Development of Campus Video-Conference System Based on Peer-To-Peer Architecture

Peer to Peer (P2P) systems inherently have high scalability, robustness and fault tolerance because there is no centralized server and the network self-organizes itself. This is achieved at the cost of higher latency for locating the resources of interest in the P2P overlay network. This paper describes the design and implementation of campus video conference system based on P2P architecture that was tested within premises of Ladoke Akintola University of Technology, Ogbomoso, Nigeria. The proposed Campus video conference system is made up of five modules which are the media stream engine, the conferencing control protocol, transmission module, TCP/UDP module and the user interface module. The media stream engine is responsible for audio/video capture and playback, the conferencing control protocol defines a set of conventions governing the structure and behavior of communication messages, the transmission module consists of a peer and a distribution network constituting of the peers also the delivery and exchange of streaming data while the audio manager and video manager use TCP/UDP to broadcast to other peer. The proposed system will offer smooth video conferencing with low delay and seldom and short freezes. It is believed that this videoconference system will bring video telephony to a new level of quality and will lead to a new trend in everyday communications in the university community

Applications Technology Satellite and Communications Technology Satellite user experiments for 1967 - 1980 reference book, volume 1

A description of each of the satellites is given and a brief summary of each user experiment is presented. A Cross Index of User Experiments sorted by various parameters and a listing of keywords versus Experiment Number are presented

Synchronizing sound from different devices over a TCP network

Nowadays, we can send audio on the Internet for multiples uses like telephony, broadcast audio or teleconferencing. The issue comes when you need to synchronize the sound from different sources because the network where we are going to work could lose packets and introduce delay in the delivery. This can also come because the sound cards could be work in different speeds. In this project, we will work with two computers emitting sound (one will simulate the left channel (mono) of a stereo signal, and the other the right channel) and connected with a third computer by a TCP network. The last computer must get the sound from both computers and reproduce it in a speaker properly (without delay). So, basically, the main goal of the project is to synchronize multi-track sound over a network. TCP networks introduce latency into data transfers. Streaming audio suffers from two problems: a delay and an offset between the channels. This project explores the causes of latency, investigates the affect of the inter-channel offset and proposes a solution to synchronize the received channels. In conclusion, a good synchronization of the sound is required in a time when several audio applications are being developed. When two devices are ready to send audio over a network, this multi-track sound will arrive at the third computer with an offset giving a negative effect to the listener. This project has dealt with this offset achieving a good synchronization of the multitrack sound getting a good effect on the listener. This was achieved thanks to the division of the project into several steps having constantly a good vision of the problem, a good scalability and having controlled the latency at all times. As we can see in the chapter 4 of the project, a lack of synchronization over c. 100μs is audible to the listener. RESUMEN. A día de hoy, podemos transmitir audio a través de Internet por varios motivos como pueden ser: una llamada telefónica, una emisión de audio o una teleconferencia. El problema viene cuando necesitas sincronizar ese sonido producido por los diferentes orígenes ya que la red a la que nos vamos a conectar puede perder los paquetes y/o introducir un retardo en las entregas de los mismos. Así mismo, estos retardos también pueden venir producidos por las diferentes velocidades a las que trabajan las tarjetas de sonido de cada dispositivo. En este proyecto, se ha trabajado con dos ordenadores emitiendo sonido de manera intermitente (uno se encargará de simular el canal izquierdo (mono) de la señal estéreo emitida, y el otro del canal derecho), estando conectados a través de una red TCP a un tercer ordenador, el cual debe recibir el sonido y reproducirlo en unos altavoces adecuadamente y sin retardo (deberá juntar los dos canales y reproducirlo como si de estéreo de tratara). Así, el objetivo principal de este proyecto es el de encontrar la manera de sincronizar el sonido producido por los dos ordenadores y escuchar el conjunto en unos altavoces finales. Las redes TCP introducen latencia en la transferencia de datos. El streaming de audio emitido a través de una red de este tipo puede sufrir dos grandes contratiempos: retardo y offset, los dos existentes en las comunicaciones entre ambos canales. Este proyecto se centra en las causas de ese retardo, investiga el efecto que provoca el offset entre ambos canales y propone una solución para sincronizar los canales en el dispositivo receptor. Para terminar, una buena sincronización del sonido es requerida en una época donde las aplicaciones de audio se están desarrollando continuamente. Cuando los dos dispositivos estén preparados para enviar audio a través de la red, la señal de sonido multi-canal llegará al tercer ordenador con un offset añadido, por lo que resultará en una mala experiencia en la escucha final. En este proyecto se ha tenido que lidiar con ese offset mencionado anteriormente y se ha conseguido una buena sincronización del sonido multi-canal obteniendo un buen efecto en la escucha final. Esto ha sido posible gracias a una división del proyecto en diversas etapas que proporcionaban la facilidad de poder solucionar los errores en cada paso dando una importante visión del problema y teniendo controlada la latencia en todo momento. Como se puede ver en el capítulo 4 del proyecto, la falta de sincronización sobre una diferencia de 100μs entre dos canales (offset) empieza a ser audible en la escucha final

