5G - A Revolutionary Future of Networking

The inherent design for Converged Next Generation Networks (NGNs) accommodates Packet Switched (PS) services. IEEE 802.16e, 802.16m, 3GPP LTE/LTE-Adv are converged IP-based Next Generation Networks (NGNs) that endorse IP Multimedia Services (IMS) support in the Core Network. Converged NGNs are mainly intended to be deployed in urban areas with wide-area coverage provided by existing 3G networks in the initial deployment. The main revenue driver for 3G operators is still Circuit Switched (CS) Voice and CS data, like SMS. Hence, there is a battle for CS Voice services support over these PS-based 4G Networks by contenders such as Voice over LTE Generic Access (VoLGA), IMS-VoIP, Media Independent Handover and CS-Fallback solutions. This paper addresses the Inter- Radio Access Technology (RAT) handover solutions in IEEE 802.16m and IMT-Advanced networks. We propose a novel solution to support backward Inter-System Handover (HO) to CS-voice from PS-Networks using VoLGA Access Network Controller (VANC) and CS Inter-Working Control to maintain CS-service growth until Next Generation Networks are deployed throughout the nation.

Normal 0 false false false MicrosoftInternetExplorer4 Normal 0 21 false false false ES-TRAD X-NONE X-NONE MicrosoftInternetExplorer4 In this paper we propose new design framework for network architecture of future 5G mobile networks, including protocols and algorithms. The proposed design is based on user-centric mobile environment with many wireless and mobile technologies on the ground. In heterogeneous wireless environment changes in all existing wireless technologies is not possible, so each solution towards the next generation mobile and wireless networks should be implemented in the service stratum, while the radio access technologies belong to the transport stratum regarding the Next Generation Networks approach. In the proposed design the user terminal has possibility to change the Radio Access Technology - RAT based on certain criteria. For the purpose of transparent change of the RATs by the mobile terminal, we introduce so-called Policy-Router as node in the core network, which establishes IP tunnels to the mobile terminal via different available RATs to the terminal. The selection of the RAT is performed by the mobile terminal by using the proposed user agent for multi-criteria decision making based on the experience from the performance measurements performed by the mobile terminal. For the process of performance measurements we introduce the QoSPRO procedure for control information exchange between the mobile terminal and the Policy Router.

PurposeThe purpose of this paper is to identify concerns developing in regard to current approaches used to define, deliver and fund universal service obligations (USOs) for telecommunications due to developments in competition, new technology, e.g. wireless, convergence and next generation networks (NGN), including voice over internet protocol (VoIP). To stimulate thinking about the policies required to deliver USOs in an IP‐enabled NGN environment.Design/methodology/approachThe paper examines the nature and scope of USOs, including the concept of “functional internet access” to address questions about whether in an NGN environment: present USOs are sustainable; USOs need to change; and whether USOs should include access to broadband? The paper also examines alternative/complementary sources of funding for USOs.FindingsWith falling revenue (due to increasing competition and declining prices) occurring alongside strong demand for funds to deploy NGN, telecommunications operators are less able to provide USOs sustained by cross‐subsidisation. Universal service funds, too, may be under pressure. Also there is likely to be an uneven migration to NGN since it is likely that NGN will be deployed first in more profitable, densely populated areas and then only later (if at all) in relatively less commercially viable rural/remote areas. The paper concludes that present USOs can be maintained in an NGN environment. It concludes that since USOs is an evolving concept, there should be regular systematic reviews of whether USOs should be extended to include broadband. The paper concludes that increased funding from general taxation revenue is warranted and would link decisions concerning the nature and scope of USOs more closely with financial responsibility for such decisions.Originality/valueThe paper stimulates rethinking about whether, and if so how, arrangements relating to USOs need to change? In adopting a longer term perspective of universal services, the paper may be a valuable complement to the reviews of USOs being conducted in various countries (since short term decisions should be consistent with longer term developments).

QoS(Quality of Service) is defined as collective effect of performance which determines the degree of satisfaction of a user of the service by ITU-T Rec. E.800. While the use of VoIP(Voice Over Internet Protocol) has been widely implemented, persistent problems with QoS are a very important sue which needs to be solved. This research is finding the assignment of VoIP QoS to deduct how to manage the control system and presenting the QoS control process and framework under NGN(Next Generation Network) environment. The trial framework is the modeling of the QoS measurement metrics, instrument, equipment, method of measurement, the series of cycle & the methodology about analysis of the result of measurement. This research underlines that the vulnerability of the VoIP protocol in relation to its QoS can be guaranteed when the product quality and management are controlled and measured systematically. Especially it`s very important time to maintain the research about VoIP QoS measurement and control because the big conversion of new network technology paradigm is now spreading. In addition, when the proposed method is applied, it can reduce an overall delay and can contribute to improved quality, in relation to signal, voice processing, filtering more effectively.

