Packet Loss Recovery Research Articles

A New Regression Model Applied for Packet Loss Recovery Technique in Real-Time Communications

A New Regression Model Applied for Packet Loss Recovery Technique in Real-Time Communications

Towards Reliable Space-Ground Integrated Networks: From System-Level Design to Implementation

The integration of space and terrestrial networks results in a promising future network architecture, which exploits the high data rate and low latency of terrestrial networks as well as the wide-range coverage of satellite networks. However, spaceground integrated networks (SGINs) face some unprecedented challenges, including the rapidly fluctuating network topology, limited resources, intermittent connections, lossy links, asymmetric bandwidth allocation, and so on. The traditional endto- end (E2E) transport protocols designed for reliable terrestrial networks exhibit inherent limitations in the context of dynamic SGIN. A promising solution is to decouple the E2E perfect reception confirmation into hop-by-hop acknowledgments, which results in prompt packet loss recovery and high transmission resilience in the face of high-dynamic topologies. Hence, in this paper we propose a reliable cache-enabled transport system, which enhances the efficiency of reliable hop-by-hop transmission, and supports multi-orbit breakpoint transmission, while at the same time facilitates the reliability of intra-satellite communication with a novel transport protocol. In addition to unveiling a compelling system-level design, we demonstrate the benefits of the proposed reliable transport system (RTS) in a real SGIN prototype relying on satellites having onboard processing capability. Our experimental results validate the feasibility of the proposed RTS in the face of lossy links, intermittent connections and intra-satellite transmission failure. Moreover, one of the satellites has been launched in April 2021, while the other two will be launched to perform on-orbit test.

Packet Loss Recovery in Broadcast for Real-Time Applications in Dense Wireless Networks

Packet loss recovery in wireless broadcast is challenging, particularly for real-time applications which have strict and short delivery deadline. To recover the maximum number of lost packets within a short time, existing packet recovery solutions often rely on instantly decodable network coding (IDNC). Some of these solutions can recover nearly the maximum number of lost packets possible at the cost of collecting feedback from all (or a large percentage of) users. This is impractical in dense networks. In addition, their runtime grows with the number of users, which is not desirable due to the urgent delivery deadline of real-time applications. In this work, we introduce random instantly decodable network coding (RIDNC), a random encoding approach to IDNC. We propose RA ndom IDN C E ncoder (RACE), a fast RIDNC encoder that can recover nearly as many lost packets as the optimal RIDNC encoder. We compare RACE with the CrowdWiFi encoder, a high performing packet loss recovery solution used in CrowdWiFi, a commercial system for broadcasting live video in dense networks. We show that RACE is up to two orders of magnitude faster than the CrowdWiFi encoder, and recovers more lost packets in practice, where there is not enough time to collect feedback from many users.

Reliability enhancement and packet loss recovery of any steganographic method in voice over IP

VoIP data is transmitted through a transport protocol called user datagram protocol (UDP) which is intrinsically unreliable. The quality of the voice or multimedia trasmitted during a VoIP session is not much affected after a few packet loss. However, if a secret message is embedded inside VoIP packets using any steganographic method, the integrity of the secret message can be undermined due to the packets being lost during transmission. In this paper, we propose a scheme which is capable of enhancing the reliability of any VoIP steganographic method. We first distribute k message bits into k successive RTP packets. Then, parity bits are used for reconstruction of lost bits caused by packet loss. The implementation of our scheme on matrix embedding using binary Hamming codes steganography results in a reasonable reliability, a good speech quality and a very high steganographic bandwidth of 3050 bps.

Investigation of Internet of Things Handover Process for Information Centric Networking and Proxy Mobile Internet Protocol

IoT (Internet of Things) technically connects billions of entities to the Internet. The IoT is divided between the technology and the service itself. As a result, great efforts are needed to join data from many contexts and services. This reason has motivated proposals to develop solutions that can overcome existing issues of limitations for mobility, security, reliability and scalability of IoT. These billions of devices are interconnected to each other either using unicast, multicast or broadcast communications, and mixture of static and mobile communications. This paper aims to investigate the parameters of mobility performance in handover process for mobile multicast IoT environment. Investigation is done quantitatively by evaluating the parameters of handover process for IoT in two networking protocols that are possible to support acceptable mobility performance for IoT. The protocols are ICN (Information Centric Networking) and Proxy Mobile Internet Protocol. The evaluation parameters include packet loss and service recovery time. The metrics are extracted from the handover process flow for each network protocol topology. The service recovery time parameter is assumed as the time duration for each message to travel from sender to receiver, while packet loss parameter depends on the packet arrival rate and service recovery time. The results show that the ICN performs better than Proxy Mobile Internet Protocol.

