Packet Loss

TCP is the most popular and widely used network transmission protocol. All most 90% of the Internet connections make use of TCP for communication. TCP is reliable for wired networks and it considers all packet timeouts in wired networks as due to network congestion and not because of bit errors. However, TCP suffers from performance degradation over error-prone wireless links, as it has no technique to distinguish error deficits from congestion deficits, with networking becoming more divergent, with wired and wireless topologies. It considers all packet deficits are due to congestion and subsequently reduces the packet burst transmission, at the same time decreasing the network throughput. In this paper a new TCP congestion control mechanism is proposed that is suitable and applicable for wireless and also for wired networks and is capable of distinguishing congestion deficits from error deficits. The proposed technique uses the reserved field of the TCP header to indicate whether t...

Résumé: Many networked applications use end-to-end congestion control protocols to avoid the occurrence of congestion collapse phenomena. However, such transport protocols may suffer from link underutilization when there are wireless links along the path: the inability to distinguish wireless losses from congestion losses often results in unnecessary throughput decreases. Since networks are becoming increasingly heterogeneous, consisting of a mix of wired and wireless links, it is important to have some means of correctly identifying the ...

Active Queue Management (AQM) is receiving wide attention as a promising technique to prevent and avoid congestion collapse in packet-switched networks. By providing advanced warning of incipient congestion, end nodes can respond to congestion before router buffer overflows and hence ensure improved performance. Random Early Discard (RED) is an IETF recommended active queue management scheme that is expected to provide several Internet performance advantages such as minimizing packet loss and router queuing delay, avoiding global synchronization of sources, guaranteeing high link utilization and fairness. It tends to drop packets from each connection in proportion to the transmission rate the flow has on the output link. It does not minimize the number of dropped packets as expected, but it manages to achieve improved performance when compared to the Tail Drop. In this paper, extensive experimental analysis has been carried out on RED using Network Simulator (NS-2) in relation to congestion control and decision has been settled where RED can perform better.

Internet is expanding rapidly over the entire globe, TCP/IP is inseparable part of Internet. The increased use of wireless links for providing connectivity in remote areas and facilitating mobility in Internet brought out some serious performance issues of TCP, which was designed for wired links. TCP Westwood (TCPW) is a sender-side modification of the TCP congestion control algorithm that improves upon the performance of TCP Reno in wired as well as wireless networks. It continuously measure the bandwidth used by the connection by monitoring the rate of returning ACKs. The estimate is then used to compute congestion window and slow start threshold after a loss episode. This is to avoid over-shrinking cwnd after loss. In this paper the basic algorithm for TCP Westwood and it's characteristics like fairness friendliness are discussed. Also how error rate, RTT, buffer size and other parameters cause the effect on performance measures like throughput, congestion window utilization and ssthresh. Review of modifications done in basic algorithm based on different parameters is included with scope of improvement.

Web is extending quickly over the whole globe, TCP/IP is as one piece of Internet. The expanded utilization of remote connections for giving availability in remote territories and encouraging portability in Internet drew out a few genuine execution issues of TCP, which was intended for wired links. Behaviours of different TCPs have been broke down over wired and wireless connections. How they initialize their congestion window during slow start and congestion avoidance phase and what is the effect in the performance in wired and wireless networks. Here the investigation of TCP rendition called TCP Westwood which alters the congestion window is incorporated. It controls the congestion window by utilizing end-to-end data transfer capacity estimation. TCP Westwood persistently evaluates the end to end accessible transmission capacity on the premise of ACK entry rate that is utilized to reset the window to coordinate the accessibility of data transfer capacity. The algorithm is called bandwidth estimation algorithm. This makes TCP Westwood very powerful to the sporadic losses as a result of remote channel issues. In this paper execution of TCP Westwood has been assessed in the vicinity of more widely used variants of TCP over wired systems utilizing a system test system apparatus, NS-2.35. It is seen from the simulations that TCP Westwood uses the accessible bandwidth in most proficient way.

