Wireless interworking architectures to support mobile IP over ATM

Have competitive conditions in the voice services market changed sufficiently to justify lifting most current economic and social regulation on the public switched telephone network (PSTN)? Yes, argues Richard Clarke, describing in detail the ongoing changes in the supply and demand structures of the voice market and their impact on reducing the economic power of the traditional incumbent local exchange carriers (ILECs). Based on this argument, and on evolving technology, Clarke notes that both the PSTN and the voice services it supports are “inexorably going away,” to be replaced by new services and technologies, making current regulatory requirements superfluous.

No Dialtone: The End of the Public Switched Telephone Network

Narrowing the 'Digital Divide': The Role of Complementarities between Fixed and Mobile Data in South Africa

This paper deals with the study of the uplink mechanisms of medium access within an UTRA system which offers the possibility of holding on to voice and data services. The voice service is beamed off through a dedicated channel whereas a common channel is used for a very widespread service nowadays like the Web traffic. The first investigations on homogeneous systems reveal that the access channel is longer occupied in the data system than in the voice system. The simulations related to the integrated voice and data system show that, in spite of the fact that the voice experiences a greater delay than in the homogeneous system, the QoS parameters for voice and data are guaranteed with no need to use all the available system resources. Following this, the integrated system allows for certain losses in the common RACH (random access channel).

The word ‘telematics’ was used for the first time in France in 1978 by Simon Nora and Alain Minc; it is a contraction of the French words ‘tétécommunications’ and ‘informatique’, and defines a marriage between computers and data transmission networks. This paper describes the five systems which will be implemented to provide telematic services via satellite in Europe during the next few years: EUTELSAT, TELECOM 1, UNISAT, DFS and ITALSAT. The main characteristics of the ‘special service’ transponders are presented, showing that, in Europe, (a) 2 to 5 transponders per satellite will be devoted to the special (telematic) services; (b) 4 satellites out of the 5 use the 12.5–12.75 GHz down‐link band for this purpose (only ITALSAT uses the 20 GHz down‐link for the same purpose), and (c) e.i.r.p.s at 12 GHz are in the range 40–49 dBW.The ground and space segments corresponding to this new service are presented. The main peculiarity of the ground segment is the exclusive use of ‘small’ earth‐stations equipped with fixed antennas and SCPC or 25–60 Mb/s TDMA multiple access equipment. Telematic services include videoconference at 2048 kb/s, voice services, computer‐to‐computer transfer of data, data distribution, remote printing, electronic mail, high‐speed facsimile, offshore voice and data services, digital network overflow, transatlantic digital business etc…Customer bit rates start from 2.4, 4.8, 9.6 and 32 kb/s (for TELECOM 1) up to 2048 kb/s by 64 kb/s steps. Bit error rates are maintained below 10−6 (standard) or 10−10 (higher grade) during 99 per cent of the year using FEC. The five networks for telematics via satellite in Europe are independent. A large efficiency improvement of the different services should be obtained through a future interconnection of the various networks via intersatellite links.

Power allocation across users in two adjacent cells is studied for a wireless code division multiple access (CDMA) network with mixed voice and data services. We assume that each user has a utility function that measures the user's satisfaction, or utility, as a function of the received signal-to-interference-plus-noise ratio (SINR). Each particular service (voice or data) is associated with a different utility function. We consider the forward link. Our objective is to allocate transmitted power to maximize the total utility summed over all active users subject to rate and power constraints. We show that the maximum utility can be achieved with a pricing scheme. We characterize the solution to a one-cell utility maximization problem with fixed interference from the other cell. For two-cell utility maximization, the two cells must cooperate to achieve the maximum utility.

Development of Unstructured Architecture for Voice and Data Services in Mobile Communication

The telecommunications industry (TI) is challenged by a significant increase in the complexity of information transfer due to a recent proliferation of data mining technologies, techniques and applications. As the result, TI is facing a fundamental paradigm shift, with the convergence of voice and data services as well as ever expanding technologies to its users. This technological movement towards a convergence of telephony and computer technologies, web-based networks, and wired and wireless services is creating areas of tremendous opportunities. These areas of opportunity are for continuous quality improvements and applications of the voice and data convergence mining techniques and their implementations. The TI’s implementation of the data mining algorithms reduces information overload, increases data integrity and accuracy, and effectively manages its global networks.

Voice-over-IP (VoIP) systems are increasingly prevalent in our lives. These systems come in a wide variety of flavors such as desktop-based software applications (e.g., Skype [26]), systems that replace public switched telephony network (PSTN) as the primary line voice service (e.g., Vonage [29]), and more recently, VoIP over smart phones. In November 2010, the number of concurrent VoIP users on Skype exceeded 25million [2]. A study estimates that as of February 2010, there are approximately 110million VoIP hard phone subscribers in the world [11]. Another study estimates that the number of mobile VoIP users will exceed 100million by 2012 [15]. With such a large existing user base of VoIP and expected user growth, and with constantly increasing costs of energy, we ask ourselves what is the energy efficiency of these systems. To answer that question, we gather information about existing VoIP systems and architectures, build energy models for these systems, and evaluate their power consumption and relative energy efficiency through analysis and a series of experiments. The core function of a VoIP system is to provide mechanisms for storing and locating the network addresses of user agents and for establishing voice and video

