Optimizing The MPLS Support For Real Time IPv6-Flows Using MPLS-PHS Approach.

The huge coverage space of IPv6 addresses and providing guaranteed support for the ever increasing customer demand, results in the dealing with bigger packet header-size compared to the payload-size especially in some real time video and audio applications, consequently more bandwidth is wasting

Performance evaluation of AAL2 over IP in the UMTS access network Iub interface

Bibliography: leaves 84-86.In this study, we proposed to retain AAL2 and lay it over IP (AAL2IIP). The IP-based lub interface is therefore designed to tunnel AAL2 channels from the Node B to the RNC. Currently IP routes packets based on best-effort which does not guarantee QoS, To provide QoS, MPLS integrated with DiffServ is proposed to support different QoS levels to different classes of service and fast forward the IP packets within the lub interface. To evaluate the performance of AAL2!IP in the Iub interface, a test-bed was created

QoS SOLUTIONS FORVIDEOCONFERENCING

This project is intended to gain knowledge and apply the theory leamt about the need of QoS in videoconferencing and the various options available. Today's conferencing applications are now IP friendly, it can run on either dedicated lines (like ISDN or telephone lines) or IP networks. However, as most network administrators know, conferencingapplications can wreak havoc on unprepared corporate networks. The key to successfully deploying conferencing applications is the activation of Quality of Service (QoS). QoS refers to a network's ability to reliably and consistently provide a certain level of throughput and performance. QoS for conferencing typically involves network availability, bandwidth, end-to-end delay, jitter, and packet loss. Simply stated, if the network doesn't conform to the minimum requirements in any of these areas, the conferences are doomed to fail. QoS can be achieved in a variety of ways, including over-provisioning (deploying additional bandwidth), data prioritization, and the use of QoS-enabled overlay or converged networks. Organizations have two main options for deploying QoS within their organizations; convergence or overlay. Convergence requires the use of QoS-capable WAN links throughout the organization. In many cases, this requires a fork-lift upgrade and migration of all network resources, which can place convergence out of reach of many cost-sensitive organizations. On the other hand, overlay networks allow a step-by-step migration from a non-QoS to a QoS network without the high cost and inherent risk of major network reconfigurations. In this way, overlay networks are a first step toward convergenc

Quality of Service over Specific Link Layers: state of the art report

The Integrated Services concept is proposed as an enhancement to the current Internet architecture, to provide a better Quality of Service (QoS) than that provided by the traditional Best-Effort service. The features of the Integrated Services are explained in this report. To support Integrated Services, certain requirements are posed on the underlying link layer. These requirements are studied by the Integrated Services over Specific Link Layers (ISSLL) IETF working group. The status of this ongoing research is reported in this document. To be more specific, the solutions to provide Integrated Services over ATM, IEEE 802 LAN technologies and low-bitrate links are evaluated in detail. The ISSLL working group has not yet studied the requirements, that are posed on the underlying link layer, when this link layer is wireless. Therefore, this state of the art report is extended with an identification of the requirements that are posed on the underlying wireless link, to provide differentiated Quality of Service

A traffic engineering system for DiffServ/MPLS networks

This thesis presents an approach to traffic engineering that uses DiffServ and MPLS technologies to provide QoS guarantees over an IP network. The specific problem described here is how best to route traffic within the network such that the demands can be carried with the requisite QoS while balancing the load on the network. A traffic engineering algorithm that determines QoS guaranteed label-switched paths (LSPs) between specified ingress-egress pairs is proposed and a system that uses such an algorithm is outlined. The algorithm generates a solution for the QoS routing problem of finding a path with a number of constraints (delay, jitter, loss) while trying to make best of resource utilisation. The key component of the system is a central resource manager responsible for monitoring and managing resources within the network and making all decisions to route traffic according to QoS requirements. The algorithm for determining QoS-constrained routes is based on the notion of effective bandwidth and cost functions for load balancing. The network simulation of the proposed system is presented here and simulation results are discussed

An optimized framework for header suppression of real time IPV6 traffic in multiprotocol label switching (MPLS) networks.

Pensuisan Label Multiprotokol (MPLS) dengan IPv6 telah dinyatakan oleh Pasukan Petugas Kejuruteraan Internet (IETF) sebagai mampu diskalakan dan sangat sesuai untuk jenis-jenis trafik yang berlainan seperti VoIP dan Video. Namun, kepala IP yang besar melahirkan overhed kepala yang berlebihan dalam rangkaian MPLS, mengakibatkan kesesakan trafik lalu menjejaskan prestasi rangkaian tulang belakang. Multiprotocol Label Switching (MPLS) with IPv6 has been defined by the Internet Engineering Task Force (IETF) as highly scalable and well suited for different types of traffic such as VoIP and Video. However, large IP headers create excessive header overhead in a MPLS network leading to traffic congestion degrading the backbone network performance

Rohc-Mpls Tunnel Architecture For Wireless Mesh

Natural or human-made disasters are sudden events that can cause significant damage, especially to the network communication infrastructure. In these events, a rapid deployment of network communication systems is required in order to relay or receive the communication among the people in the disaster areas to conduct relief and rescue efforts. Wireless mesh networks have emerged and has been recognised for its potential for rapid deployment and last mile coverage of network infrastructure, which is highly suitable for emergency response management. While wireless mesh networks have beneficial attributes, it also introduces some crucial problems. During data transmission, the path recovery time is significantly higher resulting in the loss of data if node and link failures occur

ADAPTIVE SPEECH QUALITY IN VOICE-OVER-IP COMMUNICATIONS

The quality of VoIP communication relies significantly on the network that transports the voice packets because this network does not usually guarantee the available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage the voice-over-IP stream dynamically, changing parameters as needed to assure quality. The main objective of this dissertation is to develop an adaptive speech encoding system that can be applied to conventional (telephony-grade) and wideband voice communications. This comprehensive study includes the investigation and development of three key components of the system. First, to manage VoIP quality dynamically, a tool is needed to measure real-time changes in quality. The E-model, which exists for narrowband communication, is extended to a single computational technique that measures speech quality for narrowband and wideband VoIP codecs. This part of the dissertation also develops important theoretical work in the area of wideband telephony. The second system component is a variable speech-encoding algorithm. Although VoIP performance is affected by multiple codecs and network-based factors, only three factors can be managed dynamically: voice payload size, speech compression and jitter buffer management. Using an existing adaptive jitter-buffer algorithm, voice packet-size and compression variation are studied as they affect speech quality under different network conditions. This study explains the relationships among multiple parameters as they affect speech transmission and its resulting quality. Then, based on these two components, the third system component is a novel adaptive-rate control algorithm that establishes the interaction between a VoIP sender and receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average voice quality than traditional VoIP