Web is extending quickly over the whole globe, TCP/IP is as one piece of Internet. The expanded utilization of remote connections for giving availability in remote territories and encouraging portability in Internet drew out a few genuine execution issues of TCP, which was intended for wired links. Behaviours of different TCPs have been broke down over wired and wireless connections. How they initialize their congestion window during slow start and congestion avoidance phase and what is the effect in the performance in wired and wireless networks. Here the investigation of TCP rendition called TCP Westwood which alters the congestion window is incorporated. It controls the congestion window by utilizing end-to-end data transfer capacity estimation. TCP Westwood persistently evaluates the end to end accessible transmission capacity on the premise of ACK entry rate that is utilized to reset the window to coordinate the accessibility of data transfer capacity. The algorithm is called bandwidth estimation algorithm. This makes TCP Westwood very powerful to the sporadic losses as a result of remote channel issues. In this paper execution of TCP Westwood has been assessed in the vicinity of more widely used variants of TCP over wired systems utilizing a system test system apparatus, NS-2.35. It is seen from the simulations that TCP Westwood uses the accessible bandwidth in most proficient way.

This paper describes a joint approach for control and communication network design for the application of bilateral teleoperation system (BTS). By ensuring a QoS oriented network architecture, a better quality of control (QoC) can be guaranteed despite the presence of time delays and packet losses. In this work, a joint approach is presented for the co-design problem to observe the improvements in QoC by network QoS for the bilateral teleoperation application.

Transmission Control Protocol (TCP) is known to suffer from performance degradation in mobile wireless environments, as such environments are prone to packet losses due to high bit error rates and mobility induced disconnections, because TCP is well reliable protocol for wired networks. In wired network packet loss due to congestion. Proposed scheme is a true end-to-end approach, based on the idea of exclusive handover message and is used for alleviating the degrading effect of host mobility on TCP performance. Experiments are performed using the Network Simulator (NS-2). The simulator has been extended to incorporate wireless link characteristics.

New deployment scenarios tend to consider the requirement of session continuity for service provisioning, especially multimedia streaming, to limited heterogeneous portable devices roaming among wireless localities. In particular, multimedia streaming should not experience any interruption while clients roam in wired-wireless integrated networks based on the standard besteffort Internet. The paper proposes an application-level middleware approach to proactively overcome Wi-Fi handoff and maintain multimedia session continuity in the wireless Internet by exploiting mobile proxies running on the wired network. Mobile middleware proxies locally support resource-limited clients, avoid packet losses during handoffs, pre-fetch local buffers with multimedia contents before handoff occurrence, and possibly reconfigure/renegotiate ongoing sessions after handoffs. Experimental results show that, notwithstanding the application-level implementation, mobile proxies can avoid streaming discontinuities with good efficiency in wireless-wired integrated networks even if their pro-activity is based on simple and lightweight handoff prediction techniques.

Segment Routing over IPv6 (SRv6 in short) is a networking solution for IP backbones and datacenters. The SRv6 standardization, implementation and research are actively progressing and SRv6 has already been adopted in a number of large scale network deployments. Effective Performance Monitoring (PM) solutions for SRv6 networks are strongly needed. The design, implementation and deployment of such PM solutions span the different planes of a networking architecture: Performance Measurements data (packet loss and delay) needs to be measured (in the Data Plane), the monitored nodes needs to be controlled (in the Control Plane), the measured data needs to be collected (in the Control/Management Planes), then the Data must be processed and stored, using Big-Data processing solutions. We focus on Loss Monitoring, by considering a solution capable of tracking single packet loss events in near-real time (e.g. with a delay in the order of 20 seconds). We describe SRv6-PM, a solution for Perfor...

Inter-vehicle communication is regarded as one of the major applications of mobile ad hoc networks (MANETs). In these so called vehicular ad hoc networks (VANETs) security and privacy are crucial factors for successful deployment. In a scenario, where each vehicle would have a unique identifier, eavesdroppers could easily accumulate location profiles. As a solution approach, several authors suggest using changeable pseudonyms as temporary vehicle identifiers. If a vehicle changes its pseudonym from time to time, long-term tracking can be avoided. However, as we show in this paper, changing identifiers has detrimental effects on routing efficiency and increases packet loss. So, designers of VANET systems need to aim for a balance between privacy protection on the one and performance on the other hand. The results of this paper provide advise on how to achieve this balance.

Video content providers put stringent requirements on the quality assessment methods realized on their services. They need to be accurate, real-time, adaptable to new content, and scalable as the video set grows. In this letter, we introduce a novel automated and computationally efficient video assessment method. It enables accurate real-time (online) analysis of delivered quality in an adaptable and scalable manner. Offline deep unsupervised learning processes are employed at the server side and inexpensive no-reference measurements at the client side. This provides both real-time assessment, as well as performance comparable to the full reference counter-part, while maintaining its no-reference characteristics. We tested our approach on the LIMP Video Quality Database (an extensive packet loss impaired video-set) obtaining a correlation between 78% and 91% to the FR benchmark (the Video Quality Metric, VQM). Due to its unsupervised learning essence, our method is flexible, dynamically adaptable to new content and scalable with the number of videos.