Spatial Error Concealment in Ad-hoc Audio Conferencing Systems

In this work we consider an ad-hoc audio conferensing system based on VoIP services in which the participants connect to the conference using mobile communication devices with wireless connectivity. To overcome possible quality problems in the wireless link, we propose improvements to the existing conferencing systems. Some networking modifications are suggested to increase the channel capacity and robustness from the conference server to multiple clients. On the other hand, for the improvement of the uplink quality, we suggest a new spatial error concealment method, where a backup device captures and sends the audio signals to the server together with the primary device. In the server the lost frames from the primary channel are estimated based on the backup signal. Several methods for estimating the primary signal based on the backup signal are studied. The results of the methods are evaluated by a psychoacoustic error metric based on Zwicker’s loudness model. An informal subjective test is also performed to compare the results of these methods in order to chose one for implementing on the real- time conferencing setup. Both objective and subjective tests show consistent results and confirm that usage of spatial error concealment improves significantly the audio quality in the primary signal

Integration of the White Sands Complex into a Wide Area Network

The NASA White Sands Complex (WSC) satellite communications facility consists of two main ground stations, an auxiliary ground station, a technical support facility, and a power plant building located on White Sands Missile Range. When constructed, terrestrial communication access to these facilities was limited to copper telephone circuits. There was no local or wide area communications network capability. This project incorporated a baseband local area network (LAN) topology at WSC and connected it to NASA's wide area network using the Program Support Communications Network-Internet (PSCN-I). A campus-style LAN is configured in conformance with the International Standards Organization (ISO) Open Systems Interconnect (ISO) model. Ethernet provides the physical and data link layers. Transmission Control Protocol and Internet Protocol (TCP/IP) are used for the network and transport layers. The session, presentation, and application layers employ commercial software packages. Copper-based Ethernet collision domains are constructed in each of the primary facilities and these are interconnected by routers over optical fiber links. The network and each of its collision domains are shown to meet IEEE technical configuration guidelines. The optical fiber links are analyzed for the optical power budget and bandwidth allocation and are found to provide sufficient margin for this application. Personal computers and work stations attached to the LAN communicate with and apply a wide variety of local and remote administrative software tools. The Internet connection provides wide area network (WAN) electronic access to other NASA centers and the world wide web (WWW). The WSC network reduces and simplifies the administrative workload while providing enhanced and advanced inter-communications capabilities among White Sands Complex departments and with other NASA centers

Web Based GUI Management for FlexiNT22 SHDSL.bis Modem

The purpose of this thesis was to design and implement a prototype of a web based GUI management for FlexiNT22 SHDSL.bis modem at Nokia Siemens Networks BBA NBMS division. Web based GUI management gives an administrator the ability to configure and monitor FlexiNT22 over the Internet using a web browser. The most direct way to accomplish this is to embed a web server (Embedded web Server) into the mo-dem, and use that server to provide web-based management user interface con-structed using HTML language. The project involved familiarizing with the operations and characteristics of Flex-iNT, searching for an appropriate web server, examining the features and the compatibility with the software, evaluation and implementing a demo version. The evaluation consists mainly of three parts: Surveying web servers and further choosing the most suitable web servers for the evaluation. In the third place, we were concerned with defining criteria of the embedded web server features for evaluation. The project was carried out using KLone embedded web server, which is open source software. The study describes how to program HTML pages in C language and how to implement web pages. As result pages could be embedded into a sin-gle executable binary file that contained KLone’s HTTP/S server.Tämän opinnäytetyön tavoitteena oli suunnitella ja toteuttaa graafinen web-pohjainen käyttöliittymä prototyyppi FlexiNT22 SHDSL.bis päätelaitteeseen Nokia Siemens Networksin NBMS (Narrowband Multiservic) osastolle. Web-hallintasovellus antaa järjestelmänvalvojalle mahdollisuuden määrittää ja valvoa FlexiNT22 modeemia internetin kautta käyttäen web-selainta.Toteuttaakseen tämän tarvitaan sulautettu Web-palvelin joka upotetaan modeemin sisään, ja käyttäen tätä palvelinta luodaan web-hallintasovellus HTML-kielellä. Projekti jakautuu kahteen osaan: tutkiminen ja implementaatio. Tutkimukseen kuului perehtyminen FlexiNT:n toimintaan ja ominaisuuksiin, sopivan web-palvelimen etsiminen ja sen ominaisuuksien tutkiminen ja soveltuvuus ohjelmiston kanssa. Sulautettu web-palvelin on ideaali tähän projektiin. Valintamenetelmiin kuului valita kolme sopivinta palvelinta ja tutkia niiden ominaisuudet. Implementaatioon kuului suunnitella ja toteuttaa toimiva web-hallintasovellus runko. Työ toteutettiin käyttäen KLonen web-palvelinta, joka on avoimen lähdekoodin ohjelmisto. Tässä työssä kuvataan, miten ohjelmoidaan HTML-sivuja käyttäen C-kieltä ja miten ne toteutetaan. Lopputuloksena web sivut voidaan upottaa yhteen binääri ohjelmatiedostoon, joka sisältää KLone HTTP/S-palvelimen

