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IP Networks (ip + network)
Kinds of IP Networks Selected AbstractsTrunking of TDM and narrowband services over IP NetworksINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 1 2003Dr James Aweya The recent interest in IP as the vehicle for transporting TDM and narrowband services stems from the possibility of using a common transport network for voice, video, and data, and the flexibility with which new services can be introduced. A key step in the evolution of networks towards a ,broadband' IP-based environment is the ,graceful' interworking of the IP networks with the existing networks and services, particularly with the circuit switched telephone network. A ,graceful' interworking approach is one whose complexity is minimal and preserves the user's perceived quality of service (QoS). To interwork with a circuit switched network whose services are pre-dominantly time-sensitive, the IP network must essentially behave as a transparent ,link' in the end-to-end connection. This paper presents an overview of the main technical problems to be addressed when trunking TDM and narrowband services over IP networks. Copyright © 2002 John Wiley & Sons, Ltd. [source] Clock synchronization for packet networks using a weighted least-squares error filtering technique and enabling circuit emulation serviceINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 6 2007James Aweya Abstract Circuit emulation service (CES) allows time-division multiplexing (TDM) services (T1/E1 and T3/E3 circuits) to be transparently extended across a packet network. With circuit emulation over IP, for instance, TDM data received from an external device at the edge of an IP network is converted to IP packets, sent through the IP network, passed out of the IP network to its destination, and reassembled into TDM bit stream. Clock synchronization is very important for CES. This paper presents a clock synchronization scheme based on a double exponential filtering technique and a linear process model. The linear process model is used to describe the behaviour of clock synchronization errors between a transmitter and a receiver. In the clock synchronization scheme, the transmitter periodically sends explicit time indications or timestamps to a receiver to enable the receiver to synchronize its local clock to the transmitter's clock. A phase-locked loop (PLL) at the receiver processes the transmitted timestamps to generate timing signal for the receiver. The PLL has a simple implementation and provides both fast responsiveness (i.e. fast acquisition of transmitter frequency at a receiver) and significant jitter reduction in the locked state. Copyright © 2006 John Wiley & Sons, Ltd. [source] Smoothing and transporting video in QoS IP networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 7 2006Khaled Shuaib Abstract Real-time traffic such as voice and video, when transported over the Internet, demand stringent quality of service (QoS) requirements. The current Internet as of today is still used as a best effort environment with no quality guarantees. An IP-based Internet that supports different QoS requirements for different applications has been evolving for the past few years. Video streams are bursty in nature due to the instant variability of the video content being encoded. To help mitigate the transport of bursty video streams with minimal loss of information, rate-adaptive shapers (RASs) are usually being used to reduce the burstiness and therefore help preserve the desired quality. When transporting video over a QoS IP network, each stream is classified under a specific traffic profile to which it must conform, to avoid packet loss and picture quality degradation. In this paper we study, evaluate and propose RASs for the transport of video over a QoS IP network. We utilize the encoding video parameters for choosing the appropriate configuration needed to support the real-time transport of Variable Bit Rate (VBR) encoded video streams. The performance evaluation of the different RASs is based on the transport of MPEG-4 video streams encoded as VBR. The performance is studied based on looking at the effect of various parameters associated with the RASs on the effectiveness of smoothing out the burstiness of video and minimizing the probability of packet loss. Copyright © 2005 John Wiley & Sons, Ltd. [source] Call admission control for voice over IPINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 4 2006Huseyin Uzunalioglu Abstract Voice over Internet protocol (VoIP) is a technology that enables the transmission of voice over an IP network. Recent years have witnessed heavy investment in this area in the commercial world. For VoIP to replace Public Switched Telephone Network (PSTN), it should provide voice quality comparable to circuit-switched PSTN networks. This paper addresses the mechanisms to guarantee VoIP quality of service (QoS). The