The main stages of the historical development of telecommunication networks are shown. It is noted that today the most common concepts are NGN (Next Generation Network), supporting the functionality of Triple-Play Services (triple services – voice, video and data transmission), as well as IMS (IP Multimedia Subsystem), which combines cellular technology and landline communications. They are able to provide a wide range of services: basic, additional, intelligent. In addition, IMS is able to provide mobile services. The IMS architecture layers are presented – transport layer, control layer; layer of service and application servers. Particular attention is paid to the layer of service and application servers. Based on the analysis of ITU recommendations, the main telecommunication services quality indexes are determined, namely, the time of service provision, the length of the queue that the service request falls into to wait for service on the server, and the probability of refusing to provide services. Based on the proposed mathematical models of the NGN intelligent superstructure, analytical expressions are presented for calculating the quality indexes of intelligent services in NGN for one type of service and for K types of services, taking into account the self-similarity of traffic, and without taking into account. To evaluate the quality of services in IMS, it is proposed to use the approaches of queuing theory and tensor analysis. Tensor analysis offers a mathematical apparatus for transforming coordinate systems, considering IMS as a set of geometric objects whose projections are different in different coordinate systems, but the physical properties of the objects themselves do not change. Expressions are proposed that make it possible to calculate the following QoS indexes in IMS: average queue length, average residence time of an application in the system for all types of services in IMS. It is considered that the incoming traffic is the simplest. Further development of the work is to take into account the self-similar nature of traffic, which is created by the flow of requests for services that come to IMS, as well as the limitations of the server buffer, which will allow a more accurate assessment of the quality of services in IMS.

The commercial proliferation of cellular voice and data service has placed a new challenge for mobile communication systems. Next-generation wireless systems are envisioned to have an all-IP-based infrastructure with the support of heterogeneous access technologies (Akyildiz et al., 2004). Under the circumstance, the Internet Protocol (IP) is selected as the common interconnection protocol to integrate disparate wireless systems, so that mobile users can roam among multiple wireless networks, regardless of the underlying different radio access technologies (Akyildiz et al., 2005; Makaya & Pierre, 2008; Mohanty & Xie, 2007). However, with the advent of new value-added services (video-conference, multimedia streaming, etc.) and novel concepts introduced into Long Term Evolution (LTE) architecture of the 4th Generation (4G) networks, provisioning efficient mobilitymanagement with quality of service guarantees and seamless handoff feature become even more important for next-generation wireless network design. Generally, mobility management allows mobile communication systems to locate roaming terminals for voice/data delivery as well as maintaining network connectivity when the terminal moves into a new service area (Akyildiz et al., 1999). Typically, such process contains two aspects: location management and handoff management (Quintero et al., 2004; Zhang et al., 2010). Location management enables telecommunication systems to find out the network attachment points of roaming nodes for call/data delivery. It usually contains two components: location update and call delivery (or data delivery). The former requires mobile nodes to provide the system with their location information, while the latter indicates that the system is queried for the location information of specific mobile nodes, and then services are delivered to them while they are away from their home network (Zhang et al., 2010). Handoff management aims to maintain network connectivity when mobile nodes change their network attachment points or access points. Obviously, handoff protocols need to preservemobile users’ network connectivity as they move from one network to another, while simultaneously reducing disruption to the ongoing call/data sessions. Therefore, reducing handoff delay and maximizing session continuity are always the primary goals of handoff management (Dimopoulou et al., 2005). Generally, handoff seamlessness means lower packet Performance Analysis of Seamless Handover in Mobile IPv6-based Cellular Networks 13

Service convergence, digitization of the content and packet based information transport, in addition to economies of scale, rapid and flexible service delivery, reduction of capital and operating costs, changes in telecom policy and regulation, and ever increasing competition have been key factors in the evolution of Next Generation Networks (NGN). Telecom service providers worldwide are migrating their legacy networks towards, or deploying new next generation networks to meet the market and competitive demands. The IP-centric converged next generation networks aim to provide a multitude of services over a single integrated network infrastructure, rather than multiple segmented and overlay networks as have existed before. It is also expected that the legacy network will co-exist with next generation networks in the foreseeable future. However, the deployment of next generation networks has brought its unique challenges for network managers and planners, as network has to be planned in a cost effective way without over or under building network resources. Moreover, as the next generation networks is a single converged network, the traffic load in one domain of the network has major impact in other domains and overall network as well. This paper addresses the challenges related to the planning of the next generation networks and a reference architecture enabling integrated end-to-end planning of next generation network covering multitude of domains and layers.