Delivering video services over IP networks

The main goal pursued in this Thesis is to contribute towards the design and development of an end-to-end solution/system that would assist in reliable, consistence, less packet-loss delivery of high-quality video signals of pre-recorded presentations, training lectures, live events such as seminars over standard IP networks. This Thesis will focus on the existing Internet Service Provider, Oman Telecommunications Company (Omantel) and its best delivery of high-bandwidth data such as video to its Local and regional offices and departments over IP networks. This video-over-IP system aims to accumulate the technical scientific knowledge required to be able to offer high-quality video, which is fully scalable over IP networks. It aims to convert this knowledge into experimental prototypes, which, after the Thesis, can be developed into an integrated generic environment for Video-over-IP service development and content production. The objective is to initially define the functionality of content Services that can be incorporated into the operations of Oman telecommunications company networks. Then define the functional characteristics and system requirements necessary for the deployment of content streaming services over Omantel IP based networks. The design of this system would be combined with streaming high-quality video, while maintaining scalability and bandwidth efficiencies required for large-scale enterprise deployment. The design would encompass various components that are needed to capture, store and deliver streaming video to desktops. It will investigate on what is required to deliver quality video over Omantel IP networks and will recommend the actual products and solutions for achieving the end result

Communication Architecture For Distributed Interactive Simulation (CADIS): Rationale Document Draft

Report on necessary communication system protocol data unit standards which must be accepted and adopted for supporting distributed interactive simulation

A comprehensive VoIP system with PSTN connectivity.

