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Author: Eric Coll
Publisher: Teracom Training Institute
ISBN: 198813823X
Category : Technology & Engineering
Languages : en
Pages : 67
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Book Description
SIP, Softswitches, and VoIP Call Setup This module is all about SIP, and how it is used to set up phone calls. You will understand: • What SIP is, • How it works, • Jargon like URI, proxy server and location server, and • How SIP fits in with softswitches and call managers. We'll trace the establishment of a VoIP phone call using SIP and DNS, step by step. VoIP Module 3 Detailed Outline 3. SIP, Softswitches and VoIP Call Setup 3.1 What SIP Is and What It Can Do ...... 3.1.1 The Session Initiation Protocol ...... 3.1.2 The Essential Function ...... 3.1.3 Supports Five Aspects of Communications ...... 3.1.4 SIP Servers: Proxies ...... 3.1.5 SIP Addresses 3.2 Relationship to Other Protocols ...... 3.2.1 Layer 5 of the OSI Model ...... 3.2.2 Session Description Protocol ...... 3.2.3 What SIP Doesn’t Do 3.3 SIP URIs: “Telephone Numbers” ...... 3.3.1 Uniform Resource indicators ...... 3.3.2 Address of Record ...... 3.3.3 Relating URIs to Telephones ...... 3.3.4 Not Just Telephones ...... 3.3.5 DNS Lookup ...... 3.3.6 URI Structure ...... 3.3.7 Encryption 3.4 Register: Update The Location Database ...... 3.4.1 First, Registration ...... 3.4.2 Location Database ...... 3.4.3 Call Disposition Rules and Filters 3.5 INVITE: “Dialing” and Outbound Proxy ...... 3.5.1 Dialing ...... 3.5.2 Outbound Proxy 3.6 Location Service: Finding the Far End ...... 3.6.1 Far-End Proxy ...... 3.6.2 Call Disposition ...... 3.6.3 Ringing ...... 3.6.4 Answer ...... 3.6.5 Call Type or Media Negotiation 3.7 The SIP Trapezoid ...... 3.7.1 Getting Ready To Send and Receive Calls ...... 3.7.2 Establishing a Call 3.8 SIP Messages and Session Description Protocol ...... 3.8.1 Plain Text Messages ...... 3.8.2 Message Type, Header and Payload 3.9 How SIP Relates to Softswitches and Call Managers ...... 3.9.1 SIP Software ...... 3.9.2 Authentication, Authorization and Accounting Software ...... 3.9.3 Voicemail and Other Features
Author: Eric Coll
Publisher: Teracom Training Institute
ISBN: 198813823X
Category : Technology & Engineering
Languages : en
Pages : 67
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Book Description
SIP, Softswitches, and VoIP Call Setup This module is all about SIP, and how it is used to set up phone calls. You will understand: • What SIP is, • How it works, • Jargon like URI, proxy server and location server, and • How SIP fits in with softswitches and call managers. We'll trace the establishment of a VoIP phone call using SIP and DNS, step by step. VoIP Module 3 Detailed Outline 3. SIP, Softswitches and VoIP Call Setup 3.1 What SIP Is and What It Can Do ...... 3.1.1 The Session Initiation Protocol ...... 3.1.2 The Essential Function ...... 3.1.3 Supports Five Aspects of Communications ...... 3.1.4 SIP Servers: Proxies ...... 3.1.5 SIP Addresses 3.2 Relationship to Other Protocols ...... 3.2.1 Layer 5 of the OSI Model ...... 3.2.2 Session Description Protocol ...... 3.2.3 What SIP Doesn’t Do 3.3 SIP URIs: “Telephone Numbers” ...... 3.3.1 Uniform Resource indicators ...... 3.3.2 Address of Record ...... 3.3.3 Relating URIs to Telephones ...... 3.3.4 Not Just Telephones ...... 3.3.5 DNS Lookup ...... 3.3.6 URI Structure ...... 3.3.7 Encryption 3.4 Register: Update The Location Database ...... 3.4.1 First, Registration ...... 3.4.2 Location Database ...... 3.4.3 Call Disposition Rules and Filters 3.5 INVITE: “Dialing” and Outbound Proxy ...... 3.5.1 Dialing ...... 3.5.2 Outbound Proxy 3.6 Location Service: Finding the Far End ...... 3.6.1 Far-End Proxy ...... 3.6.2 Call Disposition ...... 3.6.3 Ringing ...... 3.6.4 Answer ...... 3.6.5 Call Type or Media Negotiation 3.7 The SIP Trapezoid ...... 3.7.1 Getting Ready To Send and Receive Calls ...... 3.7.2 Establishing a Call 3.8 SIP Messages and Session Description Protocol ...... 3.8.1 Plain Text Messages ...... 3.8.2 Message Type, Header and Payload 3.9 How SIP Relates to Softswitches and Call Managers ...... 3.9.1 SIP Software ...... 3.9.2 Authentication, Authorization and Accounting Software ...... 3.9.3 Voicemail and Other Features
