VoIP Telephony and You: A Guide to Design and Build a Resilient Infrastructure for Enterprise Communications Using the VoIP Technology (English Edition) 9788194837732, 8194837731

Citation preview

VoIP Telephony and You

A Guide to Design and Build a Resilient Infrastructure for Enterprise Communications Using the VoIP Technology

Rashmi Nanda

www.bpbonline.com

FIRST EDITION 2021 Copyright © BPB Publications, India ISBN: 978-81-94837-73-2

All Rights Reserved. No part of this publication may be reproduced, distributed or transmitted in any form or by any means or stored in a database or retrieval system, without the prior written permission of the publisher with the exception to the program listings which may be entered, stored and executed in a computer system, but they can not be reproduced by the means of publication, photocopy, recording, or by any electronic and mechanical means.

LIMITS OF LIABILITY AND DISCLAIMER OF WARRANTY

The information contained in this book is true to correct and the best of author’s and publisher’s knowledge. The author has made every effort to ensure the accuracy of these publications, but publisher cannot be held responsible for any loss or damage arising from any information in this book.

All trademarks referred to in the book are acknowledged as properties of their respective owners but BPB Publications cannot guarantee the accuracy of this information.

Distributors:

BPB PUBLICATIONS

20, Ansari Road, Darya Ganj

New Delhi-110002

Ph: 23254990/23254991

MICRO MEDIA Shop No. 5, Mahendra Chambers,

150 DN Rd. Next to Capital Cinema, V.T. (C.S.T.) Station, MUMBAI-400 001

Ph: 22078296/22078297 DECCAN AGENCIES 4-3-329, Bank Street,

Hyderabad-500195 Ph: 24756967/24756400

BPB BOOK CENTRE

376 Old Lajpat Rai Market, Delhi-110006

Ph: 23861747

Published by Manish Jain for BPB Publications, 20 Ansari Road, Darya Ganj, New Delhi-110002 and Printed by him at Repro India Ltd, Mumbai

www.bpbonline.com

Dedicated to

Mrs. Bharati Nanda and Mr. Ashok Kumar Nanda My parents, who have made me into what I am today.

About the Author

Rashmi Nanda is an engineer by profession and a writer by passion. She has passed Class 10 (ICSE) from Mount Carmel Convent School, Rourkela, Odisha, in the year 1998 and Class 12 (CBSE) in Science stream from Deepika English Medium School, Rourkela, Odisha, in the year 2000. She has done her B.Tech in Electronics and Instrumentation from Purushottam Institute of Engineering and Technology (PIET), Rourkela, Odisha, in the year 2004. She has more than eight years of experience in freelance content writing. Outside work, she takes an interest in listening to music, gardening, walking, doing yoga and mentoring people.

Her LinkedIn Profile: https://www.linkedin.com/in/rashmi-nanda7bb0b995/

About the Reviewer

Amit Gumber has 20 years of experience in Information Technology and Networking. He is involved in handling multiple platforms and enhancing skills in design, implementation, and management of large complex infrastructures. He is a trusted and award winning prominent member of Avaya, Ciena, Juniper, and Cisco forums and actively participates and shares knowledge within the community and peers. His technical certifications include CCIE, MCSE, CCNP, HP Networking, Arista Alcatel-Lucent, Aruba Wireless, Avaya Design, Juniper Routing and Switching, Juniper Security, Cisco Firewall, Radware Load Balancer, BIG-IP F5 application delivery, Fortinet, Palo Alto, Cisco IoT, SDWAN, SD Access, Nutanix Certified Professional, Dell EMC Storage and Vblock, and many others. He has worked with companies like Almoayed Group, UAE. He is currently working as a consultant in HCL Technologies, Noida.

Acknowledgements

I would like to acknowledge ‘BPB PUBLICATIONS’ for giving me another chance to showcase my expertise and skills in VoIP explanatory text. I would like to acknowledge the inputs and contributions of Mr Manish Jain who has gone out of his way to help me in writing this book.

I would like to thank the organizational staff of ‘BPB PUBLICATIONS’ for contributing towards the reviewing of the VoIP paper and providing valuable insights and inputs. I would also like to thank the associates and collaborators of ‘BPB PUBLICATIONS’ for their generous volunteering of expertise and time.