Yuen Ka-nang.Thesis (M.Phil.)--Chinese University of Hong Kong, 2001.Includes bibliographical references (leaves 133-135).Abstracts in English and Chinese.Abstract --- p.iAcknowledgement --- p.iiiChapter 1. --- INTRODUCTION --- p.1Chapter 1.1. --- Background --- p.1Chapter 1.2. --- Objectives --- p.1Chapter 1.3. --- Overview of Thesis --- p.2Chapter 2. --- NETWORK ASPECT OF THE VOIP TECHNOLOGY --- p.3Chapter 2.1. --- VoIP Overview --- p.3Chapter 2.2. --- Elements in VoIP --- p.3Chapter 2.2.1. --- Call Setup --- p.3Chapter 2.2.2. --- Media Capture/Playback --- p.4Chapter 2.2.3. --- Media Encoding/Decoding --- p.4Chapter 2.2.4. --- Media Transportation --- p.5Chapter 2.3. --- Performance Factors Affecting VoIP --- p.6Chapter 2.3.1. --- Network Bandwidth --- p.6Chapter 2.3.2. --- Latency --- p.6Chapter 2.3.3. --- Packet Loss --- p.7Chapter 2.3.4. --- Voice Quality --- p.7Chapter 2.3.5. --- Quality of Service (QoS) --- p.7Chapter 2.4. --- Different Requirements of Intranet VoIP and Internet VoIP --- p.8Chapter 2.4.1. --- Packet Loss/Delay/Jitter --- p.8Chapter 2.4.2. --- Interoperability --- p.9Chapter 2.4.3. --- Available Bandwidth --- p.9Chapter 2.4.4. --- Security Requirement --- p.10Chapter 2.5. --- Some Feasibility Investigations --- p.10Chapter 2.5.1. --- Bandwidth Calculation --- p.10Chapter 2.5.2. --- Simulation --- p.12Chapter 2.5.3. --- Conclusion --- p.17Chapter 2.5.4. --- Simulation Restrictions --- p.17Chapter 3. --- SOFTWARE ASPECT OF THE VOIP TECHNOLOGY --- p.19Chapter 3.1. --- VoIP Client in JMF --- p.19Chapter 3.1.1. --- Architecture --- p.20Chapter 3.1.2. --- Incoming Voice Stream Handling --- p.23Chapter 3.1.3. --- Outgoing Voice Stream Handling --- p.23Chapter 3.1.4. --- Relation between Incoming/Outgoing Voice Stream Handling --- p.23Chapter 3.1.5. --- Areas for Further Improvement --- p.25Chapter 3.2. --- Capture/Playback Enhanced VoIP Client --- p.26Chapter 3.2.1. --- Architecture --- p.27Chapter 3.2.2. --- Native Voice Playback Mechanism --- p.29Chapter 3.2.3. --- Native Voice Capturing Mechanism --- p.31Chapter 3.3. --- Win32 C++ VoIP Client --- p.31Chapter 3.3.1. --- Objectives --- p.32Chapter 3.3.2. --- Architecture --- p.33Chapter 3.3.3. --- Problems and Solutions in Implementation --- p.37Chapter 3.4. --- Win32 DirectSound C++ VoIP Client --- p.38Chapter 3.4.1. --- Architecture --- p.39Chapter 3.4.2. --- DirectSound Voice Playback Mechanism --- p.40Chapter 3.4.3. --- DirectSound Voice Capturing Mechanism --- p.44Chapter 3.5. --- Testing VoIP Clients --- p.45Chapter 3.5.1. --- Setup of Experiment --- p.45Chapter 3.5.2. --- Experiment Results --- p.47Chapter 3.5.3. --- Experiment Conclusion --- p.48Chapter 3.6. --- Real-time Voice Stream Mixing Server --- p.48Chapter 3.6.1. --- Structure Overview --- p.48Chapter 3.6.2. --- Experiment --- p.53Chapter 3.6.3. --- Conclusion --- p.54Chapter 4. --- EXPERIMENTAL STUDIES --- p.55Chapter 4.1. --- Pure IP-side VoIP-based Call Center ´ؤ VoIP in Education --- p.55Chapter 4.1.1. --- Architecture --- p.55Chapter 4.1.2. --- Client Structure --- p.56Chapter 4.1.3. --- Client Applet User Interface --- p.58Chapter 4.1.4. --- Observations --- p.63Chapter 4.2. --- A Simple PBX Experiment --- p.63Chapter 4.2.1. --- Structural Overview --- p.63Chapter 4.2.2. --- PSTN Gateway Server Program --- p.64Chapter 4.2.3. --- Problems and Solutions in Implementation --- p.66Chapter 4.2.4. --- Experiment 1 --- p.66Chapter 4.2.5. --- Experiment 2 --- p.68Chapter 5. --- A COMPREHENSIVE VOIP PROJECT 一 GRADUATE SECOND PHONE (GSP) --- p.72Chapter 5.1. --- Overview --- p.72Chapter 5.1.1. --- Background --- p.72Chapter 5.1.2. --- Architecture --- p.76Chapter 5.1.3. --- Technologies Used --- p.78Chapter 5.1.4. --- Major Functions --- p.80Chapter 5.2. --- Client --- p.84Chapter 5.2.1. --- Structure Overview --- p.85Chapter 5.2.2. --- Connection Procedure --- p.89Chapter 5.2.3. --- User Interface --- p.91Chapter 5.2.4. --- Observations --- p.92Chapter 5.3. --- Gateway --- p.94Chapter 5.3.1. --- Structure Overview --- p.94Chapter 5.3.2. --- Connection Procedure --- p.97Chapter 5.3.3. --- Caller ID Simulator --- p.97Chapter 5.3.4. --- Observations --- p.98Chapter 5.4. --- Server --- p.101Chapter 5.4.1. --- Structure Overview --- p.101Chapter 5.5. --- Details of Major Functions --- p.103Chapter 5.5.1. --- Secure Local Voice Message Box --- p.104Chapter 5.5.2. --- Call Distribution --- p.106Chapter 5.5.3. --- Call Forward --- p.112Chapter 5.5.4. --- Call Transfer --- p.115Chapter 5.6. --- Experiments --- p.116Chapter 5.6.1. --- Secure Local Voice Message Box --- p.117Chapter 5.6.2. --- Call Distribution --- p.118Chapter 5.6.3. --- Call Forward --- p.121Chapter 5.6.4. --- Call Transfer --- p.122Chapter 5.6.5. --- Dial Out --- p.124Chapter 5.7. --- Observations --- p.125Chapter 5.8. --- Outlook --- p.126Chapter 5.9. --- Alternatives --- p.127Chapter 5.9.1. --- Netmeeting --- p.127Chapter 5.9.2. --- OpenH323 --- p.128Chapter 6. --- CONCLUSIONS --- p.129Bibliography --- p.13