Extending AES with DH Key-Exchange to Enhance VoIP Encryption in Mobile Networks

Recently,the evolution and progress have become significant in the field of information technology and mobile technology, especially inSmartphone applications that are currently widely spreading. Due to the huge developments in mobile and smartphone technologies in recent years, more attention is given to voice data transmission such as VoIP (Voice overIP) technologies– e.g. (WhatsApp, Skype, and Face Book Messenger). When using VoIP services over smartphones, there are always security and privacy concerns like the eavesdropping of calls between the communicating parties. Therefore, there is a pressing need to address these risks by enhancing the security level and encryption methods. In this work, we use scheme to encrypt VoIP channels using (128, 192 & 256-bit) enhanced encryption based on the Advanced Encryption Standard (AES) algorithm, by extending it with the well-known Diffie-Hellman (DH) key exchange method. We have performed a series of real tests on the enhanced (AES-DH) algorithm and compared its performance with the generic AES algorithm. The results have shown that we can get a significant increase in the encryption strength at a very small overhead between 4% and 7% of execution timebetween AES and AEScombine withDH for all scenario which was incurred by added time of encryption and decryption. Our approach uses high security and speed and reduces the voice delay.In dealing with sound transfer process via the internet, we use the SIP server to authenticate the communication process between the two parties. The implementation is done on a mobile device (Which is operated by (Android) system) because it has recently been widely used among different people around the world.اصبحت الثورة والتطور كبيرة حديثاً في حقول تكنولوجيا االتصاالت واليواتف النقالة، وخصوصا في تطبيقات اليواتف الذكية التي تنتشر حاليا بشكل واسع. وتم اعطاء المزيد من االىتمام لنقل البيانات الصوتية مثل تكنولوجيا االتصال عبر بروتكول االنترنت، عمى سبيل المثال: )الواتساب، السكايب، الفيس بوك، والماسنجر(. ويعزى ذلك لمتطور الكبير في تكنولوجيا اليواتف النقالة والذكية في السنوات االخيرة. عند استخدام خدمات االتصال عبر بروتكول االنترنت VoIP ،ىناك مخاوف دائمة حول الحماية والخصوصية كالتجسس عمى المكالمات بين جيات االتصال. ولذلك ىنالك حاجة ماسة لمعالجة ىذه المخاطر عن طريق تحسين مستوى الحماية وطرق التشفير. في ىذا العمل، نستخدم/ نجمع بين اثنتين من الخوارزميات لتشفير قنوات االتصال عبر بروتوكول االنترنت )128 ، 192 ،و 256 بت( عبر خوارزمية AESوتمديدىا عبر طريقة تبادل ديفي ىيممان الرئيسية المعروفة. وقمنا باداء العديد من التجارب الحقيقية عمى DH-AES ، وقمنا بمقارنة ادائيا مع اداء خوارزمية معيار التشفير المتقدم العامة. اظيرت النتائج انو بامكاننا الحصول عمى زيادة كبيرة في قوة التشفير بنسبة صغيرة جدا بين 4 %و7 %من وقت التنفيذ بين AESو DH/AES لجميع السيناريو والتي تم تكبدىا من قبل الوقت المضاف لمتشفير وفك التشفير. يستخدم نيجنا درجة عالية من الحماية والسرعة ويقمل من تأخير الصوت، ونستخدم في التعامل مع عممية نقل الصوت عبر االنترنت Server SIPلتوثيق عممية االتصال بين الجيتين. وتم التنفيذ عمى ىاتف نقال يعمل عمى نظام اندرويد؛ النو استخدم بشكل واسع مؤخرا بين مختمف الناس حول العالم

Developing Best Practices for Securing VoIP Communication for a non-profit Organization

Voice over Internet Protocol (VoIP) is the most widely used service around the world. The proficiency of it utilizing the web has increased awesome ubiquity in the current years. With this notoriety, there is expanding worry about the wellbeing of the system. The robbery or loss of the information being exchanged is great concern. For example, a basic problem for researchers who are developing safeguards for VoIP systems is the level of threats and other issues experienced by the non-profit organizations while implementing VoIP communication. This problem originated when non-profits received pressure from their donors not to implement VoIP communication because it will record important and valuable information of their bank account, including their bank balance, and consequently, exposing them to the public. Other dangers include safeguarding secrecy, respectability, and accessibility of the system, known as CIA. dangers. To battle these dangers, some security conventions and calculations have been produced. For example, the H.235 has been investigated, their calculations updated, and it is currently regarded as the most recent and effective system for security of the VoIP system. Another method for battling issues and concerns, and one that is the most proficient due to bigger budgets than non-profits, is VoIP being utilized in new structures and the IT work force. Fortunately, the expanding interest of VoIP has guaranteed and emphasized the requirement for more research to build up the effective security structures and countermeasures of CIA threats. This investigation examines the methods by which such security issues concerning VoIP can be set out to give an appropriate, secure and effective method for correspondence and data trade. In this postulation, the analyst will profoundly examine the relief of VoIP security issues

