IMS-Based Services INTERNATIONAL INTERCONNECTION FORUM
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Fully Eliminated the Language Barrier and Enable Ease of Communication Through This Application
IOSR Journal of Computer Engineering (IOSR-JCE) e-ISSN: 2278-0661, p- ISSN: 2278-8727Volume 16, Issue 2, Ver. XI (Mar-Apr. 2014), PP 113-119 www.iosrjournals.org Alltalk™- A Windows Phone Messenger with Cross Language Communication Shruti Shetye1, Akhil Abraham2, Royston Pinto3, Sonali Vaidya4 1(BE-IT Student, Information Technology, St. FrancisInstitute of Technology, India) 2(BE-IT Student, Information Technology, St. Francis Institute of Technology, India) 3(BE-IT Student, Information Technology, St. Francis Institute of Technology, India 4(Lecturer, Information Technology, St. Francis Institute of Technology, India) __________________________________________________________________________________ Abstract:In day to day life, messengers or chatting applications provide facility for instant messaging over the internet. Exchange of messages takes place in universally used languages like English, French, etc. where both the users know how to communicate in a common language. Thus chatting on mobile phones is a luxury when both the parties involved know a common language. Hence we have implemented ALLTALK™ which is a Windows 8 phone based chatting application which makes cross language communication possible using mobile programming and networking technology.This application will enable the communication between two persons irrespective of the language each user wishes to use individually. The various modes of communication available in this messenger are through text and voice. Due to the best processing power provided among the available smartphones and high battery life we choose to work on windows 8 platform. Thus we have successfully eliminated the language barrier and enable ease of communication through this application. Keywords: Cross Language communication, instant messenger, socket connection, translator,Windows phone app. -
IM and Presence Service Features and Functions
IM and Presence Service Features and Functions • IM and Presence Service Components, on page 1 • IM and Presence Service Feature Deployment Options, on page 5 • Deployment models, on page 7 • User Assignment, on page 9 • End User Management, on page 9 • Availability and Instant Messaging, on page 10 • LDAP Integrations, on page 13 • Third-Party Integrations, on page 14 • Third-Party Client Integration, on page 15 • IM Address Schemes and Default Domain, on page 16 • Security, on page 19 • Single Sign-On, on page 19 IM and Presence Service Components Main Components The following figure provides an overview of an IM and Presence Service deployment, including the main components and interfaces between Cisco Unified Communications Manager and IM and Presence Service. IM and Presence Service Features and Functions 1 IM and Presence Service Features and Functions SIP Interface Figure 1: IM and Presence Service Basic Deployment SIP Interface A SIP connection handles the presence information exchange between Cisco Unified Communications Manager and Cisco Unified Presence. To enable the SIP connection on Cisco Unified Communications Manager, you must configure a SIP trunk pointing to the Cisco Unified Presence server. On Cisco Unified Presence, configuring Cisco Unified Communications Manager as a Presence Gateway will allow Cisco Unified Presence to send SIP subscribe messages to Cisco Unified Communications Manager over the SIP trunk. Note Cisco Unified Presence does not support clients (Cisco clients or third party) connecting to Cisco Unified Presence using SIP/SIMPLE interface over TLS. Only a SIP connection over TCP is supported. Related Topics SIP Trunk Configuration on Cisco Unified Communications Manager Presence Gateway Configuration Option IM and Presence Service Features and Functions 2 IM and Presence Service Features and Functions AXL/SOAP Interface AXL/SOAP Interface The AXL/SOAP interface handles the database synchronization from Cisco Unified Communications Manager and populates the IM and Presence Service database. -
Presence Enabled Services
Presence-Enabled Services Improves communication efficiency by providing end users with the ability to control access to their availability and location Enhanced value remains the driving force behind Voice over Internet Protocol services This white paper addresses: • Applicable standards work • New presence-enabled services • Lucent’s plans to support presence-enabled services Contents Abstract .............................................................................................3 Introduction ......................................................................................3 Presence Framework .........................................................................4 Implementation of Presence-Enabled Services ...........................................9 Conclusion .......................................................................................11 Appendix..........................................................................................12 Instant Messaging and Location Services Overview..................................12 Glossary ...........................................................................................13 2 Abstract The need for enhanced value remains the driving force behind Voice over Internet Protocol (VoIP) services. Communications services should be accessible from many places – home, office and on-the-go, independent of the type of communication device deployed. Enhanced value means simplified, efficient communications and improved productivity. Lucent satisfies this need for value by -