Peer-to-peer systems have considerably evolved since their original conception, in the 90's. The idea of distributing files using the user's terminal as a relay has now been widely extended to embrace virtually any form of resource (e.g., computational and storage resources), data (e.g. files and real-time streams) and service (e.g., IP telephony, IP TV, collaboration). Robustness, resilience and autonomic management are the key evolutionary step which makes them best fitted in dynamic, large-scale, decentralized environments. Peer-to-Peer systems provide network applications the opportunity to leap over the boundaries of the standardized and best effort protocols and the constraints of administrative domains. We take here the opportunity to warmly thank all the members of the AP2PS 2011 Technical Program Committee, as well as the numerous reviewers. The creation of such a broad and high quality conference program would not have been possible without their involvement. We al...

Abstract: Peer-to-peer (P2P) is a promising technology for video streaming, and offers advantages in terms of re-configurability and scalability. It gains advantage from and share the resources owned by the end-users who are distributed around the Internet. P2P has shown an alternative solution for the traditional Client-Server approach limitations. However, due to the churn of peers, issues of video quality arise such as packet loss. This in turn degrades the QoS, then the QoE. Moreover, in current networking conditions, congestions ...

Pada dasarnya antivirus digunakan untuk membersihkan dan melindungi sistem komputer dari infeksi virus komputer. Namun saat ini beberapa jenis virus telah berevolusi sehingga dapat lolos dari pendeteksian antivirus, oleh karena antivirus yang digunakan belum bisa menyimpan signature dari virus tersebut. Pada penelitian ini dikembangkan sebuah aplikasi antivirus yang diberi nama Vici Antivirus yang dapat mendeteksi virus menggunakan signature dan dilengkapi dengan metode pendeteksian heuristic ganda dan sistem realtime protector. Hal yang perlu diperhatikan dalam penelitian ini untuk mengetahui bagaimana kinerja Sistem Realtime protector dalam mengoptimalkan proses scanning virus  yaitu, menjalankan software antivirus guna memindai file program yang akan dijalankan dan akan menganalisis tingkat bahaya-nya. Jika terdeteksi sebagai sesuatu yang bisa membahayakan sistem, maka program antivirus akan membatalkan eksekusi dan menginformasikan adanya bahaya ke pengguna.

I. INTRODUCTION Signature-based intrusion detection systems, such as Snort [1], are useful to detect events without consideration of the context in which they occur. The need to keep track of the context in intrusion detection systems has already been cited [2], [3], [4], [5]. Even in ...

In the recent years, WiFi standard has been used to develop different kind of wireless networks due to its flexibility and the availability of cheap off the shelf hardware. Even if the standard itself lacks mobility support, it has been used in networks with mobile nodes. When mobility is involved, a fast handoff is of paramount importance, especially with multimedia applications. The current IEEE 802.11 standard does not provide any specification about how the handoff should take place. In this paper we propose a handoff scheme optimized for networks providing wireless connection to mass transit vehicles. The structure of the procedure is redesigned to minimize delay as well as assure reliability. A reliable handoff triggering mechanism is also designed exploiting the results obtained from a set of preliminary outdoor experiments. The proposed scheme was implemented and deployed in an experimental testbed, and several indoor tests were run in order to demonstrate its reliability and efficiency.

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Due to dense deployment and innumerable amount of traffic flow in wireless sensor networks (WSNs), congestion becomes more common phenomenon from simple periodic traffic to unpredictable bursts of messages triggered by external events. Even for simple network topology and periodic traffic, congestion is a likely event due to time varying wireless channel condition and contention caused due to interference by concurrent transmissions. Congestion causes huge packet loss and thus hinders reliable event perception. In this paper, we present a congestion avoidance protocol that includes source count based hierarchical medium access control (HMAC) and weighted round robin forwarding (WRRF). Simulation results show that our proposed schemes avoid packet drop due to buffer overflow and achieves more than 90% delivery ratio even under bursty traffic condition, which is good enough for reliable event detection.