Unequal loss protection scheme using a quality prediction model in a Wi-Fi broadcasting system

Wireless local area network–based broadcasting techniques are a type of mobile Internet Protocol television technology that simultaneously transmits multimedia content to local users. Contrary to the existing wireless local area network–based multimedia transmission systems, which transmit multimedia data to users using unicast packets, a wireless local area network–based broadcasting system is able to transmit multimedia data to many users in a single broadcast packet. Consequently, network resources do not increase with the increase in the number of users. However, IEEE 802.11 does not provide a packet loss recovery algorithm for broadcast packet loss, which is unavoidable. Therefore, the forward error correction technique is required to address the issue of broadcast packet loss. The broadcast packet loss rate of a wireless local area network–based broadcasting system that transmits compressed multimedia data is not proportional to the quality deterioration of the received video signals; therefore, it is difficult to predict the quality of the received video while also considering the effect of broadcast packet loss. In this scenario, allocating equal forward error correction packets to compressed frames is not an effective method for recovering broadcast packet loss. Thus, several studies on unequal loss protection have been conducted. This study proposes an effective, prediction-based unequal loss protection algorithm that can be applied to wireless local area network–based broadcasting systems. The proposed unequal loss protection algorithm adopts a novel approach by adding forward error correction packets to every transmission frame while considering frame loss. This algorithm was used as a new metric to predict video quality deterioration, and an unequal loss protection structure was designed, implemented, and verified. The effectiveness of the quality deterioration model and the validity of the unequal loss protection algorithm were demonstrated through experiments.

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

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

An SDN-Based Video Multicast Orchestration Scheme for 5G Ultra-Dense Networks

Predictions indicate that mobile video streams will dominate mobile traffic in the near future. Heterogeneous UDN is a promising technique for 5G mobile networks. This technique offers high access speeds for video watchers through the dense deployment of small cells. However, the backhaul of a UDN will become a bottleneck with the rapid increase in mobile video traffic. Additionally, the considerable video traffic generated from and to UDNs induces significant stress on the core network of a 5G network. Therefore, it is necessary to optimize the distribution of video content for 5G UDNs. Multicasting is considered to be a good communication method for distributing video content. This article examines a video multicast orchestration scheme based on SDN, named HCM, for 5G UDNs. To adapt well to a heterogeneous UDN environment in 5G networks and effectively save spectrum resources, HCM implements wireless multicast functionality based on a common unicast channel. HCM performs concurrent multicasting to clearly reduce the bandwidth consumption of a 5G network, particularly the backhaul, and further controls video traffic passed through the backhaul using a traffic-aware method. During the packet loss recovery of HCM, SDN-based elastic area multicast is used to more effectively retransmit recovered packets. The experimental results indicate that our proposed scheme can distribute video content with high effectiveness.

Software Architecture for Streaming and Packet Loss Recovery Schemes in Road Monitoring Surveillance Camera Control Systems

Background/Objectives: This paper is about software architecture for security system design using surveillance control system. Various challenges are faced in surveillance control systems using monitoring cameras like packet loose, latency, quality of Closed-Circuit Television (CCTV) data, inconsistency and conflict among multiple sources etc. In this paper an attempt is made to solve some of the problems which are faced by surveillance control systems like road monitoring systems. Methods/Statistical analysis: Experimentation setup is proposed for the given software architecture and algorithm is developed for system. The Digital Video (DV) streaming system is a client-server model for the real-time delivery of DV streams over Internet Protocol (IP) networks based on Real-time Transport Protocol (RTP) / Real-time Transport Control Protocol (RTCP) protocol. The system includes two PCs, one at the server side and one at the client side, connected together with a Fast Ethernet LAN. Server has got a fire wire interface for capturing the data. Capturing device is connected to a DV converter (Canopus ADVC 1000) through a composite conversion cable. Converted DV stream is given to the server through IEEE 1394 interface. In the road monitoring systems such architecture is very much needed to recover from packet losses and it will help in reducing road accidents. Findings: In this paper attempt is made to propose the best possible architecture for arranging surveillance camera for getting the best streaming. Packet loss recovery schemes are proposed in the camera control systems which are used for monitoring roads and it can minimize the death rate using these monitoring camera systems. Applications/Improvements: The proposed software architecture is used in packet loss recovery systems of surveillance systems and very useful application for security systems which are used for monitoring road and it can reduce deaths caused due to road accidents.

Packet loss recovery in audio multimedia streaming by using compressive sensing

The aim of this study is to introduce a new scheme, based on a compressive sampling technique, for the reconstruction of lost data in multimedia streaming. The audio streaming data are encapsulated in different packets, at the sender, by using an interleaving technique. The compressive sampling technique is used to recover audio information in case of lost packets, at the receiver. Experimental results are presented for speech and musical audio signals which illustrate the performances and the capabilities of the proposed methodology.

