Improved Packet Loss Research Articles

Integrating Smart Computility for Subflow Orchestration in Remote Virtual Services.

The burgeoning domain of the metaverse has sparked significant interest from a diverse array of industries, including healthcare services. However, the metaverse and its associated applications present various challenges. This could strain the comprehensive capacity of existing networks. In this paper, we have investigated vital network demands of healthcare services within the metaverse. First, to meet the increasing demands of the metaverse, there is a need for enhanced bandwidth, reduced latency, and improved packet loss control. Furthermore, the transmission mechanism should exhibit flexibility to automatically adapt to the diverse hybrid needs of different healthcare services. Considering the aforementioned challenges, a transmission paradigm tailored for the metaverse-based healthcare services is developed. Multipath transmission has the potential to effectively enhance network performance in multiple aspects. Significantly, we devise an orchestration framework to reconcile edge-side subflow management with diverse healthcare applications. Using machine learning techniques, the framework can produce near-optimal subflow adjustment strategies for client nodes and miscellaneous services. Comprehensive experiments are performed on applications with diverse requirements to validate the adaptability of the framework to the application needs. The experimental results demonstrate that the proposed method enables the network to autonomously adapt to changing network conditions and service requirements. This includes applications' preferences for high throughput, low delay, and high stability. Moreover, the test results show that the proposed approach can notably decrease the occurrences of network quality falling below the minimum requirement. Given its adaptability and impact on network quality, this work paves the way for future metaverse-based healthcare services.

Transmission Timing Control among Both Aperiodic and Periodic Flows for Reliable Transfer by Restricted Packet Loss and within Permissible Delay in Wireless Sensor Networks

Nowadays, in various wireless sensor networks, both aperiodically generated packets like event detections and periodically generated ones for environmental, machinery, vital monitoring, etc. are transferred. Thus, packet loss caused by collision should be suppressed among aperiodic and periodic packets. In addition, some packets for wireless applications such as factory IoT must be transferred within permissible end-to-end delays, in addition to improving packet loss. In this paper, we propose transmission timing control of both aperiodic and periodic packets at an upper layer of medium access control (MAC). First, to suppress packet loss caused by collision, transmission timings of aperiodic and periodic packets are distributed on the time axis. Then, transmission timings of delay-bounded packets with permissible delays are assigned within the bounded periods so that transfer within their permissible delays is possible to maximally satisfy their permissible delays. Such control at an upper layer has advantages of no modification to the MAC layer standardized by IEEE 802.11, 802.15.4, etc. and low sensor node cost, whereas existing approaches at the MAC layer rely on MAC modifications and time synchronization among all sensor nodes. Performance evaluation verifies that the proposed transmission timing control improves packet loss rate regardless of the presence or absence of packet's periodicity and permissible delay, and restricts average transfer delay of delay-bounded packets within their permissible delays comparably to a greedy approach that transmits delay-bounded packets to the MAC layer immediately when they are generated at an upper layer.

Energy‐aware dynamic‐link load balancing method for a software‐defined network using a multi‐objective artificial bee colony algorithm and genetic operators

Information and communication technology (ICT) is one of the sectors that have the highest energy consumption worldwide. It implies that the use of energy in the ICT must be controlled. A software-defined network (SDN) is a new technology in computer networking. It separates the control and data planes to make networks more programmable and flexible. To obtain maximum scalability and robustness, load balancing is essential. The SDN controller has full knowledge of the network. It can perform load balancing efficiently. Link congestion causes some problems such as long transmission delay and increased queueing time. To overcome this obstacle, the link load balancing strategy is useful. The link load-balancing problem has the nature of NP-complete; therefore, it can be solved using a meta-heuristic approach. In this study, a novel energy-aware dynamic routing method is proposed to solve the link load-balancing problem while reducing power consumption using the multi-objective artificial bee colony algorithm and genetic operators. The simulation results have shown that the proposed scheme has improved packet loss rate, round trip time and jitter metrics compared with the basic ant colony, genetic-ant colony optimisation, and round-robin methods. Moreover, it has reduced energy consumption.

