The Impact of Frame Rate on Securing Real Time Transmission of Video over IP Networks NOUR EL DEEN M. KHALIF A, HESHAM N. ELMAHDY Information technology Department Faculty o/Computers and Information - Cairo University 5 Dr. Ahmed Zwail Street, Orman Giza, Postal code 12613. EGYPT nourmahmoud@gmail.com ehesham®mailer.eun.eg Abstract- The nature of playing video streams over a network in a real time requires that the transmitted frames are sent in a bounded delay. Also, video frames need to be displayed at a certain rate; therefore, sending and receiving encrypted packets must be sent in a certain amount of time in order to utilize the admissible delay. There are two main components for securing multimedia transmissions. Key management protocols such as Transport Layer Security (TLS) protocol is used as cryptographic protocol that provides security and data integrity for communications over IP Networks. Advanced Encryption Standard (AES) is used as a symmetric block cipher, that can encrypt (encipher) and decrypt (decipher) video frames. The Real-time Transport Protocol (RTP) protocol is used as a standardized packet format for delivering video over the IP Networks. This makes secure video encryption feasible for real-time applications without any extra dedicated hard-ware. In this paper; we studied the impact of frame rate on a secured real time transmission of video over IP networks. Through this paper, we will try to find the most suitable frame rate in order to achieve better data rate and fewer frame and packet loss. This is done via java as a programming language for implementation. Keywords: Transport Layer Security (TLS); Real-time Transport Protocol (RTP); Advanced Encryption Standard (AES); Frame Rate. 1. INTRODUCTION The advent of networked multimedia systems will make continuous media streams, such as real time audio and video, increasingly pervasive in future computing and communications environments. It is thus important to secure real time video streaming over IP networks from potential threats such as hackers, eavesdroppers, etc [1]. This work was originally done at section 5, 6. Consequently, we study the impact of frame rate on securing video streams. - Also, we study the performance of encryption and decryption algorithms such as AES for real time video streams. We present our results in diagrams, which shows the encryption performance between different frame rates and what we have to pay to grantee a secure transmission of video over IP networks. In short, the foremost goal of this research is to cover these points: • Determining how key management protocols such as Transport Layer Security (TLS) protocol can be implemented and what are the advantages and disadvantages of this protocol. 978-1-4244-3778-8/09/$25.00 ©2009 IEEE 57 • Determining how encryption and decryption can be implemented for the real time application and study the impacts on received video frames. • Determining what is the best frame rate that achieves better data rate and fewer frame and packet loss. • Comparing our results with the previous results in video encryption. • Computing performance and overheads of multimedia security. 2. REAL-TIME TRANSPORT PROTOCOL (RTP) The Real-time Transport Protocol (RTP) is a standardized packet format designed to deliver media contents (mostly video and audio, but also images) over an IP network [2]. It was first defined in 1996 as RFC 1889, which was replaced by RFC 3550 in 2003. It was originally thought as an add-on for the UDP protocol in media environments, although it can be also delivered over a TCP layer. It has no defined port to be delivered, so it is usually delivered in the wide area of non defined ports. However, it is said in the standard that RTP must use an even port and the Real-time Transport Control Protocol (RTCP) use the next odd port [2]. An RTP packet is divided into a 12-bytes header and a payload (see Figure I). The payload would carry the data of the packet and in the header the following information is provided [2]: • RTP Version [V] (2 bits): indicates the version of the protocol. The current version is 2. • Padding [PJ (l bit): indicates if there is extra information at the end of the packet, in order to exactly fit on an 8 bit alignment. • Extension [X] (l bit): indicates if any extension of the protocol is used in the packet. • Contributors [NCSRC] (2 bits): indicates the number of contributors to that session. It is used to read the CSRC optional headers. • Marker bit [NJ (I bit): it is used to inform the application that something special is in that packet. Usually is always off in audio streams and I in video packets containing the last information of a frame. Authorized licensed use limited to: UNIVERSITY OF MELBOURNE. Downloaded on June 19, 2009 at 14:11 from IEEE Xplore. Restrictions apply.