Modern packet-switched networks are increasingly capable of offering high-quality voice services such as Voice over LTE (VoLTE) which have the potential to surpass the Public Switched Telephone Network (PSTN) in terms of quality. To ensure this development is sustained, it is important that suitable quality evaluation methods exist in order to help measure and identify the effect of network impairments on voice quality. In this paper, a single-ended, objective voice quality evaluation model is proposed, utilizing a Convolutional Neural Network with regression-style output (CQCNN) to predict mean opinion scores (MOS) of speech samples impaired by a VoLTE network emulation. The results of this experiment suggest that a deep-learning approach using CNNs is highly successful at predicting MOS values for both narrowband (NB) and super-wideband (SWB) samples with an accuracy of 91.91% and 82.50% respectively.

In this paper we consider the problem of resource management in GSM/GPRS cellular networks offering not only mobile telephony services, but also data services for the wireless access to the Internet. In particular we investigate channel allocation policies that can provide a good tradeoff between the QoS guaranteed to voice and data services end users, considering three different alternatives, and developing analytical techniques for the assessment of their relative merits. The first channel allocation policy is called voice priority, since it gives priority to voice in the access to radio channels; we show that this policy cannot provide acceptable performance to data services, and we discuss the reasons for this shortcoming. The second channel allocation policy is called R-reservation; it statically reserves a fixed number of channels to data services, thus drastically improving their performance, but subtracting resources from voice users, even when these are not needed for data, thus inducing an unnecessary performance degradation for voice services. The third channel allocation policy is called dynamic reservation; as the name implies, it dynamically allocates channels to data when necessary, using the information about the queue length of GPRS data units within the base station. A threshold on the queue length is used in order to decide when channels must be allocated to data. Numerical results, show that the dynamic reservation channel allocation policy can provide very effective performance tradeoffs for data and voice services, with the additional advantage of being easily managed through the setting of the threshold value.

This chapter provides an overview of Digital Mobile Radio (DMR) technology, a low‐cost replacement of analog systems. DMR's technical specifications, network architecture, interfaces and protocols, and services are discussed. The standardization process and its future are also included. DMR reference architecture includes air and wireline network interfaces. The air, trunking, and data protocols are specified in the DMR standards documents. The DMR air interface and the protocols on it apply to the communications between base stations and the radio stations and between two radio stations for direct mode. The services offered by a DMR network can be categorized under voice services and data services. DMR voice services include individual voice calls and group voice calls. DMR offers two kinds of data services: Packet Data Services (PDS) and Short Data Services (SDS). The DMR standards cover a wide range of the frequency spectrum, from 30 MHz all the way up to 1 GHz.

The increasing need for service providers makes the performance of network management must also be improved. In this study, the author builds a clustering controller using the Open Network Operating System (ONOS) controller on an Software Defined Network (SDN) network. ONOS has a cluster feature that allows it to distribute traffic loads to other controllers to increase performance. The performance of clustering controller is compared with single controller and multi controller without clustering by measuring Quality of Service (QoS) on delay, jitter, and throughput parameters for data, voice, and video services. The test is done by adding background traffic. After being tested and analyzed, the performance of the clustering controller has the most superior QoS with an average delay in data services of 0.047026667 ms, jitter of 0.015414943 ms, throughput of 37.56360736 kbit/s. In the voice service, the delay value is 0.03909 ms, the jitter is 0.01828 ms, the throughput is 72.79049 kbit/s. In the video service, the delay value is 0,12072 ms, the jitter is 0,042013 ms, and the throughput is 5314,059 kbit/s. While in single and multi controller the average value obtained is not much different from the clustering controller. This is due to the single controller architecture, the network control center is only centered on 1 controller, as well as multi controllers, so that it affects the performance of service distribution.

Asynchronous transfer mode (ATM) products are introduced in today's telecom network at several places. Network operators are confronted with the difficult problem of how to organize the cohabitation of the new broadband network elements with the existing traditional network elements offering speech and data services. Basically, there are two network options to this problem. A first solution is to adopt an overlay solution, thereby creating a separate broadband network and providing the necessary interfaces and service interworking functions towards the existing speech and data networks. A second solution exists in the extension of the existing investments to accommodate in an integrated way both broadband and narrowband services and proper interworking functions. The second method is proposed as the selected evolution scenario for the existing A1000 S12 narrowband exchanges. An overview of the basic steps in the evolution scenario is presented and the benefits of this approach compared to an overlay solution are discussed.

In this paper we propose an adaptive transmission algorithm over an IEEE 802.11 WLAN that supports integrated voice and data services, where data traffic is transmitted with the Distributed Coordination Function (DCF), while voice transmission is carried out with the Point Coordination Function (PCF). The proposed algorithm defines an updating period. At the beginning of each period and based on the transmission statistics of the previous one, our algorithm adjusts the protocol parameters to improve data transmission without compromising voice transmissions, i.e., maximum allowable delay is respected. To show the efficiency of the proposed algorithm, we compare it with a previous work we did in Li et al. (2006). The simulation results are provided and show the advantage of using such algorithm in term of protocol performance.