Analysis of accuracy of packet loss estimation algorithm based on passive measurement

Estimating the packet loss rate in the border network helps network managers better understand network performance and achieve effective estimation of network performance. Taking the packet loss rate as an entry point, the research work analyzes the feasibility of the existing packet loss estimation algorithm applied to the network boundary, and conducts an accuracy-oriented comparative analysis based on a standard data set. On this basis, the author locates the cause of the error in the existing packet loss estimation algorithm and proposes a targeted solution. The test results show that the accuracy of the improved packet loss estimation algorithm has been significantly improved. Finally, the improved packet loss estimation algorithm is applied to the four campus networks of the CERNET and CERNET2 Nanjing master nodes, and the IPv4 and IPv6 packet loss rates of these four campus networks are measured respectively. The results show the packet loss rate in the IPv6 networks is lower than the packet loss rate in IPv4 networks.

HybTGR: a hybrid routing protocol based on topological and geographical information in vehicular ad hoc networks

Due to rapid changes in topology and high mobility of vehicles, routing in vehicular ad hoc networks needs special mechanisms. Due to these features, traditional routing protocols that are mainly developed for mobile ad hoc networks do not function well in these type of networks. To improve the efficiency of data routing, geographical routing method has been proposed by the researchers that use geographical information of nodes in the routing process. But this method relies only on the geographical information of network nodes and usually uses a greedy idea that may not perform well in some network situations. The main objective of this paper is to propose a hybrid routing protocol based on both geographical and topological information in Vehicular ad hoc networks and selecting the optimal route for data transmission. In the proposed protocol, based on the parameters like moving speed, link lifetime, the number of vehicles present in the vicinity of the node and distance to the destination node, a weight value is assigned to each network node. In this method, if the weight of a neighbor node is higher than that of the source node, geographical routing is used and otherwise, topology-based routing method will be used. The simulation results show that the proposed protocol due to the use of geographical and topological information, has improved packet loss rate, throughput, end-to-end delay and packet delivery ratio compared to LSGO, GpsrJ+ and GyTAR routing protocols.

Urgency-Based Fair Scheduling for LTE to Improve Packet Loss and Fairness: Design and Evaluation

Classic scheduling algorithms for the LTE uplink in the literature may cause severe packet loss, poor fairness, and inefficient resource allocation. To alleviate packet loss, delay budget rather than channel condition plays a key factor in our proposed algorithm that allows the user equipment (UE) with the least left delay budget to be scheduled first. Besides, the UEs with fewer allocated resources can be served earlier to improve fairness. For reaching efficient resource utilization, our proposed algorithm further poses a constraint in sequential resource allocation. Via extensive simulations, we successfully demonstrate that our proposed algorithm significantly outperforms the closely related algorithms in terms of packet loss ratio and fairness index at the cost of raised end-to-end delay but still within the delay bound. For our proposed algorithm, it also gains improvement in throughput due to its reduction in the packet loss ratio and bounded relaxation in the end-to-end delay.

An Improved Packet Loss Recovery of Audio Signals Based on Frequency Tracking

An Improved Packet Loss Recovery of Audio Signals Based on Frequency Tracking

A reinforcement learning-based link quality estimation strategy for RPL and its impact on topology management

Over the last few years, standardisation efforts are consolidating the role of the Routing Protocol for Low-Power and Lossy Networks (RPL) as the standard routing protocol for IPv6-based Wireless Sensor Networks (WSNs). Although many core functionalities are well defined, others are left implementation dependent. Among them, the definition of an efficient link-quality estimation (LQE) strategy is of paramount importance, as it influences significantly both the quality of the selected network routes and nodes’ energy consumption. In this paper, we present RL-Probe, a novel strategy for link quality monitoring in RPL, which accurately measures link quality with minimal overhead and energy waste. To achieve this goal, RL-Probe leverages both synchronous and asynchronous monitoring schemes to maintain up-to-date information on link quality and to promptly react to sudden topology changes, e.g. due to mobility. Our solution relies on a reinforcement learning model to drive the monitoring procedures in order to minimise the overhead caused by active probing operations. The performance of the proposed solution is assessed by means of simulations and real experiments. Results demonstrated that RL-Probe helps in effectively improving packet loss rates, allowing nodes to promptly react to link quality variations as well as to link failures due to node mobility.

