IP Packet Loss Research Articles

QUALITY OF EXPERIENCE FOR VOICE OVER INTERNET PROTOCOL (VoIP)

Today, the service of Voice over Internet Protocol or (VoIP) is one of the most used service around the globe in many fields, especially with video conferencing applications. Different applications and also many communications companies around the world use the VoIP service. Communication stakeholders aim to keep old customers happy and attract new customers. There was a need for some kind of service quality measures. Metrics are used to quantify the quality of transmission link are known Quality of Service (QoS). Whereas, user satisfaction level is identified by using Quality of Experience (QoE). This paper conducted a simulation model for VoIP service over heterogeneous, using OPNET modeler. This work aims to compare the user satisfaction level of the VoIP service by using different coding schemes in the application level of the User Equipment (UE). The measured QoS parameters over the heterogeneous transmission link were IP packet delay, IP Packet jitter, IP packet loss, MAC layer delay, PHY layer throughput in the network. Additionally, in this simulation model the use of E-Model to assess QoE level using Mean Opinion Score (MOS) metric in predicting the VoIP call quality. The results using the vocoder G.729A during the (30) sec of the voice call simulation time were as follows: Delay 0.57 mSec, Throughput of 290 Kbits/sec, Jitter of 0.02 uSec, MOS value of 3.76. The results using the vocoder G.711 during the same aforementioned time were as follows: Delay 0.25 mSec, Throughput of 70 Kbits/sec, Jitter of 0 Sec), MOS value of 3.08.

Transfer Error and Correction Approach in Mobile Network

Abstract With the development of information technology and social progress, human demand for information has become increasingly diverse, wherever and whenever people want to be able to easily, quickly and flexibly via voice, data, images and video and other means to communicate. Visual information to the people direct and vivid image, image / video transmission also been widespread attention. Although the third generation mobile communication systems and the emergence and rapid development of IP networks, making video communications is becoming the main business of the wireless communications, however, the actual wireless and IP channel will lead to error generation, such as: wireless channel multi- fading channels generated error and blocking IP packet loss and so on. Due to channel bandwidth limitations, the video communication compression coding of data is often beyond the data, and compress data after the error is very sensitive to error conditions caused a serious decline in image quality.

A study of TCP performance in wireless environment using fixed-point approximation

Loss of IP packets occurring as a result of imperfect local error correction is one of the major reasons for TCP performance degradation in wireless networks. These losses are misinterpreted by TCP senders as congestion indications and subsequently lead to the sharp decrease of the sending rate. In addition to wireless losses, packets can be lost as a result of buffer overflows at the IP layer. To study performance experienced by TCP sessions sharing a wireless channel, we adopt the fixed-point approximation. We distinguish between two modes of operation: (i) packet losses are mainly caused by imperfect error correction, (ii) the packet loss process is dominated by the buffer overflow. Applying different TCP models for these two regimes we approximate TCP throughput as a function of underlying layers’ parameters. Using this approach we study the effect of various protocol parameters on TCP throughput. We also investigate the effect of the queuing system involved in our analysis and demonstrate that usage of complex models with correlated arrival processes and generally distributed service time does not qualitatively affect the estimated TCP throughput. This observation allows to use simple queuing models having closed form solutions for performance metrics of interest, e.g. M/M/1/K or Geo/Geo/1/K. Finally, we address the question of choosing appropriate queuing model for quantitative TCP analysis.

Toward Lossless Video Transport

Temporary outages in IP packet delivery can occur during network reconvergence events, and conventional error-control methods that end points use can't effectively deal with such extended blocks of packet loss. Service providers can employ a number of network technologies to minimize the IP packet loss during such outages, ultimately resulting in lossless video transport.

Modeling TCP SACK performance over wireless channels with semi-reliable ARQ/FEC

Providing reliable data communications over wireless channels is a challenging task because time-varying wireless channel characteristics often lead to bit errors. These errors result in loss of IP packets and, consequently, TCP segments encapsulated into these packets. Since TCP cannot distinguish packet losses due to bit corruption from those due to network congestion, any packet loss caused by wireless channel impairments leads to unnecessary execution of the TCP congestion control algorithms and, hence, sub-optimal performance. Automatic Repeat reQuest (ARQ) and Forward Error Correction (FEC) try to improve communication reliability and reduce packet losses by detecting and recovering corrupted bits. Most analytical models that studied the effect of ARQ and FEC on TCP performance assumed that the ARQ scheme is perfectly persistent (i.e., completely reliable), thus a frame is always successfully transmitted irrespective of the number of transmission attempts it takes. In this paper, we develop an analytical cross-layer model for a TCP connection running over a wireless channel with a semi-reliable ARQ scheme, where the amount of transmission attempts is limited by some number. The model allows to evaluate the joint effect of stochastic properties of the wireless channel characteristics and various implementation-specific parameters on TCP performance, which makes it suitable for performance optimization studies. The input parameters include the bit error rate, the value of the normalized autocorrelation function of bit error observations at lag 1, the strength of the FEC code, the persistency of ARQ, the size of protocol data units at different layers, the raw data rate of the wireless channel, and the bottleneck link buffer size.