The use of technology has impacted on communication in so many ways. The advent of voice over internet protocols (VoIP) has made the world a global village where one can reach out to any part of the universe. But a challenge exists as to how to make communication and data transmission faster, the volume of traffic, bandwidth and latency in networks, that has to be transmitted between the sender and the receiver. The overall customer experience can be improved by the use of technology, which also makes it simpler to collect client information. Data packets are addressed and routed by the Internet Protocol (IP). This research aimed at deploying jitter, throughput, network traffic delay and bandwidth (JiTTraB) as a performance metrics to measure voice over internet protocols (VoIP) to measure the Quality of Service (QoS) of networks. This method prioritizes network traffic going through a router and providing acceptable service to most users in a quest to address VoIP concerns. In comparison to an existing university campus network (UCN), JiTTraB has a higher throughput, more data transmission at a given time with a minimal delay of about 0.001 seconds. However, UCN queues data with a high rate of delay before transmission. Data seem to crash considerably on the UCN due to network overload above its throughput margin and growth capacity. Thus, the proposed network design based on JiTTraB outperforms UCN in terms of transmission rate, throughput, bandwidth, delay and jitter. In addition, the JiTTraB communication network enhances QoS by performing better in close-quarters conversations than in far-quarters conversations. With an expansion reserve of 30%, the JiTTraB can handle up to 350 conversations securely while maintaining VoIP QoS standards without degrading the functionality of other network services or applications.

Information technology is developing rapidly, especially in the field of computer science, telecommunications and information systems. In the early days of telecommunications, voice networks could not be combined with data networks. However, more recently it became possible to integrate data, audio and video services onto a single network, known as a multimedia network. The capabilities of the Internet are also rapidly increasing, and now, in addition to data packets (email, web browsing) the Internet transfers audio and video packets. Voice over Internet Protocol (VoIP) is a multimedia service. This technology can be used to transmit audio, video, and data packets. Consequently, it is becoming the most preferred communications technology. VoIP will eventually replace the use of traditional telephony – Public Switched Telephone Networks (PSTN), although the switchover process is challenging. A chief concern it that, while PSTN transmits audio packets over a closed network, VoIP sends audio packets using an open network – the Internet. In PSTN, eavesdroppers must have direct access to physical network to obtain information from communications. Whereas with VoIP, eavesdroppers can monitor data packets from they are connected to the Internet. Hence, security and privacy are important considerations when switching to VoIP systems. One solution is anonymous systems, which can be used to implement security and privacy protocols. However, this typically reduces the Quality of Service (QoS) of the VoIP. This empirical research study examines VoIP performance in an anonymous network – The Onion Routing (Tor). Two scenarios are implemented using the real Tor network to investigate three QoS metrics for VoIP: latency, jitter and packet loss. As recommended, latency in VoIP should be less than 400 ms, jitter should be less than 50 ms and packet loss should not be more than 5%. In addition, the research calculates the probability of attackers in the two scenarios implemented and evaluate Tor network forecasting. Experiments were conducted in reference to two scenarios. The first scenario is VoIP calls routed through a Tor network with three Tor relays (default Tor). And the second scenario is VoIP calls routed through a Tor network with two Tor relays. Experiments were performed in three periods; December 2012, July 2013, and October 2013. During each experimental time period, a hundred calls were captured for each scenario. Experimental results show that the QoS of VoIP over the two Tor relays was better than the VoIP over the three Tor relays. However, a probability of attackers calculation found that the VoIP with three Tor relays returned better anonymity than that with two Tor relays. The best results were returned over the experimental period in July 2013, when the acceptable calls using two Tor relays reached 64 calls at 5% packet loss. Meanwhile, the worst experimental results were returned in October 2013, when acceptable calls numbered 11 using three Tor relays at 1% packet loss. At the end of 2013, the actual data for relay numbers and bandwidth approached forecast result with time series analysis. Meanwhile, the numbers of Tor users differed from the Tor users forecast. In August 2013, Tor users increased dramatically; this is attributed to the fact that BotNet attack was using the Tor network to attack their target leading to a fivefold increase.

