Voice Over Internet Protocol

Implementations of Voice over Internet Protocol (VoIP) have focused, up to now, mainly on the need to transport data in real-time, often at the expense of security. The neglect of secure VoIP is often intentional, as developers are striving to minimise overheads and ...

This work presents an analysis based on a triple play transmission generated on two combined access technologies, PLC (Power Line Communications) and ADSL2+ (Asymmetric Digital Subscriber Line), called of pDSL. New services are based on multimedia applications, as voice over internet protocol (VoIP), video on demand (VoD), and internet protocol television (IPTV). Such services use digital broadband networks such as

Telecommunication technology is increasingly con- verging towards software-intensive solutions. As the role of software increases, it naturally follows that traditional software development procedures need to be reassessed. In South Africa, this need is especially urgent because of recent legislation that presages an increase in the number of competitors. This paper assesses two important contemporary software development ap- proaches - namely

Implementations of Voice over Internet Protocol (VoIP) have focused, up to now, mainly on the need to transport data in real-time, often at the expense of security. The neglect of secure VoIP is often intentional, as developers are striving to minimise overheads and delays. The Secure Real-Time Protocol (SRTP) has the potential to secure real-time streams without exacting too high a performance price. SRTP is the addition of security to the audio/video profile used in the Real-Time Transport Protocol (RTP). SRTP adds confidentiality, integrity and optionaly authenticity to RTP media streams. This paper focuses on the integration of SRTP into Asterisk, an open-source VoIP PBX. SRTP support has recently been added to Asterisk by Mikael Magnusson. This paper analyses Magnusson’s implementation, contrasting it to a proof-of-concept implementation developed independently at Rhodes University. The interoperability of SRTP implementations cannot be taken for granted, given the relatively r...

In this work, it has been demonstrated through experiments that Long Range Dependence (LRD) in internet traffic is indeed affecting the queueing performance. Queueing performance is being degraded as the self-similarity of the packet arrival process increases. Here, the link buffer capacity and queueing delay exhibits super-linear relationship. This in turn has led to the proposal of resource provisioning strategy to boost the queueing performance. The inherent self-similar nature of the internet made the possibility of carrying voice traffic accompanied by voice quality degradation. Self-similar or variable nature of packet delay process implies voice quality degradation due to inter packet arrival variations. Delay and delay variation (jitter) behave erratically due to the self-similar nature of the packet arrival and delays. As the self-similarity of the packet round trip delay increases, TCP throughput performance is degraded.

Realization of cost effective packet telephony solutions over mobile ad-hoc networks is one of the attended area by the computer science research fraternity. As with other ideas it is difficult to test new proposals on mobile ad-hoc networks using a real time environment. So researchers need to resort to simulation based environment. To model packet telephony over mobile ad-hoc network generation of relevant traffic is central to the success. Authors in this work are proposing an algorithm to model the application layer of a mobile ad-hoc network carrying packet telephony, thereby generation of relevant telephony data to be transported over the network.

Voice over Internet Protocol (VoIP) technology has observed rapid growth in the world of telecommunications. VoIP offers high-rate voice services at low cost with good flexibility, typically in a Wireless Local Area Network (WLAN). In a voice conversation, each client works either as a sender or a receiver depending on the direction of traffic flow over the network. A VoIP technologically requires high throughput, less packet loss and a high fairness index over the network. The packets of VoIP streaming may experience drops because of competition among the different kinds of traffic flow over the network. A VoIP application is also sensitive to delays and requires voice packets to arrive on time from the sender to the receiver without any delay over a WLAN.  To date, scheduling of VoIP traffic is still an unresolved problem. The objectives of this survey paper to  discuss fundamental principles of VoIP-related schedulers and identify current scheduler issues.  This survey paper also identifies the importance of the scheduling techniques over WLANs. Related research work for real-time applications specifically for VoIP will also be highlighted.

In this paper, we consider the evolution of telephone networks from time-division multiplexing circuit switching to packet switching and, in particular, to packet switching-based on Internet Protocol (IP-supported telephony). We analyze IP-supported telephony design solutions by proposing a layered reference model in which each layer is associated to a subset of the functions that support telephony. We use the reference model to establish a terminology and a framework for the comparison of the design solutions. We group the design solutions in scenarios and compare them in terms of the reference model proposed. We then focus on IP telephony, in which IP is used in telephone company networks, and on Internet telephony, in which the Internet is used to support telephony. We show that they both can be seen as implementations of the same architecture, which consists of a set of components, associated to functions, and of the interactions among these components. We then consider the issue of voice-data integration and analyze the variety of design solutions that can be adopted to integrate voice and data.

Voice over Internet protocol (VoIP) is one of the most important applications for the IEEE 802.11 wireless local area networks (WLANs). For network planners who are deploying VoIP over WLANs, one of the important issues is the VoIP capacity. VoIP bandwidth consumption over a WAN is one of the most important factors to consider when building a VoIP infrastructure. Failure to account for VoIP bandwidth requirements will severely limit the reliability of a VoIP system and place a huge burden on the WAN infrastructure. Less bandwidth utilization is the key reasons for reduced number of channel accesses in VOIP. But in the QoS point of view the free bandwidth of atleast 1-5% will improve the voice quality. This proposal utilizes the maximum bandwidth by leaving 1-5% free bandwidth. A Bandwidth Data rate Moderation (BDM) algorithm has been proposed which correlates the data rate specified in IEEE802.11b with the free bandwidth. At each time BDM will calculate the bandwidth utilization bef...

The telecommunications network operators have been under increasing attacks from new technologies, which are "stealing away" their customers and reducing revenues from the traditional fixed telephony services. One of those new technologies is VOIP (Voice over Internet Protocol). Operators responded by acting in the content distribution segment. In this context, a new technology, called IPTV (Internet Protocol Television) emerged. The

Hacking and threats are increasing with the enhanced population of wireless traffic. Authentication of mobile subscribers and network is an extremely challenging issue to the researchers. Now mobile communication system (3-G or advanced) has been developed to speed up the data communication. 3-G mobile communications use two different switching techniques: circuit switching for voice and low speed data communications and

Converged networks, the merging of voice, video and data networks onto a single infrastructure, and in particular IP telephony offer the construction industry cheaper calling and the ability to log all voice communication in an efficient manner. The emergence of wireless networks on the construction site allow IP telephony to be deployed on the site replacing the mobile phone network, where it was discovered that UK construction companies would spend over£ 1 Million a year on communication. This paper looks at the ...

Summary The Session Initiation Protocol (SIP) is one of the most widely used protocols for Voice over Internet Protocol (VoIP). Mobility is a very sophisticated service in VoIP. VoIP mobility performance depends on the handoff delay in the Data Link layer. As VoIP is application layer application, application layer mobility support has a vital role in supporting VoIP mobility. This paper presents an overview of different modes of VoIP mobility at both the application layer and data link layer. Different aspects of these modes are explored.

Distance learning and internet-based delivery of educational content are becoming very popular as an alternative to real face-to-face delivery. Clinical-based discussions still remain greatly face-to-face despite the advancement of remote communication and internet sharing technology. In this study we have compared three communication modalities between a learner and educator: audio and video using voice over internet protocol (VoIP) alone [AV];