Author: Eric Coll
Publisher: Teracom Training Institute
ISBN: 189488793X
Category : Technology & Engineering
Languages : en
Pages : 148
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Book Description
VoIP 101 is designed for the non-engineering professional needing an overview and update, and for those new to the business needing to get up to speed quickly. VoIP 101 covers all major topics in Voice over IP – in plain English. From fundamentals to softswitches and SIP trunking, we’ll explain the jargon and the technologies, and more importantly, the underlying ideas. Based on training courses developed, refined and tuned over many years, the selection of material, its order and emphasis in VoIP 101 is proven and field-tested to bring you the knowledge of Voice over IP you need. Understanding what everything does and how it all works together allows you to be more accurate, eliminates frustration with buzzwords, and gives you confidence to make meaningful contributions. You have the added advantage of knowing what someone is talking about, even if you’re not familiar with the exact details of the variation they’re discussing. Don't be the one person in the room who isn't up to speed! Your employer (or prospective employer) will be interested to hear you have this solid foundation of knowledge that project- and product-specific knowledge can be built on, getting you up to speed quickly… then having the versatility to subsequently work on different projects. VoIP 101 has six chapters covering all aspects of Voice over IP, including all the different ways VoIP is implemented, how calls are set up with softswitches and SIP, how voice is packetized, the factors affecting sound quality, connecting to carriers and SIP trunking, and network quality with MPLS, Service Level Agreements and Class of Service: 1 Fundamentals of Voice over IP A complete introduction to everything Voice over IP. You'll learn the fundamental ideas and principles of a VoIP telephone system, the jargon and buzzwords, and how it all works together. 2 VoIP Architectures and Implementation Choices Next is a comprehensive overview of the many flavors of VoIP, comparing and contrasting the various implementation and architecture choices. 3 Softswitches, SIP, Call Setup and SIP Trunking This chapter is all about SIP, and how it is used to set up phone calls: what SIP is, how it works, jargon like proxy server and location server, how SIP fits in with softswitches and call managers, and trace the establishment of a VoIP phone call using SIP and DNS, step by step. 4 Voice Packetization, Codecs and Voice Quality The "nuts and bolts" of Voice over IP: how the voice is digitized and coded, time stamps applied with the RTP protocol, and how the result is carried in UDP, IP packets and MAC frames. 5 SIP Trunking & Carrier Connections This chapter is all about connecting to carriers to communicate VoIP phone calls, both carrier-to-carrier connections and business-to-carrier SIP trunking. 6 IP Network Quality: CoS, QoS, MPLS and SLAs The last chapter in VoIP 101 focuses on the network service provided by carriers to move VoIP packets, and how network traffic is identified, managed and prioritized, resulting in Class of Service offerings to meet Service Level Agreements. VoIP 101 is intended to be read sequentially, building ideas on top of ideas. The detailed Table of Contents makes it also useful as a day-to-day reference handbook. Our goal is to explain the big picture, the jargon and buzzwords, and put in place a very solid base of VoIP knowledge spanning fundamentals to the latest technologies and how they are deployed – in plain English. Let’s get started!
Author: Ted Wallingford
Publisher: "O'Reilly Media, Inc."
ISBN: 0596008686
Category : Computers
Languages : en
Pages : 505
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Book Description
Based on real-world experience, this handy solutions manual addresses the most common VoIP migration challenges. Find out how to build your own VoIP system, install it, and begin making calls - so you can start saving today. Ideal for IT managers, network engineers, and system administrators.
