Enterprise network convergence: path to cost optimization

During the past two decades, telecommunications has evolved a great deal. In the eighties, people were using television, radio and telephone as their communication systems. Eventually, the introduction of the Internet and the WWW immensely transformed the telecommunications industry. This internet revolution brought about a huge change in the way businesses communicated and operated. Enterprise networks now had an increasing demand for more bandwidth as they started to embrace newer technologies. The requirements of the enterprise networks grew as the applications and services that were used in the network expanded. This stipulation for fast and high performance communication systems has now led to the emergence of converged network solutions. Enterprises across the globe are investigating new ways to implement voice, video, and data over a single network for various reasons – to optimize network costs, to restructure their communication system, to extend next generation networking abilities, or to bridge the gap between their corporate network and the existing technological progress. To date, organizations had multiple network services to support a range of communication needs. Investing in this type of multiple communication infrastructures limits the networks ability to provide resourceful bandwidth optimization services throughout the system. Thus, as the requirements for the corporate networks to handle dynamic traffic grow day by day, the need for a more effective and efficient network arises. A converged network is the solution for enterprises aspiring to employ advanced applications and innovative services. This thesis will emphasize the importance of converging network infrastructure and prove that it leads to cost savings. It discusses the characteristics, architecture, and relevant protocols of the voice, data and video traffic over both traditional infrastructure and converged architecture. While IP-based networks present excellent quality for non real-time data networking, the network by itself is not capable of providing reliable, quality and secure services for real-time traffic. In order for IP networks to perform reliable and timely transmission of real-time data, additional mechanisms to reduce delay, jitter and packet loss are required. Therefore, this thesis will also discuss the important mechanisms for running real-time traffic like voice and video over an IP network. Lastly, it will also provide an example of an enterprise network specifications (voice, video and data), and present an in depth cost analysis of a typical network vs. a converged network to prove that converged infrastructures provide significant savings

Energy-efficient wireless communication

In this chapter we present an energy-efficient highly adaptive network interface architecture and a novel data link layer protocol for wireless networks that provides Quality of Service (QoS) support for diverse traffic types. Due to the dynamic nature of wireless networks, adaptations in bandwidth scheduling and error control are necessary to achieve energy efficiency and an acceptable quality of service. In our approach we apply adaptability through all layers of the protocol stack, and provide feedback to the applications. In this way the applications can adapt the data streams, and the network protocols can adapt the communication parameters

Optimizing IETF multimedia signaling protocols and architectures in 3GPP networks : an evolutionary approach

