming a more important obligation in VoIP solutions. The purpose of this final year project is to study and analyze VoIP and implement the security aspect using Secure Real-time Transport Protocol (SRTP) end-to-end media encryption in the Universiti Teknologi PETRONAS (UTP) laboratory. Extensive research, evaluation of case studies, literature reviews, network analysis, as well as testing and experimentation are the methods employed in achieving a secure and reliable VoIP network. With the given time frame and adequate resources, the study and analysis of VoIP and implementation of SRTP should prove to be very successful

SECURING USER INTERACTION CHANNELS ON MOBILE PLATFORM USING ARM TRUSTZONE

Smartphones have become an essential part of our lives, and are used daily forimportant tasks like banking, shopping, and making phone calls. Smartphones provide several interaction channels which can be affected by a compromised mobile OS. This dissertation focuses on the user interaction channels of UI input and audio I/O. The security of the software running on smartphones has become more critical because of widespread smartphone usage. A technology called TEE (Trusted Execution Environment) has been introduced to help protect users in the event of OS compromise, with the most commonly deployed TEE on mobile devices being ARM TrustZone. This dissertation utilizes ARM TrustZone to provide secure design for user interactionchannels of UI input (called Truz-UI) and Audio I/O for VoIP calls (called Truz-Call). The primary goal is to ensure that the design is transparent to mobile applications. During research based on TEE, one of the important challenges that is encountered is the ability to prototype a secure design. In TEE research one often needs to interface hardware peripherals with the TEE OS, which can be challenging for non-hardware experts, depending on the available support from the TEE OS vendor. This dissertation discusses a simulation based approach (called Truz-Sim) that reduces setup time and hardware experience required to build a hardware environment for TEE prototyping

Performance evaluation of a technology independent security gateway for Next Generation Networks

With the all IP based Next Generation Networks being deployed around the world, the use of real-time multimedia service applications is being extended from normal daily communications to emergency situations. However, currently different emergency providers utilise differing networks and different technologies. As such, conversations could be terminated at the setup phase or data could be transmitted in plaintext should incompatibility issues exit between terminals. To this end, a novel security gateway that can provide the necessary security support for incompatible terminals was proposed, developed and implemented to ensure the successful establishment of secure real-time multimedia conversations. A series of experiments were conducted to evaluate the security gateway through the use 40 Boghe softphone acting as the terminals. The experimental results demonstrate that the best performance of the prototype was achieved by utilising a multithreading and multi-buffering technique, with an average of 582 microseconds processing overhead. Based upon the ITU-Ts 150 milliseconds one way delay recommendation for voice communications, it is envisaged that such a marginal overhead will not be noticed by users in practice

Securing media streams in an Asterisk-based environment and evaluating the resulting performance cost

When adding Confidentiality, Integrity and Availability (CIA) to a multi-user VoIP (Voice over IP) system, performance and quality are at risk. The aim of this study is twofold. Firstly, it describes current methods suitable to secure voice streams within a VoIP system and make them available in an Asterisk-based VoIP environment. (Asterisk is a well established, open-source, TDM/VoIP PBX.) Secondly, this study evaluates the performance cost incurred after implementing each security method within the Asterisk-based system, using a special testbed suite, named DRAPA, which was developed expressly for this study. The three security methods implemented and studied were IPSec (Internet Protocol Security), SRTP (Secure Real-time Transport Protocol), and SIAX2 (Secure Inter-Asterisk eXchange 2 protocol). From the experiments, it was found that bandwidth and CPU usage were significantly affected by the addition of CIA. In ranking the three security methods in terms of these two resources, it was found that SRTP incurs the least bandwidth overhead, followed by SIAX2 and then IPSec. Where CPU utilisation is concerned, it was found that SIAX2 incurs the least overhead, followed by IPSec, and then SRTP

Subjective Audio Quality over a Secure IEEE 802.11n Draft 2.0 Wireless Local Area Network

This thesis investigates the quality of audio generated by a G.711 codec and transmission over an IEEE 802.11n draft 2.0 wireless local area network (WLAN). Decline in audio quality due to additional calls or by securing the WLAN with transport mode Internet Protocol Security (IPsec) is quantified. Audio quality over an IEEE 802.11n draft 2.0 WLAN is also compared to that of IEEE 802.11b and IEEE 802.11g WLANs under the same conditions. Audio quality is evaluated by following International Telecommunication Union Telecommunication Standardization Sector (ITU-T) Recommendation P.800, where human subjects rate audio clips recorded during various WLAN configurations. The Mean Opinion Score (MOS) is calculated as the average audio quality score given for each WLAN configuration. An 85% confidence interval is calculated for each MOS. Results suggest that audio quality over an IEEE 802.11n draft 2.0 WLAN is not higher than over an IEEE 802.11b WLAN when up to 10 simultaneous G.711 calls occur. A linear regression of the subjective scores also suggest that an IEEE 802.11n draft 2.0 WLAN can sustain an MOS greater than 3.0 (fair quality) for up to 75 simultaneous G.711 calls secured with WPA2, or up to 40 calls secured with both WPA2 and transport mode IPsec. The data strongly suggest that toll quality audio (MOS ≥ 4.0) is not currently practical over IEEE 802.11 WLANs secured with WPA2, even with the G.711 codec

A Comprehensive Survey of Voice over IP Security Research

We present a comprehensive survey of Voice over IP security academic research, using a set of 245 publications forming a closed cross-citation set. We classify these papers according to an extended version of the VoIP Security Alliance (VoIPSA) Threat Taxonomy. Our goal is to provide a roadmap for researchers seeking to understand existing capabilities and to identify gaps in addressing the numerous threats and vulnerabilities present in VoIP systems. We discuss the implications of our findings with respect to vulnerabilities reported in a variety of VoIP products. We identify two specific problem areas (denial of service, and service abuse) as requiring significant more attention from the research community. We also find that the overwhelming majority of the surveyed work takes a black box view of VoIP systems that avoids examining their internal structure and implementation. Such an approach may miss the mark in terms of addressing the main sources of vulnerabilities, i.e., implementation bugs and misconfigurations. Finally, we argue for further work on understanding cross-protocol and cross-mechanism vulnerabilities (emergent properties), which are the byproduct of a highly complex system-of-systems and an indication of the issues in future large-scale systems