Cisco IP Phone Portfolio Brochure
Total Page:16
File Type:pdf, Size:1020Kb
Load more
Recommended publications
-
Title: Communicating with Light: from Telephony to Cell Phones Revision
Title: Communicating with Light: From Telephony to Cell Phones Revision: February 1, 2006 Authors: Jim Overhiser, Luat Vuong Appropriate Physics, Grades 9-12 Level: Abstract: This series of six station activities introduces the physics of transmitting "voice" information using electromagnetic signals or light. Students explore how light can be modulated to encode voice information using a simple version of Bell's original photophone. They observe the decrease of the intensity of open-air signals by increasing the distance between source and receiver, and learn the advantage of using materials with different indices of refraction to manipulate and guide light signals. Finally, students are introduced to the concept of bandwidth by using two different wavelengths of light to send two signals at the same time. Special Kit available on loan from CIPT lending library. Equipment: Time Required: Two 80-minute periods NY Standards 4.1b Energy may be converted among mechanical, electromagnetic, Met: nuclear, and thermal forms 4.1j Energy may be stored in electric or magnetic fields. This energy may be transferred through conductors or space and may be converted to other forms of energy. 4.3b Waves carry energy and information without transferring mass. This energy may be carried by pulses or periodic waves. 4.3i When a wave moves from one medium into another, the waves may refract due a change in speed. The angle of refraction depends on the angle of incidence and the property of the medium. 4.3h When a wave strikes a boundary between two media, reflection, transmission, and absorption occur. A transmitted wave may be refracted. -
IP Telephony Fundamentals: What You Need to Know to “Go to Market”
IP Telephony Fundamentals: What You Need to Know to “Go To Market” Ken Agress Senior Consultant PlanNet Consulting What Will Be Covered • What is Voice over IP? • VoIP Technology Basics • How Do I Know if We’re Ready? • What “Real” Cost Savings Should I Expect? • Putting it All Together • Conclusion, Q&A 2 What is Voice Over IP? • The Simple Answer – It’s your “traditional” voice services transported across a common IP infrastructure. • The Real Answer – It’s the convergence of numerous protocols, components, and requirements that must be balanced to provide a quality voice experience. 3 Recognize the Reality of IP Telephony • IP is the catalyst for convergence of technology and organizations • There are few plan templates for convergence projects • Everybody seems to have a strong opinion • Requires an educational investment in the technology (learning curve) – Requires an up-front investment in the technology that can be leveraged for subsequent deployments • Surveys indicate deployment is usually more difficult than anticipated • Most implementations are event driven (that means there is a broader plan) 4 IP Telephony vs. VoIP • Voice over IP – A broad technology that encompasses many, many facets. • IP Telephony – What you’re going to implement to actually deliver services across your network – Focuses more on features than possibilities – Narrows focus to specific implementations and requirements – Sets appropriate context for discussions 5 Why Does Convergence Matter? • Converged networks provide a means to simplify support structures and staffing. • Converged networks create new opportunities for a “richer” communications environment – Improved Unified Messaging – Unified Communications – The Promise of Video • Converged networks provide methods to reduce costs (if you do things right) 6 The Basics – TDM (vs. -
What Is the Impact of Mobile Telephony on Economic Growth?
What is the impact of mobile telephony on economic growth? A Report for the GSM Association November 2012 Contents Foreword 1 The impact of mobile telephony on economic growth: key findings 2 What is the impact of mobile telephony on economic growth? 3 Appendix A 3G penetration and economic growth 11 Appendix B Mobile data usage and economic growth 16 Appendix C Mobile telephony and productivity in developing markets 20 Important Notice from Deloitte This report (the “Report”) has been prepared by Deloitte LLP (“Deloitte”) for the GSM Association (‘GSMA’) in accordance with the contract with them dated July 1st 2011 plus two change orders dated October 3rd 2011 and March 26th 2012 (“the Contract”) and on the basis of the scope and limitations set out below. The Report has been prepared solely for the purposes of assessing the impact of mobile services on GDP growth and productivity, as set out in the Contract. It should not be used for any other purpose or in any other context, and Deloitte accepts no responsibility for its use in either regard. The Report is provided exclusively for the GSMA’s use under the terms of the Contract. No party other than the GSMA is entitled to rely on the Report for any purpose whatsoever and Deloitte accepts no responsibility or liability or duty of care to any party other than the GSMA in respect of the Report or any of its contents. As set out in the Contract, the scope of our work has been limited by the time, information and explanations made available to us. -
ATA User's Manual