Preface

Today, an increasing number of organizations are offering facilities of remote working to their workforce. Some businesses do not have an office but they have staff in a distributed manner. Remote working has multiple benefits, but the biggest challenge faced by it is in the scenario of collaboration and communication. There are four great tools of interaction which are as follows:

Mobile App/Webphone: Call from whichever possible place

With the phone system of the hosted VoIP, a person can make calls using a phone through the Internet. VoIP helps employees to get flexibility with their used phone in a better manner. Other than IP phones or desk phones, remote workers can utilize mobile apps or web phones. With both of these, employees can use their extension numbers from any locality within a specified time span of their possessing the active internet. VoIP home-based or remote workers can make work line calls. They can also maintain inbound calls in realm of customer support or sales. For the employees, there are important reasons of not using their numbers, instead of using working lines for better quality control and security. It is easy to use mobile phones for calls, but the mobile apps create an easy factor for the workers to their line of office phones. Mobile apps of VoIP are user-friendly and the employees keep their working and personal calls separate. Mobile apps include: Call Conference, Call Recording, Company Directory, Do-not-disturb (DND), and Music on Hold. A webphone has been built into the

user interface of VoIP for ones who prefer to operate from their computers. People make calls by utilizing a headset or microphone of the computer. There is no requirement of downloading anything for using webphones. We just require a password and username for logging in.

Live Reporting: Remote agent’s better management

With the aid of live reporting, people are able to see all active calls and phone lines in real time. You can check how many are busy and active agents by putting the caller’s number and average caller’s hold time on hold. With in-depth information, you can improve the experience of the consumer and manage agents. Other than reports that are live, the dashboard of VoIP displays analytics for analyzing patterns and call usage further. So, you can handle adequately peak times of calls.

Call Recording:

VoIP system providers offer on-demand and automatic call recording. All recordings need to be saved on the dashboard, downloaded and then, one can listen to all recordings in a direct manner from the dashboard. Managers give better feedback to the agents and listen to the call recordings. With workers who are remote, you can have regular training check-ins by particularly changing the landscape by adding features to the product or platform. Go to the manager and employees of call recordings, and you can both discuss suggestions and feedback for improvement.

Virtual Fax:

Virtual fax is more secure when compared with a fax machine on the landline. Virtual faxing has a lot of easy implementation. With VoIP, total faxes have online storage, and you can access these via the online dashboard. Providers provide two ways of sending the fax that involves document uploading through dashboards and sending faxes with email attachments. Chapter introduces Voice over Internet Protocol (VoIP) applications that are multi-faceted with multimedia and voice-based telecommunication facilities. Amongst these the successful market segments are standardization, infrastructure management, regulation, bandwidth and Quality of Service (QoS).

Chapter discusses Corona and its ramification for WFH practitioners with respect to VoIP and video conferencing.

Chapter deals with a lot of economic and technical advantages for packet switching that are available due to VoIP.

Chapter explains the overview of the system level VoIP that consists of POTS networking by Digital Subscriber Line (DSL) modems or cables. It also explains VoIP processing that supports successful transmission of dual tone multiple frequencies, which has presence in bands. Chapter covers Telcos that takes the IP telephony challenge very seriously. It is a threat and an opportunity. It covers the SWOT

analysis for strengths, we have, telecommunications which become largely custom centric. Even the start-up companies have large venture capitals and technical knowledge.

Chapter shows the VoIP technology that has widespread deployment and the main reason being assurance of voice quality. Voice quality is voice perception of an end user depending on Mean Opinion Scores (MOS). Wireless Local Area Networks (WLANs) are basically used for better quality of voice. VoIP networking has several issues like management of traffic, failure of network switches, routers, virus attacks, Denial of Service (DoS), failure of VoIP gateways, and call servers. Chapter provides security in the VoIP environment that has authentication, integrity, secrecy, non-repudiation, and DoS protection. IETF protocols are SIP, Megaco, and MGCP, deploying many methodologies for finding out the fatal flaws. Chapter talks about Network Address Translation (NAT) which is the standard of IP on LAN where the address key translation is kept separate from the public view. Customers, providers, and Internet are segregated by a security layer. Chapter presents TCP/IP which is a industrial protocol suite used for communicating and accessing over the transmission medium. The various layers are Application Layer, Transport Layer, Internet Layer, and Physical Layer.

Chapter deals with IP end-points, that is, PCs and phones are required VoIP hardware. Currently, smart phones are very much in vogue. Personal Digital Assistants (PDAs) are utilized in combination with service providers.

Chapter talks about the in-depth knowledge of the SYSteam Nat AB, a premier re-seller product of Cisco that provides cabling installation and wireless communication with internet accessibility and fiber optic measures.

Chapter discusses the best practices which are open standard’s deployment, WAN links, provisioning of bandwidth, prioritizing, and voice trafficking over data. It also discusses facility upgrades, controllers of session border, port-cards, media gateways, test equipment, soft switches, and systems of back office are the economics of scale. Chapter concludes with competition and demands that determine market. There are numerous factors for VoIP to gain widespread acceptance. QoS is VoIP service’s pre-requisite for business provisions of mass market.