Driveuc Mobility for Ios User Guide
Business Cloud Voice LOGIX Communicator for iOS User Guide Contents Getting Started....................................................................................................................................................... 5 Installation .......................................................................................................................................................... 5 Device Setting Options for Mobility ................................................................................................................. 5 Sign In ................................................................................................................................................................. 6 Mobility Feature Overview ................................................................................................................................ 7 Messages and Chat ............................................................................................................................................... 8 View a Chat ........................................................................................................................................................ 8 Add a Chat .......................................................................................................................................................... 8 Call from a Chat ................................................................................................................................................. 9 Group Chat -
Surround Sound Processed by Opus Codec: a Perceptual Quality Assessment
28. Konferenz Elektronische Sprachsignalverarbeitung 2017, Saarbrücken SURROUND SOUND PROCESSED BY OPUS CODEC: APERCEPTUAL QUALITY ASSESSMENT Franziska Trojahn, Martin Meszaros, Michael Maruschke and Oliver Jokisch Hochschule für Telekommunikation Leipzig, Germany [email protected] Abstract: The article describes the first perceptual quality study of 5.1 surround sound that has been processed by the Opus codec standardised by the Internet Engineering Task Force (IETF). All listening sessions with up to five subjects took place in a slightly sound absorbing laboratory – simulating living room conditions. For the assessment we conducted a Degradation Category Rating (DCR) listening- opinion test according to ITU-T P.800 recommendation with stimuli for six channels at total bitrates between 96 kbit/s and 192 kbit/s as well as hidden references. A group of 27 naive listeners compared a total of 20 sound samples. The differences between uncompressed and degraded sound samples were rated on a five-point degradation category scale resulting in Degradation Mean Opinion Score (DMOS). The overall results show that the average quality correlates with the bitrates. The quality diverges for the individual test stimuli depending on the music characteristics. Under most circumstances, a bitrate of 128 kbit/s is sufficient to achieve acceptable quality. 1 Introduction Nowadays, a high number of different speech and audio codecs are implemented in several kinds of multimedia applications; including audio / video entertainment, broadcasting and gaming. In recent years the demand for low delay and high quality audio applications, such as remote real-time jamming and cloud gaming, has been increasing. Therefore, current research objectives do not only include close to natural speech or audio quality, but also the requirements of low bitrates and a minimum latency. -
Contents Introduction
Frequently Asked Questions (FAQ) Skype for Business (Skype4B) Contents Introduction .................................................................................................................................................. 2 Planning, Setup, and Implementation .......................................................................................................... 2 What is the Enterprise Skype for Business service? ................................................................................. 2 What features are included with each of the three Enterprise Skype for Business service offerings?.... 3 What are use cases supported by the Enterprise Skype45B service? ...................................................... 5 Will Skype for Business Online (in O365) still be available? ..................................................................... 6 Do I need to be on Office 365 to use the Enterprise Skype4B service? .................................................... 6 What is the purpose of the site survey / onboarding workbook? ............................................................ 6 What is the AT&T Migration Commitment Letter? .................................................................................. 7 Do I need to utilize the ACCOUNTS domain to use the Enterprise Skype4B service? .............................. 7 Will the AT&T Hosted Skype4B service work with Network Access Control (NAC)? ................................ 7 How will the IP phones be managed in the AT&T Hosted Skype4B service? .......................................... -
