SECURITY MEASUREMENT FOR LTE/SAE NETWORK DURING SINGLE RADIO VOICE CALL CONTINUITY (SRVCC).

Voice has significant place in mobile communication networks. Though data applications have extensively gained in importance over the years but voice is still a major source of revenue for mobile operators. It is obvious that voice will remain an important application even in the era of Long Term Evolution (LTE). Basically LTE is an all-IP data-only transport technology using packet switching. Therefore, it introduces challenges to satisfy quality of service expectations for circuit-switched mobile telephony and SMS for LTE capable smartphones, while being served on the LTE network. Since 2013, mobile operators have been busy deploying Voice Over LTE (VoLTE). They are relying on a VoLTE technology called Single Radio Voice Call Continuity (SRVCC) for seamless handover between packet-switch domain to circuit-switch domain or vice versa. The aim of thesis is to review and identify the security measurement during SRVCC and verify test data for ciphering and integrity algorithm.fi=Opinnäytetyö kokotekstinä PDF-muodossa.|en=Thesis fulltext in PDF format.|sv=Lärdomsprov tillgängligt som fulltext i PDF-format

SECURITY MEASUREMENT FOR LTE/SAE NETWORK DURING SINGLE RADIO VOICE CALL CONTINUITY (SRVCC).

Voice has significant place in mobile communication networks. Though data applications have extensively gained in importance over the years but voice is still a major source of revenue for mobile operators. It is obvious that voice will remain an important application even in the era of Long Term Evolution (LTE). Basically LTE is an all-IP data-only transport technology using packet switching. Therefore, it introduces challenges to satisfy quality of service expectations for circuit-switched mobile telephony and SMS for LTE capable smartphones, while being served on the LTE network. Since 2013, mobile operators have been busy deploying Voice Over LTE (VoLTE). They are relying on a VoLTE technology called Single Radio Voice Call Continuity (SRVCC) for seamless handover between packet-switch domain to circuit-switch domain or vice versa. The aim of thesis is to review and identify the security measurement during SRVCC and verify test data for ciphering and integrity algorithm.fi=Opinnäytetyö kokotekstinä PDF-muodossa.|en=Thesis fulltext in PDF format.|sv=Lärdomsprov tillgängligt som fulltext i PDF-format

Evaluating the Performance of VoIP Over Different Wireless Environment

WiFi and WiMAX are the most technologies for broadband wireless access are used nowadays. The excessive demand for providing mobile users with broadband wireless access has attracted tremendous investment from the telecommunications industry in the development and deployment of WiMAX and WiFi networks. Voice over IP (VoIP) over these technologies will be one of the killer applications for rapid deployment of WiMAX and WiFi networks. The legal desire for bundling voice and data will increase the portion of voice traffic in these networks. Therefore, VoIP, as the current technology for making voice calls through packet switch networks, will be a key application in WiMAX and WiFi networks. The increase of Voice over IP (VoIP) applications such as Skype, Google Talk, and MSN Messenger along with emerging deployment of WiMAX networks is making VoIP over WiMAX an attractive market and a driving force for both carriers and equipment suppliers in capturing and spurring the next wave of telecommunications innovation, though challenges remain. However, the enhancement on the hardware and application sides only seems inadequate. The chosen of proper network environment is also crucial in order to deliver the voice communication and multimedia session over the Internet. Optimization of the VoIP call capacity over WiMAX and WiFi networks is one such crucial challenge and remains an open research issue. Hence, in this project, we present the performance evaluation of VoIP in two wireless network protocols, WiMAX and WiFi as a baseline to evaluate the VoIP performance

Mobile VoIP : managing , scheduling and refining voice packets to and from mobile phones

This thesis report is submitted in partial fulfillment of the requirements for the degree of Bachelor of Science in Computer Science and Engineering, 2006.Cataloged from PDF version of thesis report.Includes bibliographical references (page 19).Voice over IP (VoIP) is rapidly gaining acceptance. VoIP has already become a multi billion dollar industry in the IT world. On the other hand the use of mobile phones in the world has exceeded all estimations. Given these two technologies, we are unfortunate that there is no VoIP support available for mobile phones. The bridge between VoIP and mobile phones is a long way, but an initiative like this paper would surely lay down the roadmap. Mobile phones have several annoying limitations. Limited data transfer rate, processor speed, lack of protocol implementation etc. are the relevant issues for VoIP. This paper addresses the challenges of VoIP and tries to explore ways to overcome those challenges in mobile phones. The language we used for our experiment is JavaME for its simplicity and wide acceptance.Mohammad Abdus SalamTapan Biswas,B. Computer Science and Engineerin

Quality of experience evaluation of voice communication: an affect-based approach

Voice communication systems such as Voice-over IP (VoIP), Public Switched Telephone Networks, and Mobile Telephone Networks, are an integral means of human tele-interaction. These systems pose distinctive challenges due to their unique characteristics such as low volume, burstiness and stringent delay/loss requirements across heterogeneous underlying network technologies. Effective quality evaluation methodologies are important for system development and refinement, particularly by adopting user feedback based measurement. Presently, most of the evaluation models are system-centric (Quality of Service or QoS-based), which questioned us to explore a user-centric (Quality of Experience or QoE-based) approach as a step towards the human-centric paradigm of system design. We research an affect-based QoE evaluation framework which attempts to capture users\u27 perception while they are engaged in voice communication. Our modular approach consists of feature extraction from multiple information sources including various affective cues and different classification procedures such as Support Vector Machines (SVM) and k-Nearest Neighbor (kNN). The experimental study is illustrated in depth with detailed analysis of results. The evidences collected provide the potential feasibility of our approach for QoE evaluation and suggest the consideration of human affective attributes in modeling user experience

A study of mobile VoIP performance in wireless broadband networks

Voice service is to date still the killer mobile service and the main source for operator revenue for years to come. Additionally, voice service will evolve from circuit switched technologies towards packet based Voice over IP (VoIP). However, using VoIP over wireless networks different from 3GPP cellular technologies makes it also a disruptive technology in the traditional telecommunication sector. The focus of this dissertation is on determining mobile VoIP performance in different wireless broadband systems with current state of the art networks, as well as the potential disruption to cellular operators when mobile VoIP is deployed over different access networks. The research method is based on an empirical model. The model and experiments are well documented and based on industry standards for voice quality evaluation. The evaluation provides results from both experiments in a controlled laboratory setup as well as from live scenarios. The research scope is first, evaluate each network technology independently; second, investigate vertical handover mobility cases; third, determine other aspects directly affecting end user experience (e.g., call setup delay and battery lifetime). The main contribution of this work is a systematic examination of mobile VoIP performance and end user experience. The research results point out the main challenges for achieving call toll quality, and how derive the required changes and technological performance roadmap for improved VoIP service. That is, investigate how the performance and usability of mobile VoIP can eventually be improved to be a suitable substitute for circuit switched voice. In addition, we evaluate the potential disruption to cellular operators that mobile VoIP brings when deployed over other access networks. This research extends the available knowledge from simulations and provides an insight into actual end user experience, as well as the challenges of using embedded clients in handheld devices. In addition, we find several issues that are not visible or accounted for in simulations in regard to network parameters, required retransmissions and decreased battery lifetime. The conclusion is that although the network performance of several wireless networks is good enough for near toll quality voice in static scenarios, there are still a number of problems which make it currently unfeasible to use as a primary voice service. Moreover, under mobility scenarios performance is degraded. Finally, there are other issues apart from network performance such as energy consumption, hardware limitations and lack of supporting business models (e.g., for WiFi mesh) that further limit the possibility of rolling out mobile VoIP services