Downloading the coloured images:

Please follow the link to download the Coloured Images of the book: https://rebrand.ly/6b7ba6

Errata

We take immense pride in our work at BPB Publications and follow best practices to ensure the accuracy of our content to provide with an indulging reading experience to our subscribers. Our readers are our mirrors, and we use their inputs to reflect and improve upon human errors, if any, that may have occurred during the publishing processes involved. To let us maintain the quality and help us reach out to any readers who might be having difficulties due to any unforeseen errors, please write to us at :

[email protected]

Your support, suggestions and feedbacks are highly appreciated by the BPB Publications’ Family.

Did you know that BPB offers eBook versions of every book published, with PDF and ePub files available? You can upgrade to the eBook version at www.bpbonline.com and as a print book customer, you are entitled to a discount on the eBook copy. Get in touch with us at [email protected] for more details.

At you can also read a collection of free technical articles, sign up for a range of free newsletters, and receive exclusive discounts and offers on BPB books and eBooks.

BPB is searching for authors like you

If you're interested in becoming an author for BPB, please visit www.bpbonline.com and apply today. We have worked with thousands of developers and tech professionals, just like you, to help them share their insight with the global tech community. You can make a general application, apply for a specific hot topic that we are recruiting an author for, or submit your own idea.

The code bundle for the book is also hosted on GitHub at In case there's an update to the code, it will be updated on the existing GitHub repository.

We also have other code bundles from our rich catalog of books and videos available at Check them out!

PIRACY

If you come across any illegal copies of our works in any form on the internet, we would be grateful if you would provide us with the location address or website name. Please contact us at [email protected] with a link to the material. If you are interested in becoming an author

If there is a topic that you have expertise in, and you are interested in either writing or contributing to a book, please visit

REVIEWS

Please leave a review. Once you have read and used this book, why not leave a review on the site that you purchased it from? Potential readers can then see and use your unbiased opinion to make purchase decisions, we at BPB can understand what you think about our products, and our authors can see your feedback on their book. Thank you! For more information about BPB, please visit

Table of Contents

1. Introduction to Voice Over Internet Protocol Introduction Structure Objective VoIP Introduction VoIP regulation VoIP methods VoIP equipment IP phones Gateways of VoIP Market of equipment Conclusion Points to remember Multiple-choice questions Answers to MCQs Questions

2. VoIP Video Conferencing and Coronavirus Introduction Structure Objective VoIP video/audio conferencing VoIP server Asterisk server FreePBX server Endpoints of VoIP Linphone softphone

Mumble

Software of video conferencing FreeSWITCH server TeamTalk Video surveillance and streaming IP video cameras Software of video display iSpy MotionEye ZoneMinder Shinobi Coronavirus in relation with VoIP, video conferencing Choosing a platform of video conferencing Current WFH situation in India and worldwide Using platform of e-mail provider for video conferencing in coronavirus times Office 365 Google Meet Different platforms of e-mail Using platform of provider of VoIP in COVID-19 times for video conferencing Usingthird-partypaid vendor for video conferencing in coronavirus times Situation in India – timeline Governance initiatives – opportunity in sickness and misery by the Government of India ArogyaSetu app PM CARES Fund Ayushman Bharat Aatmanirbhar Bharat Abhiyan

Vocal for Local Jandhan Conclusion Points to remember Multiple-choice questions Answers to MCQs Questions

3. VoIP’s Challenges and Benefits, VoIP independent Providers Introduction Structure Objective VoIP’s advantages VoIP challenges Independent providers of VoIP market Conclusion Points to remember Multiple-choice questions Answers to MCQs Questions 4. Overview of Systems Level Introduction Structure Objective Overview POTS mimicking Echo dealings Encoding of voice Voice detection

Interfaces of POTS Playout Feature implementation Telephonic and packet network signaling PSTN interfaces and analog phones Device provisioning and supplementary services Data functions of the voice gateway Elements of security of VoIP Putting everything together Conclusion Points to remember Multiple-choice questions Answers to MCQs Questions 5. Interfaces of VoIP Telephony Introduction Structure Objective VoIP and mobile service provider Telcos in VoIP Strengths Weaknesses Opportunities Threats Functionalities of the FXO and FXS PSTN interfaces and analog phones FXS is mirrored by the FXO Lifeline for failure of power Re-direction of calls

