Mostafa Hashem Sherif

SET and SSL: electronic payments on the Internet

This paper presents a service architecture that combines SET and SSL to provide a lightweight secure electronic commerce system. Although both protocols secure data over an open network like the Internet, they differ in their mode of operation. SET defines secure transaction-oriented exchanges for payment systems whereas SSL secures point-to-point communications. SET and SSL are complementary and the combined architecture permits a gradual introduction of SET services by reducing the need for cardholder certification.

Overview and performance of CCITT/ANSI embedded ADPCM algorithms

Embedded adaptive differential pulse coded modulation (ADPCM) algorithms quantize the differences between the input signal and the estimated signal into core bits and enhancement bits. CCITT Recommendation G.727, which describes embedded ADPCM encoding algorithms with 5, 4, 3, and 2 core bits, is virtually identical to the corresponding ANSI standard T1.310. The main features of G.727 and T1.310 and performance results are presented. A formal subjective evaluation of the speech performance of embedded ADPCM algorithms indicates that a midrise quantizer provides better voice transmission performance than its midtread counterpart when two core bits are used. The subjective data also show that the performance of the 40-kb/s midrise ADPCM algorithm with two feedback bits is indistinguishable from that of 64-kb/s pulse code modulation (PCM) for up to four tandem encodings. Embedded algorithms are therefore recommended for flexible congestion control of integrated traffic in multinode networks. >

Overview of CCITT embedded ADPCM algorithms

G.EMB is a draft CCITT recommendation for embedded ADPCM (adaptive differential pulse code modulation) encoding with five, four, three, and two core bits. The main properties of the algorithms of G.EMB are reviewed, and their performance for speech traffic is discussed. The embedded ADPCM algorithms allow the dropping of up to three least-significant bits to relieve congestion at any point between the transmitter and the receiver without the need to inform the receiving end. This bit dropping does not cause decoder divergence. Test results indicate that a mid-rise quantizer provides better voice transmission performance than its mid-tread counterpart when two bits are used in the feedback path. Subjective data show that the 40 kb/s mid-rise ADPCM algorithm with two feedback bits performs just as well as 64 kb/s PCM for up to four tandem encodings. G.EMB embedded algorithms are thus suitable for congestion control in multipoint and multinode network configurations.<<ETX>>

Evaluation of the Casner-Jacobson algorithm for compressing the RTP/UDP/IP headers

Protocols such as the real-time transport protocol (RTP), are used to provide suitable voice/video communication channels over the Internet. RTP transports real-time media data along with synchronization information over a datagram protocol such as the user datagram protocol (UDP). The Casner-Jacobson (see Internet-Draft, 1997) RTP/UDP/IP header compression algorithm can significantly reduce the overhead and increase the throughput of a connection. We evaluate the behavior of header compression in noisy links. It is shown that error indication and error correction can improve the resilience of header compression over noisy links.

Wavelet Based RDNN for Software Reliability Estimation

Using the wavelet basis in Recurrent Dynamic Neural Network (RDNN) can improve the failure event estimation of software defect tracking in telecommunications. Non-linearity of the system is represented by proper selection of the wavelet function. This RDNN handles noisy data and enhances the speed of convergence as compared with alternate approaches. A new adaptive RDNN is presented where software deployment testing observations are used to synthesize intrinsic model parameters.

Overview of speech packetization

There have been significant advances in the design and development of toll-quality packetization systems. The paper presents various issues that affect the packetization of speech such as robustness to noise, reconstitution of the flow of speech and congestion control. The target audience consists of designers and implementers of speech packetization systems as well as service providers concerned with the quality of service and efficiency of transmission.

Adaptive (T/sub 1/, T/sub 2/)-multiplexing transmission schemes for voice/data integrated networks

Integrated multiplexing schemes are needed to optimize the use of transmission bandwidth in integrated networks. In this paper, we introduce the idea of a dynamically controlled (T/sub 1/, T/sub 2/) multiplexing scheme. The advantages of this transmission policy is that it allocates the transmission capacity of the channel to two traffic types according to the instantaneous needs. We consider only two types of traffic, voice band traffic and digital data. The channel access times for voice and data are T/sub 1/ and T/sub 2/, respectively. In this paper, we present some preliminary results of our proposed scheme. In our simulations, we observe that a reduction in blocking probability of 15% or more is possible.

Expérimentation sur site ďun système de multiplication de circuits par mise en paquets

RésuméCet article décrit les principaux résultats de V expérimentation menée par France Télécom etatt du système ďaccès et de brassage intégré (iacs). Ce système de multiplication de circuits par mise en paquets a subi les tests suivants: évaluation subjective de la parole (interviews ďabonnés), performances des transmissions de données en bande vocale, résistance du système à un taux ďerreurs en ligne, test du dispositif de démodulation de télécopie.AbstractThis article describes the main results of the field trial conducted by France Telecom andatt of theiacs. This packetized circuit multiplication equipment has been submitted to the following tests: subjective evaluation of speech (subscribers interview), performances of voice band data transmission, performances with a bit error rate on the bearer, test of the fax demodulation process.

Les paquets de bande élargie: une nouvelle technique de transmission

RésuméCet article décrit les principes et la réalisation d’une nouvelle technique de transmission par mise en paquet du trafic multimédia. Cette intégration permet une utilisation plus efficace du réseau et réduit les coûts d’exploitation et de maintenance. On présente ensuite un système d’accès et de brassage intégré (IACS) développé par AT&T comme un exemple de mise en œuvre de la technique décrite.AbstractThe purpose of this article is to describe the principles and applications of a new transmission technology for multimedia traffic based on wideband packets. This integration allows a more efficient utilization of the network and reduces the operation and maintenance costs. Next, we present AT&T’s integrated access and cross-connect system (IACS,) as an example of an implementation of this technology.

Impact of network transmission delay and echo on Group 3 facsimile performance

This paper provides information on a laboratory study conducted by COMSAT Laboratories to evaluate the effect of echo and delay on facsimile call retention rates. The term call retention is defined as the ability of a pair of facsimile terminals to maintain a circuit connection until all pages have been successfully transmitted after negotiation of the high speed signaling rate. The results of this study indicate that the effect of network echo has the potential of degrading facsimile call retention. In addition, propagation delay reduces retention rates even further. The results of this work are currently under consideration by the International Telecommunication Union (formerly CCITT) in the formulation of test methodologies for assessing facsimile terminal implementation performance.