Abstract: Over the past decade, mobile has experienced a
revolution that will ultimately change the way we communicate.All
these technologies have a common denominator exploitation of
computer information systems, but their operation can be tedious
because of problems with heterogeneous data sources.To overcome
the problems of heterogeneous data sources, we propose to use a
technique of adding an extra layer interfacing applications of
management or supervision at the different data sources.This layer
will be materialized by the implementation of a mediator between
different host applications and information systems frequently used
hierarchical and relational manner such that the heterogeneity is
completely transparent to the VoIP platform.
Abstract: In a recent year usage of VoIP subscription has increased tremendously as compare to Public Switching Telephone System(PSTN). A VoIP subscriber would like to know the exact tariffs of the calls made using VoIP. As the usage increases, the rate of fraud is also increases, causing users complain about excess billing. This in turn hampers the growth of VoIP .This paper describe the common frauds and attack on VoIP based system and make an attempt to solve the billing attack by creating secured channel between caller and callee.
Abstract: Insufficient Quality of Service (QoS) of Voice over
Internet Protocol (VoIP) is a growing concern that has lead the need
for research and study. In this paper we investigate the performance
of VoIP and the impact of resource limitations on the performance of
Access Networks. The impact of VoIP performance in Access
Networks is particularly important in regions where Internet
resources are limited and the cost of improving these resources is
prohibitive. It is clear that perceived VoIP performance, as measured
by mean opinion score [2] in experiments, where subjects are asked
to rate communication quality, is determined by end-to-end delay on
the communication path, delay variation, packet loss, echo, the
coding algorithm in use and noise. These performance indicators can
be measured and the affect in the Access Network can be estimated.
This paper investigates the congestion in the Access Network to the
overall performance of VoIP services with the presence of other
substantial uses of internet and ways in which Access Networks can
be designed to improve VoIP performance. Methods for analyzing
the impact of the Access Network on VoIP performance will be
surveyed and reviewed. This paper also considers some approaches
for improving performance of VoIP by carrying out experiments
using Network Simulator version 2 (NS2) software with a view to
gaining a better understanding of the design of Access Networks.
Abstract: In the recent years multimedia traffic and in particular
VoIP services are growing dramatically. We present a new algorithm
to control the resource utilization and to optimize the voice codec
selection during SIP call setup on behalf of the traffic condition
estimated on the network path.
The most suitable methodologies and the tools that perform realtime
evaluation of the available bandwidth on a network path have
been integrated with our proposed algorithm: this selects the best
codec for a VoIP call in function of the instantaneous available
bandwidth on the path. The algorithm does not require any explicit
feedback from the network, and this makes it easily deployable over
the Internet. We have also performed intensive tests on real network
scenarios with a software prototype, verifying the algorithm
efficiency with different network topologies and traffic patterns
between two SIP PBXs.
The promising results obtained during the experimental validation
of the algorithm are now the basis for the extension towards a larger
set of multimedia services and the integration of our methodology
with existing PBX appliances.
Abstract: The purpose of this research is to develop a security model for voice eavesdropping protection over digital networks. The proposed model provides an encryption scheme and a personal secret key exchange between communicating parties, a so-called voice data transformation system, resulting in a real-privacy conversation. The operation of this system comprises two main steps as follows: The first one is the personal secret key exchange for using the keys in the data encryption process during conversation. The key owner could freely make his/her choice in key selection, so it is recommended that one should exchange a different key for a different conversational party, and record the key for each case into the memory provided in the client device. The next step is to set and record another personal option of encryption, either taking all frames or just partial frames, so-called the figure of 1:M. Using different personal secret keys and different sets of 1:M to different parties without the intervention of the service operator, would result in posing quite a big problem for any eavesdroppers who attempt to discover the key used during the conversation, especially in a short period of time. Thus, it is quite safe and effective to protect the case of voice eavesdropping. The results of the implementation indicate that the system can perform its function accurately as designed. In this regard, the proposed system is suitable for effective use in voice eavesdropping protection over digital networks, without any requirements to change presently existing network systems, mobile phone network and VoIP, for instance.
Abstract: Voice over Internet Protocol (VoIP) application or commonly known as softphone has been developing an increasingly large market in today-s telecommunication world and the trend is expected to continue with the enhancement of additional features. This includes leveraging on the existing presence services, location and contextual information to enable more ubiquitous and seamless communications. In this paper, we discuss the concept of seamless session transfer for real-time application such as VoIP and IPTV, and our prototype implementation of such concept on a selected open source VoIP application. The first part of this paper is about conducting performance evaluation and assessments across some commonly found open source VoIP applications that are Ekiga, Kphone, Linphone and Twinkle so as to identify one of them for implementing our design of seamless session transfer. Subjective testing has been carried out to evaluate the audio performance on these VoIP applications and rank them according to their Mean Opinion Score (MOS) results. The second part of this paper is to discuss on the performance evaluations of our prototype implementation of session transfer using Linphone.
Abstract: This paper presents the findings of two experiments that were performed on the Redundancy in Wireless Connection Model (RiWC) using the 802.11b standard. The experiments were simulated using OPNET 11.5 Modeler software. The first was aimed at finding the maximum number of simultaneous Voice over Internet Protocol (VoIP) users the model would support under the G.711 and G.729 codec standards when the packetization interval was 10 milliseconds (ms). The second experiment examined the model?s VoIP user capacity using the G.729 codec standard along with background traffic using the same packetization interval as in the first experiment. To determine the capacity of the model under various experiments, we checked three metrics: jitter, delay and data loss. When background traffic was added, we checked the response time in addition to the previous three metrics. The findings of the first experiment indicated that the maximum number of simultaneous VoIP users the model was able to support was 5, which is consistent with recent research findings. When using the G.729 codec, the model was able to support up to 16 VoIP users; similar experiments in current literature have indicated a maximum of 7 users. The finding of the second experiment demonstrated that the maximum number of VoIP users the model was able to support was 12, with the existence of background traffic.
Abstract: This article discusses the questions concerning of creating small packet networks for energy companies with application of high voltage power line carrier equipment (PLC) with functionality of IP traffic transmission. The main idea is to create converged PLC links between substations and dispatching centers where packet data and voice are transmitted in one data flow. The article contents description of basic conception of the network, evaluation of voice traffic transmission parameters, and discussion of header compression techniques in relation to PLC links. The results of exploration show us, that convergent packet PLC links can be very useful in the construction of small packet networks between substations in remote locations, such as deposits or low populated areas.