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

A novel mechanism for anonymizing Global System for Mobile Communications calls using a resource-based Session Initiation Protocol community network

Considering the widespread adoption of smartphones in mobile communications and the well-established resource sharing use in the networking community, we present a novel mechanism to achieve anonymity in the Global System for Mobile Communications (GSM). We propose a Voice over Internet Protocol infrastructure using the Session Initiation Protocol (SIP) where a smartphone registers on a SIP registrar and can start GSM conversation through another smartphone acting as a GSM gateway, by using a SIP intermediate without an extra cost. The testbed that we developed for empirical evaluation revealed no significant quality of service degradation

Integrating Voice over IP Solution in IPv6 and IPv4 Networks to Increase Employee Productivity: A Case Study of Cameroon Telecommunications (Camtel), North-West

Telecommunications organizations have to follow the rapid innovation of technology if they want to face challenges raised by competition. The challenge to respond to the huge market demand of updated products and services from customers requires that the organization‘s working environment be equipped with tools and communication facilities that contribute to ameliorating productivity. Cameroon Telecommunications (Camtel) is facing a digital telephony and Internet Protocol strategic management challenge. Successful implementation cannot be achieved if the employees are still depending on the ageing public switched telephone network (PSTN) as their primary communication system, despite the frequent loss of dial tone experience in a day which can last up to a week, with serious repercussions on business activities and revenues. This study is designed to provide a solution to the telecommunications challenge. The fundamental question is how to integrate a digital telephony system that will provide telephony services in the existing IPv4 data network while prioritizing IPv6 traffic forwarding. This study proposes and implements solutions that integrate a Voice over IP solution with IPv6 as an alternative communication system that relies on the existing IPv4 data network. VoIP is deemed as one of the driving forces behind the adoption of IPv6. The purpose is to offer to workers an option that will free them from the poor Quality of Service (QoS) of their existing PSTN based solution, hopefully enhancing the overall productivity. This paper follows two research methodologies: Qualitative Research in Applied Situations and Engineering design process. The first part of this study reports the results of the evaluation of how much such a solution can enhance workers’ productivity. As it is important to provide an environment where IPv4 and IPv6 networks and applications/devices can interoperate in the context of VoIP; the second part describes practically a simulation environment where various configurations of network entities are done following a Dual-Stack transition approach. Document and records were used to gather information related to the structure, operations, and topological update of the Camtel’s existing IP data network. The findings demonstrated that VoIP can be an effective communication solution for Camtel and its implementation with IPv6 will be preferable. However, for this to be efficient there must be a provision of sufficient bandwidth and usage of types of equipment and transmission mediums that minimizes processing and propagation delays. Findings also reveal that better productivity will be achieved if workers are fully trained for the exploitation. This research article tries to highlight, discuss a required transition roadmap and extend the local knowledge and practice on IPv6. Future expansion of this research work will consist of deploying Dual-Stack VoIP in the remaining 9 regional offices for full integration in the corporate communication system of Camtel

Audio steganography with AES for real-time covert Voice over Internet Protocol communications

As a popular real-time service on the Internet, Voice over Internet Protocol (VoIP) communication attracts more and more attention from the researchers in the information security field. In this study, we proposed a VoIP steganographic algorithm with variable embedding capacities, incorporating AES and key distribution, to realize a real-time covert VoIP communication. The covert communication system was implemented by embedding a secret message encrypted with symmetric cryptography AES-128 into audio signals encoded by PCM codec. At the beginning of each VoIP call, a symmetric session key (SK) was assigned to the receiver with a Session Initiation Protocol-based authentication method. The secret message was encrypted and then embedded into audio packets with different embedding algorithms before sending them, so as to meet the real-time requirements of VoIP communications. For each audio packet, the embedding capacity was calculated according to the specific embedding algorithm used. The encryption and embedding processes were almost synchronized. The time cost of encryption was so short that it could be ignored. As a result of AES-based steganography, observers could not detect the hidden message using simple statistical analysis. At the receiving end, the corresponding algorithm along with the SK was employed to retrieve the original secret message from the audio signals. Performance evaluation with state-of-the-art network equipment and security tests conducted using the Mann-Whitney-Wilcoxon method indicated that the proposed steganographic algorithm is secure, effective, and robust