Session Management in Advanced Telecommunication Services - Anand Ranganathan, Dipanjan Chakraborty
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Cisco Unified Communications Manager System Guide, Release 9.1(1) First Published: December 20, 2012 Last Modified: September 08, 2015
Cisco Unified Communications Manager System Guide, Release 9.1(1) First Published: December 20, 2012 Last Modified: September 08, 2015 Americas Headquarters Cisco Systems, Inc. 170 West Tasman Drive San Jose, CA 95134-1706 USA http://www.cisco.com Tel: 408 526-4000 800 553-NETS (6387) Fax: 408 527-0883 Text Part Number: OL-27946-01 THE SPECIFICATIONS AND INFORMATION REGARDING THE PRODUCTS IN THIS MANUAL ARE SUBJECT TO CHANGE WITHOUT NOTICE. ALL STATEMENTS, INFORMATION, AND RECOMMENDATIONS IN THIS MANUAL ARE BELIEVED TO BE ACCURATE BUT ARE PRESENTED WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED. USERS MUST TAKE FULL RESPONSIBILITY FOR THEIR APPLICATION OF ANY PRODUCTS. THE SOFTWARE LICENSE AND LIMITED WARRANTY FOR THE ACCOMPANYING PRODUCT ARE SET FORTH IN THE INFORMATION PACKET THAT SHIPPED WITH THE PRODUCT AND ARE INCORPORATED HEREIN BY THIS REFERENCE. IF YOU ARE UNABLE TO LOCATE THE SOFTWARE LICENSE OR LIMITED WARRANTY, CONTACT YOUR CISCO REPRESENTATIVE FOR A COPY. The Cisco implementation of TCP header compression is an adaptation of a program developed by the University of California, Berkeley (UCB) as part of UCB's public domain version of the UNIX operating system. All rights reserved. Copyright © 1981, Regents of the University of California. NOTWITHSTANDING ANY OTHER WARRANTY HEREIN, ALL DOCUMENT FILES AND SOFTWARE OF THESE SUPPLIERS ARE PROVIDED “AS IS" WITH ALL FAULTS. CISCO AND THE ABOVE-NAMED SUPPLIERS DISCLAIM ALL WARRANTIES, EXPRESSED OR IMPLIED, INCLUDING, WITHOUT LIMITATION, THOSE OF MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT OR ARISING FROM A COURSE OF DEALING, USAGE, OR TRADE PRACTICE. IN NO EVENT SHALL CISCO OR ITS SUPPLIERS BE LIABLE FOR ANY INDIRECT, SPECIAL, CONSEQUENTIAL, OR INCIDENTAL DAMAGES, INCLUDING, WITHOUT LIMITATION, LOST PROFITS OR LOSS OR DAMAGE TO DATA ARISING OUT OF THE USE OR INABILITY TO USE THIS MANUAL, EVEN IF CISCO OR ITS SUPPLIERS HAVE BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGES. -
Long Distance Voice Services Introduction
LONG DISTANCE VOICE SERVICES 1. GENERAL 1.1 Service Definition 1.2 Platforms 1.3 Standard Service Features 1.4 Descriptions of Features and Feature Packages 2. AVAILABLE VERSIONS 2.1 Interstate Services 2.2 International Services 2.3 LD Virtual VoIP Service 2.4 Switched Digital Services 2.5 Intrastate Inbound and Outbound Service 3. SUPPLEMENTAL TERMS 4. FINANCIAL TERMS 5. DEFINITIONS 1. GENERAL 1.1 Service Definition. Long Distance Voice Services enable Customer to make telephone calls beyond their local calling area. Verizon offers several types of Long Distance Voice Services, including Interstate, International, LD Virtual VoIP, Switched Digital, and Intrastate. Customers choose one of four Feature Options, which defines the Base Features included with that option and optional features. Feature Option A (formerly Feature Option 1). Feature Option A (Option A) offers inbound and outbound service. Feature Option B (formerly Feature Option 2). Feature Option B (Option B) offers inbound and outbound service. Feature Option C-1 (formerly Feature Option 3A). Feature Option C-1 (Option C-1) offers outbound service only and is characterized by a private dialing plan. Feature Option C-2 (formerly Feature Option 3B). Feature Option C-2 (Option C-2) offers inbound toll free service only. 1.2 Platforms. These terms apply to non-Optimized Long Distance Voice Services only. 1.3 Standard Service Features. Customers receive the Base Features associated with the Feature Options Customer selects, as reflected in the Feature Availability Table below. Feature Options may be available either on a stand-alone basis or as part of a combined feature package. -
Telephone Switching