Signaling in Next Generation IP-based networks heavily relies in the family of multimedia signaling protocols defined by IETF. Two of these signaling protocols are RTSP and SIP, which are text-based, client-server, request-response signaling protocols aimed at enabling multimedia sessions over IP networks. RTSP was conceived to set up streaming sessions from a Content / Streaming Server to a Streaming Client, while SIP was conceived to set up media (e.g.: voice, video, chat, file sharing, …) sessions among users. However, their scope has evolved and expanded over time to cover virtually any type of content and media session. As mobile networks progressively evolved towards an IP-only (All-IP) concept, particularly in 4G and 5G networks, 3GPP had to select IP-based signaling protocols for core mobile services, as opposed to traditional SS7-based protocols used in the circuit-switched domain in use in 2G and 3G networks. In that context, rather than reinventing the wheel, 3GPP decided to leverage Internet protocols and the work carried on by the IETF. Hence, it was not surprise that when 3GPP defined the so-called Packet-switched Streaming Service (PSS) for real-time continuous media delivery, it selected RTSP as its signaling protocol and, more importantly, SIP was eventually selected as the core signaling protocol for all multimedia core services in the mobile (All-)IP domain. This 3GPP decision to use off-the-shelf IETF-standardized signaling protocols has been a key cornerstone for the future of All-IP fixed / mobile networks convergence and Next Generation Networks (NGN) in general. In this context, the main goal of our work has been analyzing how such general purpose IP multimedia signaling protocols are deployed and behave over 3GPP mobile networks. Effectively, usage of IP protocols is key to enable cross-vendor interoperability. On the other hand, due to the specific nature of the mobile domain, there are scenarios where it might be possible to leverage some additional “context” to enhance the performance of such protocols in the particular case of mobile networks. With this idea in mind, the bulk of this thesis work has consisted on analyzing and optimizing the performance of SIP and RTSP multimedia signaling protocols and defining optimized deployment architectures, with particular focus on the 3GPP PSS and the 3GPP Mission Critical Push-to-Talk (MCPTT) service. This work was preceded by a detailed analysis work of the performance of underlying IP, UDP and TCP protocol performance over 3GPP networks, which provided the best baseline for the future work around IP multimedia signaling protocols. Our contributions include the proposal of new optimizations to enhance multimedia streaming session setup procedures, detailed analysis and optimizations of a SIP-based Presence service and, finally, the definition of new use cases and optimized deployment architectures for the 3GPP MCPTT service. All this work has been published in the form of one book, three papers published in JCR cited International Journals, 5 articles published in International Conferences, one paper published in a National Conference and one awarded patent. This thesis work provides a detailed description of all contributions plus a comprehensive overview of their context, the guiding principles beneath all contributions, their applicability to different network deployment technologies (from 2.5G to 5G), a detailed overview of the related OMA and 3GPP architectures, services and design principles. Last but not least, the potential evolution of this research work into the 5G domain is also outlined as well.Els mecanismes de Senyalització en xarxes de nova generació es fonamenten en protocols de senyalització definits per IETF. En particular, SIP i RTSP són dos protocols extensibles basats en missatges de text i paradigma petició-resposta. RTSP va ser concebut per a establir sessions de streaming de continguts, mentre SIP va ser creat inicialment per a facilitar l’establiment de sessions multimèdia (veu, vídeo, xat, compartició) entre usuaris. Tot i així, el seu àmbit d’aplicació s’ha anat expandint i evolucionant fins a cobrir virtualment qualsevol tipus de contingut i sessió multimèdia. A mesura que les xarxes mòbils han anat evolucionant cap a un paradigma “All-IP”, particularment en xarxes 4G i 5G, 3GPP va seleccionar els protocols i arquitectures destinats a gestionar la senyalització dels serveis mòbils presents i futurs. En un moment determinat 3GPP decideix que, a diferència dels sistemes 2G i 3G que fan servir protocols basats en SS7, els sistemes de nova generació farien servir protocols estandarditzats per IETF. Quan 3GPP va començar a estandarditzar el servei de Streaming sobre xarxes mòbils PSS (Packet-switched Streaming Service) va escollir el protocol RTSP com a mecanisme de senyalització. Encara més significatiu, el protocol SIP va ser escollit com a mecanisme de senyalització per a IMS (IP Multimedia Subsystem), l’arquitectura de nova generació que substituirà la xarxa telefònica tradicional i permetrà el desplegament de nous serveis multimèdia. La decisió per part de 3GPP de seleccionar protocols estàndards definits per IETF ha representat una fita cabdal per a la convergència del sistemes All-IP fixes i mòbils, i per al desenvolupament de xarxes NGN (Next Generation Networks) en general. En aquest context, el nostre objectiu inicial ha estat analitzar com aquests protocols de senyalització multimèdia, dissenyats per a xarxes IP genèriques, es comporten sobre xarxes mòbils 3GPP. Efectivament, l’ús de protocols IP és fonamental de cara a facilitar la interoperabilitat de solucions diferents. Per altra banda, hi ha escenaris a on és possible aprofitar informació de “context” addicional per a millorar el comportament d’aquests protocols en al cas particular de xarxes mòbils. El cos principal del treball de la tesi ha consistit en l’anàlisi i optimització del rendiment dels protocols de senyalització multimèdia SIP i RTSP, i la definició d’arquitectures de desplegament, amb èmfasi en els serveis 3GPP PSS i 3GPP Mission Critical Push-to-Talk (MCPTT). Aquest treball ha estat precedit per una feina d’anàlisi detallada del comportament dels protocols IP, TCP i UDP sobre xarxes 3GPP, que va proporcionar els fonaments adequats per a la posterior tasca d’anàlisi de protocols de senyalització sobre xarxes mòbils. Les contribucions inclouen la proposta de noves optimitzacions per a millorar els procediments d’establiment de sessions de streaming multimèdia, l’anàlisi detallat i optimització del servei de Presència basat en SIP i la definició de nous casos d’ús i exemples de desplegament d’arquitectures optimitzades per al servei 3GPP MCPTT. Aquestes contribucions ha quedat reflectides en un llibre, tres articles publicats en Revistes Internacionals amb índex JCR, 5 articles publicats en Conferències Internacionals, un article publicat en Congrés Nacional i l’adjudicació d’una patent. La tesi proporciona una descripció detallada de totes les contribucions, així com un exhaustiu repàs del seu context, dels principis fonamentals subjacents a totes les contribucions, la seva aplicabilitat a diferents tipus de desplegaments de xarxa (des de 2.5G a 5G), així una presentació detallada de les arquitectures associades definides per organismes com OMA o 3GPP. Finalment també es presenta l’evolució potencial de la tasca de recerca cap a sistemes 5G.Postprint (published version