Emerging practical schemes indicate that algebraic mixing of different packets by means of random linear network coding can increase the throughput and robustness of streaming services over wireless networks. However, concerns with the security of wireless video, in particular when only some of the users are entitled to the highest quality, have uncovered the need for a network coding scheme capable of ensuring different levels of confidentiality under stringent complexity requirements. We show that the triple goal of hierarchical fidelity levels, robustness against wireless packet loss and efficient security can be achieved by exploiting the algebraic structure of network coding. The key idea is to limit the encryption operations to a critical set of network coding coefficients in combination with multi-resolution video coding. Our contributions include an information-theoretic security analysis of the proposed scheme, a basic system architecture for hierarchical wireless video with network coding and simulation results.

∗Dept. of Electrical Engineering, Princeton University, Princeton, NJ 08544. yunnanwu,kung@princeton.edu †Microsoft Research, One Microsoft Way, Redmond, WA 98052-6399. pachou@microsoft.com ... Microsoft Research Microsoft Corporation One ...

A mechanism to discover the geographical positions of mobile nodes in ad hoc wireless networks through the implementation of a mobile-multi-agent based framework is investigated in this work. In other words, we have developed an agent based information exchange and navigation protocol in order to make each node in the network aware of the positions of other nodes. Our primary aim is to collect position-related information from each node in the network and distribute them periodically (as updates) to other nodes through mobile agents. The notion of stigmergic communication has been used through the implementation of a shared information cache in each node. We have used the GPS (Global Positioning System) support at each node for the extraction of geographical co-ordinates, velocity and direction of movement of each node. We have also incorporated a predictive mechanism which helps each node to predict the instantaneous positions of other nodes, based on the information that agents ha...

In this paper, we present a model for TCP/IP congestion control mechanism. The rate at which data is transmitted increases linearly in time until a packet loss is detected. At this point, the transmission rate is divided by a constant factor. Losses are generated by some exogenous random process which is assumed to be stationary ergodic. This allows us to account for any correlation and any distribution of inter-loss times. We obtain an explicit expression for the throughput of a TCP connection and bounds on the throughput when there is a limit on the window size. In addition, we study the effect of the Timeout mechanism on the throughput. A set of experiments is conducted over the real Internet and a comparison is provided with other models that make simple assumptions on the inter-loss time process. The comparison shows that our model approximates well the throughput of TCP for many distributions of inter-loss times.

Loss Differentiation Algorithms (LDA) are currently used to determine the cause of packet losses with an aim of improving TCP performance over wireless networks. In this work, we propose a cross-layer solution based on two LDA in order to classify the loss origin on an 802.11 link and then to react consequently. The first LDA scheme, acting at the MAC layer, allows differentiating losses due to signal failure caused by displacement or by noise from other loss types. Moreover, in case of signal failure, it adapts the behavior of the MAC layer to avoid a costly end-to-end TCP resolution. The objective of the second LDA scheme, which acts at the TCP layer, is to distinguish a loss due to interferences from those due to congestions and to adapt consequently the TCP behavior. The efficiency of each LDA scheme and of the whole cross-layer solution are then demonstrated through simulations.

The bandwidth-intensive network applications with aggressive quality of service (QoS) requirements requires fast and efficient networks. The wireless network performance is impacted due to multitude of data transport at uneven transmission rates on various channels and line losses leading to congestion. It is a big challenge to achieve the required QoS by managing delay, jitter, bandwidth and packet loss parame-ters on a network. This paper highlights the major causes affecting QoS and proposes an optimization technique which allocates the channel dynamically by integrating all the parameters affecting QoS across network layer, medium access control (MAC) layer and physical layer. The proposed algorithm utilizes the feedback parameters namely queueing delay, packet priority and timeout, MAC layer contention delay and packet loss ratio as inputs and a closed loop processing control for the scheduler based on fuzzy logic control (FLC). Hence, the algo-rithm is more realistic and consi...

This technical report describes the process of implementing a testbed, which was used in [1] to experimentally evaluate the performance of FAST under typical 'edge of network' scenarios involving DOCSIS (Data Over Cable Service Interface Specification) cable system.