Improving Packet Loss Rate of EtherCAT Master Dependent on Hardware Performance

Improving Packet Loss Rate of EtherCAT Master Dependent on Hardware Performance

An Improved Packet Loss Concealment Method for Mobile Audio Coding

In this paper, we propose an improved packet loss concealment (PLC) algorithm for ACELP-based speech coders in order to improve the quality of decoded speech at the receiver. Conventional receiver-based PLC algorithms in the AMR-WB+ speech codec are usually based on speech correlations to reconstruct the decoded speech of the lost frames by using parameter information obtained from the previous or adjacent correctly received frames. However, this approach has difficulties in reconstructing accurate voice signals when consecutive packet loss and bursts of packet loss occur. Thus, we propose a new PLC approach based on data-driven principle to recover the Immittance Spectral Frequen- cy (ISF) coefficients of lost packet, referring to pitch period parameters of adjacent correctly received voice packets. The proposed PLC algorithm is designed for ACELP-based audio codec. We make the performance comparison between the data-driven method and the PLC algorithm currently employed in AMR-WB+ speech codec through the PESQ objective evaluation test and MOS subjective evaluation test. The results prove that our algorithm can achieve better speech recov- ery quality than original PLC algorithm employed in AMR-WB+ speech codec.

Scalable Speech Coding for IP Networks: Beyond iLBC

High quality speech at low bit rates makes code excited linear prediction (CELP) the dominant choice for a narrowband coding technique despite the susceptibility to packet loss. One of the few techniques which received attention after the introduction of CELP coding technique is the internet low bitrate codec (iLBC) because of inherent high robustness to packet loss. Addition of rate flexibility and scalability makes the iLBC an attractive choice for voice communication over IP networks. In this paper, performance improvement schemes of multi-rate iLBC and its scalable structure are proposed, and the proposed codec enhanced from the previous work is re-designed based on the subjective listening quality instead of the objective quality. In particular, perceptual weighting and the modified discrete cosine transform (MDCT) with short overlap in weighted signal domain are employed along with the improved packet loss concealment (PLC) algorithm. The subjective evaluation results show that the speech quality of the proposed codec is equivalent to that of state-of-the-art codec, G.718, under both a clean channel condition and lossy channel conditions. This result is significant considering that development of the proposed codec is still in early stage.

Analysis and optimization of a cross-layer adaptation mechanism for real-time applications in wireless networks

The theoretical analysis of a cross-layer mechanism for improving the quality of service of real-time applications in wireless networks is presented. The mechanism coordinates adaptations of the modulation order at the Physical layer and the media encoding mode at the Application layer, to improve packet loss rate, throughput and mean delay. With the use of Continuous Flow Modeling, the system is considered as a "fluid" queue with inflow and outflow rates representing its traffic generation and service rates, respectively. Each data source is modeled as a Markov chain, from the steady-state of which the optimal adaptation thresholds of the cross-layer mechanism are derived. Extensive performance evaluation results show that the optimized operation of the mechanism attains a significant performance improvement compared to both the sub-optimal case, and a legacy system, which adjusts the modulation order and encoding mode separately and independently of each other.

Concurrent multipath transmission combining forward error correction and path interleaving for video streaming

Video streaming is a popular application on next generation networks (NGNs). However, video streaming over NGNs has many challenges due to the high bit error rates of these networks. Forward error correction (FEC) is often applied to improve the quality of video streaming. However, continuous lost packets decrease the recovery performance of FEC protection in NGNs. To disperse continuous lost packets to different FEC blocks, we propose a concurrent multipath transmission that combines FEC with path interleaving. Our proposed control scheme adaptively adjusts the FEC block length and concurrently sends data interleaved over multiple paths. Experimental results with our approach show improved packet loss and signal to noise ratio performance.