Satellite Networks: Architectures, Applications, and Technologies

Since global satellite networks are moving to the forefront in enhancing the national and global information infrastructures due to communication satellites' unique networking characteristics, a workshop was organized to assess the progress made to date and chart the future. This workshop provided the forum to assess the current state-of-the-art, identify key issues, and highlight the emerging trends in the next-generation architectures, data protocol development, communication interoperability, and applications. Presentations on overview, state-of-the-art in research, development, deployment and applications and future trends on satellite networks are assembled

Quality of service over ATM networks

PhDAbstract not availabl

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

Call admission and routing in telecommunication networks.

by Kit-man Chan.Thesis (M.Phil.)--Chinese University of Hong Kong, 1994.Includes bibliographical references (leaves 82-86).Chapter 1 --- Introduction --- p.1Chapter 1.1 --- Overview of Integrated Service Digital Networks --- p.1Chapter 1.2 --- Multirate Loss Networks --- p.5Chapter 1.3 --- Previous Work --- p.7Chapter 1.4 --- Organization --- p.11Chapter 1.5 --- Publications --- p.12Chapter 2 --- Call Admission in Multirate Loss Networks --- p.13Chapter 2.1 --- Introduction --- p.13Chapter 2.2 --- Two Adaptive Routing Rules --- p.15Chapter 2.3 --- Call Admission Policies --- p.17Chapter 2.4 --- Analysis of Call Admission Policies --- p.25Chapter 2.4.1 --- "The CS, LO, GB and the EB Policies" --- p.25Chapter 2.4.2 --- The DP Policy --- p.29Chapter 2.5 --- Performance Comparisons --- p.32Chapter 2.6 --- Concluding Remarks --- p.35Chapter 3 --- Least Congestion Routing in Multirate Loss Networks --- p.41Chapter 3.1 --- Introduction --- p.41Chapter 3.2 --- The M2 and MTB Routings --- p.42Chapter 3.2.1 --- M2 Routing --- p.43Chapter 3.2.2 --- MTB Routing --- p.43Chapter 3.3 --- Bandwidth Sharing Policies and State Aggregation --- p.45Chapter 3.4 --- Analysis of M2 Routing --- p.47Chapter 3.5 --- Analysis of MTB Routing --- p.50Chapter 3.6 --- Numerical Results and Discussions --- p.53Chapter 3.7 --- Concluding Remarks --- p.56Chapter 4 --- The Least Congestion Routing in WDM Lightwave Networks --- p.60Chapter 4.1 --- Introduction --- p.60Chapter 4.2 --- Architecture and Some Design Issues --- p.62Chapter 4.3 --- The Routing Rule --- p.66Chapter 4.4 --- Analysis of the LC Routing Rule --- p.67Chapter 4.4.1 --- Fixed Point Model --- p.67Chapter 4.4.2 --- Without Direct-link Priority --- p.68Chapter 4.4.3 --- With Direct-link Priority --- p.72Chapter 4.5 --- Performance Comparisons --- p.73Chapter 4.6 --- Concluding Remarks --- p.75Chapter 5 --- Conclusions and Future Work --- p.79Chapter 5.1 --- Future Work --- p.8