NATIONAL UNIVERSITY OF ENGINEERING COLLEGE OF ELECTRICAL AND ELECTRONICS ENGINEERING TELECOMMUNICATIONS ENGINEERING PROGRAM IT535 – TELEPHONE SWITCHING I. GENERAL INFORMATION CODE : IT535 – Telephone Switching SEMESTER : 9 CREDITS : 03 HOURS PER WEEK : 04 (Theory – Practice) PREREQUISITES : IT515 – Telecommunications III CONDITION : Mandatory II. COURSE DESCRIPTION The purpose of this course is to provide the student with the knowledge about the evolution and development of the telephony for the analog and digital switching systems, including the hardware and software of different telephone systems, integration of technologies such as ATM and IP including Voice over IP (VoIP) and photonic switching technologies. III. COURSE OUTCOMES At the end of the course the student will: Know the criteria for the design of analog and digital telephone switching systems. Know the operation and use of software and hardware used in different telephone systems. Know how to integrate technologies such as ATM and IP over classic telephone systems. IV. LEARNING UNITS 1. INTRODUCTION Communication channels. Switching channels. Networks of POTS (Plain Old Telephone Service). The Telecommunications Industry. Evolution of technology, Evolution of architecture. Evolution of telephone systems. 2. LINES AND TRUNKS Lines. The telephone. Subscriber signaling. Telephone exchange. Subscriber extension Functions per line. Trunks: Network hierarchy between telephone switching centers. Trunks Trunk circuits. Signaling between telephone switching centers. 3. TRAFFIC ENGINEERING Traffic measurements. Network management. Quality of telephone service. Telephone demand projections. Routing plan. Interconnection of telephone switching centers. 4. PUBLIC AND PRIVATE ANALOG SWITCHING Analog switching Architecture. Line Finders. Selectors. Crossbar. Block of lines Trunk blocks. Call progression, call routing. 1 5. PUBLIC AND PRIVATE DIGITAL SWITCHING SYSTEMS Concepts of pulse code modulation (PCM). -
Data Communications Via Powerlines I I (B) (3)-P.L
UNCLASSIFIED Cryptologic Quarterly Data Communications Via Powerlines I I (b) (3)-P.L. 86-36 The author is a member ofNSA Cohort 11 at bine, such as in nuclear- or coal-powered electric the Joint Military Intelligence College. Many of power plants, or a low-speed turbine, such as is the ideas presented in this paper were developed used in hydroelectric power plants). The power is as a class research paper at the Joint Military transferred to the transmission system via a volt Intelligence College. age step-up transformer.3 Typical voltages in this The views expressed in this paper are those of stage range from 138 kV to 500 kV or more. Bulk the author and do not reflect the official policy power is delivered from the generating plants via or position of the Department of Defense or the this intercity transmission system (which can U.S. government. span several states) to the transmission substa tions where the power is transferred to a sub The hunger for increased bandwidth is driv transmission system whose voltages range from ing individuals, corporations, and organizations 38 kV to 138 kV; power transference is made via to seek new methods for delivering Internet serv a step-down transformer. The subtransmission ice to customers. Many of these methods are well system delivers the high voltage throughout a city known: radio-frequency (or wireless) communi or large region. Power is delivered to the con cations (such as the IEEE 802.11 Wireless LAN, sumers via the distribution system. Transference Bluetooth, and the HomeRF and SWAP from the subtransmission system to the distribu Protocols), infrared communications (IrDA), tion system is made within regions called distri fiber-optic channels, high-speed telephone con bution substations, likewise using step-down nections (such as DSL and ISDN or the more transformers. -
Mediatrix G7 Series Datasheet
ediatrix G7 Series The Mediatrix G7 Series is a reliable and secure VoIP Analog Adaptor and Media Gateway platform for SMBs. Featuring PRI, FXS, and FXO interfaces; the Mediatrix G7 Series provides the best solution to connect legacy equipmentedia to cloud telephonytrix services and IP PBX systems to PSTN landlines. Widely interoperable with SIP softswitch and IMS vendors, the Mediatrix G7 Series provides transparent integration of legacy PBX systems for SIP Trunking and PSTN replacement applications. Interconnects any device to SIP Highly reliable Fax and Modem Transmissions over IP The Mediatrix G7 Series links any analog or digital With T.38 and clear channel fax and modem pass-through connection to an IP network and delivers a rich capabilities, the Mediatrix G7 Series ensures seamless feature set for a comprehensive VoIP solution. transport of voice and data services over IP networks. PSTN access and legacy PBX system gateway Advanced Mass Management With FXS, FXO, configurable NT/TE PRI ports, local Our advanced provisioning capabilities deliver call switching, and user-defined call properties remarkable benefits to Mediatrix customers. (including caller/calling ID), Mediatrix G7 Series Mediatrix enables centralised CPE management, a gateways smoothly integrate into legacy PBXs and definite advantage to monitor the network, ensure incumbent PSTN networks. service, and reduce operational costs. ediatrix G7 Series Applicationsediatrix Operators ✓ Connect legacy equipment in PSTN replacement/TDM replacement projects ✓ Provide SIP termination -
