An analytical model for capacity evaluation of VoIP on HCCA and TCP file transfers over EDCA in an IEEE 802.11e WLAN

In the context of the IEEE 802.11e standard for WLANs, we provide an analytical model for obtaining the maximum number of VoIP calls that can be supported on HCCA, such that the delay QoS constraint of the accepted calls is met, when TCP downloads are coexistent on EDCA. In this scenario, we derive the TCP download throughput by using an analytical model for the case where only TCP sessions are present in the WLAN. We show that the analytical model for combined voice and TCP transfers provides accurate results in comparison with simulations (using ns-2).

An analytical model for the capacity estimation of combined VoIP and TCP file transfers over EDCA in an IEEE 802.11e WLAN

In this paper we develop and numerically explore the modeling heuristic of using saturation attempt probabilities as state dependent attempt probabilities in an IEEE 802.11e infrastructure network carrying packet telephone calls and TCP controlled file downloads, using Enhanced Distributed Channel Access (EDCA). We build upon the fixed point analysis and performance insights in [1]. When there are a certain number of nodes of each class contending for the channel (i.e., have nonempty queues), then their attempt probabilities are taken to be those obtained from saturation analysis for that number of nodes. Then we model the system queue dynamics at the network nodes. With the proposed heuristic, the system evolution at channel slot boundaries becomes a Markov renewal process, and regenerative analysis yields the desired performance measures.The results obtained from this approach match well with ns2 simulations. We find that, with the default IEEE 802.11e EDCA parameters for AC 1 and AC 3, the voice call capacity decreases if even one file download is initiated by some station. Subsequently, reducing the voice calls increases the file download capacity almost linearly (by 1/3 Mbps per voice call for the 11 Mbps PHY).

An analytical model for an IEEE~802.11 WLAN using DCF with two types of VoIP calls

In this paper we develop and numerically explore the modeling heuristic of using saturation attempt probabilities as state dependent attempt probabilities in an IEEE 802.11e infrastructure network carrying packet telephone calls and TCP controlled file downloads, using enhanced distributed channel access (EDCA). We build upon the fixed point analysis and performance insights. When there are a certain number of nodes of each class contending for the channel (i.e., have nonempty queues), then their attempt probabilities are taken to be those obtained from saturation analysis for that number of nodes. Then we model the system queue dynamics at the network nodes. With the proposed heuristic, the system evolution at channel slot boundaries becomes a Markov renewal process, and regenerative analysis yields the desired performance measures. The results obtained from this approach match well with ns2 simulations. We find that, with the default IEEE 802.11e EDCA parameters for AC 1 and AC 3, the voice call capacity decreases if even one file download is initiated by some station. Subsequently, reducing the voice calls increases the file download capacity almost linearly (by 1/3 Mbps per voice call for the 11 Mbps PHY)

A Scalable Architecture for VoIP Conferencing

Real-Time services are traditionally supported on circuit switched network. However, there is a need to port these services on packet switched network. Architecture for audio conferencing application over the Internet in the light of ITU-T H.323 recommendations is considered. In a conference, considering packets only from a set of selected clients can reduce speech quality degradation because mixing packets from all clients can lead to lack of speech clarity. A distributed algorithm and architecture for selecting clients for mixing is suggested here based on a new quantifier of the voice activity called “Loudness Number” (LN). The proposed system distributes the computation load and reduces the load on client terminals. The highlights of this architecture are scalability, bandwidth saving and speech quality enhancement. Client selection for playing out tries to mimic a physical conference where the most vocal participants attract more attention. The contributions of the paper are expected to aid H.323 recommendations implementations for Multipoint Processors (MP). A working prototype based on the proposed architecture is already functional.

Telefonia na Internet : a "voz sobre IP" (VOIP) e seus impactos nas operadoras de telefonia

Consultoria Legislativa - Área XIV Comunicação Social, Informática, Telecomunicações, Ciência Postal, Ciência e Tecnologia. Inclui gráficos.

Assessment of the VoIP Transmission Quality over Digital Power Line Carrier Channels

This article is devoted to the research of VoIP transmission quality over Digital Power Line Carrier channels. Assessment of quality transmission is performed using E-model. Paper considers the possibility of joint using of Digital Power Line carrier equipment with different architecture in one network. As a result of the research, the rule for constructing of multi-segment Digital Power Line Carrier channels was formulated. This rule allows minimizing the transmission delay and saving frequency resources of high voltage Power Line Carrier range.

