A secure archive for Voice-over-IP conversations

An efficient archive securing the integrity of VoIP-based two-party conversations is presented. The solution is based on chains of hashes and continuously chained electronic signatures. Security is concentrated in a single, efficient component, allowing for a detailed analysis.Comment: 9 pages, 2 figures. (C) ACM, (2006). This is the author's version of the work. It is posted here by permission of ACM for your personal use. Not for redistribution. The definitive version was published in Proceedings of VSW06, June, 2006, Berlin, German

Mobile Multimedia Streaming Library

In recent years, multimedia has become a commonly used tool for presenting contents to the users. The employment of multimedia is no longer limited to only the entertainment industry, but spans in other areas as well. In academics, lectures are recorded to audio and video for storage and distribution to students. Free online multimedia hosting services are popularly cherished, such as “youtube.com” and “yahoo video”, and with the increasing affordability of digital camera, hundreds, or maybe thousands, of home-made videos and music audio are created daily and published online. Low-cost digital recorders such as webcams also help promote the use of video for surveillance, both for commercial and personal use. Suddenly, there comes the need for digital multimedia delivery, which happens naturally with the advancement in Internet bandwidth and the popularity of multimedia sharing. Multimedia delivery comes in two methods: downloading and streaming. Streaming requires more complex structure, but rewards with better user experience. Although streaming is the method of choice today, downloading is still useful in ad-hoc situation where streaming is not feasible. This project aims to provide streaming-like capability to mobile devices. Since mobile gadgets are limited in resources compared to personal computers (PC), streaming sometimes is the only way to deliver media contents to user. This work targets devices in the so-called “ad-hoc situation”, and also seeks to save the cost associated with multimedia streaming, which traditionally uses the operator wireless network, by using a LAN-connected proxy and the Bluetooth medium. It is also to serve the educational purpose in learning about multimedia streaming on cellular phones. This project experiments with several approaches to implement streaming on mobile phones. It discusses each approach in details. Finally, a library and a sample application are implemented to demonstrate the solution

DSL-based triple-play services

This research examines the triple play service based on the ADSL technology. The voice over IP will be checked and combined with the internet data by two monitoring programs in order to examine the performance that this service offers and then will be compared with the usual method of internet connection.This research examines the triple play service based on the ADSL technology. The voice over IP will be checked and combined with the internet data by two monitoring programs in order to examine the performance that this service offers and then will be compared with the usual method of internet connection.

Videopuhelun tallentaminen ja toisto

Recording a video call recording is beneficial in cases like job interviews and business meetings. Skype or Google Hangouts for example have no recording implemented without additional plugins. MP4 file container is capable of containing different types of media tracks, is widely used and is supported by media players and has a feature called RTP/RTCP Reception Hint Tracks. Hint tracks contain media transmission instructions, which can be used, e.g., to record RTP stream into a file. Without this information the video call session cannot be replayed afterwards. The purpose of this Thesis is to implement and verify the usage of RTP and RTCP Reception Hint Tracks in video call recording. No open-source MP4 multiplexing library or a media player with support for RTP/RTCP Reception Hint Tracks was found, so the support had to be implemented in both the library and the media player. The setup includes Linphone and L-SMASH for recording and VLC media player for playback. The created MP4 file has two RTP Reception Hint Tracks, two RTCP Reception Hint Tracks, and two media tracks. The GSM audio is chosen because it is supported by Linphone, L-SMASH, and VLC media player. H.264/AVC is chosen for video, because it is the best available codec supported by the three software. Tests were carried out using two laptops with both having recording enabled. From the tests it is concluded that using the RTP Reception Hint Track increases the total CPU usage by less than 1% and the size of the recorded video call by 4% over the conventional media tracks. The implementation shows that RTP Reception Hint Tracks meet well the needs of implementations with choice of different codecs

An Experimental Analysis of the Call Capacity of IEEE 802.11b Wireless Local Area Networks for VoIP Telephony

The use of the Internet to make phone calls is growing in popularity as the Voice over Internet protocol (VoIP) allows users to make phone calls virtually free of charge. The increased uptake of broadband services by domestic users will further increase the use of VoIP telephony. Furthermore, the emergence of low cost wireless networks (namely IEEE 802.11a/b/g WLANs) is expected to bring wireless VoIP into the mainstream. As the number of wireless hotspots increases more users will want to use VoIP calls wherever possible by connecting to open access points (AP). A major concern with VoIP is Quality of Service (QoS). In order for VoIP to be truly successful users must enjoy a similar perceived QoS as a call made over a traditional telephone network. There are many factors that influence QoS which include: throughput, packet delay, delay variation (or jitter), and packet loss. This thesis is an experimental study of the call capacity of an IEEE 802.11b network when using VoIP telephony. Experiments included increasing the number of VoIP stations and also increasing the level of background traffic until network saturation occurs. Results show that the network is capable of supporting at least 16 VoIP stations. Due to the operation of the IEEE 802.11 medium access control (MAC) mechanism, the AP acts as a bottleneck for all traffic destined for wireless stations, in that significant delays can be incurred by VoIP packets which can lead to a poor perceived QoS by users. Consequently the performance of the AP downlink is the critical component in determining VoIP call capacity

Guidelines for using the multiplexing features of RTP to support multiple media streams