Author: Christina Hattingh
Publisher: Pearson Education
ISBN: 1587059479
Category : Computers
Languages : en
Pages : 639
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Book Description
The first complete guide to planning, evaluating, and implementing high-value SIP trunking solutions Most large enterprises have switched to IP telephony, and service provider backbone networks have largely converted to VoIP transport. But there’s a key missing link: most businesses still connect to their service providers via old-fashioned, inflexible TDM trunks. Now, three Cisco® experts show how to use Session Initiation Protocol (SIP) trunking to eliminate legacy interconnects and gain the full benefits of end-to-end VoIP. Written for enterprise decision-makers, network architects, consultants, and service providers, this book demystifies SIP trunking technology and trends and brings unprecedented clarity to the transition from TDM to SIP interconnects. The authors separate the true benefits of SIP trunking from the myths and help you systematically evaluate and compare service provider offerings. You will find detailed cost analyses, including guidance on identifying realistic, achievable savings. SIP Trunking also introduces essential techniques for optimizing network design and security, introduces proven best practices for implementation, and shows how to apply them through a start-to-finish case study. Discover the advanced Unified Communications solutions that SIP trunking facilitates Systematically plan and prepare your network for SIP trunking Generate effective RFPs for SIP trunking Ask service providers the right questions—and make sense of their answers Compare SIP deployment models and assess their tradeoffs Address key network design issues, including security, call admission control, and call flows Manage SIP/TDM interworking throughout the transition This IP communications book is part of the Cisco Press® Networking Technology Series. IP communications titles from Cisco Press help networking professionals understand voice and IP telephony technologies, plan and design converged networks, and implement network solutions for increased productivity.
Author: Stephen P. Olejniczak
Publisher: John Wiley & Sons
ISBN: 0470440066
Category : Computers
Languages : en
Pages : 376
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Book Description
So you’re in charge of implementing a VoIP phone system for your organization? VoIP Deployment For Dummies is a crash course in Voice over Internet Protocol implementation! Here’s how to analyze your network and implement a VoIP phone system, manage and maintain it, keep it secure, and troubleshoot problems. You’ll learn how to plan the rollout, work with Session Initiation Protocol (SIP), handle fax issues, and keep your users happy. Understand how VoIP works, common misconceptions about it, and the pros and cons for your organization Compare and comprehend hardware and software choices Discover the options for touch tones and faxing via VoIP systems Analyze network devices, IP addresses, connections to remote sites, and other aspects that will affect VoIP implementation Draw up a test plan, check out both voice and fax transmission, get a report, and schedule the installation Investigate SIP call generation, identify the elements, understand cancelled calls, and re-INVITE calls Troubleshoot your system, identify call variables, trace the source of a problem, manage trouble tickets, and resolve failures Manage latency, jitter, and flap, and take advantage of Wireshark Find out what to expect when your system goes live Written by an expert with extensive real-world experience in VoIP implementation and management, VoIP Deployment For Dummies provides the know-how you need. You’ll be able to implement your system and manage any issues proactively, which is sure to look good to your boss!
Author: Flavio E. Goncalves
Publisher: Packt Publishing Ltd
ISBN: 1785288679
Category : Computers
Languages : en
Pages : 384
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Book Description
Build high-speed and highly scalable telephony systems using OpenSIPS About This Book Install and configure OpenSIPS to authenticate, route, bill, and monitor VoIP calls Gain a competitive edge using the most scalable VoIP technology Discover the latest features of OpenSIPS with practical examples and case studies Who This Book Is For If you want to understand how to build a SIP provider from scratch using OpenSIPS, then this book is ideal for you. It is beneficial for VoIP providers, large enterprises, and universities. This book will also help readers who were using OpenSER but are now confused with the new OpenSIPS. Telephony and Linux experience will be helpful to get the most out of this book but is not essential. Prior knowledge of OpenSIPS is not assumed. What You Will Learn Learn to prepare and configure a Linux system for OpenSIPS Familiarise yourself with the installation and configuration of OpenSIPS Understand how to set a domain and create users/extensions Configure SIP endpoints and make calls between them Make calls to and from the PSTN and create access control lists to authorize calls Install a graphical user interface to simplify the task of provisioning user and system information Implement an effective billing system with OpenSIPS Monitor and troubleshoot OpenSIPS to keep it running smoothly In Detail OpenSIPS is a multifunctional, multipurpose signalling SIP server. SIP (Session Initiation Protocol) is nowadays the most important VoIP protocol and OpenSIPS is the open source leader in VoIP platforms based on SIP. OpenSIPS is used to set up SIP Proxy servers. The purpose of these servers is to receive, examine, and classify SIP requests. The whole telecommunication industry is changing to an IP environment, and telephony as we know it today will completely change in less than ten years. SIP is the protocol leading this disruptive revolution and it is one of the main protocols on next generation networks. While a VoIP provider is not the only kind of SIP infrastructure created using OpenSIPS, it is certainly one of the most difficult to implement. This book will give you a competitive edge by helping you to create a SIP infrastructure capable of handling tens of thousands of subscribers. Starting with an introduction to SIP and OpenSIPS, you will begin by installing and configuring OpenSIPS. You will be introduced to OpenSIPS Scripting language and OpenSIPS Routing concepts, followed by comprehensive coverage of Subscriber Management. Next, you will learn to install, configure, and customize the OpenSIPS control panel and explore dialplans and routing. You will discover how to manage the dialog module, accounting, NATTraversal, and other new SIP services. The final chapters of the book are dedicated to troubleshooting tools, SIP security, and advanced scenarios including TCP/TLS support, load balancing, asynchronous processing, and more. A fictional VoIP provider is used to explain OpenSIPS and by the end of the book, you will have a simple but complete system to run a VoIP provider. Style and approach This book is a step-by-step guide based on the example of a VoIP provider. You will start with OpenSIPS installation and gradually, your knowledge depth will increase.