Le support de VoIP dans les réseaux maillés sans fil WiMAX en utilisant une approche de contrôle et d'assistance au niveau MAC

Les réseaux maillés sans fil (RMSF), grâce à leurs caractéristiques avantageuses, sont considérés comme une solution efficace pour le support des services de voix, vidéo et de données dans les réseaux de prochaine génération. Le standard IEEE 802.16-d a spécifié pour les RMSF, à travers son mode maillé, deux mécanismes de planifications de transmission de données; à savoir la planification centralisée et la planification distribuée. Dans ce travail, on a évalué le support de la qualité de service (QdS) du standard en se focalisant sur la planification distribuée. Les problèmes du système dans le support du trafic de voix ont été identifiés. Pour résoudre ces problèmes, on a proposé un protocole pour le support de VoIP (AVSP) en tant qu’extension au standard original pour permettre le support de QdS au VoIP. Nos résultats préliminaires de simulation montrent qu’AVSP offre une bonne amélioration au support de VoIP.

An Upgrade of Network Traffic Recognition System for SIP/VoIP Traffic Recognition

The purpose of this project is to update the tool of Network Traffic Recognition System (NTRS) which is proprietary software of Ericsson AB and Tsinghua University, and to implement the updated tool to finish SIP/VoIP traffic recognition. Basing on the original NTRS, I analyze the traffic recognition principal of NTRS, and redesign the structure and module of the tool according to characteristics of SIP/VoIP traffic, and then finally I program to achieve the upgrade. After the final test with our SIP data trace files in the updated system, a satisfactory result is derived. The result presents that our updated system holds a rate of recognition on a confident level in the SIP session recognition as well as the VoIP call recognition. In the comparison with the software of Wireshark, our updated system has a result which is extremely close to Wireshark’s output, and the working time is much less than Wireshark. In the aspect of practicability, the memory overflow problem is avoided, and the updated system can output the specific information of SIP/VoIP traffic recognition, such as SIP type, SIP state, VoIP state, etc. The upgrade fulfills the demand of this project.

VOIP : um estudo experimental

Voz sobre IP (VoIP) é uma tecnologia que permite a digitalização e a codificação da voz e o empacotamento em pacotes de dados IP para a transmissão em uma rede que utilize o protocolo TCP/IP. Devido ao volume de dados gerados por uma aplicação VoIP, esta tecnologia se encontra em funcionamento, em redes corporativas privadas. Mas se a rede base para o transporte desta aplicação for a Internet, certamente, não deve ser utilizada para fins profissionais, pois o TCP/IP não oferece padrões de QoS (Qualidade de Serviço) comprometendo desta forma a qualidade da voz. A qualidade da voz fica dependente do tráfego de dados existentes no momento da conversa. Para realizar um projeto de VoIP é necessário conhecer todo o tráfego existente na rede e verificar o quanto isto representa em relação à banda total da rede. Também se deve conhecer o tipo de aplicação que se deseja implantar, verificando a banda a ser utilizada por esta, e então projetar como a rede deverá ser estruturada. Para auxiliar no projeto de VoIP, pretende-se mostrar o que está sendo desenvolvido para que o protocolo TCP/IP ofereça QoS e uma ferramenta para análise do tráfego de voz sobre redes TCP/IP e também análises dos resultados obtidos em experimentos simulando diversas situações práticas.

Estudo da qualidade de serviço de uma aplicação VoIP em ambientes wireless com handoff

This work deals with experimental studies about VoIP conections into WiFi 802.11b networks with handoff. Indoor and outdoor network experiments are realised to take measurements for the QoS parameters delay, throughput, jitter and packt loss. The performance parameters are obtained through the use of software tools Ekiga, Iperf and Wimanager that assure, respectvely, VoIP conection simulation, trafic network generator and metric parameters acquisition for, throughput, jitter and packt loss. The avarage delay is obtained from the measured throughput and the concept of packt virtual transmition time. The experimental data are validated based on de QoS level for each metric parameter accepted as adequated by the specialized literature

Uma arquitetura baseada em SCTP e SIP para suporte a aplicações VoIP móveis e a especificação formal do seu módulo de controle

New versions of SCTP protocol allow the implementation of handover procedures in the transport layer, as well as the supply of a partially reliable communication service. A communication architecture is proposed herein, integrating SCTP with the session initiation protocol, SIP, besides additional protocols. This architecture is intended to handle voice applications over IP networks with mobility requirements. User localization procedures are specified in the application layer as well, using SIP, as an alternative mean to the mechanisms used by traditional protocols, that support mobility in the network layer. The SDL formal specification language is used to specify the operation of a control module, which coordinates the operation of the system component protocols. This formal specification is intended to prevent ambiguities and inconsistencies in the definition of this module, assisting in the correct implementation of the elements of this architecture