VoIP remote calling Echo Standards Conclusion Points to remember Multiple-choice questions Answers to MCQs Questions 6. Assurance of Voice Quality for Voice Over Internet Protocol Networks Introduction Structure Objective Voice quality assurance Issues of VoIP networking Delay Jitter Loss of voice packets Echoes Vocoder Voice activity detection Issues of VoIP quality in relation with example deployment Measurement tools of quality of voice Intrusive/ Active tests Passive tests Solution aspects of practical VQM Reporting block of statistic summary Block of VoIP metrics Conclusion

Points to remember Multiple-choice questions Answers to MCQs Questions 7. Implementation of VoIP Security Introduction Structure Objective Security Services of VoIP Network services Services of retails Virtual private networks Fax over Internet Protocol (FoIP) VoIP security Demand for security of VoIP VoIP security areas Components of security of VoIP Measurement of performance of VoIP security Protocols for encryption (Triple) Data encryption standard (3DES/DES) Advanced encryption standard Rivest cipher (RC4) Protocol for Voice Encryptions – Secure RTP (SRTP) Key exchanging methods Symmetric key Public key Hybrid key Diffie-Helman keys

Security association Configuration security of VoIP VoIP security for call controlling process Internet security (IPSec) Transport layer security (TLS) Voice processing security Denial of Services (DoS) Open issues Conclusion Points to remember Multiple choice questions Answers to MCQs Questions

8. Functionality of Data Router Introduction Structure Objective Data routing Basics of information router Network address translation (NAT) Firewall Factors of performance of router Designing process Targeting performance Options that are architectural for routing Designing factors that is additional Conclusion Points to remember Multiple choice questions

Answers to MCQs Questions

9. Technical Description Related with VoIP Introduction Structure Objective TCP/IP Primer Transport layer Internet layer Physical Layer Implementation that seems to be decentralized Physical media Types of media Data Voice Video Transport of media and call control separation Mobility and virtual locations Architecture options of internet telephony Trunk replacement Hybrid (hop-off/hop-on) End-to-end, direct IP connection Usage methods PBX replacement PBX extension IP Centrex Residential service API / Programmatic interfaces

Session initiated protocol – Common gateway interface (SIP-CGI)

Call processing language JAIN SIP Servlets Conclusion Points to remember Multiple choice questions Answers to MCQs Questions 10. VoIP Hardware and Software Components Introduction Structure Objective Components of hardware End-points of IP (that is PCs and phones) Access gateways Integrated Access Devices (IAD) Components of software Signaling Conversion Application server (that is billing and accounting) Media server Signaling server gateways Policy server Media encoding (Layer 1) Physical transport header PPP Header IP Header UDP Header RTP Header Protocols H.323

SIP Megaco/ H.248 Media Gateway Control Protocol (MGCP) Others Reliability Reliability of trunk replacement Reliability for hybrid architecture VoIP reliability that is end-to-end Mapping of telephone number Conclusion Points to remember Multiple choice questions Answers to MCQs Questions

11. Business Model and Market Model in Relation with Internet Telephony Introduction Structure Objective Use-case scenario Current situation of SYSteam Nat in model of market and business Market model Students housing facilities Societies owned by tenants Models of business Build-it-Yourself

Resale Outsourcing Recommendation

Attitudes and requirements of customers Goals of ISP Competitors of Broadband Blixtvik Internet och Telefoni AB Bredbandsbolaget (B2Bredband AB) Com Hem AB Tele2 AB TeliaSoneraSverige AB Uppsala Stadsnat AB (Uppsala Metropolitan Area Network) Skype Microsoft Customer proposition of ISP internet telephony To price VoIP that is residential Cost free Possible charges that are extra Costs that are extra Cost of internet telephonies for consumers Conclusion Points to remember Multiple choice questions Answers to MCQs Questions

12. Technology, Economics, In Practice, to be Concerned with IP Telephony Introduction Structure Objective Situation of SYSteam Nat presently Solutions of IP telephony in realm of technology General Build-it-Yourself

Outsourcing Resale Recommending provider of IP telephonies Solutions of internet telephony related with economics Build-it-Yourself SYSteam Nat’s Financial Business case: Outsourcing Resale Suggesting/recommending giver of telephony of IP In practice when concerned with internet telephony Tests done with SYSteam Nat AB Software of test Call Report of analysis Quality of Call Statistics of call: List following statistics for all channels of currently selected calls: Automated approach of test Conclusion Points to remember Multiple choice questions Answers to MCQs Questions 13. VoIP to be Concluded Introduction

Structure Objective Conclusion in general VoIP’s conclusion with respect to business and market model Conclusion of VoIP with regards to technology

Ending VoIP when concerned with economics Concluding VoIP in relation with Covid-19 Conclusion Points to remember Multiple choice questions Answers to MCQs Questions Key terms