Incorporating packet semantics in scheduling of real-time multimedia streaming

In this work, we develop a novel packet scheduling algorithm that properly incorporates the semantics of a packet. We find that improvement in overall packet loss does not necessarily coincide with improvement in user perceivable QoS. The objective of this work is to develop a packet scheduling mechanism which can improve the user perceivable QoS. We do not focus on improving packet loss, delay, or burstiness. We develop a metric called, "Packet Significance," that effectively quantifies the importance of a packet that properly incorporates the semantics of a packet from the perspective of compression. Packet significance elaborately incorporates inter-frame, intra-frame information dependency, and the transitive information dependency characteristics of modern compression schemes. We apply packet significance in scheduling the packet. In our context, packet scheduling consists of two technical ingredients: packet selection and interval selection. Under limited network bandwidth availability, it is desirable to transmit the subset of the packets rather than transmitting the entire set of packets. We use a greedy approach in selecting packets for transmission and use packet significance as the selection criteria. In determining the transmission interval of a packet, we incorporate the packet significance. Simulation based experiments with eight video clips were performed. We embed the decoding engine in our simulation software and examine the user perceivable QoS (PSNR). We compare the performance of the proposed algorithm with best effort scheduling scheme and one with simple QoS metric based scheduling scheme. Our Significance-Aware Scheduling scheme (SAPS) effectively incorporates the semantics of a packet and delivers best user perceivable QoS. SAPS can result in more packet loss or burstier traffic. Despite these limitations, SAPS successfully improves the overall user perceivable QoS.

HMIPv6에서 MN의 지연을 최소화하는 멀티미디어 컨텐츠 서비스 방법

모바일 단말기를 사용하여 모바일 웹 서비스를 제공하는 과정에서 고려해야 할 사항은 끊김없는 서비스와 품질 보장형 서비스 이다. HMIIPv6는 끊김으로 인해 발생되는 패킷 손실과 지연 현상을 개선하기 위하여 MAP을 둔다. 하지만 MN을 위하여 패킷을 수신하고 전달하기 때문에 부하가 집중된다. 이는 보다 빨리 처리해야 할 실시간 데이터를 빨리 처리하지 못하는 결과는 낳는다. 뿐만 아니라 품질 보장 서비스를 위해서는 적응적 모바일 서비스 방법이 필요하다. 하지만 이 방법은 다양한 단말기의 하드웨어적인 차이점으로 인한 컨텐츠의 서비스 응답시간비용이 필요하다. 따라서 본 논문에서는 끊김없는 서비스를 위하여 MAP에 큐를 두어 실시간 데이타 처리 성능을 향상시켰다. 또한 응답시간비용을 줄이기 위하여 컨텐츠의 구성요소를 이용하여 컨텐츠를 재사용할 수 있는 캐시를 갖는 모바일 웹서비스 방법을 제안한다. 수식 분석과 시뮬레이션 결과, 제안방법인MAP에 큐를 두는 방식과 모바일 노드의 이동성을 고려한 캐시 방법이 다른 다양한 시스템 조건에서 우수함을 알수 있었다. The issues that we should consider in the process of providing mobile web service using a mobile device are seamless service and QoS-guaranteed service. HMIPv6 has MAP because of improving packet loss and transmission delay due to disconnection. However, a load is concentrated on HMIPv6 because of receiving and delivering packet for MN. Owing to this, real time data fails to be processed quickly, and also adaptive mobile service is required for QoS guaranteed service. However, this method demands the response time cost of contents service owing to the hardware differences of various devices. Therefore, we improve the process performance of real time data by applying a queue in MAP for seamless service in this paper. For decreasing response time cost, we propose mobile web service method which has reusable cache of contents using the elements of contents. The result of a numerical formula and simulation shows that our proposed method is superior under various system conditions.

An Online Buffer Management Algorithm for QoS-Sensitive Multimedia Networks

In this letter, we propose a new online buffer management algorithm to simultaneously provide diverse multimedia traffic services and enhance network performance. Our online approach exhibits dynamic adaptability and responsiveness to the current traffic conditions in multimedia networks. This approach can provide high buffer utilization and thereby improve packet loss performance at the time of congestion.