focus is given to the call admission control, which blocks voice calls when the required resources are not available to guarantee the QoS for the call. We review call admission control approaches that can be applied to VoIP, and describe the advantages and disadvantages of each approach. In the second part of the paper, we present a measurement-based admission control scheme that achieves QoS in an efficient and scalable manner. The scheme uses voice traffic load measurements at each router link to compute link-level blocking policies for new call attempts. Then, these policies are translated into path-level blocking policies, which are applied to new call set-up requests. The performance of the scheme is presented for single and multiple-priority voice calls. Copyright © 2006 John Wiley & Sons, Ltd. [source] An adaptive path routing scheme for satellite IP networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 1 2003Jing Chen Abstract Mobile satellites can be considered as the promising solution to the global IP network. In order to provide quality of service (QoS) in future networks, mobile satellite can be integrated with the asynchronous transfer mode (ATM) to switch IP datagrams in the space. For such a network, new and sophisticated routing and handoff algorithms are essential. In this paper, a new scheme called adaptive path routing scheme (APRS) is proposed. It is shown that the APRS can provide superior performance for routing and handoff in mobile satellite networks compared with conventional schemes. Copyright © 2003 John Wiley & Sons, Ltd. [source] AAA architecture for mobile IPv6 based on WLANINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 5 2004R. C. Wang Mobility support for Internet devices is quite important for consumer electronics. The number of the hand-held devices is growing quickly. However, there are not enough IP addresses for the number of the rapidly growing devices in the All-IP generation. Internet Protocol version 6 (IPv6) was therefore adopted to solve these problems. Our purposed structure is based on IEEE 802.11. However, IEEE 802.11 has a serious security drawback. Further, from the Internet Service Providers' point of view, accounting is a potential problem. A mechanism combining Mobile IPv6 and AAA based on IEEE 802.11 to overcome these problems is essential. Both Internet Protocol version 4 (IPv4) and IPv6 support IP security (IPsec) when data packets are exchanged across the IP network. IPsec operates at the IP layer. It can support system authentication and authorization, However, it lacks a system accounting function. Therefore ISPs cannot establish correct billing for their services. This is the reason why we chose to combine the wireless network and AAA functions. In this paper, the AAA mechanism is used to protect security, with the architecture having authentication, authorization, and accounting functions. We will discuss the benefits of AAA and state the reason why we choose to combine AAA with the mobility architecture.,Copyright © 2004 John Wiley & Sons, Ltd. [source] Trunking of TDM and narrowband services over IP NetworksINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 1 2003Dr James Aweya The recent interest in IP as the vehicle for transporting TDM and narrowband services stems from the possibility of using a common transport network for voice, video, and data, and the flexibility with which new services can be introduced. A key step in the evolution of networks towards a ,broadband' IP-based environment is the ,graceful' interworking of the IP networks with the existing networks and services, particularly with the circuit switched telephone network. A ,graceful' interworking approach is one whose complexity is minimal and preserves the user's perceived quality of service (QoS). To interwork with a circuit switched network whose services are pre-dominantly time-sensitive, the IP network must essentially behave as a transparent ,link' in the end-to-end connection. This paper presents an overview of the main technical problems to be addressed when trunking TDM and narrowband services over IP networks. Copyright © 2002 John Wiley & Sons, Ltd. [source] Dimensioning of data networks: a flow-level perspectiveEUROPEAN TRANSACTIONS ON TELECOMMUNICATIONS, Issue 6 2009Pasi Lassila Traditional network dimensioning formulations have applied the Erlang model where the connections reserve capacity in the network. Until recently, tractable stochastic network models where the connections share the capacity in the network did not exist. The latter are becoming increasingly important as they can be applied to characterise file transfers in current data networks (e.g. IP networks). In particular, they can be utilised for dimensioning of networks with respect to the file transfer performance. To this end, we consider a model where the traffic consists of elastic flows (i.e. file transfers). Flows arrive randomly and