Mobile Audiovisual Terminal: System Design and Subjective Testing in DECT and UMTS networks

It is anticipated that there will shortly be a requirement for multimedia terminals that operate via mobile communications systems. This paper presents a functional specification for such a terminal operating at 32 kb/s in a digital European cordless telecommunications (DECT) and universal mobile telecommunications system (UMTS) radio network. A terminal has been built, based on a PC with digital signal processor (DSP) boards for audio and video coding and decoding. Speech coding is by a phonetically driven code-excited linear prediction (CELP) speech coder and video coding by a block-oriented hybrid discrete cosine transform (DCT) coder. Separate channel coding is provided for the audio and video data. The paper describes the techniques used for audio and video coding, channel coding, and synchronization. Methods of subjective testing in a DECT network and in a UMTS network are also described. These consisted of subjective tests of first impressions of the mobile audio–visual terminal (MAVT) quality, interactive tests, and the completion of an exit questionnaire. The test results showed that the quality of the audio was sufficiently good for comprehension and the video was sufficiently good for following and repeating simple mechanical tasks. However, the quality of the MAVT was not good enough for general use where high-quality audio and video was needed, especially when transmission was in a noisy radio environment

Proceedings of the Second International Mobile Satellite Conference (IMSC 1990)

Presented here are the proceedings of the Second International Mobile Satellite Conference (IMSC), held June 17-20, 1990 in Ottawa, Canada. Topics covered include future mobile satellite communications concepts, aeronautical applications, modulation and coding, propagation and experimental systems, mobile terminal equipment, network architecture and control, regulatory and policy considerations, vehicle antennas, and speech compression

Media gateway utilizando um GPU

Mestrado em Engenharia de Computadores e Telemátic

Proceedings of the Third International Mobile Satellite Conference (IMSC 1993)

Satellite-based mobile communications systems provide voice and data communications to users over a vast geographic area. The users may communicate via mobile or hand-held terminals, which may also provide access to terrestrial cellular communications services. While the first and second International Mobile Satellite Conferences (IMSC) mostly concentrated on technical advances, this Third IMSC also focuses on the increasing worldwide commercial activities in Mobile Satellite Services. Because of the large service areas provided by such systems, it is important to consider political and regulatory issues in addition to technical and user requirements issues. Topics covered include: the direct broadcast of audio programming from satellites; spacecraft technology; regulatory and policy considerations; advanced system concepts and analysis; propagation; and user requirements and applications

A MODEL FOR PREDICTING THE PERFORMANCE OF IP VIDEOCONFERENCING

With the incorporation of free desktop videoconferencing (DVC) software on the majority of the world's PCs, over the recent years, there has, inevitably, been considerable interest in using DVC over the Internet. The growing popularity of DVC increases the need for multimedia quality assessment. However, the task of predicting the perceived multimedia quality over the Internet Protocol (IP) networks is complicated by the fact that the audio and video streams are susceptible to unique impairments due to the unpredictable nature of IP networks, different types of task scenarios, different levels of complexity, and other related factors. To date, a standard consensus to define the IP media Quality of Service (QoS) has yet to be implemented. The thesis addresses this problem by investigating a new approach to assess the quality of audio, video, and audiovisual overall as perceived in low cost DVC systems. The main aim of the thesis is to investigate current methods used to assess the perceived IP media quality, and then propose a model which will predict the quality of audiovisual experience from prevailing network parameters. This thesis investigates the effects of various traffic conditions, such as, packet loss, jitter, and delay and other factors that may influence end user acceptance, when low cost DVC is used over the Internet. It also investigates the interaction effects between the audio and video media, and the issues involving the lip sychronisation error. The thesis provides the empirical evidence that the subjective mean opinion score (MOS) of the perceived multimedia quality is unaffected by lip synchronisation error in low cost DVC systems. The data-gathering approach that is advocated in this thesis involves both field and laboratory trials to enable the comparisons of results between classroom-based experiments and real-world environments to be made, and to provide actual real-world confirmation of the bench tests. The subjective test method was employed since it has been proven to be more robust and suitable for the research studies, as compared to objective testing techniques. The MOS results, and the number of observations obtained, have enabled a set of criteria to be established that can be used to determine the acceptable QoS for given network conditions and task scenarios. Based upon these comprehensive findings, the final contribution of the thesis is the proposal of a new adaptive architecture method that is intended to enable the performance of IP based DVC of a particular session to be predicted for a given network condition

