Traditional digital watermarking schemes are mainly based on spatial-domain or transform-domain, such as discrete cosine transform and discrete wavelet transform. The watermark is extracted or detected at any point where identification or integrity is concerned. For these reasons, it is important to make information systems secure to protect data and resources from malicious acts. Due to rapid progress in wireless communication systems, extreme prevalence mobile systems, and smart card technology, information is more vulnerable to abuse.
The specific information is known as watermark and usually used for ownership identification, authentication purpose and protection of integrity of data. I INTRODUCTION Digital watermarking is the technique of inserting specific information into signal, data, image or video. Receiver receives that signal and decrypts both signal and watermark.
In proposed method voice signal is compressed, watermarked, encrypted and sent through a transmission medium from the transmitting end. To generate more secured random number, nibble bit of noise signal is Xored with hardware-generated random number. Change has brought at the generation of random numbers. FPGA can perform a large number of operations concurrently. Conventional processors contain small number of registers and perform large operations in multiple cycles. FPGA is an efficient device for these operations on realtime signals. There are many encryption techniques, but for realtime encryption, fast encryption is needed. Watermark and random key is embedded at the vacant places. To maintain same data-rate compression is performed at first. Data-size is one of the major concerns of cryptographic systems. This paper presents a Field-Programmable Gate Array (FPGA) based secured speech communication system. Also revealing that VoIP-supported codecs are faster and have a higher threshold in terms of the number of calls before the e2e delay and the rate of packet loss exceeds the acceptable limit for encrypted and plain VoIP e2e delay and packet loss rate. Through simulation of appropriate scenarios, results indicates that each of the encryption algorithms (AES, DES and 3DES) append additional overhead on the e2e delay and rate of packet loss during VoIP transmission. This study sought to determine the combination of cryptographic algorithms, cipher mode and voice codec that holds the uppermost threshold point, before the latency and rate of packet loss of active calls goes past ITU acceptable standards for one way latency in both plain and encrypted VoIP traffic of 150ms and 200ms respectively and 5% packet loss rate. Mindful of the fact that other components like voice codecs and network bandwidth also contributed delay capabilities on VoIP traffic, and additions of security overheads, there exists a threshold point where an increase in call volume exerts a negative effect on pre-established calls with respect to time and the rate of packet loss. This delay levied is dependent on the mode of operation of the cryptographic algorithms.
Nonetheless, the uses of cryptographic algorithms yet imposes a delay overhead and packet size overhead on VoIP, which is unconnected to the processing time required to encrypt/decrypt bits or blocks of data and the increase in packet size due the block size of the encryption algorithm. The use of publically verified cryptographic algorithms to ensure confidentiality of VoIP traffic transmitted over insecure public networks as the Internet cannot be overemphasized. However, just like many other new Information technology trends, VoIP introduces both security risks and opportunities for the IT world, viable solutions of which are required. VoIP as a packet switched system is clearly one of the most important evolving trends in computing and telecommunications.
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