New security and control protocol for VoIP based on steganography and digital watermarking

In this paper new security and control protocol for Voice over Internet Protocol (VoIP) service is presented. It is the alternative for the IETF's (Internet Engineering Task Force) RTCP (Real-Time Control Protocol) for real-time application's traffic. Additionally this solution offers authentication and integrity, it is capable of exchanging and verifying QoS and security parameters. It is based on digital watermarking and steganography that is why it does not consume additional bandwidth and the data transmitted is inseparably bound to the voice content.Comment: 8 pages, 4 figures, 1 tabl

Error resilient packet switched H.264 video telephony over third generation networks.

Real-time video communication over wireless networks is a challenging problem because wireless channels suffer from fading, additive noise and interference, which translate into packet loss and delay. Since modern video encoders deliver video packets with decoding dependencies, packet loss and delay can significantly degrade the video quality at the receiver. Many error resilience mechanisms have been proposed to combat packet loss in wireless networks, but only a few were specifically designed for packet switched video telephony over Third Generation (3G) networks. The first part of the thesis presents an error resilience technique for packet switched video telephony that combines application layer Forward Error Correction (FEC) with rateless codes, Reference Picture Selection (RPS) and cross layer optimization. Rateless codes have lower encoding and decoding computational complexity compared to traditional error correcting codes. One can use them on complexity constrained hand-held devices. Also, their redundancy does not need to be fixed in advance and any number of encoded symbols can be generated on the fly. Reference picture selection is used to limit the effect of spatio-temporal error propagation. Limiting the effect of spatio-temporal error propagation results in better video quality. Cross layer optimization is used to minimize the data loss at the application layer when data is lost at the data link layer. Experimental results on a High Speed Packet Access (HSPA) network simulator for H.264 compressed standard video sequences show that the proposed technique achieves significant Peak Signal to Noise Ratio (PSNR) and Percentage Degraded Video Duration (PDVD) improvements over a state of the art error resilience technique known as Interactive Error Control (IEC), which is a combination of Error Tracking and feedback based Reference Picture Selection. The improvement is obtained at a cost of higher end-to-end delay. The proposed technique is improved by making the FEC (Rateless code) redundancy channel adaptive. Automatic Repeat Request (ARQ) is used to adjust the redundancy of the Rateless codes according to the channel conditions. Experimental results show that the channel adaptive scheme achieves significant PSNR and PDVD improvements over the static scheme for a simulated Long Term Evolution (LTE) network. In the third part of the thesis, the performance of the previous two schemes is improved by making the transmitter predict when rateless decoding will fail. In this case, reference picture selection is invoked early and transmission of encoded symbols for that source block is aborted. Simulations for an LTE network show that this results in video quality improvement and bandwidth savings. In the last part of the thesis, the performance of the adaptive technique is improved by exploiting the history of the wireless channel. In a Rayleigh fading wireless channel, the RLC-PDU losses are correlated under certain conditions. This correlation is exploited to adjust the redundancy of the Rateless code and results in higher Rateless code decoding success rate and higher video quality. Simulations for an LTE network show that the improvement was significant when the packet loss rate in the two wireless links was 10%. To facilitate the implementation of the proposed error resilience techniques in practical scenarios, RTP/UDP/IP level packetization schemes are also proposed for each error resilience technique. Compared to existing work, the proposed error resilience techniques provide better video quality. Also, more emphasis is given to implementation issues in 3G networks

Satellite system performance assessment for in-flight entertainment and air traffic control

Concurrent satellite systems have been proposed for IFE (In-Flight Entertainment) communications, thus demonstrating the capability of satellites to provide multimedia access to users in aircraft cabin. At the same time, an increasing interest in the use of satellite communications for ATC (Air Traffic Control) has been motivated by the increasing load of traditional radio links mainly in the VHF band, and uses the extended capacities the satellite may provide. However, the development of a dedicated satellite system for ATS (Air Traffic Services) and AOC (Airline Operational Communications) seems to be a long-term perspective. The objective of the presented system design is to provide both passenger application traffic access (Internet, GSM) and a high-reliability channel for aeronautical applications using the same satellite links. Due to the constraints in capacity and radio bandwidth allocation, very high frequencies (above 20 GHz) are considered here. The corresponding design implications for the air interface are taken into account and access performances are derived using a dedicated simulation model. Some preliminary results are shown in this paper to demonstrate the technical feasibility of such system design with increased capacity. More details and the open issues will be studied in the future of this research work