Presence Service; Stage 1 (3GPP TS 22.141 Version 7.0.0 Release 7)
ETSI TS 122 141 V7.0.0 (2005-12) Technical Specification Universal Mobile Telecommunications System (UMTS); Presence service; Stage 1 (3GPP TS 22.141 version 7.0.0 Release 7) 3GPP TS 22.141 version 7.0.0 Release 7 1 ETSI TS 122 141 V7.0.0 (2005-12) Reference RTS/TSGS-0122141v700 Keywords UMTS ETSI 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE Tel.: +33 4 92 94 42 00 Fax: +33 4 93 65 47 16 Siret N° 348 623 562 00017 - NAF 742 C Association à but non lucratif enregistrée à la Sous-Préfecture de Grasse (06) N° 7803/88 Important notice Individual copies of the present document can be downloaded from: http://www.etsi.org The present document may be made available in more than one electronic version or in print. In any case of existing or perceived difference in contents between such versions, the reference version is the Portable Document Format (PDF). In case of dispute, the reference shall be the printing on ETSI printers of the PDF version kept on a specific network drive within ETSI Secretariat. Users of the present document should be aware that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at http://portal.etsi.org/tb/status/status.asp If you find errors in the present document, please send your comment to one of the following services: http://portal.etsi.org/chaircor/ETSI_support.asp Copyright Notification No part may be reproduced except as authorized by written permission. -
Tr 126 959 V15.0.0 (2018-07)
ETSI TR 126 959 V15.0.0 (2018-07) TECHNICAL REPORT 5G; Study on enhanced Voice over LTE (VoLTE) performance (3GPP TR 26.959 version 15.0.0 Release 15) 3GPP TR 26.959 version 15.0.0 Release 15 1 ETSI TR 126 959 V15.0.0 (2018-07) Reference DTR/TSGS-0426959vf00 Keywords 5G ETSI 650 Route des Lucioles F-06921 Sophia Antipolis Cedex - FRANCE Tel.: +33 4 92 94 42 00 Fax: +33 4 93 65 47 16 Siret N° 348 623 562 00017 - NAF 742 C Association à but non lucratif enregistrée à la Sous-Préfecture de Grasse (06) N° 7803/88 Important notice The present document can be downloaded from: http://www.etsi.org/standards-search The present document may be made available in electronic versions and/or in print. The content of any electronic and/or print versions of the present document shall not be modified without the prior written authorization of ETSI. In case of any existing or perceived difference in contents between such versions and/or in print, the only prevailing document is the print of the Portable Document Format (PDF) version kept on a specific network drive within ETSI Secretariat. Users of the present document should be aware that the document may be subject to revision or change of status. Information on the current status of this and other ETSI documents is available at https://portal.etsi.org/TB/ETSIDeliverableStatus.aspx If you find errors in the present document, please send your comment to one of the following services: https://portal.etsi.org/People/CommiteeSupportStaff.aspx Copyright Notification No part may be reproduced or utilized in any form or by any means, electronic or mechanical, including photocopying and microfilm except as authorized by written permission of ETSI. -
Enhanced Voice Services (EVS) Codec Management of the Institute Until Now, Telephone Services Have Generally Failed to Offer a High-Quality Audio Experience Prof
TECHNICAL PAPER Fraunhofer Institute for Integrated Circuits IIS ENHANCED VOICE SERVICES (EVS) CODEC Management of the institute Until now, telephone services have generally failed to offer a high-quality audio experience Prof. Dr.-Ing. Albert Heuberger due to limitations such as very low audio bandwidth and poor performance on non- (executive) speech signals. However, recent developments in speech and audio coding now promise a Dr.-Ing. Bernhard Grill significant quality boost in conversational services, providing the full audio bandwidth for Am Wolfsmantel 33 a more natural experience, improved speech intelligibility and listening comfort. 91058 Erlangen www.iis.fraunhofer.de The recently standardized Enhanced Voice Service (EVS) codec is the first 3GPP commu- nication codec providing super-wideband (SWB) audio bandwidth for improved speech Contact quality already at 9.6 kbps. At the same time, the codec’s performance on other signals, Matthias Rose like music or mixed content, is comparable to modern audio codecs. The key technology Phone +49 9131 776-6175 of the codec is a flexible switching scheme between specialized coding modes for speech [email protected] and music signals. The codec was jointly developed by the following companies, repre- senting operators, terminal, infrastructure and chipset vendors, as well as leading speech Contact USA and audio coding experts: Ericsson, Fraunhofer IIS, Huawei Technologies Co. Ltd, NOKIA Fraunhofer USA, Inc. Corporation, NTT, NTT DOCOMO INC., ORANGE, Panasonic Corporation, Qualcomm Digital Media Technologies* Incorporated, Samsung Electronics Co. Ltd, VoiceAge and ZTE Corporation. Phone +1 408 573 9900 [email protected] The objective of this paper is to provide a brief overview of the landscape of commu- nication systems with special focus on the EVS codec. -
Presence and IM: Reduce Distractions and Increase Productivity