Demonstration of 10 Gbit Ethernet/Optical-Packet Converter for IP Over Optical Packet Switching Network

We developed novel network interfaces, for example 10 Gbit Ethernet to 80 Gbit/s optical-packet (10 GbitE-80 GbitOP) or 80 Gbit/s optical-packet to 10 Gbit Ethernet (80 GbitOP-10 GbitE) converters (collectively called as 10 GbitE/80 GbitOP converters), to connect optical packet switching (OPS) networks with IP technology-based networks. By using newly developed arrayed burst-mode optical packet transmitters/receivers together, the 10 GbitE-80 GbitOP converter at the ingress edge node of the OPS network encapsulates an IP packet into an 80(8lambda×10) Gbit/s dense wavelength division multiplexing (DWDM)-based optical packets and generates an optical label based on a lookup table and the destination addresses of the IP packet. The 80 GbitOP-10 GbitE converter at the egress edge node decapsulates the IP packet from the optical packet and generates a 10 GbitE frame accommodating the IP packet according to a lookup table. By using these network interface devices and OPS system based on multiple optical label processing, we achieved, for the first time, 74-km single-mode fiber transmission, switching, and buffering of 80(8lambda×10) Gbit/s DWDM-based optical packets encapsulating almost 10 Gbit/s IP packets with error-free operation (IP packet loss rate < 10 <sup xmlns:mml="http://www.w3.org/1998/Math/MathML" xmlns:xlink="http://www.w3.org/1999/xlink">-6</sup> ).

Not All Packets Are Equal, Part 2: The Impact of Network Packet Loss on Video Quality

For pt. 1 see ibid., vol. 13, no. 1, p.70-5 (2009). In this second part of a two-part article, the authors highlight the impact that different durations of IP packet loss have on the quality of experience for IP-based video streaming services. They describe the visual impairments that result from such packet losses and present the results of testing and analysis to compare impairments for different loss durations for both MPEG-2-encoded standard and high-definition services.

IP Packet Loss Prevention Scheme with Bicast and Forwarding for Handover in Mobile Communications

This paper proposes a scheme for hard handover (HO) between base stations (BSs) that combines bicast with forwarding; it realizes packet-lossless HO as well as low HO control delay. The proposed scheme observes the status of the current channel condition and initiates bicasting, the simultaneous transfer of IP packets from the access router to both the old BS and the new BS, when the probability of HO becomes high; this reduces the control delay imposed by hard HO. When HO becomes unavoidable, only those IP packets remaining in the old BS buffer that are not shared with the new BS are forwarded to the MS; this prevents the loss of IP packets. Computer simulations show that the proposed scheme reduces the HO control delay at 95% cumulative distribution function (CDF) by approximately 1700 (300) msec, 340 (320) msec, and 170 (330) msec compared to the conventional forwarding scheme (the conventional bicast scheme) when the number of users is 80 and the maximum Doppler frequency (fdmax) is 5.55Hz and data rate (D) on the wired propagation channel is 10, 50, and 100Mbps, respectively. The results confirm the superiority of the proposed scheme as an IP packet loss prevention scheme for hard HO.

Monitoring and Troubleshooting in Operational IP-TV System

When deploying large scale of QoS guaranteed IP-TV services over Triple-Play networks, there is a growing demand to access to the live network services, monitor in real time the broadcast chains, acquire the dynamic network parameters to ensure an uninterrupted, high quality service. Considering the difference of IP packet switching network and the de-facto transport stream (TS) standard of broadcast digital video network, this paper describes design consideration and implementation of a powerful embedded IP-TV monitor for real time monitoring the IP-TV QoS and using this equipment to construct a payload-aware monitoring network for on-line troubleshooting when the QoS deteriorate. The monitor should be designed to have ability to monitor and troubleshoot the blended combination of network status and digital TV quality in real time and cross layer way, meanwhile the procedures of monitoring does not interfere with the on-line services. This monitor can be deployed in the networks to measure the three ultimate metrics: IP cumulative jitter, packet loss and MPEG TS errors and provide helpful information about the network and service when service failed. Implementation and experimental results show promise of being practical in the real IP-TV trial networks.

A new technique for minimizing network loss from users’ perspective

Loss control is an essential issue for video transportation over IP networks. Buffer management is a traditional network technique to protecting video data against IP packet loss. In this paper, a new innovative use of packet scheduling to meet the loss constraints of video communication is proposed. A packet-scheduling scheme at a router that assures that intra and inter-stream loss requests for video transportation are met was developed and evaluated using real video data. The results demonstrate the advantages of the proposed scheme, in terms of loss performance and fairness.

Opinion Model Using Psychological Factors for Interactive Multimodal Services

We propose the concept of an opinion model for interactive multimodal services and apply it to an audiovisual communication service. First, psychological factors of an audiovisual communication service were extracted by using the semantic differential (SD) technique and factor analysis. Forty subjects participated in subjective tests and performed point-to-point conversational tasks on a PC-based video phone that exhibited various network qualities. The subjects assessed those qualities on the basis of 25 pairs of adjectives. Two psychological factors, i.e., an aesthetic feeling and a feeling of activity, were extracted from the results. Then, quality impairment factors affecting these two psychological factors were analyzed. We found that the aesthetic feeling was affected by IP packet loss and video coding bit rate, and the feeling of activity depended on delay time, video packet loss, video coding bit rate, and video frame rate. Using this result, we formulated an opinion model derived from the relationships among quality impairment factors, psychological factors, and overall quality. The validation test results indicated that the estimation error of our model was almost equivalent to the statistical reliability of the subjective score.

A novel framework for IP DiffServ over optical burst switching networks

This paper presents a novel framework for IP Differentiated Services (DiffServ) over optical burst switching (OBS), namely, DS-OBS. The network architecture, functional model of edge nodes and core nodes, the control packet format, a novel burst, assembly scheme at ingress nodes and scheduling algorithm of core nodes are presented. The basic idea is to apply DiffServ capable burst assembly at ingress nodes and perform different per hop behavior (PHB) electronic treatments for control packets of different QoS class services at core nodes. Simulation results show that the proposed schemes can provide the best differentiated service for expedited forwarding (EF), assured forwarding (AF) and best, effort (BE) services in terms of end-to-end delay, throughput and IP packet loss probability.