The Aruba CX 10000 Series Switch represents a cutting-edge solution for data center networking, offering advanced performance capabilities combined with built-in security and services, powered by AMD Pensando technology. The architecture of the Aruba CX 10000 Series Switch is designed with embedded Pensando Distributed Services Modules (DSMs), featuring two DSMs per switch to create a Distributed Services Switch (DSS). The Aruba CX 10000 Series Switch is perfectly suited for data centers and edge deployments requiring high-performance switching, built-in firewall and telemetry capabilities, and a streamlined approach to reduce appliance sprawl by integrating services directly into the switch.
One limitation of the Aruba 10000 Series switch is that flow synchronization is managed using VSX, a proprietary solution. This synchronization takes place over the ISL (Inter-Switch Link), which can pose challenges in non-VSX environments, especially EVPN standard-based Multihoming. This paper addresses this issue by presenting an alternative method for flow synchronization that operates independently of VSX, enabling wider applicability in modern data center architectures.

3GPP's Long Term Evolution is defined by the standardization body's Release 8 and 9, and provides more than a substrate for 3GPP's IMT-Advanced Candidate, namely LTE-Advanced, which is due to be defined in Release 10. Both LTE and LTE-Advanced have SC-FDMA in their uplink, a multi-carrier access technique requiring contiguous subcarrier allocations for each UE. No scheduling algorithm, however, is dictated by the standard and several proposals have hence been presented to be implemented by vendors. A definite scheduling requirement is the support of QoS attributes of different types of uplink traffic. Our intent in this study is to evaluate the connectionlevel performance of representative scheduling proposals, with focus on QoS aspects. Specifically, we utilize a mixed type of traffic flows and evaluate the schedulers in terms of per-user throughput, packet loss and fairness.

The primary goals of Quality of Service (QoS) are bandwidth management, controlled jitter, latency and improved packet loss characteristics to provide satisfactory services for users. Shaping network optimization is crucial for the service provider. To implement the network QoS, optimizing the current network physical and logical architectures is among the best practice. In this work, an attempt has been made to investigate the end-to-end QoS parameters of EthioTelecom service level agreement (SLA) customers network by using differentiated service (DiffServ) model, to manage end-to-end traffic delay, jitter, and packet loss. The traffics are classified and marked depending on their priorities. The proposed network architecture has used weighted fair queueing for congestion management and weighted random early detection for congestion avoidance method. The eNSP and Wireshark have used been to design, demonstrate and evaluate the network architectures. When the results of the existing network are compared with the proposed network architecture that is designed using the DiffServ model; delay, jitter, and packet loss have decreased whereas the traffic utilization increased.

The intent of H.264 (MPEG-4 Part 10) was to achieve equivalent quality to previous standards (e.g., MPEG-2) at no more than half the bit-rate. H.264 is commonly felt to have achieved this objective. This document presents results of an HDTV subjective experiment that compared the perceptual quality of H.264 to MPEG-2. The study included both the coding-only impairment case and a coding plus packet loss case, where the packet loss was representative of a well managed network (0.02% random packet loss rate). Subjective testing results partially uphold the commonly held claim that H.264 provides quality similar to MPEG-2 at no more than half the bit rate for the coding-only case. However, the advantage of H.264 diminishes with increasing bit rate and all but disappears when one reaches about 18 Mbps. For the packet loss case, results from the study indicate that H.264 suffers a large decrease in quality whereas MPEG-2 undergoes a much smaller decrease.

This paper focuses on a comparative statistical analysis of the performance of VoIP calls under various situations, namely, during a handover and under different background traffic conditions at a wireless access point (AP). Using empiricalbased measurements, it demonstrates that these network conditions exhibit distinct statistical behaviour, in terms of SNR, packet losses and end-to-end delays, and thus, impact the VoIP user quality in a different manner. The analysis shows that both network conditions and codec type, as well as their interaction, have a significant effect on the PESQ MOS values. Moreover, it indicates statistically highly significant differences between the estimations of the PESQ and E-model. Finally, it highlights the benefits of the packet loss concealment of the AMR 12.2kb/s under these network conditions.

Wireless sensor networks are often composed of large number of wireless sensor nodes that are mostly using battery supplies. Nodes are small and equipped with sensors that cooperatively monitor physical world. Communication, sensing and computing have the most influence on performance and power consumption of wireless sensor networks. Various optimization methods are used to decrease power consumption and improve performance of wireless sensor networks. One of the novel methods is cross-layer approach. In this paper, first we present the advantages of cross-layer approach. The articles that are using a cross-layer approach for improving performance and energy preservation in wireless sensor networks are reviewed and state of the art is presented. In their work, scientists propose various optimization methods that are improving performance and decreasing power consumption. One of the fundamental problems is how to systematically make the performance evaluation and get the real contribution of these proposed optimization methods. In this paper, we develop benchmark methodology and performance evaluation model of various optimization methods for wireless sensor networks. Furthermore, we present the benchmark application for collecting and comparing performance measurement results of various optimization methods.