Unequal forced-intra-refresh for robust video streaming

For video streaming over lossy channels, intra refresh can mitigate the error-propagation effect caused by packet losses. Besides some intra-mode macroblocks (MBs) generated by the “Lagrangian rate-distortion” or “Sum of absolute difference” mode decision, the encoder or transcoder possibly needs to increase some “forced” intra-mode MBs for robust video streaming. Based on the error-propagation analysis in a group of pictures (GOP), we propose an unequal Forced-Intra-Refresh (FIR) scheme to improve packet loss resilience of video streaming. According to a GOP-level error-propagation model, the proposed unequal FIR scheme can optimally increase the unequal number of forced intra-mode MBs for different frames in a GOP. Simulation results showed that the proposed scheme can effectively enhance the robustness of video streaming under different channel conditions, and achieve about 0.1∼0.9 dB gains over the average FIR scheme in H.264/AVC tools.

Speech Quality Measurement Methods with Applying PLC Algorithms on Real-time Transmission Control Scheme for VoIP Service

According to the discovery of available and suitable audiovisual services over IP, researches on the Quality of Service (QoS) for various multimedia application services are required. This paper presents evaluation of speech quality applying improved Packet Loss Concealment (PLC) algorithms with an end-to-end bidirectional transmission control scheme. We propose real-time transmission control scheme to perform better quality monitoring of VoIP service. In order to evaluate the newly proposed RTCP-XR scheme for VoIP services, we measure in real-time the quality degree of the received speech signal while applying our improved Packet Loss Concealment (PLC) algorithms in G.711 and G.729A codecs using an implemented Session Initiation Protocol (SIP)- based SoftPhone and a 1-port gateway systems. Overall, for the evaluation with scientific exactitude of speech quality applied our PLC algorithms we examine with made phone-calls through the heterogeneous network systems. The experimental results confirm the applied PLC algorithms for improvement of speech quality over IP in real environments with the QoSfactor transmission control mechanism.

Packet loss resilient MPEG-4 compliant video coding for the Internet

Targeting multimedia communications over the Internet, this paper describes a set of complementary techniques in the direction of both improved packet loss resiliency of video-compressed streams and e$cient usage of available network resources. Aiming rst at a best trade-o! between compression e$ciency and packet loss resiliency, a procedure for adapting the video coding modes to varying network characteristics is introduced. The coding mode selection is based on a rate-distortion procedure with global distortion metrics incorporating channel characteristics under the form of a two-state Markov model. This procedure has been incorporated in an MPEG-4 video encoder. It has been observed that, in error-free environments, the channel adaptive mode selection technique does not bring any penalty in terms of compression, with respect to the initial MPEG-4 encoder, while allowing a signicant gain with respect to simple conditional replenishment. On the other hand, under the same loss conditions, it is shown that this procedure signicantly improves the encoder’s performance with respect to the original MPEG-4 encoder, to approach the robustness of conditional replenishment mechanisms. This intrinsic robustication of the encoder allows to minimize the e!ects of packet losses on the visual quality of the received video; however, it does not avoid losses. A rate-based #ow control mechanism is then developed and introduced into the encoder, in order to match the bandwidth requirements of the source to the bandwidth available over the path of the connection, for both &social’ and &individual’ benets. The control mechanism developed combines an RTT-based control loop allowing early reaction to congestion and a TCPfriendly rate prediction model getting into play under lossy conditions. This hybrid control mechanism allows full rate control (even in loss-free conditions) and smooth rate variations together with high responsiveness. The introduction of the rate control in the MPEG-4 compliant encoder allows to maintain a stable PSNR and visual quality while decreasing signicantly the source throughput, hence reducing congestion and loss provoked by the same video source at a constant bit-rate. ( 1999 Elsevier Science B.V. All rights reserved.

Performance of the COD: A transparent optical TDM statistical packet multiplexor for scalable terabit networks

We have constructed an optically transparent, memoryless 2:1 statistical TDM packet multiplexor for constant packet-length input. This multiplexor is the fundamental building block in a class of Cascaded Optical Delay (COD) multiplexors intended for wide-area terabit networks. The multiplexor does not require slot-aligned traffic. This multiplexor can be cascaded to build multiplexors with improved packet loss rate. In this article we present the experimental packet loss rate of the memoryless multiplexor as well as the calculated packet loss rate of larger, buffered COD multiplexors. Furthermore, we outline an optical loss compensation scheme for buffered COD multiplexors.