Multifrequency Compelled Signaling Fundamentals Avaya Communication Server 1000
Multifrequency Compelled Signaling Fundamentals Avaya Communication Server 1000 7.5 NN43001-284, 05.02 November 2011 © 2011 Avaya Inc. Copyright All Rights Reserved. Except where expressly stated otherwise, no use should be made of materials on this site, the Documentation, Software, or Hardware Notice provided by Avaya. All content on this site, the documentation and the Product provided by Avaya including the selection, arrangement and While reasonable efforts have been made to ensure that the design of the content is owned either by Avaya or its licensors and is information in this document is complete and accurate at the time of protected by copyright and other intellectual property laws including the printing, Avaya assumes no liability for any errors. Avaya reserves the sui generis rights relating to the protection of databases. You may not right to make changes and corrections to the information in this modify, copy, reproduce, republish, upload, post, transmit or distribute document without the obligation to notify any person or organization of in any way any content, in whole or in part, including any code and such changes. software unless expressly authorized by Avaya. Unauthorized reproduction, transmission, dissemination, storage, and or use without Documentation disclaimer the express written consent of Avaya can be a criminal, as well as a “Documentation” means information published by Avaya in varying civil offense under the applicable law. mediums which may include product information, operating instructions and performance specifications that Avaya generally makes available Third-party components to users of its products. Documentation does not include marketing Certain software programs or portions thereof included in the Product materials. -
Global Call E1/T1 CAS/R2 Technology Guide
Global Call E1/T1 CAS/R2 Technology Guide July 2005 05-2445-001 INFORMATION IN THIS DOCUMENT IS PROVIDED IN CONNECTION WITH INTEL® PRODUCTS. NO LICENSE, EXPRESS OR IMPLIED, BY ESTOPPEL OR OTHERWISE, TO ANY INTELLECTUAL PROPERTY RIGHTS IS GRANTED BY THIS DOCUMENT. EXCEPT AS PROVIDED IN INTEL'S TERMS AND CONDITIONS OF SALE FOR SUCH PRODUCTS, INTEL ASSUMES NO LIABILITY WHATSOEVER, AND INTEL DISCLAIMS ANY EXPRESS OR IMPLIED WARRANTY, RELATING TO SALE AND/OR USE OF INTEL PRODUCTS INCLUDING LIABILITY OR WARRANTIES RELATING TO FITNESS FOR A PARTICULAR PURPOSE, MERCHANTABILITY, OR INFRINGEMENT OF ANY PATENT, COPYRIGHT OR OTHER INTELLECTUAL PROPERTY RIGHT. Intel products are not intended for use in medical, life saving, or life sustaining applications. Intel may make changes to specifications and product descriptions at any time, without notice. This Global Call E1/T1 CAS/R2 Technology Guide as well as the software described in it is furnished under license and may only be used or copied in accordance with the terms of the license. The information in this manual is furnished for informational use only, is subject to change without notice, and should not be construed as a commitment by Intel Corporation. Intel Corporation assumes no responsibility or liability for any errors or inaccuracies that may appear in this document or any software that may be provided in association with this document. Except as permitted by such license, no part of this document may be reproduced, stored in a retrieval system, or transmitted in any form or by any means without express written consent of Intel Corporation. -
On Common Channel Signaling Number 7 and Its Comparison with Multi-Frequency Coded Signaling and Internet Protocol