Bibliography Image credits Index

CHAPTER 1 Introduction to Voice Over Internet Protocol

Introduction

Voice over Internet Protocol (VoIP) applications are multi-faceted with multimedia and voice-based telecommunication facilities. Among these, the successful market segments are standardization, infrastructure management, regulation, bandwidth and Quality of Service (QoS) . VoIP gateways are interfaces between Public Switched Telephone Network (PSTN) and IP networks, which moves data across a Local Area Network (LAN) through router and interfaces. The worldwide marketing of VoIP gateways reaches 7 billion dollars by 2020. VoIP gateways are implemented by methods like PC-based servers, router-based voice modules, and voice modules by concentrators. The VoIP equipment are IP phones, which belong to IP protocol functions having connection to the PSTN. Fax over IP (FoIP) is much better than fax because there is no real-time transmission.

Structure

In this chapter, we will cover the following topics: VoIP introduction

VoIP regulation

VoIP methods VoIP equipment

Objective

The first chapter deals with methodologies related to VoIP regulation, VoIP equipment, and introduces the various functionalities that enable the successful implementation of VoIP in our daily lives. Among these, the successful market segments are standardization, infrastructure management, regulation, bandwidth, and This is an introductory chapter, which walks us through VoIP.

VoIP Introduction

VoIP has acquired too much of attention recently because of numerous reasons. The most notable reason is that it possesses the potentiality for significant cost reduction of international and long-distance communication of voice. It introduces totally enhanced and new communicating ways. The following figure 1.1 shows how the basic VoIP system works:

Figure 1.1: VoIP

Fundamental VoIP applications are multimedia, voice, data and fax. traditionally has suggested telecommunications of voice alone. But, over is the generalized term used toward all realms of traditional telephony, with further applications offered over IP private and over public internet networks. The result is that the real-time information, fax, and multimedia services are regarded as VoIP sub-segments.

VoIP had a difficult beginning, partly due to the limitations that were technological and partly because of slower acceptance of newer technology characterized of being of poor reliability and quality. Improvements technologically with stronger public demands for low tariffs of phone result in the widespread VoIP marketing acceptances.

The services of VoIP are rapidly becoming a practical alternative to old services of telephony. PSTN services regulated by the government artificially with high price have been left vulnerable due to stepping in of these newer technologies. Quality and different VoIP challenges are addressed with worldwide booming in the usage of the internet, making it a conceivable substitute, resulting possibly in an integration predicted longer of information and voice networks. The following figure 1.2 shows a VoIP gateway with PSTN and IP phones:

Figure 1.2: PSTN, IP Phone, VoIP Gateway

The current VoIP technological applications are focused primarily around alternated service of long-distance voices. The providers of incumbent services recognize opportunity of VoIP as telcos of next generation and non-traditional givers of a service. Voice over packets is researched since 1980s and 1970s, but the real VoIP developments had not begun till 1995. VocalTec, which is a firm in Israel, initiated IP telephonic market in year 1995 together with software allowing voice connections among 2 PCs over network based on an IP. Numerous packages started emerging, and in year 1997, Delta Three launched the first service

of phone-to-phone for commercial use. The following figure 1.3 depicts an IP telephony system:

Figure 1.3: IP Telephony There are current segments and niches of market where VoIP does prevail, but there are key obstacles to be overcome prior to it becoming successful mass market-wise. This includes bandwidth, regulation, network and infrastructure management, standardization, and QoS.

VoIP regulation

Regulations of VoIPs are largely questionable. Telephonic services traditionally have been regulated heavily. Numerous governments do operate on monopolies of telecommunications, with deregulation observed nowadays in others (1998 in lots of the EU and 1996 in the US).

Many government policies (in both developing and developed nations) toward the Internet have encouraged competition and growth by adoption of attitude of laissez-faire. The internet’s rapid development largely has been because of a lack of regulatory intervention. The introduction of most regulation forms truly retards internet innovation, thereby creating a barrier for entry for start-up organizations. Operators stand to get profit from this because they have cherished the monopoly position for numerous years. Both the European Commission and the Federal Communications Commission (FCC) welcome this provision to force down international tariffs of phone. The constraints of regulations related specifically to the internet put newer entrants at a severe disadvantageous place, in turn assisting continued overpricing of international and longer-distance telephonies. So, the regulation of telecoms generally has advantages for smaller providers of VoIPs. The following figure 1.4 shows the basic structure of a VoIP software:

Figure 1.4: VoIP Regulation and Software, H.323 The main barriers to change telephone companies for consumers are to convert numbers of phone and associated inconvenience. Portability of number allows customers to manage numbers while changing industries. When VoIP of phone-to-phone becomes commonly placed and convenient, the abilities for offering VoIP total services with no conversion of phone numbers work toward the advantage of small providers.