Author: Eric Coll
Publisher: Teracom Training Institute
ISBN: 1894887166
Category : Technology & Engineering
Languages : en
Pages : 200
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Book Description
Certification study guide and companion reference textbook for the TCO Certified VoIP Analyst (CVA) courses and exams. The CVA Study Guide corresponds directly to the CVA online courses and exams. For each lesson, detailed text notes are provided along with the main graphic. Many people find they learn better with a companion book. Printed in color! Get a complete understanding of Voice over IP and SIP, with CVA Certification to prove it. CVA covers all aspects of Voice over IP, including all the different ways VoIP is implemented, how calls are set up with softswitches and SIP, how voice is packetized and the factors affecting sound quality, connecting to carriers and SIP trunking, and network quality with MPLS, Service Level Agreements and Class of Service. The Certified VoIP Analyst Certification includes six online courses and exams: 2221 Fundamentals of Voice over IP 2222 VoIP Architectures and Implementation Choices 2223 Softswitches, SIP, Call Setup and SIP Trunking 2224 Voice Packetization, Codecs and Voice Quality 2225 SIP Trunking and Carrier Connections 2226 IP Network Quality: CoS, QoS, MPLS AND SLAs This knowledge enables a CVA to stand out from the rest, with demonstrated broad and deep vendor-agnostic knowledge of VoIP systems and best practices. This kind of knowledge enables higher-paying positions performing analysis, writing reports, making recommendations and providing effective, value-added contributions in project management, business and product development, software design, sales, marketing and finance. Invest in yourself with the CVA Certification Package from Teracom Training Institute! The CVA Certification Package Unlimited Plan includes the six CVA online courses and the TCO CVA Certification Exam, both with unlimited repeats – which means guaranteed to pass, and refresh your knowledge anytime. Get up to speed on all major topics, at your own pace. Understand the fundamentals, technologies, jargon, buzzwords and most importantly, the underlying ideas ... and how it all fits together. Plus, get your Telecommunications Certification Organization (TCO) Certified VoIP Analyst (CVA) Certification to prove it! Upgrade your skills – and your résumé – with this training and certification today!
Author: Syed A. Ahson
Publisher: CRC Press
ISBN: 1420070215
Category : Technology & Engineering
Languages : en
Pages : 471
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Book Description
The number of worldwide VoIP customers is well over 38 million. Thanks to the popularity of inexpensive, high-quality services, it's projected to increase to nearly 250 million within the next three years. The VoIP Handbook: Applications, Technologies, Reliability, and Security captures the state of the art in VoIP technology and serves as the comprehensive reference on this soon-to-be ubiquitous technology. It provides: A step-by-step methodology to evaluate VoIP performance prior to network implementation An invaluable overview of implementation challenges and several VoIP multipoint conference systems Unparalleled coverage of design and engineering issues such VoIP traffic, QoS requirements, and VoIP flow As this promising technology’s popularity increases, new demands for improved quality, reduced cost, and seamless operation will continue to increase. Edited by preeminent wireless communications experts Ahson and Illyas, the VoIP Handbook guides you to successful deployment.