LTE (Long Term Evolution), a 4G mobile communication standard has a support for higher data rates and lower latency. Since LTE is completely an Internet Protocol (IP) based technology, Network operators of LTE can handle voice via a “fall-back” technique by using previously existed circuit switched networks. To avoid this, Voice over IP (VoIP) services need to be used to deliver the voice over IP networks, such as Internet. Voice Codec is one of the critical components in VoIP used to convert analog signals to compressed digital packets and vice-versa that can be sent over the internet. But there are different kinds of codecs exist that optimize the voice stream based on bandwidth and application requirement. Unfortunately, there is no standard codec that is acceptable for all circumstances, some codecs provides better quality, some other uses lesser bandwidth. All these features will effect on the ultimate selection of codec by an organization. This paper focuses on the quality of service (QOS) analysis of VoIP traffic over LTE by configuring different voice codecs such as G.711, G.722.2, G.723.1 and G.729. To analyze the performance of these codecs ns-3 simulator is being used since its core has a support for LTE models. The comparative study of above selected VoIP codecs over LTE network in terms of Packet Loss, End to End delay, Jitter, and Throughput in normal and error prone scenarios has been done. From the analyzed results and the performance evaluated, the selection of codec will become easy to provide better VoIP services for customer satisfaction.

A major change in telecommunication industry is Voice over Internet Protocol (VoIP). VoIP offers interactive communications. It differs from conventional circuit switched networks. It allows people to communicate with each other at very low rates. The transmission of Real time voice data is not as easy as ordinary text data. The real time voice trans- mission faces lot of difficulties. It suffers from packet loss, delay, security and quality. These factors will affect the com- munication, degrades the performance and quality of a VoIP. This paper addresses the security aspects of VoIP to improve the quality.

The lecture will present a test of the VoIP (Voice over Internet Protocol) network at the University of Applied Sciences (FHTW) Berlin based on Cisco VoIP phones model 7970G as an example in test of security policy of VoIP networks to find out the security gaps and to support improving security of the FHTW's VoIP campus network. Configuration options of the VoIP phone used at the University of Applied Sciences Berlin will be presented. The finding, installing and configuring products for testing the security of VoIP network and VoIP phones have to be dicussed and the security test of Cisco VoIP phones presented offer ways for make more secure configuration and steps to improve the overall security of VoIP networks. The presented results are backed by the long term ERASMUS TN co-operation between the University of Applied Sciences Berlin and the University of Rousse.

Systems employing the Internet and Voice over Internet Protocol (VoIP) are alternative to the legacy landline telephony system. The services offered by these systems allow users to communicate with their family members, friends, banks and business partners whenever they are online. Particularly, the VoIP service is popular with the Internet users because charges are bound into usually fixed access costs making the price of the long distance calls themselves appear economical and even free. However, VoIP services have several disadvantages. One of the disadvantages of VoIP is, it is exposed to Internet security vulnerabilities, threats and attacks. The attacks come in many forms. One of the methods to attack the VoIP service, is by sending a Spam over Internet Telephony (SPIT). SPIT is normally executed by malicious parties initiating automated, unsolicited, and unwanted communications that used VoIP or video conferencing services, just like an email SPAM. Solutions to mitigate SPIT are still lacking. To defense a VoIP system from SPIT, we have come out with the idea of Automated Intrusion Detection and Prevention System over SPIT (AIDPoS) that can detect and prevent SPIT from external parties to enter the VoIP network. AIDPoS is a combination system of Intrusion Detection System (IDS) and Intrusion Prevention System (IPS) on Voice over IP (VoIP) server. This is an experimental research to find a solution to combat SPIT if it happens in the future. Our result shows that an automated Intrusion detection and Prevention system over SPIT attack is feasible. Our work is on an open source VoIP network. Other vulnerabilities and threats are important, but beyond our research scope and is not covered in this paper.