With a local loop of high-speed technology, the introduction ability for accessing local loops becomes important for providing competitive services. Regulation results in unbundling of local loops in a few European countries, the US, and Hong Kong. Recently, in 1999 July, BT did order open-up of their local loop. This gives a high-speed accessibility provision opportunity to those providers that are small.

There is a impediment of growth and sustainability which caused huge ethos of disappointment for Telecom companies. Telcos does not wish to obliterate a profitable business that exists to bank on regulation while assuming VoIP threat for petering out like Internet Telephony Service Providers (ITSPs) becoming classified such as provider of voice telephones. Many others have no desires for investing a solution of VoIP till its technical troubles are overcome. The following figure 1.5 shows an ITSP.

Figure 1.5: ITSP

Data services that are enhanced are the ones having an economic provision by givers of independent services (non-facility-based service providers) to have competition with networking provider as longer as they gain accessibility to networks. An example of service enhanced is the provision of contents for premium rating service and retail internet service provision. Internet Service Providers (ISPs), Enhanced Service Providers (ESPs), and ITSPs generally are not subjected to longer-distanced charge of accessibility. Telcos urges regulatory positions for changing. ITSPs become competitive increasingly with telcos, so they argue for identical classification. Simultaneously, there is a difficulty for differentiating between varied internet traffic types. So, the methods for levying accessibility costs on ITSPs do seem impossible. The following figure 1.6 captures the very essence of ISPs for internet users:

Figure 1.6: ISP

The following figure 1.7 shows the ESPs data packets:

Figure 1.7: ESP

The European Commission defines voice telephony as provisions for publics of speech switching and direct transportation in real-time among termination points of public switched networks in turn enabling whichever user to utilize connected equipment for such network termination points for communicating with different points of

So, the following conditions should be met: Services offered should be part of commercial offers. Services are provided to all members of the public. Services involve speech and direct transport in real time. Services are given between termination points of public-switched networks in fixed telephonic networks. VoIP lacks a few of these factors, notably the most

being transmission in real time. This falls short of traditional telephony, which is not fully transparent, reliable, and ubiquitous. VoIP generally is not an available service of voice transmission, but it is an application available for the ones who already subscribe to the basic package of the internet. The European Commission does not force ITSPs for the payment of accessibility charges because of such closed user group offerings. VoIP is seen as an effective method for introduction of competition because of restriction of telephones in Europe. Additional points suggested is that added movements of regulation affects industry of VoIP. Such things include rules of entry and requirements of market entries:

Providers are not subjected to required factors of market entry and restrictive licensing, which have applicability to varied providers of telecommunications because of the classification of many internet services as non-voice or non-basic services. Regulators begin to impose licensing, registration, or various ITSP requirements because VoIP becomes a serious competitor for older telephonies. Such processes are time-consuming, thereby increasing time-to-market of newer entrants. If ITSPs have been classified as a carrier of basic service of voices, there are requirements for having compliance with routing rules and restrictions. This restricts those services that in turn bypass the systems of accounting. But, with provision of complex internet topology, there are virtual impossibilities for monitoring

packets that are individual with imposition of rules of routing on them.

VoIP methods

VoIP had been initially dismissed easily. Software of client provided poor quality that resembled a CB radio, and that allowed only party numbered one to speak in one instance. Communicating party numbered two needs to have an online service together with compatible running software on their PC. But, potentiality for these services was visualized by vendors like and All of these launched items of VoIP in the year 1996. Till this time, two of the internet backbone and computing power capacity had been improved. Much better quality of voice had got VoIP availability. Simultaneously, communications of VoIP received standardization of the International Telecommunications Union (ITU). By 1997, commodities based on standards had achieved presence. The basic necessities of all earlier products of VoIP were users who at all ends had a connection of the internet. VoIP gateway development has modified this such that phone-to-phone and PC-to-phone calls could be made. This removes one among most core barriers to broad acceptance in the market. With installed gateways, there is no need for users to have special software, PCs, or even internet connection for placing calls mainly routed over the public internet. Gateways do allow users for circumventing PSTN with its tariffs, utilizing internet for variable quality voice communications and for significant reduced prices.

Numerous newer operators and vendors/manufacturers of equipment have emerged ever since the release of internet phone.

The first scalable and robust commodity of VoIP and VoIP initial service had been given by Delta Three in the late 1995. Client software has been presently supported by many operating systems like and

VoIP’s original form is PC-to-PC telephony. Both users must be online prior to whichever setting up of connection and usage of compatible multimedia computers and software. This does not have practicality for tariffs separation or regulation of such VoIP forms because it is critical with the probability of being counterproductive and trying for distinguishing the bits of audio from different bits. There are difficulties for differentiating among the full-duplex real-time communication and time-insensitive store and forward information. Unless the internet has been regulated in wholesome, there are no feasibilities for regulating this VoIP form.