VoIP Analog Telephone Adapter VIP-156 VIP-157 User’s manual 1 Copyright Copyright (C) 2006 PLANET Technology Corp. All rights reserved. The products and programs described in this User’s Manual are licensed products of PLANET Technology, This User’s Manual contains proprietary information protected by copyright, and this User’s Manual and all accompanying hardware, software, and documentation are copyrighted. No part of this User’s Manual may be copied, photocopied, reproduced, translated, or reduced to any electronic medium or machine-readable form by any means by electronic or mechanical. Including photocopying, recording, or information storage and retrieval systems, for any purpose other than the purchaser's personal use, and without the prior express written permission of PLANET Technology. Disclaimer PLANET Technology does not warrant that the hardware will work properly in all environments and applications, and makes no warranty and representation, either implied or expressed, with respect to the quality, performance, merchantability, or fitness for a particular purpose. PLANET has made every effort to ensure that this User’s Manual is accurate; PLANET disclaims liability for any inaccuracies or omissions that may have occurred. Information in this User’s Manual is subject to change without notice and does not represent a commitment on the part of PLANET. PLANET assumes no responsibility for any inaccuracies that may be contained in this User’s Manual. PLANET makes no commitment to update or keep current the information in this User’s Manual, and reserves the right to make improvements to this User’s Manual and/or to the products described in this User’s Manual, at any time without notice. -
Cisco ATA 192 Multiplatform Analog Telephone Adapter Data Sheet
Data Sheet Cisco ATA 192 Multiplatform Analog Telephone Adapter The Cisco® ATA 192 Multiplatform Analog Telephone Adapter is a 2-port handset-to- Ethernet adapter that brings traditional analog devices into the IP world. Product Overview The Cisco ATA 192 Multiplatform Analog Telephone Adapter turns traditional telephone, fax, and overhead paging communications devices into IP devices for greater cost-effectiveness. Customers can take advantage of IP telephony applications by connecting their analog devices to Cisco analog telephone adapters. The ATA 192 is the preferred solution to address the needs of customers who connect to enterprise networks, small offices, or unified communications as a service from the cloud. It has two standard FXS ports, which can be configured independently as two Session Initiation Protocol (SIP) registrations. It also has two 100BASE-T ports with an integrated high-performance router to extend local network connectivity. With the ATA 192, customers can protect and extend their existing investment in analog systems, as well as smooth their migration to pure voice over IP in a more affordable and reliable way. The ATA 192 is designed to work with third-party call control systems and does not work with Cisco call control systems. Features and Benefits Feature Benefit Voice quality Offers clear, natural-sounding voice quality via advanced preprocessing, high-performance echo cancellation, voice activity detection, and comfort noise generation Cloud provisioning Enables zero-touch provisioning via TR-069 and XML configuration files Security Provides a complete security solution for both media and signaling Problem reporting (PRT) Improves serviceability with a dedicated PRT button for problem reporting and log collection IPv6 Enables IPv6 dual stack to help with migration to IPv6 Platform Support Information The Cisco ATA 192 Multiplatform Analog Telephone Adapter is designed to work with third-party call control systems. -
SIP Analog Telephone Adapter Key Features
VIP-156 SIP Analog Telephone Adapter Key Features PRODUCT FEATURES • Feature-rich telephone service over home or office Internet / Intranet connection • Web-Based and telephone keypad machine configuration • Remote administrator authentication • Easy access with PLANET Dynamic DNS • Voice prompt for machine configurations • Cost effective, field proven compatibility and stability Based on years of VoIP manufacturing experiences, PLANET Technology VoIP total solutions are known for advanced implementation of standard VOIP FEATURES based telephony with mass deployment capability. • SIP 2.0 (RFC3261) compliant • Peer-to-Peer / SIP proxy calls Cost-effective, easy-to-install and simple-to-use, the VIP-156 converts • Supports STUN, Outbound Proxy standard telephones to IP-based network communication. It supports • Up to 3 Proxy server registrations the service providers and enterprises enhance the traditional telephony communication services via the existing broadband connection to the • T.38 FAX transmission over IP network Internet or corporation network. • Call Hold / Forward / Transfer / Waiting • Incoming / Outgoing / Miss Call record With the VIP-156, home users and companies are able to save the • Local Phone book (Export / Upload) installation cost of voice-over-IP deployment and extend their past investments in telephones, conference and speakerphones. The VIP-156 • In band and out-of-band DTMF support can be the bridge between traditional analog systems and IP network • Voice codec support: G.711, G.723.1, G.729A, G.729B with an extremely affordable investment. • Voice processing: Voice Active Detection, DTMF detection / generation, G.168 echo cancellation (16mSec.), Comfort noise generation The VIP-156 and PLANET IP PBX system integration is the ideal application to your office daily communications. -
HT813 FXS/FXO Port Analog Telephone Adaptors User Guide