Compressive sampling and adaptive dictionary learning for the packet loss recovery in audio multimedia streaming

In this work, a scheme based on a compressive sampling technique and a fast dictionary learning approach for reconstructing audio content in multimedia streaming is introduced. Audio streaming data are encapsulated in different packets by means of an interleaving technique. The compressive sampling technique is used to reconstruct audio information in case of lost packets, with a sparsifying basis provided by a greedy adaptive dictionary learning algorithm. In order to assess the performance of the methodology, several experiments on speech and musical audio signals are presented.

A dynamic MBS zone framework for cost-effective inter-MBS zone handover in WiMAX networks

To synchronize multicast broadcast service (MBS) zone data transmissions, the WiMAX standard defines a coordination mechanism to coordinate data transmissions over the WiMAX network; however, the packet loss recovery procedures, which are parts of the coordination mechanism, enlarge packet transmission latency and packet buffer pool requirement. Therefore, we propose an in-frame control (IFC) scheme to decrease packet error rate and packet retransmission count, so as to reduce the packet transmission latency and packet buffer pool requirement. To support level-2 frame-offset coordination, we also propose a dynamic MBS zone (DMZ) framework that can provide data continuity between any two adjacent MBS zones. Based on the proposed DMZ framework, a seamless dynamic inter-MBS zone handover (called DMZ HO) scheme is proposed to resolve the data discontinuity (or packet loss) problem during inter-MBS zone HO. Compared to the WiMAX standard scheme, the proposed IFC scheme reduces packet error rate and packet retransmission count by 49.8 and 49.73 %, respectively. The proposed DMZ HO scheme also outperforms an overlapping zones (OLZ) scheme in terms of channel occupation time and channel bandwidth consumption (in terms of channel idle ratio). Therefore, by resolving data discontinuity, the proposed DMZ HO scheme is more cost-effective and more feasible in providing smooth multimedia content delivery services without disruption to multiple mobile stations (MSs) during HO, and thus will enhance user experience.

Scalable Video Transmission Over Wireless Networks Based on Loss Distribution and Layer Information

We propose an efficient scalable video transmission method using unequal forward error correction (FEC) scheme based on loss distribution and layer information. The actual packet loss rates over wireless networks are variable due to the clustered packet losses. First, we deduce the expected packet loss recovery rate from the estimation of the effective packet loss rate using loss distribution statistics. Second, layer-based distortion metric is presented including error propagation effects from hierarchical prediction structure in compressed scalable video packets. Third, the performance metric for FEC assignment is proposed from the layer-based distortion metric and the expected packet loss recovery rate. Finally, the FEC assignment algorithm is proposed for maximizing the performance metric. The proposed unequal FEC assignment method demonstrates robustness and significantly improved performance for scalable video transmission in various channel conditions compared to previous FEC assignment schemes.

Constant Weight Codes From Constant Dimension Codes and Their Applications

This paper presents a geometric approach to the construction of constant weight codes from constant dimension codes. Two classes of constant weight codes are constructed based on the incidence structures in projective and Euclidean geometries, respectively. It is shown that some known optimal codes can be re-obtained in this way. As an application, the constructed codes are used in store-and-forward networks for packet loss recovery. Their decoding is translated into that of the corresponding constant dimension codes. The decoding capability is beyond that of the bounded-distance decoding. Remarkably, for constant weight codes from Koetter-Kschischang codes, the minimum-distance decoding is achieved.

Enhanced packet loss recovery for real time PC-based GigE vision AVI systems

GigE vision could be a viable solution for seamless and cost effective integration of machine vision in manufacturing processes. Transmission delays, inherent to streaming and lost packets protocol implemented within the GigE, often jeopardise the implementation of networked PC-based machine vision. This is related to GigE vision open standard handling data integrity and error control through control and stream protocols (GVSP and GVCP) that implement data resend functions. In this work, we study the effect of data loss on the real time performance of distributed GigE vision AVI systems. We also introduce an optimised error recovery function based on definition of packets of interest (POIs) that allows for selective lost packet recovery based on their pertinence to subsequent machine vision processing. This scheme alleviates the traffic overhead by avoiding non-useful retransmissions. This mechanism allows reducing the latency and improving the predictability of such systems.