Enterprise network convergence: path to cost optimization

During the past two decades, telecommunications has evolved a great deal. In the eighties, people were using television, radio and telephone as their communication systems. Eventually, the introduction of the Internet and the WWW immensely transformed the telecommunications industry. This internet revolution brought about a huge change in the way businesses communicated and operated. Enterprise networks now had an increasing demand for more bandwidth as they started to embrace newer technologies. The requirements of the enterprise networks grew as the applications and services that were used in the network expanded. This stipulation for fast and high performance communication systems has now led to the emergence of converged network solutions. Enterprises across the globe are investigating new ways to implement voice, video, and data over a single network for various reasons – to optimize network costs, to restructure their communication system, to extend next generation networking abilities, or to bridge the gap between their corporate network and the existing technological progress. To date, organizations had multiple network services to support a range of communication needs. Investing in this type of multiple communication infrastructures limits the networks ability to provide resourceful bandwidth optimization services throughout the system. Thus, as the requirements for the corporate networks to handle dynamic traffic grow day by day, the need for a more effective and efficient network arises. A converged network is the solution for enterprises aspiring to employ advanced applications and innovative services. This thesis will emphasize the importance of converging network infrastructure and prove that it leads to cost savings. It discusses the characteristics, architecture, and relevant protocols of the voice, data and video traffic over both traditional infrastructure and converged architecture. While IP-based networks present excellent quality for non real-time data networking, the network by itself is not capable of providing reliable, quality and secure services for real-time traffic. In order for IP networks to perform reliable and timely transmission of real-time data, additional mechanisms to reduce delay, jitter and packet loss are required. Therefore, this thesis will also discuss the important mechanisms for running real-time traffic like voice and video over an IP network. Lastly, it will also provide an example of an enterprise network specifications (voice, video and data), and present an in depth cost analysis of a typical network vs. a converged network to prove that converged infrastructures provide significant savings

A study on the impact of AL-FEC techniques on TV over IP Quality of Experience

Abstract In this contribution, an evaluation of the effectiveness of Application Layer-Forward Error Correction (AL-FEC) scheme in video communications over unreliable channels is presented. In literature, several AL-FEC techniques for reducing the effect of noisy transmission on multimedia communication have been adopted. Recently, their use has been proposed for inclusion in TV over IP broadcasting international standards. The objective of the analysis performed in this paper is to verify the effectiveness of AL-FEC techniques in terms of perceived Quality of Service (QoS) and more in general of Quality of Experience (QoE), and to evaluate the trade-off between AL-FEC redundancy and video quality degradation for a given packet loss ratio. To this goal, several channel error models are investigated (random i.i.d. losses, burst losses, and network congestions) on test sequences encoded at 2 and 4 Mbps. The perceived quality is evaluated by means of three quality metrics: the full-reference objective quality metric NTIA-VQM combined with the ITU-T Rec. G.1070, the full-reference DMOS-KPN metric, and the pixel-wise error comparison performed by using the PSNR distortion measure. A post-processing synchronization between the original and the reconstructed stream has also been designed for improving the fidelity of the performed quality measures. The experimental results show the effectiveness and the limits of the Application Layer protection schemes

A study on the impact of AL-FEC techniques on TV over IP Quality of Experience

Abstract In this contribution, an evaluation of the effectiveness of Application Layer-Forward Error Correction (AL-FEC) scheme in video communications over unreliable channels is presented. In literature, several AL-FEC techniques for reducing the effect of noisy transmission on multimedia communication have been adopted. Recently, their use has been proposed for inclusion in TV over IP broadcasting international standards. The objective of the analysis performed in this paper is to verify the effectiveness of AL-FEC techniques in terms of perceived Quality of Service (QoS) and more in general of Quality of Experience (QoE), and to evaluate the trade-off between AL-FEC redundancy and video quality degradation for a given packet loss ratio. To this goal, several channel error models are investigated (random i.i.d. losses, burst losses, and network congestions) on test sequences encoded at 2 and 4 Mbps. The perceived quality is evaluated by means of three quality metrics: the full-reference objective quality metric NTIA-VQM combined with the ITU-T Rec. G.1070, the full-reference DMOS-KPN metric, and the pixel-wise error comparison performed by using the PSNR distortion measure. A post-processing synchronization between the original and the reconstructed stream has also been designed for improving the fidelity of the performed quality measures. The experimental results show the effectiveness and the limits of the Application Layer protection schemes

Designing and optimization of VOIP PBX infrastructure

In the recent decade, communication has stirred from the old wired medium such as public switched telephone network (PSTN) to the Internet. Present, Voice over Internet Protocol (VoIP) Technology used for communication on internet by means of packet switching technique. Several years ago, an internet protocol (IP) based organism was launched, which is known as Private Branch Exchange "PBX", as a substitute of common PSTN systems. For free communication, probably you must have to be pleased with starting of domestic calls. Although, fairly in few cases, VoIP services can considerably condense our periodical phone bills. For instance, if someone makes frequent global phone calls, VoIP talk service is the actual savings treat which cannot achieve by using regular switched phone. VoIP talk services strength help to trim down your phone bills if you deal with a lot of long-distance (international) and as well as domestic phone calls. However, with the VoIP success, threats and challenges also stay behind. In this dissertation, by penetration testing one will know that how to find network vulnerabilities how to attack them to exploit the network for unhealthy activities and also will know about some security techniques to secure a network. And the results will be achieved by penetration testing will indicate of proven of artefact and would be helpful to enhance the level of network security to build a more secure network in future