share the network resources resulting in stochastically varying transmission rates for flows. Our contribution is to develop efficient methods for capacity planning to meet the performance requirements expressed in terms of the average transmission rate of flows on a given route, i.e. the per-flow throughput. These methods are validated using ns2 simulations. We discuss also the effects of access rate limitations and how to combine the elastic traffic requirements with those of real-time traffic. Finally, we outline how the methods can be applied in wireless mesh networks. Our results enable a simple characterisation of the order-of-magnitude of the required capacities, which can be utilised as a first step in practical network planning and dimensioning. Copyright © 2008 John Wiley & Sons, Ltd. [source] Experimental analysis of the impact of peer-to-peer applications on traffic in commercial IP networksEUROPEAN TRANSACTIONS ON TELECOMMUNICATIONS, Issue 6 2004Nadia Ben Azzouna To evaluate the impact of peer-to-peer (P2P) applications on traffic in wide area networks, we analyze measurements from a high speed IP backbone link carrying TCP traffic towards several ADSL areas. The first observations are that the prevalent part of traffic is due to P2P applications (almost 80% of total traffic) and that the usage of network becomes symmetric in the sense that customers are not only clients but also servers. This latter point is observed by the significant proportion of long flows mainly composed of ACK segments. When analyzing the bit rate created by long flows, it turns out that the TCP connections due to P2P applications have a rather small bit rate and that there is no evidence for long range dependence. These facts are intimately related to the way P2P protocols are running. We separately analyze signaling traffic and data traffic. It turns out that by adopting a suitable level of aggregation, global traffic can be described by means of usual tele-traffic models based on M/G/, queues with Weibullian service times. Copyright © 2004 AEI [source] QoS in IntServ-based IP networks: the peak rate policingEUROPEAN TRANSACTIONS ON TELECOMMUNICATIONS, Issue 4 2003Lorenzo Battaglia In the last few years, IP has moved towards resource reservation, with the task to guarantee in the future Quality of Service (QoS). This has led to flow admission control algorithms based on the negotiation of standardised traffic parameters. QoS can be guaranteed in any network, a priori from the used technology, only if the used admission control algorithm wisely shares the network's resources among the users. Any admission control algorithm on its turn can do so, only if every user respects the negotiated traffic parameters. Since any user could, maliciously or not, send at a higher rate than negotiated, i.e. use a higher share of resources than the negotiated one, in every network in which admission control is performed, a policing algorithm is used. An ideal policer should guarantee to reject no packet of a well-behaved user and police contract violation as rigidly as possible. All this independently of the characteristics of the monitored stream and of the background traffic. This holds also for Integrated Services (IS) based IP networks. In these networks, every user negotiates a peak and an average rate. In this paper we present the solution to the peak rate policing issue. We adapt the Generic Cell Rate Algorithm (GCRA), well-known policer used in ATM networks, to police the peak rate of flows of packets with variable length. We intuitively call this modified GCRA Generic Packet Rate Algorithm (GPRA) and dimension its parameters so that independently of the characteristics of the policed flow and of the background traffic, no packets of a well-behaved user are rejected and that the flows of any misbehaving user are rigidly policed. Copyright © 2003 AEI. [source] Reliable and efficient multicast protocol for mobile IP networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 7 2008Sang-Jo Yoo Abstract To provide a multicasting service, several multicast protocols for mobile hosts (MHs) have been proposed. However, all of these protocols have faults, such as non-optimal delivery routes and data loss when hosts move to another network, resulting in insecure multicast data transmissions. Thus, this paper presents a new reliable and efficient multicast routing protocol for mobile IP networks. The proposed protocol provides a reliable multicast transmission by compensating the data loss from the previous mobile agent when a MH moves to another network. In addition, an additional function allows for direct connection to the multicast tree according to the status of agents, thereby providing a more efficient and optimal multicast path. The performance of the proposed protocol is confirmed based on simulations under various conditions. Copyright © 2008 John Wiley & Sons, Ltd. [source] Performance of