Presence and IM: Reduce Distractions and Increase Productivity Table of Contents 1. Executive Summary Communications and collaboration via Unified Communications (UC) has 1. Executive Summary .............. 1 become a major tool in fostering greater productivity in the enterprise. 2. Why Care About Presence Leveraging near real-time communications such as Presence technology and and IM ....................................... 2 Instant Messaging (IM) is a goal of most enterprise users. In addition, social networks are invading the enterprise and cannot be ignored. Enterprises will 3. Avaya Presence and IM need to embrace social networks to help customers and employees Strategy .................................... 4 communicate and to improve contact center satisfaction. 4. Avaya Presence and IM The communications industry is moving toward integrating multiple forms of Product Overview ................. 5 media into one cohesive, flexible platform that a combination of disparate platforms cannot deliver. Two components of UC capabilities in this movement 6. The Avaya Vision ................... 9 are rich Presence information and Instant Messaging (IM). Both, of which, are fundamental components of the Avaya Aura® architecture and the broader UC framework. Avaya’s vision is to integrate and interoperate with other vendor’s communications capabilities for Presence information and IM outside of the enterprise network. The Avaya Aura® platform is designed to integrate with third party platforms, not to block their use. The average user has two or more communications devices and networks operating simultaneously. Presence information can be delivered about a person, organization, knowledge expert or an enterprise function. Presence and identity are the glue that connects people and collaborative experiences to drive accelerated business processes. Avaya Presence technology aggregates these devices and networks into an efficient enterprise communications solution. -
A Survey of Open Source Products for Building a SIP Communication Platform
Hindawi Publishing Corporation Advances in Multimedia Volume 2011, Article ID 372591, 21 pages doi:10.1155/2011/372591 Research Article A Survey of Open Source Products for Building a SIP Communication Platform Pavel Segec and Tatiana Kovacikova Department of InfoCom Networks, University of Zilina, Univerzitna 8215/1, 010 26 Zilina, Slovakia Correspondence should be addressed to Tatiana Kovacikova, [email protected] Received 29 July 2011; Revised 31 October 2011; Accepted 15 November 2011 Academic Editor: T. Turletti Copyright © 2011 P. Segec and T. Kovacikova. This is an open access article distributed under the Creative Commons Attribution License, which permits unrestricted use, distribution, and reproduction in any medium, provided the original work is properly cited. The Session Initiation Protocol (SIP) is a multimedia signalling protocol that has evolved into a widely adopted communication standard. The integration of SIP into existing IP networks has fostered IP networks becoming a convergence platform for both real- time and non-real-time multimedia communications. This converged platform integrates data, voice, video, presence, messaging, and conference services into a single network that offers new communication experiences for users. The open source community has contributed to SIP adoption through the development of open source software for both SIP clients and servers. In this paper, we provide a survey on open SIP systems that can be built using publically available software. We identify SIP features for service deve- lopment and programming, services and applications of a SIP-converged platform, and the most important technologies support- ing SIP functionalities. We propose an advanced converged IP communication platform that uses SIP for service delivery. -
Parametric Coding for Spatial Audio
Master Thesis : Parametric Coding for Spatial Audio Author : Bertrand Fatus Supervisor at Orange : St´ephane Ragot Supervisor at KTH : Sten Ternstr¨om July-December 2015 1 2 Abstract This thesis presents a stereo coding technique used as an extension for the Enhanced Voice Services (EVS) codec [10] [8]. EVS is an audio codec recently standardized by the 3rd Generation Partnership Project (3GPP) for compressing mono signals at chosen rates from 7.2 to 128 kbit/s (for fixed bit rate) and around 5.9 kbit/s (for variable bit rate). The main goal of the thesis is to present the architecture of a parametric stereo codec and how the stereo extension of EVS may be built. Parametric stereo coding relies on the transmission of a downmixed signal, sum of left and right channels, and the necessary audible cues to synthesize back the stereo image from it at the decoding end. The codec has been implemented in MATLAB with use of the existing EVS codec. An important part of the thesis is dedicated to the description of the implementation of a robust downmixing technique. The remaining parts present the parametric coding architecture that has been adapted and used to develop the EVS stereo extension at 24.4 and 32 kbit/s and other open researches that have been conducted for more specific situ- ations such as spatial coding for stereo or binaural applications. Whereas the downmixing algorithm quality has been confronted to subjective testing and proven to be more efficient than any other existing techniques, the stereo extension has been tested less extensively.