A QoE study of different stream and layout configurations in video conferencing under limited network conditions

One particular problem of QoE research in video conferencing is, that most research in the past concentrated on one-to-one video conferencing or simply video consumption. However, video conferencing with two people (one-to-one) and within a group (multi-party) is different. Particularly, limitations of one participant might have an effect on the QoE of the whole group. This possible effect however is not well studied. Therefore, this paper aims to better understand the impact of individual limitations towards the groups QoE. To do so, we show a study about different video stream configurations and layouts for multi-party conferencing in respect to individual network limitations. For this, we conduct a user study with 20 participants in 5 groups, in a semi-controlled setup. Such a setup, combines supervising participants locally while still using our software infrastructure deployed in the internet. Furthermore, we use an asymmetric experiment design, by putting every participant under a different condition, as this proposes a more realistic scenario. Within our study, we look at three different factors: layout, video quality and network limitations. To foster conversation between participants, the group engaged in a discussion about different survival questions. Our findings show that packet loss and the resulting distortions have a greater impact on the QoE as reducing the video quality by its resolution. Furthermore, our findings indicate that participants are more satisfied in a visually equal layout (showing participants in a similar size) and a more balanced stream configuration

Comparison and optimization of packet loss repair methods on VoIP perceived quality under bursty loss

Packet loss degrades the perceived quality of voice over IP (VoIP). In addition, packet loss in the Internet tends to come in bursts, which may further degrade audio quality. Using the Gilbert loss model, we infer that changing the packet interval affects loss burstiness, which in turn influences forward error correction (FEC) performance. Next, we perform subjective listening tests based on Mean Opinion Score (MOS) to evaluate the effect of bursty loss on VoIP perceived quality. Then, we compare the perceived quality achieved by two major loss repair methods: FEC and low bit-rate redundancy (LBR). Our MOS test results show that FEC is much preferred over LBR. In addition, our MOS results reveal that, under bursty loss, FEC quality is much better with a moderately large packet interval. Finally, because FEC introduces an extra delay proportional to the packet interval, we present a method of optimizing the packet interval to maximize FEC MOS by considering the delay impairment in ITU’s E-model standard

Evaluación del impacto de los mecanismos de control de error en la calidad de servicio de telefonía IP basado en Asterisk sobre una red inalámbrica de banda ancha en la provincia de Tayacaja - Huancavelica

En la presente tesis de maestría de telecomunicaciones titulada “Evaluación del Impacto de los Mecanismos de Control de Error en la Calidad de Servicio de Telefonía IP basado en Asterisk sobre una Red Inalámbrica de Banda Ancha en la Provincia de Tayacaja - Huancavelica” se describe el proceso y los elementos de la transmisión de VoIP en las redes LAN inalámbricas, se evalúa los diversos mecanismos de control de error existentes y particularmente se indica el más adecuado para la red LAN inalámbrica. Por otro lado, se analizan los parámetros de calidad de servicio y su relación en el servicio de VoIP, identificando los elementos que degradan la calidad de la voz que un usuario percibe en una comunicación como el retardo, la pérdida de paquetes, variación de retardo, latencia extremo a extremo, errores binarios en el canal de transmisión. En ese sentido, también se describe el proceso de medición de calidad de la VoIP basado en los estándares de la UIT. Se propone unas estrategias de mejora del sistema de comunicaciones para VoIP en entorno WLAN bajo la operación mayoritaria DCF restringiendo la operación del mecanismo de retransmisión WLAN al tiempo máximo permitido por la aplicación y la memoria de reproducción del receptor. Se investigan estos mecanismos de retransmisión básicos de WLAN y los mismos interactuando con la memoria de reproducción y la limitación temporal de la aplicación. Para ello, se realiza una definición de un modelo de simulación para encontrar los resultados; los cuales indican una mejora en el rendimiento ya que mantienen el nivel de calidad en entornos inalámbricos hostiles como los de una tasa de error peor de 10-4.Tesi

Medidas de qualidade de voz em redes IP /

Orientador: Eduardo Parente RibeiroCo-orientador: Marcus Vinicius LamarDissertação (mestrado) - Universidade Federal do Paraná, Setor de Tecnologia, Programa de Pós-Graduaçao em Engenharia Elétrica. Defesa: Curitiba, 2006Inclui bibliografia e anexo

On modeling and mitigating new breed of dos attacks

Denial of Service (DoS) attacks pose serious threats to the Internet, exerting in tremendous impact on our daily lives that are heavily dependent on the good health of the Internet. This dissertation aims to achieve two objectives:1) to model new possibilities of the low rate DoS attacks; 2) to develop effective mitigation mechanisms to counter the threat from low rate DoS attacks. A new stealthy DDoS attack model referred to as the quiet attack is proposed in this dissertation. The attack traffic consists of TCP traffic only. Widely used botnets in today\u27s various attacks and newly introduced network feedback control are integral part of the quiet attack model. The quiet attack shows that short-lived TCP flows used as attack flows can be intentionally misused. This dissertation proposes another attack model referred to as the perfect storm which uses a combination of UDP and TCP. Better CAPTCHAs are highlighted as current defense against botnets to mitigate the quiet attack and the perfect storm. A novel time domain technique is proposed that relies on the time difference between subsequent packets of each flow to detect periodicity of the low rate DoS attack flow. An attacker can easily use different IP address spoofing techniques or botnets to launch a low rate DoS attack and fool the detection system. To mitigate such a threat, this dissertation proposes a second detection algorithm that detects the sudden increase in the traffic load of all the expired flows within a short period. In a network rate DoS attacks, it is shown that the traffic load of all the expired flows is less than certain thresholds, which are derived from real Internet traffic analysis. A novel filtering scheme is proposed to drop the low rate DoS attack packets. The simulation results confirm attack mitigation by using proposed technique. Future research directions will be briefly discussed

ADAPTIVE SPEECH QUALITY IN VOICE-OVER-IP COMMUNICATIONS

The quality of VoIP communication relies significantly on the network that transports the voice packets because this network does not usually guarantee the available bandwidth, delay, and loss that are critical for real-time voice traffic. The solution proposed here is to manage the voice-over-IP stream dynamically, changing parameters as needed to assure quality. The main objective of this dissertation is to develop an adaptive speech encoding system that can be applied to conventional (telephony-grade) and wideband voice communications. This comprehensive study includes the investigation and development of three key components of the system. First, to manage VoIP quality dynamically, a tool is needed to measure real-time changes in quality. The E-model, which exists for narrowband communication, is extended to a single computational technique that measures speech quality for narrowband and wideband VoIP codecs. This part of the dissertation also develops important theoretical work in the area of wideband telephony. The second system component is a variable speech-encoding algorithm. Although VoIP performance is affected by multiple codecs and network-based factors, only three factors can be managed dynamically: voice payload size, speech compression and jitter buffer management. Using an existing adaptive jitter-buffer algorithm, voice packet-size and compression variation are studied as they affect speech quality under different network conditions. This study explains the relationships among multiple parameters as they affect speech transmission and its resulting quality. Then, based on these two components, the third system component is a novel adaptive-rate control algorithm that establishes the interaction between a VoIP sender and receiver, and manages voice quality in real-time. Simulations demonstrate that the system provides better average voice quality than traditional VoIP