Simulation and analysis of adaptive routing and flow control in wide area communication networks

This thesis presents the development of new simulation and analytic models for the performance analysis of wide area communication networks. The models are used to analyse adaptive routing and flow control in fully connected circuit switched and sparsely connected packet switched networks. In particular the performance of routing algorithms derived from the L(_R-I) linear learning automata model are assessed for both types of network. A novel architecture using the INMOS Transputer is constructed for simulation of both circuit and packet switched networks in a loosely coupled multi- microprocessor environment. The network topology is mapped onto an identically configured array of processing centres to overcome the processing bottleneck of conventional Von Neumann architecture machines. Previous analytic work in circuit switched work is extended to include both asymmetrical networks and adaptive routing policies. In the analysis of packet switched networks analytic models of adaptive routing and flow control are integrated to produce a powerful, integrated environment for performance analysis The work concludes that routing algorithms based on linear learning automata have significant potential in both fully connected circuit switched networks and sparsely connected packet switched networks

IP and ATM integration: A New paradigm in multi-service internetworking

ATM is a widespread technology adopted by many to support advanced data communication, in particular efficient Internet services provision. The expected challenges of multimedia communication together with the increasing massive utilization of IP-based applications urgently require redesign of networking solutions in terms of both new functionalities and enhanced performance. However, the networking context is affected by so many changes, and to some extent chaotic growth, that any approach based on a structured and complex top-down architecture is unlikely to be applicable. Instead, an approach based on finding out the best match between realistic service requirements and the pragmatic, intelligent use of technical opportunities made available by the product market seems more appropriate. By following this approach, innovations and improvements can be introduced at different times, not necessarily complying with each other according to a coherent overall design. With the aim of pursuing feasible innovations in the different networking aspects, we look at both IP and ATM internetworking in order to investigating a few of the most crucial topics/ issues related to the IP and ATM integration perspective. This research would also address various means of internetworking the Internet Protocol (IP) and Asynchronous Transfer Mode (ATM) with an objective of identifying the best possible means of delivering Quality of Service (QoS) requirements for multi-service applications, exploiting the meritorious features that IP and ATM have to offer. Although IP and ATM often have been viewed as competitors, their complementary strengths and limitations from a natural alliance that combines the best aspects of both the technologies. For instance, one limitation of ATM networks has been the relatively large gap between the speed of the network paths and the control operations needed to configure those data paths to meet changing user needs. IP\u27s greatest strength, on the other hand, is the inherent flexibility and its capacity to adapt rapidly to changing conditions. These complementary strengths and limitations make it natural to combine IP with ATM to obtain the best that each has to offer. Over time many models and architectures have evolved for IP/ATM internetworking and they have impacted the fundamental thinking in internetworking IP and ATM. These technologies, architectures, models and implementations will be reviewed in greater detail in addressing possible issues in integrating these architectures s in a multi-service, enterprise network. The objective being to make recommendations as to the best means of interworking the two in exploiting the salient features of one another to provide a faster, reliable, scalable, robust, QoS aware network in the most economical manner. How IP will be carried over ATM when a commercial worldwide ATM network is deployed is not addressed and the details of such a network still remain in a state of flux to specify anything concrete. Our research findings culminated with a strong recommendation that the best model to adopt, in light of the impending integrated service requirements of future multi-service environments, is an ATM core with IP at the edges to realize the best of both technologies in delivering QoS guarantees in a seamless manner to any node in the enterprise