Grandstream Networks, Inc. HT813 FXS/FXO Port Analog Telephone Adaptors User Guide COPYRIGHT ©2021 Grandstream Networks, Inc. https://www.grandstream.com/ All rights reserved. Information in this document is subject to change without notice. Reproduction or transmittal of the entire or any part, in any form or by any means, electronic or print, for any purpose without the express written permission of Grandstream Networks, Inc. is not permitted. The latest electronic version of this user manual is available for download here: https://www.grandstream.com/support Grandstream is a registered trademark and Grandstream logo is trademark of Grandstream Networks, Inc. in the United States, Europe and other countries. CAUTION Changes or modifications to this product not expressly approved by Grandstream, or operation of this product in any way other than as detailed by this User Manual, could void your manufacturer warranty. WARNING Please do not use a different power adaptor with your devices as it may cause damage to the products and void the manufacturer warranty. P a g e | 1 HT813 User Guide Version 1.0.13.3 GNU GPL INFORMATION The firmware for the HT813 contains third-party software licensed under the GNU General Public License (GPL). Grandstream uses software under the specific terms of the GPL. Please see the GNU General Public License (GPL) for the exact terms and conditions of the license. Grandstream GNU GPL related source code can be downloaded from Grandstream web site from: https://blog.grandstream.com/faq/gnu-general-public-license/gnu-gpl-information-download P a g e | 2 HT813 User Guide Version 1.0.13.3 Table of Content DOCUMENT PURPOSE ................................................................................................ -
NENA Master Glossary of 9-1-1 Terminology
NENA Master Glossary Of 9-1-1 Terminology NENA Master Glossary of 9-1-1 Terminology NENA 00-001, Version 16, August 22, 2011 Prepared by: National Emergency Number Association (NENA) Committee Chairs Published by NENA Printed in USA NENA Master Glossary of 9-1-1 Terminology NENA 00-001, Version 16, August 21, 2011 NENA STANDARDS NOTICE This NENA STANDARD is published by National Emergency Number Association (NENA) as a guide for the designers and manufacturers of systems that are used for the purpose of processing emergency calls. It is not intended to provide complete design specifications or parameters nor to assure the quality of performance of such equipment. NENA reserves the right to revise this NENA STANDARD for any reason including, but not limited to, conformity with criteria or standards promulgated by various agencies, utilization of advances in the state of the technical and operational arts or to reflect changes in the design of equipment or services described herein. It is possible that certain advances in technology or operations will precede these revisions. Therefore, this NENA STANDARD should not be the only source of information used. NENA members are advised to contact their Telecommunications Carrier representative to ensure compatibility with the 9-1-1 network. Patents may cover the specifications, techniques or network interface/system characteristics disclosed herein. No license expressed or implied is hereby granted. This document is not to be construed as a suggestion to any manufacturer to modify or change any of its products, nor does this document represent any commitment by NENA or any affiliate thereof to purchase any product whether or not it provides the described characteristics. -
Voice Over Internet Protocol (VOIP): Overview, Direction and Challenges 1 U
View metadata, citation and similar papers at core.ac.uk brought to you by CORE provided by International Institute for Science, Technology and Education (IISTE): E-Journals Journal of Information Engineering and Applications www.iiste.org ISSN 2224-5782 (print) ISSN 2225-0506 (online) Vol.3, No.4, 2013 Voice over Internet Protocol (VOIP): Overview, Direction And Challenges 1 U. R. ALO and 2 NWEKE HENRY FIRDAY Department of Computer Science Ebonyi State University Abakaliki, Nigeria 1Email:- [email protected] 2Email: [email protected] ABSTRACT Voice will remain a fundamental communication media that cuts across people of all walks of life. It is therefore important to make it cheap and affordable. To be reliable and affordable over the common Public Switched Telephone Network, change is therefore inevitable to keep abreast with the global technological change. It is on this basis that this paper tends to critically review this new technology VoIP, x-raying the different types. It further more discusses in detail the VoIP system, VoIP protocols, and a comparison of different VoIP protocols. The compression algorithm used to save network bandwidth in VoIP, advantages of VoIP and problems associated with VoIP implementation were also critically examined. It equally discussed the trend in VoIP security and Quality of Service challenges. It concludes by reiterating the need for a cheap, reliable and affordable means of communication that would not only maximize cost but keep abreast with the global technological change. Keywords: Voice over Internet Protocol (VoIP), Public Switched Telephone Network (PSTN), Session Initiation Protocol (SIP), multipoint control unit 1. Introduction Voice over Internet Protocol (VoIP) is a technology that makes it possible for users to make telephone calls over the internet or intranet networks. -