Telephony of PC-to-phone arose as the PSTN-to-IP and IP-to-PSTN gateways established their presence. Gateways compress and packetize voice traffics from PSTNs, placing it on the network of IP and decompressing and assembling traffic in varied directions. The following figure 1.8 shows the gateway between PSTN and

Figure 1.8: PSTN-IP Gateway

Regulation of the VoIP of PC-to-phone is a difficult process. Call originators pay no accessibility charges because, to service providers, it looks similar to a PC-to-PC call. This reduces the fees of accessibility greatly, as the originator pays usually nearly twothirds of total charge of access. At the end of termination, the provider of service gives termination service both to long-distance and local callers, but they cannot know where particular call of VoIP has been originated, with spurring of troubles with tariffing and regulation. With VoIP of phone-to-PC, there have been no access fees at the point of termination because originators do not know where calls terminate. The following figure 1.9 deals with PCto-phone communication by VoIP.

Figure 1.9: PC to Phone Communication

VoIP of phone-to-phone does eliminate the need for PC totally, by using internet for carrying voice in between two gateways. VoIP calls of phone-to-phone now offer great quality when compared with PC-to-phone and PC-to-PC calls of VoIP because PC-to-PC offers over one dedicated network of IP. There are breakouts to PSTN at two ends, but because the originator is not aware of call termination and vice-versa, problematic factors regarding access fees and regulation exist. FoIP is superior to fax, which is traditional, because it does have no requirement for real-time transmission. Fax transmission is associated traditionally with PSTN and is not varied from sending whichever different file. Messages of a fax have an ability to absorb delays with the retransmission of packets with no negative impacts on the final data received. Conferencing of multimedia does involve data, video, and real-time voice combination transmission over the internet. Although standards have been emerging, this shall lag behind various applications of VoIP due to substantially great bandwidths needed for multimedia application.

VoIP equipment

For providing solutions of VoIP, varied networks of IP and internet must seamlessly interface with PSTN by means of gateways. Such gateways are one among the key revenue areas for vendors of VoIP equipment. IP networks must acknowledge schemes of prioritization and reservation of resource needed for guaranteeing QoS for dependent data of delay. The hardware needs to give services of VoIP ranging from standardized machinery of networking like switches and routers to IP phones and to gateways.

IP phones

Phones belonging to IP functions like regular phones operate, but in place of having connection with PSTN, they have a connection with IP messaging networks with the transmission of voice information by virtue of IP packets.

IP phones presently are too expensive, and to use VoIP is inconvenient and cumbersome. Old telephones are simpler to utilize and cheaper to buy, and costs that are marginal are closer to zero, in turn allowing prices to have huge reductions. IP phones (and VoIP generally) must be developed further for ways of easy usage and decreasing prices, prior to their invasions in space of old telephonies.

Till date, IP telephones have a difficulty for average users’ adaptation such that interfaces are inconvenient. To create ample familiarity with necessary features of older telephony to IP environments, moving toward IP networks and PSTNs seamless integration has started.

In 1998, Selsius was purchased by Cisco. It introduced phones providing all the functionality that is found in traditional phones with a direct connection to IP networks. There is a shift in industry of telephonic equipment, suggesting newer opportunities for Private Branch eXchange (PBX), and vendors of telecom services. Theoretically, IP telephones have operations as phones

that are old, while having ability for adding newer capabilities. The following figure 1.10 tells us about the PBX:

Figure 1.10: PBX IP Courier Ethernet Phone of Nokia does provide functionality of PBX (like features of call controlling and multiple lining appearances) without PBX. It is an Ethernet telephone together with familiar interfaces linking directly to IP networks. It supports telephonic features like transfer of calls, caller ID, forwarding of calls, and waiting of calls.

IP Shuttle product of Nokia is a premises device of customers, allowing standardized telephones to become plugged into network of IP. It aims at those environments that are residential and are

specifically designed to serve newer data and voice services offered by organizations of telephones and industries of cables. Siemens released (April 1999) an IP telephone with capability that is enhanced –new interface, message with regard to unanswered calls, and ability for name retrieval and address of IP caller retrieval. Their HiNet LP 5100 IP telephones cost 425 dollars, but the price might fall rapidly.

Ericsson has released Doubler telephone that allows the use of a virtual line of telephone, enabling a user to get calls of phone (over IP) without logging off to the internet. Voice traffics are carried over IPs between the gateway and the PC of the user, with rest of the calls carried on PSTN. Thus, QoS has direct dependence on the speed of user connection and on trafficking that is generated by them.