delay-sensitive traffic in multi-layered satellite IP networks with on-board processing capabilityINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 12 2007Suzan Bayhan Abstract In this article, performance of delay-sensitive traffic in multi-layered satellite Internet Protocol (IP) networks with on-board processing (OBP) capability is investigated. With OBP, a satellite can process the received data, and according to the nature of application, it can decide on the transmission properties. First, we present a concise overview of relevant aspects of satellite networks to delay-sensitive traffic and routing. Then, in order to improve the system performance for delay-sensitive traffic, specifically Voice over Internet Protocol (VoIP), a novel adaptive routing mechanism in two-layered satellite network considering the network's real-time information is introduced and evaluated. Adaptive Routing Protocol for Quality of Service (ARPQ) utilizes OBP and avoids congestion by distributing traffic load between medium-Earth orbit and low-Earth orbit layers. We utilize a prioritized queueing policy to satisfy quality-of-service (QoS) requirements of delay-sensitive applications while evading non-real-time traffic suffer low performance level. The simulation results verify that multi-layered satellite networks with OBP capabilities and QoS mechanisms are essential for feasibility of packet-based high-quality delay-sensitive services which are expected to be the vital components of next-generation communications networks. Copyright © 2007 John Wiley & Sons, Ltd. [source] Smoothing and transporting video in QoS IP networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 7 2006Khaled Shuaib Abstract Real-time traffic such as voice and video, when transported over the Internet, demand stringent quality of service (QoS) requirements. The current Internet as of today is still used as a best effort environment with no quality guarantees. An IP-based Internet that supports different QoS requirements for different applications has been evolving for the past few years. Video streams are bursty in nature due to the instant variability of the video content being encoded. To help mitigate the transport of bursty video streams with minimal loss of information, rate-adaptive shapers (RASs) are usually being used to reduce the burstiness and therefore help preserve the desired quality. When transporting video over a QoS IP network, each stream is classified under a specific traffic profile to which it must conform, to avoid packet loss and picture quality degradation. In this paper we study, evaluate and propose RASs for the transport of video over a QoS IP network. We utilize the encoding video parameters for choosing the appropriate configuration needed to support the real-time transport of Variable Bit Rate (VBR) encoded video streams. The performance evaluation of the different RASs is based on the transport of MPEG-4 video streams encoded as VBR. The performance is studied based on looking at the effect of various parameters associated with the RASs on the effectiveness of smoothing out the burstiness of video and minimizing the probability of packet loss. Copyright © 2005 John Wiley & Sons, Ltd. [source] Managing QoS requirements for video streaming: from intra-node to inter-nodeINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 5 2006Y. Bai Abstract Streaming video over IP networks has become increasingly popular; however, compared to traditional data traffic, video streaming places different demands on quality of service (QoS) in a network, particularly in terms of delay, delay variation, and data loss. In response to the QoS demands of video applications, network techniques have been proposed to provide QoS within a network. Unfortunately, while efficient from a network perspective, most existing solutions have not provided end-to-end QoS that is satisfactory to users. In this paper, packet scheduling and end-to-end QoS distribution schemes are proposed to address this issue. The design and implementation of the two schemes are based on the active networking paradigm. In active networks, routers can perform user-driven computation when forwarding packets, rather than just simple storing and forwarding packets, as in traditional networks. Both schemes thus take advantage of the capability of active networks enabling routers to adapt to the content of transmitted data and the QoS requirements of video users. In other words, packet scheduling at routers considers the correlation between video characteristics, available local resources and the resulting visual quality. The proposed QoS distribution scheme performs inter-node adaptation, dynamically adjusting local loss constraints in response to network conditions in order to satisfy the end-to-end loss requirements. An active network-based simulation shows that using QoS distribution and packet scheduling together increases the