The Real-time Transport Protocol (RTP) is a flexible protocol that can be used in a wide range of applications, networks, and system topologies. That flexibility makes for wide applicability but can complicate the application design process. One particular design question that has received much attention is how to support multiple media streams in RTP. This memo discusses the available options and design trade-offs, and provides guidelines on how to use the multiplexing features of RTP to support multiple media streams

An Internet based multimedia infrastructure for collaborative engineering

Thesis (S.M.)--Massachusetts Institute of Technology, Dept. of Civil and Environmental Engineering, 2000.Includes bibliographical references (leaves 129-131).The evolution of computer based collaborative environments has resulted in easier and more economical design efforts among geographically distributed design teams. Most of today's internet based collaborative applications allow people that are geographically dispersed to meet with each other using their computers and work together without actually having to travel. A prototype system was developed by taking two tactical planning applications and incorporating them into the collaboration model employed by CAIRO (Collaborative Agent Interaction control and synchROnization). This system was developed based on the collaboration infrastructure that was developed as a part of the Da-Vinci Society Initiative at MIT. The main focus of this research lies in the formalization of a multi-media based architecture that supplements the existing collaboration infrastructure. This architecture lays the groundwork for development of a robust collaboration system that incorporates audio/video conferencing, speech recognition and synthesis and three-dimensional virtual meeting environments in order to facilitate efficient collaboration.by Padmanabha N. Vedam.S.M

Analyzing Voice And Video Call Service Performance Over A Local Area Network

Tez (Yüksek Lisans) -- İstanbul Teknik Üniversitesi, Fen Bilimleri Enstitüsü, 2010Thesis (M.Sc.) -- İstanbul Technical University, Institute of Science and Technology, 2010Bu çalışmada, VOIP teknolojisinden ve bu teknolojiyi kablolu ve kablosuz ortamda gerçeklemenin en önemli darboğazları anlatılacaktır. Ayrıca H.323, SIP (Session Initiation Protocol), Megaco ve MGCP gibi yaygın olarak kullanılan ses iletim protokolleri ve H.261, H.263 ve H.264 gibi görüntü iletim protokollerinden bahsedilmiştir. Ses kodek seçimi ve VOIP servis kalitesine etki eden faktörleri anlatılmaktadır. Bu tezde, ses, görüntü ve veri iletişimini aynı anda bünyesinde barındıran gerçek şebekeler simüle edilecektir. Kullanıcılara rastlantısal olarak ses, görüntü ve FTP gibi birtakım uygulamalar atanmıştır. Ayrıca önerilen kablolu şebekeye, kablosuz bir şebeke ilave edilerek sonuçlar incelenecektir. Optimal servis kalitesini sağlamak için seçilen uygun kuyruklama mekanizmaları ve kodek seçimlerini içeren senaryolar incelenecek ve OPNET ile elde edilmiş simülasyon sonuçları tartışılacaktır.In this study, we present a detailed description of the VoIP and also the most common challenges of implementing voice communication into wireline or wireless networks are discussed. Common voice protocols, such as H.323, Session Initiation Protocol (SIP), Megaco, MGCP and video protocols such as H.261, H.263, H.264 are described as well. CODEC selection and factors affecting VoIP Quality of Service are analyzed. We simulate a real network which includes both voice, video and data communication simultaneously. Workstations are randomly assigned to different applications, such as voice, video and FTP. We will also implement a wireless network to our proposed system. The scenarios including selecting appropriate queuing scheme and codec selection are presented and the simulation results with OPNET are drawn.Yüksek LisansM.Sc

Quality of Service Controlled Multimedia Transport Protocol

PhDThis research looks at the design of an open transport protocol that supports a range of services including multimedia over low data-rate networks. Low data-rate multimedia applications require a system that provides quality of service (QoS) assurance and flexibility. One promising field is the area of content-based coding. Content-based systems use an array of protocols to select the optimum set of coding algorithms. A content-based transport protocol integrates a content-based application to a transmission network. General transport protocols form a bottleneck in low data-rate multimedia communicationbsy limiting throughpuot r by not maintainingt iming requirementsT. his work presents an original model of a transport protocol that eliminates the bottleneck by introducing a flexible yet efficient algorithm that uses an open approach to flexibility and holistic architectureto promoteQ oS.T he flexibility andt ransparenccyo mesi n the form of a fixed syntaxt hat providesa seto f transportp rotocols emanticsT. he mediaQ oSi s maintained by defining a generic descriptor. Overall, the structure of the protocol is based on a single adaptablea lgorithm that supportsa pplication independencen, etwork independencea nd quality of service. The transportp rotocol was evaluatedth rougha set of assessmentos:f f-line; off-line for a specific application; and on-line for a specific application. Application contexts used MPEG-4 test material where the on-line assessmenuts eda modified MPEG-4 pl; yer. The performanceo f the QoSc ontrolledt ransportp rotocoli s often bettert hano thers chemews hen appropriateQ oS controlledm anagemenatl gorithmsa re selectedT. his is shownf irst for an off-line assessmenwt here the performancei s compared between the QoS controlled multiplexer,a n emulatedM PEG-4F lexMux multiplexers chemea, ndt he targetr equirements. The performanceis also shownt o be better in a real environmentw hen the QoS controlled multiplexeri s comparedw ith the real MPEG-4F lexMux scheme