International Journal of Electronics and Communication Engineering. ISSN 0974-2166 Volume 5, Number 2 (2012), pp. 125-132 © International Research Publication House http://www.irphouse.com On Common Channel Signaling Number 7 and its Comparison with Multi-Frequency Coded Signaling and Internet Protocol Md. Shah Alam1 and Md. Rezaul Huque Khan2 Dept. of Applied Physics, Electronics and Communication Engineering, University of Chittagong, Bangladesh 1 2 E-mail: [email protected], [email protected] Abstract A message based comparative study between signaling system #7(SS7) and R2 Signaling is done. The reasons for the transition from Multi-frequency Coded (MFC) Signaling to SS7 and SS7 to Internet Protocol are also discussed. Keywords: Common Channel Signaling No.7 (CCS7), R2 signaling or Multi- frequency Coded (MFC) signaling, Internet Protocol (IP), Advance Intelligent Network (AIN). Introduction The over increasing demand of telecommunication in the world wide significantly involves telecommunication signaling systems. The Common Channel Signaling no.7 is usually termed as Signaling System No.7 (SS7). The purpose of this paper is to study the signaling systems R2 or MFC, SS7 [1] and IP, to find the limitations of the above signaling system, analysis of the signaling systems (R2 and SS7) on the basis of their message format. An overall comparison between the two systems has been studied. This paper also focuses on the transition of MFC to SS7. A distinction is made between SS7 and IP and finally reasons are shown why SS7 is moving towards IP. Common Channel Signaling No. 7 Common Channel Signaling System No. 7 (i.e., SS7 or C7) is a global standard for telecommunications defined by the International Telecommunication Union (ITU) Telecommunication Standardization Sector (ITU-T). -
Design of an ISDN Central Office, U-Interface Timothy N
Iowa State University Capstones, Theses and Retrospective Theses and Dissertations Dissertations 1991 Design of an ISDN central office, U-interface Timothy N. Toillion Iowa State University Follow this and additional works at: https://lib.dr.iastate.edu/rtd Part of the Digital Circuits Commons, Digital Communications and Networking Commons, and the Signal Processing Commons Recommended Citation Toillion, Timothy N., "Design of an ISDN central office, U-interface" (1991). Retrospective Theses and Dissertations. 16968. https://lib.dr.iastate.edu/rtd/16968 This Thesis is brought to you for free and open access by the Iowa State University Capstones, Theses and Dissertations at Iowa State University Digital Repository. It has been accepted for inclusion in Retrospective Theses and Dissertations by an authorized administrator of Iowa State University Digital Repository. For more information, please contact [email protected]. Design of an ISDN central office, U-interface by Timothy N. Toillion A Thesis Submitted to the Graduate Faculty in Partial Fulfillment of the Requirements for the Degree of MASTER OF SCIENCE Department: Electrical Engineering and Computer Engineering Major: Computer Engineering Signatures have been redacted for privacy Signatures have been redacted for privacy Iowa State University Ames, Iowa 1991 Copyright @ Timothy N. Toillion, 1991. All rights reserved. 11 TABLE OF CONTENTS INTRODUCTION. .. 1 CHAPTER 1. THE EVOLUTION OF TELECOMMUNICATIONS 4 Telephone Communications .. 4 Inventions Spawn Growth 4 Modern Telecommunication Networks. 8 Telecommunications ......... 11 Definition of Telecommunication. 11 Other Forms of Telecommunication 11 Integration of Services . 12 Evolution of the Concept . 12 The Evolution of ISDN .. 15 CCITT's Standards . 15 Major Set back. 16 Evolution of Products and Services 16 CHAPTER 2. -
Omnix: an Open Peer-To-Peer Middleware Framework
Omnix: An Open Peer-to-Peer Middleware Framework Engineering Topology- and Device-Independent Peer-to-Peer Systems Ph.D. Thesis Roman Kurmanowytsch DISSERTATION Omnix: An Open Peer-to-Peer Middleware Framework Engineering Topology- and Device-Independent Peer-to-Peer Systems ausgefuhrt¨ zum Zwecke der Erlangung des akademischen Grades eines Doktors der technischen Wissenschaften unter der Leitung von o.Univ.-Prof. Dipl.-Ing. Dr.techn. Mehdi Jazayeri Institut fur¨ Informationssysteme Abteilung fur¨ Verteilte Systeme (E184-1) eingereicht an der Technischen Universitat¨ Wien Fakultat¨ fur¨ Informatik von Univ.-Ass. Dipl.-Ing. Roman Kurmanowytsch [email protected] Matrikelnummer: 9327324 Brunnerstr. 28/12 A-1230 Wien, Osterreich¨ Wien, im Februar 2004 Omnix: An Open Peer-to-Peer Middleware Framework Engineering Topology- and Device-Independent Peer-to-Peer Systems Ph.D. Thesis at Vienna University of Technology submitted by Dipl.