Wireless Sensor Networks consisting of nodes with limited power are deployed to gather useful information from the field. In WSNs it is critical to collect the information in an energy efficient manner. Ant Colony Optimization, a swarm intelligence based optimization technique, is widely used in network routing. A novel routing approach using an Ant Colony Optimization algorithm is proposed for Wireless Sensor Networks consisting of stable nodes. Illustrative examples, detailed descriptions and comparative performance test results of the proposed approach are included. The approach is also implemented to a small sized hardware component as a router chip. Simulation results show that proposed algorithm provides promising solutions allowing node designers to efficiently operate routing tasks.

First-in-First-out (FIFO) is the most widely used scheduling protocol in packet switching network. In fact, it is one of the simplest queuing policies used to provide best effort services in packet-switched network. However, the performance of FIFO is really crucial when it related to stability i.e. question of whether there is a bound on the total size of packets in the network at all times. In this study, our primary objective is to find the optimum packet size of voice packet when using FIFO adversarial network and FIFO M/M/1 network. Our new approach is based on adversarial generation of packets so that positive results are more robust in that they do not depend on particular probabilistic assumptions about the input sequences. In this paper, we proposed the FIFO scheduling technique that uses adversarial queuing model to find the best packet size of voice packet in FIFO network. Although the simulation results show that the average packet loss is increase when the arrival packet is increased, the average packet delay is improved as compared to FIFO M/M/1 technique, studied by . The proposed algorithm can be applied in packet-switched network, with adaptive routing, in order to transfer voice packet over the Internet.

Clustering is an important research topic in wireless networks, because cluster structures can facilitate resource reuse and increase system capacity. Ad hoc networks consist of wireless hosts that communicate with each other in the absence of a fixed infrastructure. Such networks cannot rely on centralized and organized network management. A slight change in the speed of cluster head nodes has a great influence on the cluster members and even causes the cluster head to switch frequently. we present a robust self-stabilizing weight-based clustering algorithm for ad hoc networks. The robustness property guarantees that, starting from an arbitrary configuration, after one asynchronous round, the network is partitioned into clusters. The communication in VANET is at greater risk because the messages are broadcasted and a wireless medium is used for data propagation. For secure communication among these vehicles, clustering algorithms are preferred. The proposed algorithm intends to create stable clusters by reducing reclustering overhead prolonging cluster lifetime, and shortening the average distance between cluster heads and their cluster members. Many clustering schemes have been proposed by the researchers which can be classified into either mobility based clustering or non-mobility based clustering algorithms. This technique takes the distance and velocity as a parameter to create relatively a stable cluster structure. We also developed a algorithm for super cluster-head selections. .

One of the main challenges in wireless cyber-physical systems is to reduce the load of the communication channel while preserving the control performance. In this way, communication resources are liberated for other applications sharing the channel bandwidth. The main contribution of this work is the design of a remote control solution based on an aperiodic and adaptive triggering mechanism considering the current network delay of multiple robotics units. Working with the actual network delay instead of the maximum one leads to abandoning this conservative assumption, since the triggering condition is fixed depending on the current state of the network. This way, the controller manages the usage of the wireless channel in order to reduce the channel delay and to improve the availability of the communication resources. The communication standard under study is the widespread IEEE 802.11g, whose channel delay is clearly uncertain. First, the adaptive self-triggered control is validated through the TrueTime simulation tool configured for the mentioned WiFi standard. Implementation results applying the aperiodic linear control laws on four P3-DX robots are also included. Both of them demonstrate the advantage of this solution in terms of network accessing and control performance with respect to periodic and non-adaptive self-triggered alternatives.

To date, most analysis of WLANs has been focused on their operation under saturation condition. This work is an attempt to understand the fundamental performance of WLANs under unsaturated condition. In particular, we are interested in the delay performance when collisions of packets are resolved by an exponential backoff mechanism. Using a multiple-vacation queueing model, we derive an explicit expression for packet delay distribution, from which necessary conditions for finite mean delay and delay jitter are established. It is found that under some circumstances, mean delay and delay jitter may approach infinity even when the traffic load is way below the saturation throughput. Saturation throughput is therefore not a sound measure of WLAN capacity when the underlying applications are delay sensitive. To bridge the gap, we define safe-bounded-mean-delay (SBMD) throughput and safe-bounded-delay-jitter (SBDJ) throughput that reflect the actual network capacity users can enjoy when they require bounded mean delay and delay jitter, respectively. The analytical model in this paper is general enough to cover both single-packet reception (SPR) and multipacket reception (MPR) WLANs, as well as carrier-sensing and non-carrier-sensing networks. We show that the SBMD and SBDJ throughputs scale super-linearly with the MPR capability of a network. Together with our earlier work that proves super-linear throughput scaling under saturation condition, our results here complete the demonstration of MPR as a powerful capacity-enhancement technique for both delay-sensitive and delay-tolerant applications.