probability of meeting end-to-end QoS requirements of networked video. Copyright © 2005 John Wiley & Sons, Ltd. [source] Measurement-based admission control scheme with priority and service classes for application in wireless IP networks,INTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 6 2003Abbas Jamalipour Abstract Wireless IP networks will provide voice and data services using IP protocols over the wireless channel. But current IP is unsuitable to provide delay or loss bounds and insufficient to support diverse quality of service, both required by real-time applications. In order to support real-time applications in wireless IP networks, in this paper a measurement-based admission control (MBAC) with priority criteria and service classes is considered. First we have shown the suitability of MBAC in wireless IP networks by comparing its performance with a parameter-based scheme. Next, we have investigated the performance of strictly policy-based MBAC and policy plus traffic characteristic-based MBAC schemes in terms of (1) increasing the user mobility, (2) changing traffic parameters and (3) the presence of greedy users. The efficiency and fairness of each scheme are measured in terms of lower class new and handoff traffic performance. Copyright © 2003 John Wiley & Sons, Ltd. [source] An adaptive path routing scheme for satellite IP networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 1 2003Jing Chen Abstract Mobile satellites can be considered as the promising solution to the global IP network. In order to provide quality of service (QoS) in future networks, mobile satellite can be integrated with the asynchronous transfer mode (ATM) to switch IP datagrams in the space. For such a network, new and sophisticated routing and handoff algorithms are essential. In this paper, a new scheme called adaptive path routing scheme (APRS) is proposed. It is shown that the APRS can provide superior performance for routing and handoff in mobile satellite networks compared with conventional schemes. Copyright © 2003 John Wiley & Sons, Ltd. [source] An analytical simulator for deploying IP telephonyINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 1 2009K. Salah Deploying IP telephony or voice over IP (VoIP) is a major and challenging task. This paper describes an analytical design and planning simulator to assess the readiness of existing IP networks for the deployment of VoIP. The analytical simulator utilizes techniques used for network flows and queuing network analysis to compute two key performance bounds for VoIP: delay and bandwidth. The simulator is GUI-based and has an interface with drag-and-drop features to easily construct any generic network topology. The simulator has an engine that automates and implements the analytical techniques. The engine determines the number of VoIP calls that can be sustained by the constructed network while satisfying VoIP QoS requirements and leaving adequate capacity for future growth. As a case study, the paper illustrates how the simulator can be utilized to assess the readiness to deploy VoIP for a typical network of a small enterprise. We have made the analytical simulator publicly available in order to improve and ease the process of VoIP deployment. Copyright © 2008 John Wiley & Sons, Ltd. [source] An OPNET-based simulation approach for deploying VoIPINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 3 2006K. Salah These days a massive deployment of VoIP is taking place over IP networks. VoIP deployment is a challenging task for network researchers and engineers. This paper presents a detailed simulation approach for deploying VoIP successfully. The simulation uses the OPNET network simulator. Recently OPNET has gained a considerable popularity in both academia and industry, but there is no formal or known approach or methodology as to how OPNET can be used to assess the support and readiness of an existing network in deploying VoIP. Our approach and work presented in this paper predict, prior to the purchase and deployment of VoIP equipment, the number of VoIP calls that can be sustained by an existing network while satisfying QoS requirements of all network services and leaving adequate capacity for future growth. As a case study, we apply the simulation approach on a typical network of a small enterprise. The paper presents a detailed description of simulation models for network topology and elements using OPNET. The paper describes modeling and representation of background and VoIP traffic, as well as various simulation configurations. Moreover, the paper discusses many design and engineering issues pertaining to the deployment of VoIP. These issues include characteristics of VoIP traffic and QoS requirements, VoIP flow and call distribution, defining future growth capacity, and measurement and impact of background traffic.,Copyright © 2006 John Wiley & Sons, Ltd. [source] A new policy-aware terminal for