-Ing. Roman Kurmanowytsch Distributed Systems Group, Information Systems Institute, Technical University of Vienna Argentinierstr. 8/184-1 A-1040 Vienna, Austria 1st February 2004 c Copyright 2004 by Roman Kurmanowytsch Advisor: o. Univ.-Prof. Dr. Mehdi Jazayeri Second Advisor: a.o. Univ.-Prof. Dr. Gabriele Kotsis Abstract In this thesis, we present Omnix, a Peer-to-Peer (P2P) middleware framework. P2P mid- dleware provides an abstraction between the application and the underlying network by pro- viding higher-level functionality such as distributed P2P searches and direct communication among peers. P2P networks build an overlay network, a network on the network. The central idea of this dissertation is to apply the well-established ISO OSI layered model on P2P middleware systems. A layered P2P architecture helps to identify separate parts of a complex system, allows changes to the system without affecting other layers and makes maintenance easier due to modularization. -
2Wire Analog Bulk Call Generator Presentation
MAPS™ APS and ALS Analog Phone/ Line Simulator 818 West Diamond Avenue - Third Floor, Gaithersburg, MD 20878 Phone: (301) 670-4784 Fax: (301) 670-9187 Email: [email protected] Website: https://www.gl.com 1 MAPS™ Analog Phone Emulator 2 Main Features • Up to 192 independent FXO ports per 1U MAPS™ APS (More can be achieved by scaling) • Test Central Office, PBX, Gateway, Analog/Digital/VoIP Networks • Manual and Automated Bulk Analog call generation • Call monitoring and call recording • Multiple users and tests per system • Fully Automated with CLI and external control • Supports E&M (Type I, II, III, IV, V) signaling – immediate start, wink start, delay start • Full FXO Functionality via flexible scripts • Scalable to support up to 1000s of calls • Supports Interactive Voice Response (IVR) using Speech Transcription Server • Voiceband Measurement Tests using VF Ports 3 Functional Specifications FXO capabilities FXS capabilities • Up to 96 independent FXO ports per 1U MAPS™ APS • Up to 96 independent FXS ports per 1U MAPS™ APS (More can be achieved by scaling) (more can be achieved by scaling, requires FXS voice • Supports Loop Start and Ground Start signaling cards) • Full FXO Functionality via flexible scripts • Central office simulation with two-way calling • Supported call scenarios • Supports Loop Start and Ground Start signaling ➢ Caller ID • User-programmable call progress tone generation for different countries/regions: ➢ Two-way Calling ➢ Dial tone ➢ Three-way Conference Calling ➢ Ringback tone ➢ Three-way Calling with Calling Party -
Voice/Fax Over IP Networks Model MVP400 H.323 Mode User Guide
Voice/Fax Over IP Networks Model MVP400 H.323 Mode User Guide User Guide S0000008 Revision C MultiVOIP 400 (Model MVP400) This publication may not be reproduced, in whole or in part, without prior expressed written permission from Multi- Tech Systems, Inc. All rights reserved. Copyright © 2001 by Multi-Tech Systems, Inc. Multi-Tech Systems, Inc. makes no representations or warranties with respect to the contents hereof and specifically disclaims any implied warranties of merchantability or fitness for any particular purpose. Furthermore, Multi-Tech Systems, Inc. reserves the right to revise this publication and to make changes from time to time in the content hereof without obligation of Multi-Tech Systems, Inc. to notify any person or organization of such revisions or changes. Record of Revisions Revision Description A Added H.323 protocol support; covers software version 3.50. All pages at revision A. (6/5/00) B Manual updated for software version 3.51. (1/24/01) C Removed references to MultiVOIP 800 (12/26/01) Patents This Product is covered by the following U.S. Patent Numbers: 6151333, 5757801, 5682386. Other Patents Pending. TRADEMARK Multi-Tech and the Multi-Tech logo are registered trademarks and MultiVOIP is a trademark of Multi-Tech Systems, Inc. Adobe Acrobat is a trademark of Adobe Systems Incorporated. Microsoft Windows, Windows 2000, Windows 98, Windows 95, Windows NT, and NetMeeting are either registered trademarks or trademarks of Microsoft Corporation in the United States and/or other countries. Multi-Tech Systems, Inc. 2205 Woodale Drive Mounds View, Minnesota 55112 (763) 785-3500 or (800) 328-9717 Fax 763-785-9874 Technical Support (800) 972-2439 Internet Address: http://www.multitech.com Contents Chapter 1 - Introduction and Description ....................................................