Analysis and Improvement of Send-and-Wait Automatic Repeat-reQuest Protocols for Wireless Sensor Networks

Due to the characteristics of wireless sensor networks (WSNs) and the unique communication pattern, it is crucial to disseminate messages in wireless sensor networks by a reliable and energy-efficient way. Although send and wait automatic repeat-request (SW-ARQ) has been the most widely used and well studied mechanism to ensure reliable data transmission in wired network, such as cable networks. Little attention has been paid to SW-ARQ in wireless sensor networks. Unlike wired network, sensor nodes in WSNs are power limited. And the nodal lifetime is the most crucial issue for WSN design. As a result, the ARQ-based packet loss recovery mechanisms proposed for wired network cannot be directly applied to WSNs. In this paper, we present an analysis strategy that characterizes the trade-off between energy consumption and source-to-sink statistical reliability (or simply reliability). Based on the insight gained about the nodal data load and energy consumption, we propose an analysis of strategy to meet the specific requirements of a sensing application in terms of lifetime and reliability in WSNs. More importantly, we propose a new SW-ARQ protocol named $$s$$send and $$w$$wait one $$d$$data $$m$$multi-$$A$$ACK (SW-ODMA) ARQ protocol to improve the energy efficiency and statistically reliability. In this protocol, multiple ACKs are returned after receiving one data packet, although the number of ACK packets sent by nodes is increased, the times of sending data packets that consumes large energy consumption decline. So the nodal energy consumption is reduced and thus the network lifetime is improved. Also, we give the applicability of SW-ODMA, as well as the algorithm that maximizes the lifetime under the premise of the network reliability requirement. Our simulation results are found to be consistent with our theoretical results, and demonstrated that the SW-ODMA protocol boosted the energy efficiency and reliability. Therefore, SW-ODMA protocol can be widely applied in WSNs with important significance.

Performance analysis of packet loss recovery policies in P2P video streaming

Packet loss recovery is an important part of P2P video streaming networks due to inevitable packet loss in today's internet and interdependency of data units in compressed video streams. In addition, the architecture of P2P streaming networks, in which the data delivered to the receivers through chain of peers, can intensify the impact of the internet packet loss on the quality of perceived video at the receivers. FEC and ARQ are the two most important techniques that can be used to overcome the side effect of the internet packet loss in P2P video streaming networks. Based on these two techniques, different packet loss recovery strategies can be applied in different overlay hops of a given P2P streaming network. In this paper, we have modelled and analysed the performance and efficiency of variant packet loss recovery policies can be used in P2P streaming networks. Our analysis show that a hybrid packet loss recovery scheme in which the most important data units are protected by packet level FEC and the remaining lost ones are recovered through retransmission i.e., ARQ is the most efficient strategy that improves the performance of P2P video streaming networks.

Packet Loss Recovery Scheme with Uniquely-Decodable Codes for Streaming Multimedia over P2P Networks

This paper presents an effective packet loss recovery scheme for Internet Protocol (IP) based peer-to-pear (P2P) networks that deliver multimedia streaming contents. The proposed scheme consists of two key components, a packet forward error correction (FEC) scheme and a uniquely decodable coding (UDC) scheme. The FEC scheme is devised to recover network packet loss over the transport layer to improve link transmission reliability at the user-ends of the P2P networks. The UDC scheme is employed to allow multiple sources to transmit data streams simultaneously in such a way that the combined data streams can be separated uniquely without ambiguity at the user-ends. This simultaneous multiple sources data transmission increases the throughput at the user-ends. Theoretical analysis and computer simulations show that the proposed scheme can achieve the commonly acceptable of 10^{-6} end-to-end packet loss rate for input packet loss rate of up to 10%with a reasonably small bandwidth expansion factor, while maintains a high throughput at the user-ends.

A Waveform Interpolation Based Packet Loss Recovery Technique

A Waveform Interpolation Based Packet Loss Recovery Technique

Coupling Loss and Delay Differentiation to Enhance TCP Performance within Wireless Multi-hop Ad-hoc Networks

Most existing TCP variants cannot distinguish between different packet loss causes within MANETs. TCP was, mainly, developed to deal with network congestion errors. While within MANETs, there are packet loss causes other than congestion. Studying the behaviour of TCP in front of such losses, we notice that TCP doesn’t have always the optimum behaviour as it reacts, in most cases, without considering the loss cause. This misbehaviour might cause network performance degradation and resources’ waste. To overcome this problem, many LDAs have been designed. However, these LDAs were optimized for data networks where wireless link is only the last hop, meaning that they might be inadequate for MANETs. Also, the proposed LDAs deal only with losses due to wireless channel and/or congestion-induced errors. We show, in this paper, the importance of dealing with a third loss cause that is common in MANETs, which is link failure. We propose a new TCP variant that is called TCP-WELCOME. TCP-WELCOME can: (i) identify the loss cause by coupling loss and delay information, and (ii) trigger the appropriate packet loss recovery according to the identified loss cause. The performance evaluation, through both simulations and experimental tests, shows that TCP-WELCOME optimizes both energy consumption and achievable throughput. TCP-WELCOME does not change the standard and can operate with existing TCP variants.