Πειραματικές μετρήσεις της ποιότητας ομιλίας VoIP

Voice-over-IP είναι μια οικογένεια τεχνολογιών επεξεργασίας και μετάδοσης φωνής που έχει σαν στόχο να εκμεταλλευτεί τις υπάρχουσες υποδομές των δικτύων δεδομένων. Τα δίκτυα VoIP υπόσχονται να μειώσουν το κόστος των τηλεφωνικών κλήσεων και να παρέχουν νέες υπηρεσίες επιταχύνοντας έτσι την ολοκλήρωση της τηλεφωνίας με τους υπολογιστές. Για να γίνει όμως ευρέως αποδεκτό το VoIP θα πρέπει να είναι σε θέση να επιτύχει την ποιότητα σήματος αλλά και γενικά την ποιότητα συνομιλίας που παρέχουν τα δημόσια τηλεφωνικά δίκτυα και στην οποία οι χρήστες είναι συνηθισμένοι. Σε άμεση συνάρτηση με αυτή την πρόκληση είναι η πρόκληση της ενσωμάτωσης του VoIP στα υφιστάμενα δίκτυα φωνητικής τηλεφωνίας και της ομαλής συνεργασίας του με αυτά. Το E-model είναι ένα υπολογιστικό μοντέλο που προτείνεται από την ITU και χρησιμοποιεί παραμέτρους της μετάδοσης για να προβλέψει την υποκειμενική ποιότητα της πακετοποιημένης ομιλίας. Συνδυάζει τους διαφορετικούς παράγοντες υποβάθμισης βασιζόμενο στην αρχή ότι το αποτέλεσμα των υποβαθμίσεων που αντιλαμβάνεται ο χρήστης - αν μετατραπεί στην κατάλληλη ψυχοακουστική κλίμακα - είναι προσθετικό. Στα πλαίσια της εργασίας, με τη χρήση του E-model μελετάται η ενσωμάτωση του VoIP στα υφιστάμενα δίκτυα καταρχήν με την ένταξη μιας γραμμής VoIP που συνδέει τμήματα ενός PSTN δικτύου και σε δεύτερο στάδιο με τη διασύνσεση του PSTN δικτύου με το IP δίκτυο κορμού. Οι επιπτώσεις της κωδικοποίησης στην ποιότητα της ομιλίας ελέγχεται και στις δύο τοπολογίες με τη χρήση του δημοφιλούς στα συστήματα VoIP κωδικοποιητή G.729. Τέλος αναλύονται οι επιπτώσεις που έχει στην ποιότητα της ομιλίας η αύξηση του φορτίου σε μια γραμμή VoIP - που αποτελεί μέρος της διαδρομής της φωνής - καθώς αυξάνεται σταδιακά μέχρι το σημείο που το απαιτούμενο εύρος ζώνης υπερβαίνει την χωρητικότητα της γραμμής.Voice-over-IP refers to an expanding family of voice processing and transport technologies that seek to take advantage of existing data network infrastructures. VoIP networks promise to reduce the cost of telephone calls and have the potential to provide unique new services and hasten computer telephony integration. To be widely accepted and employed, however, VoIP has to match the signal and conversational quality that is consistently delivered by the Public Switched Telephone Networks and to which telephone customers have become accustomed. Related to this challenge of achieving acceptable sound and conversational quality is the technical challenge of integrating and interworking VoIP with existing voice networks. The E-model is a computational model, standardized by ITU that uses transmission parameters to predict the subjective quality of packetized voice. The E-model combines different impairments based on the principle that the perceived effect of impairments is additive, when converted to the appropriate psycho-acoustic scale. In this work, with the use of E-model, the integration of VoIP is studied first with the introduction of a VoIP trunk interconnecting parts of the PSTN network, and then by the interworking of PSTN with the IP backbone network. The effect of voice coding in speech quality is also considered in both topologies with the use of codec G.729, among the most popular in VoIP environments. In the end the effects on speech quality are analyzed, when the traffic load of a VoIP trunk in the voice path is gradually incremented until the required bandwidth exceeds the trunk capacity

Proceedings of the Fifth International Mobile Satellite Conference 1997

Satellite-based mobile communications systems provide voice and data communications to users over a vast geographic area. The users may communicate via mobile or hand-held terminals, which may also provide access to terrestrial communications services. While previous International Mobile Satellite Conferences have concentrated on technical advances and the increasing worldwide commercial activities, this conference focuses on the next generation of mobile satellite services. The approximately 80 papers included here cover sessions in the following areas: networking and protocols; code division multiple access technologies; demand, economics and technology issues; current and planned systems; propagation; terminal technology; modulation and coding advances; spacecraft technology; advanced systems; and applications and experiments

Abstracts on Radio Direction Finding (1899 - 1995)

The files on this record represent the various databases that originally composed the CD-ROM issue of "Abstracts on Radio Direction Finding" database, which is now part of the Dudley Knox Library's Abstracts and Selected Full Text Documents on Radio Direction Finding (1899 - 1995) Collection. (See Calhoun record https://calhoun.nps.edu/handle/10945/57364 for further information on this collection and the bibliography). Due to issues of technological obsolescence preventing current and future audiences from accessing the bibliography, DKL exported and converted into the three files on this record the various databases contained in the CD-ROM. The contents of these files are: 1) RDFA_CompleteBibliography_xls.zip [RDFA_CompleteBibliography.xls: Metadata for the complete bibliography, in Excel 97-2003 Workbook format; RDFA_Glossary.xls: Glossary of terms, in Excel 97-2003 Workbookformat; RDFA_Biographies.xls: Biographies of leading figures, in Excel 97-2003 Workbook format]; 2) RDFA_CompleteBibliography_csv.zip [RDFA_CompleteBibliography.TXT: Metadata for the complete bibliography, in CSV format; RDFA_Glossary.TXT: Glossary of terms, in CSV format; RDFA_Biographies.TXT: Biographies of leading figures, in CSV format]; 3) RDFA_CompleteBibliography.pdf: A human readable display of the bibliographic data, as a means of double-checking any possible deviations due to conversion