QoS, AAA and mobility managementINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 2 2004Hakima Chaouchi Policy-based management has been widely studied in recent years. The Internet Engineering Task Force (IETF) has recently introduced the policy-based networking as a means of managing IP networks according to the new constraints defined in the network, such as the guarantee of the quality of service (QoS). Network management based on policies, is modelled as a state machine, which moves from one state to another according to the enforced policy. The IETF policy-based networking is defined for application to network nodes. However, some recent work suggests extending the policy-based networking to the end-user terminals. In this paper, we present an analysis of such an extension and we propose some possible solutions to support new policy-aware terminals. In addition, we present AAA, QoS and mobility management that user such a policy-aware terminals.,Copyright © 2004 John Wiley & Sons, Ltd. [source] Network capacity allocation for traffic with time prioritiesINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 6 2003Xian Liu The packet switching techniques are undergoing evolution. The conventional ,best effort' approach will no longer be the dominant service. The next generation of IP networks must provide QoS to customers. Inadequacy is obvious when the conventional capacity allocation (CA) models are applied to the new IP architecture. In this paper, we propose several CA models that characterize: (1) the service priority scheme; (2) the service preemption scheme; and (3) the non-Poisson traffic in which the packets follow heavy tailed distributions.,Copyright © 2003 John Wiley & Sons, Ltd. [source] Trunking of TDM and narrowband services over IP NetworksINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 1 2003Dr James Aweya The recent interest in IP as the vehicle for transporting TDM and narrowband services stems from the possibility of using a common transport network for voice, video, and data, and the flexibility with which new services can be introduced. A key step in the evolution of networks towards a ,broadband' IP-based environment is the ,graceful' interworking of the IP networks with the existing networks and services, particularly with the circuit switched telephone network. A ,graceful' interworking approach is one whose complexity is minimal and preserves the user's perceived quality of service (QoS). To interwork with a circuit switched network whose services are pre-dominantly time-sensitive, the IP network must essentially behave as a transparent ,link' in the end-to-end connection. This paper presents an overview of the main technical problems to be addressed when trunking TDM and narrowband services over IP networks. Copyright © 2002 John Wiley & Sons, Ltd. [source] Quality of service for satellite IP networks: a surveyINTERNATIONAL JOURNAL OF SATELLITE COMMUNICATIONS AND NETWORKING, Issue 4-5 2003Sastri Kota Abstract The future media rich applications such as media streaming, content delivery distribution and broadband access require a network infrastructure that offers greater bandwidth and service level guarantees. As the demand for new applications increases, ,best effort' service is inadequate and results in lack of user satisfaction. End-to-end quality of service (QoS) requires the functional co-operation of all network layers. To meet future application requirements, satellite is an excellent candidate due to features such as global coverage, bandwidth flexibility, broadcast, multicast and reliability. At each layer, the user performance requirements should be achieved by implementation of efficient bandwidth allocation algorithms and satellite link impairment mitigation techniques. In this paper, a QoS framework for satellite IP networks including requirements, objectives and mechanisms are described. To fully understand end-to-end QoS at each layer, QoS parameters and the current research are surveyed. For example at physical layer (modulation, adaptive coding), link layer (bandwidth allocation), network layer (IntServ/DiffServ, MPLS traffic engineering), transport layer (TCP enhancements, and alternative transport protocols) and security issues are discussed. Some planned system examples, QoS simulations and experimental results are provided. The paper also includes the current status of the standardization of satellite IP by ETSI, ITU and IETF organizations. Copyright © 2003 John Wiley & Sons, Ltd. [source] TCP-Peach for satellite networks: analytical model and performance evaluationINTERNATIONAL JOURNAL OF SATELLITE COMMUNICATIONS AND NETWORKING, Issue 5 2001Ian F. Akyildiz Abstract Current TCP protocols have low throughput performance in satellite networks mainly due to the effects of long propagation delays and high link error rates. TCP-Peach is a new congestion control scheme for satellite IP networks based on the use of