Telecommunications Electronics Technician Competency Listing
Telecommunications Electronics Technician Competency Listing Telecommunications Electronics Technician - TCM Competency Requirements Telecommunications electronics technicians are expected to obtain knowledge of wired and wireless communications basic concepts which are then applicable to various types of voice, data and video systems. Once the CET has acquired these skills, abilities and knowledge, he or she will be able to enter employment in any part of the telecommunications field. With minimal training in areas unique to specific products, the CET should become a profitable and efficient part of the electronics-communications workforce. Telecommunications Electronics Technicians must be knowledgeable and have abilities in the following technical areas: 1.0 CABLES AND CABLING 1.1 Describe unshielded twisted pair (UTP) - List common usage locations and capabilities 1.2 Demonstrate installation and troubleshooting of RJ45/48 telephone connectors and fittings 1.3 CAT 5 wiring—Explain the differences vs.: single twisted pair and where it is most used 1.4 10 base T-explain where it is commonly used and its frequency capabilities 1.5 Describe the T568A / T568B standards 1.6 Explain how Cable TV wiring is used for data and voice services 1.7 Explain the differences between coax types RG 58, RG 59 and RG 6 1.8 Describe required grounding of electronics equipment 1.10 Describe the differences in Single and Multi-mode fiber optics 2.0 ANALOG TELEPHONY 2.1. Give a brief history of the telephone industry 2.2 Explain how basic phone systems work 2.3 Define POTS, DID, OPX, tie lines and WATS lines 2.4 Explain the benefits and usage of multiple phone lines 2.5 Define PBX and explain basic switching method 2.6 Sketch a local loop map 2.7 Define Key service units 2.8 Define Central Office and list it purposes 2.9 Explain the terms and usage of tones, loop start, ground start and wink start 2.10 Define CO, CPE. -
An Easy-To-Use 1 Port
An easy-to-use 1 port ATA HT801 The HT801 is a single port analog telephone adapter (ATA) that allows users to create a high-quality and manageable IP telephony solution for residential and office environments. Its ultra-compact size, voice quality, advanced VoIP functionality, security protection and auto provisioning options enable users to take advantage of VoIP on analog phones. It also allows service providers to offer high quality IP service to their market. The HT801 is an ideal ATA for individual use as well as commercial IP voice deployments worldwide. Supports 1 SIP profile TLS and SRTP Automated Supports 3-way voice through a single FXS security encryption provisioning options conferencing port and a single technology to include TR-069 and 10/100Mbps port protect calls and XML config files accounts Failover SIP server Supports T.38 Fax for Supports a wide Use with automatically switches creating Fax-over-IP range of caller ID Grandstream’s UCM to secondary server formats series of IP PBXs for if main server loses Zero Configuration connection provisioning Supports advanced telephony features, including call transfer, call forward, call- waiting, do not disturb, message waiting indication, multi-language prompts, flexible dial plan and more www.grandstream.com Interfaces Telephone Interfaces One (1) FXS port Network Interfaces One (1) 10/100Mbps auto-sensing ethernet port (RJ45) LED Indicators POWER, INTERNET, PHONE Factory Reset Button Yes Voice, Fax, Modem Caller ID display or block, call waiting, flash, blind or attended -
A Novel Speech Enhancement Approach Based on Modified Dct and Improved Pitch Synchronous Analysis
American Journal of Applied Sciences 11 (1): 24-37, 2014 ISSN: 1546-9239 ©2014 Science Publication doi:10.3844/ajassp.2014.24.37 Published Online 11 (1) 2014 (http://www.thescipub.com/ajas.toc) A NOVEL SPEECH ENHANCEMENT APPROACH BASED ON MODIFIED DCT AND IMPROVED PITCH SYNCHRONOUS ANALYSIS 1Balaji, V.R. and 2S. Subramanian 1Department of ECE, Sri Krishna College of Engineering and Technology, Coimbatore, India 2Department of CSE, Coimbatore Institute of Engineering and Technology, Coimbatore, India Received 2013-06-03, Revised 2013-07-17; Accepted 2013-11-21 ABSTRACT Speech enhancement has become an essential issue within the field of speech and signal processing, because of the necessity to enhance the performance of voice communication systems in noisy environment. There has been a number of research works being carried out in speech processing but still there is always room for improvement. The main aim is to enhance the apparent quality of the speech and to improve the intelligibility. Signal representation and enhancement in cosine transformation is observed to provide significant results. Discrete Cosine Transformation has been widely used for speech enhancement. In this research work, instead of DCT, Advanced DCT (ADCT) which simultaneous offers energy compaction along with critical sampling and flexible window switching. In order to deal with the issue of frame to frame deviations of the Cosine Transformations, ADCT is integrated with Pitch Synchronous Analysis (PSA). Moreover, in order to improve the noise minimization performance of the system, Improved Iterative Wiener Filtering approach called Constrained Iterative Wiener Filtering (CIWF) is used in this approach. Thus, a novel ADCT based speech enhancement using improved iterative filtering algorithm integrated with PSA is used in this approach.