Modeling Of Subscriber Station Energy Consumption Network 802.16

Worldwide Interoperability for Microwave Access (WiMAX) is a Broadband Wireless Access (BWA) technology that has high speed and wide access for multimedia communication. The reception of quality video or voice packets at the receiver is related to how efficient the energy consumption is sent by the Subscriber Station (SS). The more video or voice packets that are sent, the more energy consumption is needed. One way to calculate energy consumption at a WiMAX subscriber station is by using mathematical modeling. This final project is modeling energy consumption in WiMAX. From the simulation, it is found that the change in the bit rate affects the energy consumption of the subscriber station. Through simulation, training data is generated to obtain a mathematical model.The mathematical model contains the components of the state duration and the level of energy consumption. Mathematical models are then used to predict energy consumption in WiMAX. The model is tested through the generation of test data. The test results through simulation showthe percentage deviation of the average energy consumption of training data with mathematical modeling on average is 0.180%. Meanwhile, the energy consumption of test data and mathematical models has an average deviation of 0.187% and 0.191%. The least energy consumption is generated when the MAC Initialization state (state i0) is 0.001 Joule, while the highest energy consumption is the Uplink Frame state (state i14) of 101.017226 Joule.

Coverless VoIP Steganography Using Hash and Hash

Performing secure and robust embedding and extracting in real time voice streams without deteriorating the voice quality is a great challenge. This paper aims on hiding the secret data bits in the voice packets without modifying any data in the cover thereby improving the embedding transparency and becomes robust against the steganalysis attacks using coverless approach. Initially a hash array is built with the frame size. The cover bit position is identified from the hashing function. The hash array is marked with a flag value to indicate that the particular sample consist of the secret message bit. The hash array is attached with the VoIP samples, at the receiver side the hash table is separated, and the secret bits are extracted based on the hash array. The experimental results conducted on a VoIP prototype proved to be simpler and effective in terms of the computational complexity, undetectability and voice quality at both sender and receiver end.

Simulation and Formal Verification of SIP/ZRTP Protocol using UPPAAL

The security of voice communication over the Internet Protocol is a continuously growing research area due to the rapid rise in its usage among consumers. With the advent of Voice-over-IP Protocols, the Real Time Protocol (RTP) was used to facilitate VoIP communications. To secure this communication, Secure Real Time Protocol (SRTP) was implemented to encrypt these voice packets. The SRTP requires a session key to be shared between the communicating entities. The challenging task of establishing a new, unused session key to secure each SRTP session was overcome by the key agreement protocol, Zimmermann Real-time Transport Protocol (ZRTP) which ensures confidentiality as well as a shield against Man-in-the-Middle attack. We firstly analyze the security properties of this protocol. Formal analysis is a mathematical technique that can be used to verify the correctness of the system. We simulated the complete ZRTP Protocol with the well-known formal analysis tool, Uppaal, and verified the existing security properties such as Deadlock Prevention, Liveliness, Safety and other protocol parameters mismatch detection using the Uppaal model checker engine. Temporal logic was used to design the queries to verify the properties.

Performance Evaluation of Distributed Multi Media Wireless Sensor Network

The demand for multimedia services i.e. audio, video and data with improve QoS and optimum utilization of resources in WSN’s has posed new challenges. As the intensity of traffic increases; it demands for higher bandwidth and dedicated resources to reduce packet loss and delay. There have been analytical models proposed where priorities were assigned to video and voice packets to reduce packet loss and optimize resource utilization. In this paper distributed scheme is proposed to handle video, voice and data packets by having multiple sink nodes. There are shared sink nodes where video, voice and data packets are serviced and dedicated sink nodes only for video and voice packets. The proposed scheme has shown that the packet loss for data packets is higher than voice and video packets. The simulation results show that the performance of the network is improved when priorities are assigned to video and voice packets by giving dedicated resources.

A Statistical Approach to Adaptive Playout Scheduling in Voice Over Internet Protocol Communication

Factors like network delay, latency and bandwidth significantly affect the quality of communication using Voice over Internet Protocol. The use of jitter buffer at the receiving end compensates the effect of varying network delay up to some extent. But the extra buffer delay given for each packet plays a major role in playing late packets and thereby improving voice quality. As the buffer delay increases packet loss rate decreases, which in general is a very good sign. However, an increase of buffer delay beyond a certain limit affects the interactive quality of voice communication. In this paper, we propose a statistical framework for adaptive playout scheduling of voice packets based on network statistics, packet loss rate and availability of packets in the buffer. Experimental results show that the proposed model allocates optimal buffer delay with the lowest packet loss rate when compared with other algorithms.