The Hidden-node problem is a well-known phenomenon in IEEE 802.11 wireless networks. Most previous studies of the hiddennode problem were conducted by simulations, not experiments. In this paper, we set up real-network experiments to investigate the impact of hidden nodes in both infrastructure and multi-hop adhoc networks. Our experiments suggest transmission and carriersensing channel models different from those in the default settings of the NS2 simulator. We also study the RTS/CTS mode as a solution to the hidden-node problem. We show that using either 2Mbps or 11Mbps to transmit RTS/CTS not only can not solve the problem, but also further degrades the throughput by introducing additional overhead. In addition, our experiments confirm the existence of "re-routing instability" and "optimal offered load" in multi-hop ad-hoc networks which have only been previously "seen" in simulations.

Voice over IP (VoIP) has the potential to be integrated with other Internet applications to provide interactive multimedia communication services that are impossible (or at least very difficult) to deploy over the traditional circuit-switched wired and wireless networks. To fully exploit the benefit of service integration, it is necessary that VoIP services can be seamlessly provided with a good Quality-of-Service over several different network technologies. Due to their wide availability and significant user interest in mobile voice communications, particularly wireless network technologies will be crucial for the success of VoIP (and vice versa). However, the legacy Internet architecture is based on the "best effort" principle which does not guarantee a minimum amount of packet loss and a minimum delay of packet transmission required for voice communication. Considering a wireless access network to the Internet, QoS assurance is even more difficult. This is due to the inherently shared nature of the medium as well as its high medium error rate which add to the effect of congestion at routers known from wireline networks. In this paper we first analyze the concealment performance of the G.729 decoder as one prominent example of the CELP coder family which are typically employed for VoIP. Using this result, we then develop QoS support schemes which selectively mark packets to a higher (DiffServ) network priority at the sender dependent on the properties of the speech signal and the expected concealment performance. On the wireless access link the priorities are then mapped to a simple ARQ scheme. This layered approach is in full conformance with existing standards. Objective quality measures (ITU-T P.861A and EMBSD) show that almost the same speech quality as if all packets of the data stream would have been marked as eligible for retransmission can be achieved while significantly reducing the number of actual retransmissions.

Recently we have seen research efforts on how to protect a real-time speech signal when transmitting over an unreliable packet-switched network like the Internet by open-loop error control. Research has covered the type of Foward Error Correction (generic or voice-specific), the protocol support needed and adaptivity to the current network congestion state. However, the sender does not take into account that some segments of the signal are essential to the speech quality, while others can be extrapolated at the receiver from data received earlier in the event of a packet loss. This is especially true for modern frame-based codecs like the G.729 and G.723.1 which contain an internal loss concealment algorithm. Thus, the sender consumes additional bandwidth and aggravates the congestion in the Internet by sending unnecessary redundancy. In this paper we first analyze the concealment performance of the G.729 decoder. We find that the loss of unvoiced frames can be concealed well. Also, the loss of voiced frames is concealed well once the decoder has obtained sufficient information on them. However the decoder fails to conceal the loss of voiced frames at an unvoiced/voiced transition because it extrapolates internal state (filter coefficients and excitation) for an unvoiced sound. Moreover, once the encoder has failed to build the appropriate linear prediction synthesis filter, it takes a long time for the decoder to resynchronize with the encoder. Using this result, we then develop a new FEC scheme to support frame-based codecs, which adjusts the amount of added redundancy adaptively to the properties of the speech signal. Objective quality measures (ITU P.861A and EMBSD) show that our speech property-based FEC (SPB-FEC) scheme achieves almost the same speech quality as current FEC schemes while approximately halving the amount of necessary redundant data to adequately protect the voice flow.