low priority segments, called dummy segments. The sender transmits dummy segments to probe the availability of network resources. Dummy segments are treated as low priority segments thus, they do not effect the throughput of actual data segments. In this paper, TCP-Peach is presented along with its analytical model which is used to evaluate the throughput performance. Experiments show that TCP-Peach is robust to high link error rates as well as long propagation delays, and outperforms other TCP schemes for satellite networks. Copyright © 2001 John Wiley & Sons, Ltd. [source] Monitoring infrastructure for converged networks and servicesBELL LABS TECHNICAL JOURNAL, Issue 2 2007Shipra Agrawal Network convergence is enabling service providers to deploy a wide range of services such as Voice over Internet Protocol (VoIP), Internet Protocol television (IPTV), and push-to-talk on the same underlying IP networks. Each service has unique performance requirements from the network, and IP networks have not been designed to satisfy these diverse requirements easily. These requirements drive the need for a robust, scalable, and easy-to-use network management platform that enables service providers to monitor and manage their networks to provide the necessary quality, availability, and security. In this paper, we describe monitoring mechanisms that give service providers critical information on the performance of their networks at a per-user, per-service granularity in real time. This allows the service providers to ensure that their networks adequately satisfy the requirements of the various services. We present various methods to acquire data, which can be analyzed to determine the performance of the network. This platform enables service providers to offer carrier grade services over their converged networks, giving their customers a high-quality experience. © 2007 Alcatel-Lucent. [source] Ensuring fairness among ECN and non-ECN TCP over the InternetINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 5 2003Salahuddin Muhammad Salim Zabir Explicit Congestion Notification (ECN) has been proved to provide a fast indication of incipient congestion and thus better the performance of a TCP/IP network. In this work, we carry out investigations on gateway or router performance in providing fairnesss when both FIM ECN-capable and non-ECN-capable connections are employed. We propose a new packet-dropping scheme called Fair In-time Dropping (FID) which drops packets from a connection upon detecting an incipient indication of congestion depending on its share of gateway or router buffer occupancy. We also show that a combination of FIM and FID offers the best fairness compared with a combination of FIM along with other dropping schemes.,Copyright © 2003 John Wiley & Sons, Ltd. [source] DRED: a random early detection algorithm for TCP/IP networksINTERNATIONAL JOURNAL OF COMMUNICATION SYSTEMS, Issue 4 2002James Aweya Abstract It is now widely accepted that a RED [2] controlled queue certainly performs better than a drop-tail queue. But an inherent weakness of RED is that its equilibrium queue length cannot be maintained at a preset value independent of the number of TCP active connections. In addition, RED's optimal parameter setting is largely correlated with the number of connections, the round-trip time, the buffer space, etc. In light of these observations, we propose DRED, a novel algorithm which uses the basic ideas of feedback control to randomly discard packets with a load-dependent probability when a buffer in a router gets congested. Over a wide range of load levels, DRED is able to stabilize a router queue occupancy at a level independent of the number of active TCP connections. The benefits of stabilized queues in a network are high resources utilization, predictable maximum delays, more certain buffer provisioning, and traffic-load-independent network performance in terms of traffic intensity and number of connections. Copyright © 2002 John Wiley & Sons, Ltd. [source] A simple mechanism for stabilizing network queues in TCP/IP networksINTERNATIONAL JOURNAL OF NETWORK MANAGEMENT, Issue 4 2007James Aweya In this paper we determine the stability bounds for the DRED active queue management (AQM) algorithm using a previously developed nonlinear dynamic model of TCP. We develop a second-order linear model with time delay by linearizing the nonlinear model. Using the Pade approximation of time-delayed system e,R0s, where R0 is the delay in the system, we then determine the range of stabilizing gains of DRED when controlling the second-order system with time delay R0. We also present examples showing the stability bounds of the DRED controller gain for networks with different parameters such as link capacity, load level, and round-trip time. In addition, we describe an efficient implementation of the DRED AQM algorithm. Copyright © 2006 John Wiley & Sons, Ltd. [source] | |||||||||||||