A New Efficient Authenticated and Key Agreement Scheme for SIP Using Digital Signature Algorithm on Elliptic Curves

Voice over Internet Protocol (VoIP) has been recently one of the more popular applications in Internet technology. It benefits lower cost of equipment, operation, and better integration with data applications than voice communications over telephone networks. However, the voice packets delivered over the Internet are not protected. The session initiation protocol (SIP) is widely used signaling protocol that controls communications on the Internet, typically using hypertext transport protocol (HTTP) digest authentication, which is vulnerable to many forms of attacks. This paper proposes a new secure authentication and key agreement scheme based on Digital Signature Algorithm (DSA) and Elliptic Curve Cryptography (ECC) named (ECDSA). Security analysis demonstrates that the proposed scheme can resist various attacks and it can be applied to authenticate the users with different SIP domains.

HIDING SECRET DATA THROUGH STEGANOGRAPHY IN VOIP

Steganography is an effective way of hiding secret data, by this means of protecting the data from unauthorized or unwanted viewing. Using cryptography technique will encrypt and decrypt message to provide better security. Cryptography protects the message from being read by unauthorized parties, steganography lets the sender conceal the fact that he has even sent a message. One of the new and promising communication medium that can be used as steganography is the Voice over Internet Protocol. VOIP covers a wide range of information hiding techniques. The main idea is to use free unused fields of VOIP protocols like TCP, UDP etc. By hiding one secret text into the cover speech using steganography we can get a stego speech, which sounds indistinguishable from the original cover speech. So even if the Hackers/crackers catch the audio packets on network, they would not notice that there is some secret text hidden inside it. To develop a Voice Chat Tool, this can also enable us to send secret data hidden inside the voice packets at the same time. We used LSB method of steganography and for better security we provide encryption to the message to be sent. There is no restriction on the length of message as more the communicators talk larger the file is sent. Human auditory system (HAS) operates over a wide dynamic range. It is challenging to hide secret data inside audio.

QoS and Performance Evaluation for SIP-Based VoIP Over DMO

In this chapter, analyses for the performance metrics that define the quality of service (QoS) of SIP-based VoIP will be introduced. SIP-based VoIP applications over Direct Mode of Operation (DMO), which behaves in a way similar to Mobile Ad-hoc Network (MANET) systems, have three main performance categories related to the QoS. These categories are the SIP signaling, voice data transmission, and MANET routing. The SIP signaling controls the VoIP calls initiation, termination, and modifications. The major QoS parameters of VoIP that are managed by SIP signaling are the registration intervals, call setup time, and call termination time. These QoS parameters are increased in MANET due to the nodes' mobility that affects the routing calculations and the connectivity status. These necessitate mechanisms to reduce the delays in the MANET environment. The voice packets are transferred over the Real Time Protocol (RTP) which is encapsulated in the unreliable transport protocol using the User Datagram Protocol (UDP).

Influence of the QoS Measures for VoIP Traffic in a Congested Network

The paper revolves around the subject regarding quality of service (QoS) n a telecommunication network. The chosen scenario is based on the transmission of ata and voice packets using a WAN connection, which has a limited bandwidth and mphasize the need of implementing QoS mechanisms in order to fulfill the quality equirements of the traffic, especially for VoIP. This topology will outline the impact nd importance of the QoS implementation, illustrated by the desired quality resulted hrough VoIP traffic simultaneously with maintaining the data conectivity using a ower bandwidth for applications which require a smaller amount of QoS properties, uch as FTP.

Quality Analysis of VoIP in Real-Time Interactive Systems over Lossy Networks

Voice over Internet Protocol (VoIP) systems have been spreading massively during the recent years. However, many challenges are still facing this technology among which is the lossy behavior and the uncontrolled network impairments of the Internet. In this chapter, the authors design and implement a VoIP test-bed utilizing the Adobe Real-Time Media Flow Protocol (RTMFP) that can be used for many voice interactive applications. The test-bed was used to study the effect of changing some voice parameters, mainly the encoding rate and the number of frames per packet as function of the network packet loss. Several experiments were conducted on several voice files over different packet losses, concluding in the best combination of parameters in low, moderate, and high packet loss conditions to improve the performance of voice packets measured by the Perceptual Evaluation of Speech Quality (PESQ) values.