Present IPTV system fails to establish QoS due to packet loss, server congestion and limited bandwidth. To overcome these drawbacks an implementation of IPTV is presented here using MPLS in 3GPP LTE release 8 networks. This implementation will provide multicast capability and FEC along with retransmission to minimize packet loss. Deploying IPTV in LTE makes LTE capable of transmitting voice, data and audio/video which will ensure convergence.

Mobile HMDs must sacrifice compute performance to achieve ergonomic and power requirements for extended use. Consequently, applications must either reduce rendering and simulation complexity -along with the richness of the experience -or offload complexity to a server. Within the context of edge-computing, a popular way to do this is through render streaming. Render streaming has been demonstrated for desktops and consoles. It has also been explored for HMDs. However, the latency requirements of head tracking make this application much more challenging. While mobile GPUs are not yet as capable as their desktop counterparts, we note that they are becoming more powerful and efficient. With the hard requirements of VR, it is worth continuing to investigate what schemes could optimally balance load, latency and quality. We propose an alternative we call edge-physics: streaming at the scene-graph level from a simulation running on edge-resources, analogous to cluster rendering. Scene streaming is not only straightforward, but compute and bandwidth efficient. The most demanding loops run locally. Jobs that hit the power-wall of mobile CPUs are off-loaded, while improving GPUs are leveraged, maximising compute utilisation. In this paper we create a prototypical implementation and evaluate its potential in terms of fidelity, bandwidth and performance. We show that an effective system which maintains high consistencies on typical edge-links can be easily built, but that some traditional concepts are not applicable, and a better understanding of the perception of motion is required to evaluate such a system comprehensively.

3GPP's Long Term Evolution is defined by the standardization body's Release 8 and 9, and provides more than a substrate for 3GPP's IMT-Advanced Candidate, namely LTE-Advanced, which is due to be defined in Release 10. Both LTE and LTE-Advanced have SC-FDMA in their uplink, a multi-carrier access technique requiring contiguous subcarrier allocations for each UE. No scheduling algorithm, however, is dictated by the standard and several proposals have hence been presented to be implemented by vendors. A definite scheduling requirement is the support of QoS attributes of different types of uplink traffic. Our intent in this study is to evaluate the connectionlevel performance of representative scheduling proposals, with focus on QoS aspects. Specifically, we utilize a mixed type of traffic flows and evaluate the schedulers in terms of per-user throughput, packet loss and fairness.

A suburban ad-hoc network (SAHN) interconnects a cooperative group at broadband speed using wireless links. The nodes are stationary. A SAHN involves low installation and service costs for SAHN specific traffic. Some current routing solutions for mobile ad-hoc networks may be used in the SAHN after certain optimizations. Efficiency can be improved by using smart directional antennas. Neighboring nodes falling outside the transmission region of a directional antenna are less vulnerable to co-channel interference. Additionally, increased transmission range can be achieved by reducing noise, interference and multi-path fading. In this paper, we emphasize the interference reduction capability of directional antennas and investigate the routing performances using three antenna schemes; multiple fixed directional, multiple omnidirectional and single omnidirectional antennas. We present an estimate of the achievable performance in a SAHN through extensive simulations. We also discuss the impact of using omnidirectional antennas on routing performance where different networks, using overlapping frequency channels, exist within each other's transmission ranges.

With the demand for multimedia and computational science applications, guaranteed performance communication services have become a necessary feature of future high-speed networks. These communications services should possess a high level of sophistication so that they can easily adapt the network to the wide variety of applications soon to be seen, thereby allowing the network to increase its availability and flexibility. Availability is the ability of the network to accommodate as many real-time clients as possible without violating any client's performance guarantees, while flexibility is the ability to adapt to changing network state and client demands in order to maintain the performance guarantees and quality of service promised to the client. Flexibility also refers to the ability of the network to easily increase the variety of real-time services that it offers. It is our contention that availability and flexibility can be enhanced in a network by providing the network with the ability to modify the performance parameters and/or the route of any guaranteed performance connection in the network without violating the previously made performance contracts. In this paper, we present a scheme for dynamically managing guaranteed performance service connections and experimental results to verify the correctness and usefulness of the scheme. The motivation for this scheme, Dynamic Connection Management (DCM), is discussed, and detailed descriptions of the DCM modification contracts and algorithms are provided. A survey of guaranteed performance services protocols, architectures, and routing algorithms are presented, together with their relevance to this work. A simulator has been built, and preliminary experiments and analyses were done on the scheme. The paper concludes with a summary and some topics for future work.