In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, a number of issues have to be solved which have hampered it in the Internet. Therefore, in this paper the VoIP (Voice over Internet Protocol) network design problems are modeled as nonlinear non convex combinatorial integer mathematical programming problems. The optimization problems have great practice value for the network service providers. This problem is a constrained multicast QoS routing and is proved to be a NP complete problem. The total link capacity augmented cost should be minimized when we design a new VoIP network or when the original network could not serve all of the traffic demands. It is also known as a complicated problem. Introduction Originally, the Internet was designed to provide best-effort services for the data generated by computers. The timeliness of data communication is generally delay tolerant. Quality-of-Service (QoS) constraints are not as important as routing flexibility and connectivity. Hence, using IP to transport voice data is contradictory to the basic requirement of the voice service: a timely delivery of voice samples. Although IP was not initially designed to provide services for real-time traffic, recent technical progress has made IP have the capabilities to provide real-time services in near future. In order to provide the same or better service quality in the Internet than traditional circuit-switched telephone network, we must deal with a number of issues that have hampered it in the Internet. Voice service requirements could be discussed from two perspectives: (1) application requirements such as end-to-end delay, jitter, packet loss and overdue probability; (2) user’s perspective such as reliability, availability, and supplementary services [1][2]. Telephony service providers must guarantee the quality of service they provide e.g. maximum one way delay does not exceed 150ms. Internet telephony requires a range of protocols, ranging from those needed for transporting real-time data across to the network e.g. Real-time Transport protocol (RTP) [3], to Quality-of-Service aware routing (QoS routing), signaling protocol, resource reservation, internetworking between IP networks and PSTN, QoS-aware network management and billing protocols. ITU-T defined H.323 to provide multimedia communication in packet networks. From the perspective of network service providers, they want to optimize the network performance such as minimizing the total bandwidth consumption, maximizing throughput or total revenue [4][5] subject to user and application constraints. In this paper, we want to develop the mathematical model for VoIP network. We minimize the total bandwidth consumption under users’ QoS requirements, the network topology and the network capacity. Performance Optimization Model It is generally accepted that Internet telephony and traditional circuit-switched telephony will coexist for quite some time. The VoIP architecture must deal with interworking between IP networks and PSTN, so we need gateways between the two worlds. There are four possible models of VoIP [6]. They are PC-to-PC, Gateway-to-Gateway, PC-to-Gateway, and Gateway-to-PC models. The architecture of VoIP is shown in Fig.1. The first model of VoIP is PC-to-PC architecture, which based 2nd International Conference on Electrical, Computer Engineering and Electronics (ICECEE 2015) © 2015. The authors Published by Atlantis Press 1144 on the assumption that two or more users have access to multimedia computers that are connected to the Internet. Fig.1 VoIP Architecture The VoIP system is modeled as a graph, where the hosts, VoIP gateways, and switches are represented by nodes and communication link sets are represented by links. Let N = {1, 2,..., n} be the set of nodes and L be the set of links in the graph (network). Let G be the set of all user groups. An user group g is a voice communication session requesting for transmission in the network. An user group g may be an unicast from the source to a destination or a multicast from the source to multiple destinations. For each user group g, the traffic is transmitted exactly over one tree. Dg represents the set of destinations of user group g. g λ is the required bandwidth of the voice transmission for each user group g. Below is a verbal description of the VoIP system design problem we considered. Given : Network topology Capacities of network links Equivalent bandwidth of each multicast/unicast group Time threshold and tolerable overdue probability for each multicast/unicast group To determine : The minimum overall bandwidth consumption Routing tree for each multicast/unicast group Overdue probability of each multicast/unicast group Objective : To minimize the total bandwidth consumption Subject to : End-to-end QoS (overdue probability) constraint Tree constraint Multi-commodity flow constraint Capacity constraint Integrality constraint Hop constraint Table 1 and 2 are the notations we use in this paper . Internet VoIP Gateway VoIPGatewa Central office switch or local hub Central office switch or local hub Multimedia PC IP Router Multimedia PC IP Router Gatekeepe r

Mobile and wireless technologies provide Voice over Internet Protocol (VoIP) service to the users with the help of internet. However, there are some limitations with respect to bandwidth, delay, jitter, etc. While it meets congestion and heavy load on the network, it drastically consumes and wastes the resources unnecessarily. Likewise, Mobile Ad-hoc Networks (MANETs) is a part of ad hoc network to provide VoIP communication among the users without the aid of any internet facilities. Since the nodes are mobile in MANETs, it faces many challenges like dynamic topology, limited bandwidth and battery power in order to provide Quality of Service (QoS). This paper proposes a CBT-VoIP technique for reducing the VoIP transmission delay. Ultimately, it helps to improve QoS in MANETs.