Author: Eric Coll
Publisher: Teracom Training Institute
ISBN: 1988138221
Category : Technology & Engineering
Languages : en
Pages : 84
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Book Description
VoIP Architectures and Implementation Choices Is a comprehensive overview of the many flavors of VoIP, comparing and contrasting the various implementation and architecture choices. Progressing from talking between computers over the Internet, through Internet telephony, VoIP from carriers, PBX replacement with call manager / softswitch systems, Hosted PBX and Cloud Services, to IP Centrex, we’ll deliver to you the knowledge to confidently differentiate VoIP architectures and discuss pros and cons of different options. VoIP Module 2 Detailed Outline 2. VoIP Architectures and Implementation Choices 2.1 Internet Telephony: Computer-Computer VoIP over the Internet ...... 2.1.1 Voice in Packets Over The Internet ...... 2.1.2 Call Setup ...... 2.1.3 Voice Quality ...... 2.1.4 Proprietary Web Phone Apps 2.2 Proprietary Internet Telephony Example: Skype ...... 2.2.1 Proprietary Internet Telephony ...... 2.2.2 Communicating to the Skype App Behind a NAT ...... 2.2.3 Proprietary Closed Ecosystems 2.3 VSPs: Internet to Phone ...... 2.3.1 Voice over IP Service Providers ...... 2.3.2 Connection to the Telephone Network ...... 2.3.3 VSP Gateway Required 2.4 VSP: Phone to Phone over the Internet e.g. Vonage ...... 2.4.1 POTS Telephones for VoIP ...... 2.4.2 Adapter ...... 2.4.3 Differences from POTS 2.5 VoIP Is The New POTS ...... 2.5.1 VoIP From The LEC ...... 2.5.2 Not Internet Communications 2.6 VoIP at Carriers ...... 2.6.1 Migrating to an All-IP Network ...... 2.6.2 Ending Switched-Access Tariffs 2.7 VoIP-Enabled PBX and Migration Options ...... 2.7.1 Choices for Business VoIP Systems ...... 2.7.2 Critical Operations Infrastructure ...... 2.7.3 Upgrading a PBX ...... 2.7.4 Migration Plan ...... 2.7.5 TDM over IP: Legacy Equipment Support ...... 2.7.6 Handset Gateways and BORSCHT ...... 2.7.7 Guinea Pigs 2.8 PBX Replacement: Premise Softswitch ...... 2.8.1 Call Manager ...... 2.8.2 Essential Function ...... 2.8.3 Special-Purpose Premise Hardware ...... 2.8.4 Standards-Based? ...... 2.8.5 Many Names ...... 2.8.6 Capabilities ...... 2.8.7 System Architectures ...... 2.8.8 Redundancy and Critical Functioning 2.9 Cloud Services and Hosted PBX: Softswitch as a Service (SaaS) ...... 2.9.1 Cloud Services ...... 2.9.2 Customer Management ...... 2.9.3 Additional Features ...... 2.9.4 Numerous Advantages ...... 2.9.5 Disadvantages 2.10 IP Centrex ...... 2.10.1 Outsourcing Everything To The Phone Company ...... 2.10.2 Advantages over Legacy Centrex ...... 2.10.3 Many Service Providers and Features ...... 2.10.4 Cost Structure 2.11 Asterisk and Open-Source Softswitch Software ...... 2.11.1 Open-Source Software ...... 2.11.2 Asterisk and sipX ...... 2.11.3 Advantages ...... 2.11.4 Disadvantages 2.12 IP Phone Features and Uses ...... 2.12.1 VoIP Phone Components ...... 2.12.2 Functions ...... 2.12.3 Numerous Methods for Dialing ...... 2.12.4 Special Features and Customization
Author: Olivier Hersent
Publisher: John Wiley & Sons
ISBN: 0470023600
Category : Technology & Engineering
Languages : en
Pages : 419
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Book Description
IP (internet protocol) Telephony, enabled by softswitches, isgoing to usher in a new era in telecommunications. By putting voiceand data over one IP network, operators can enjoy lower costs andcreate new, revenue-generating "multimedia" services. This valuable reference offers a comprehensive overview of thetechnology behind IP telephony and offers essential information tonetwork engineers, designers and managers who need to understandthe protocols and explore the issues involved in migrating theexisting telephony infrastructure to an IP-based real timecommunication service. Drawing on extensive research and practical developmentexperience in VoIP from its earliest stages, the authors giveaccess to all the relevant standards and cutting-edge techniques ina single resource. IP Telephony: Deploying Voice-over-IP Protocols: Assumes a working knowledge of IP and networking and addressesthe technical aspects of real-time communication over IP. Presents a high level overview of packet media transporttechnologies, covering all the major VoIP protocols – SIP,H323 and MGCP Details specific strategies to design services for publicnetworks where endpoints cannot be trusted and can be behindfirewalls. Explores the problems that may arise from incomplete protocolimplementations, or architectures optimized for private networkswhich fail in a public environment. This amply illustrated, state-of-the art reference tool will bean invaluable resource for all those involved in the practicaldeployment of VoIP technology.