Gateways of VoIP

Gateways act like an interface in between IP networks and PSTN. When VoIP calls are made, there are breakouts usually to PSTN at all ends, for last and first legs of links, with internet or different IP networks utilized for trunk connections.

Gateways accept fax machine or traditional phone connection determining termination calling points and deciding what is the cheapest and best way for routing calls (that is how much of links are over PSTN).

Gateways originally were aimed at customers of business, and they increasingly were aimed at carriers and ISPs desiring to offer services of VoIP as competitive advantages. With rising scalability opportunities, there are supplying items to providers of services instead of small enterprises.

Gateways do not interoperate correctly; users are left to confine to networks of their providers or the ones formed through the partnerships. H.323 likely becomes the protocol of standardized interoperability, and such standardization happens soon (by end of 2000) because of pressure from the service providers. Motorola has collaborated with Vsys (June 1999) for developing VoIP gateways on the basis of Vswitch VoIP solutions for intelligent applications of network. Vswitch is a software product

based on UNIX, which provides an interface both for Signaling System no. 7 (SS7) and H.323 standards to assist fax, voice, and message. Its capabilities involve an advanced intelligent service of network like single-stage dialing and free phone numbers and provisioning and billing. The item must be available commercially later.

Gateways are implemented by means of methods that follow:

PC-based servers

Voice modules on the basis of routers Modules of voice by virtue of concentrators

Gateways based on PC move data across LAN, to router and through interfaces. Such standalone devices do not move traffic dependent of delay to the front of the router queue, but they have many advantages:

They have easy scalability. They possess router independence with installation in a mixed network of equipment.

They have no requirement of up gradation of routers.

Their market has been more competitive when compared with a varied market of gateways, thereby resulting in low prices. They are designed for these purposes.

The following figure 1.11 speaks about data flow in a local area network:

Figure 1.11: LAN

Concentrator and router-based gateways reduce the delay timing because they eliminate the extra hopping needed for PC-based gateways. This improves system quality. Traffic is directly moved to WAN interfaces and into router queues. Simple prioritization forms are employed too.

Market of equipment

Suppliers to a VoIP’s market do bridge the dynamic internet world with a more static traditional tele communication world. Organization of equipment of VoIP grows from smaller players like With the growth of industry, significant acquisition has taken place. Nortel has acquired Micom, which is a gateway vendor. Cisco, Bay Networks has acquired It incorporates modules of voice into routers. Nokia has taken into control which is a switching firm of IP. Small vendors team up with large players with which is a provider of software, which announced dealings with Ascend (which has merged with Lucent) and 3Com that in turn provides IP gateway functionalities by means of a Digital Signal Processor (DSP) in equipment of former. Such partnerships do intend building up of experience of all members. It leads to newer items with existing technology adaptation.

Although with the growth of the market, machinery of internet has not been recognized along with reliabilities as the equipment of PSTN. With the movement of the telecommunication world toward IP, newer machinery is in the form of equipment of data. So, the information machinery updates for handling of voice instead of vice-versa factors. This gives advantages to the vendors of message networking equipment because this is where they possess experiences.

Operators of telecoms have own accessibility networks, with main consumers of traditional and old suppliers like and Supplier

relationships built up across the span of time with main providers in countries of theirs is there.

Suppliers of data networking do understand networking of IP better than older suppliers, but they may not satisfy management and quality company requirements as strong as telcos. Such industries have attained huge experienced growth in the near present, thereby needing application that is new for continuing these growths. VoIP has greatest marketing opportunities for these. With tremendous numbers of new entrants, such suppliers develop a connection and contacts with older telco suppliers of machinery, and either by marketing, acquisition, or alliance, are making contact with telcos themselves.

Larger VoIP and information vendors have greater experience along with packeted switching to have connection with ITSPs and ISPs. They need building of expertise in communications of realtime. Key challenges for these (and different business that operate in such a realm) are to be aware of information and voice integration with the ability for realizing such integrations both economically and technologically. Lots of vendors of networking of data have added capabilities of VoIP in their equipment like remote and router accessibility machinery. They extend their networks for enabling functions that are more sophisticated too. It is the strategy for information and voice networking integration for creating multi-servicing networks.

From the perspective of a provider, deployment of gateways of VoIP is costlier than PSTN switching deployment equipment on the basis of perport. VoIP gateways costs 1,800 dollars per port on average in 1997, and in 1999, it has fallen to 600 dollars perport. The same capacity (that is per line) for older telephony cost nearly 100–200 dollars. Carriers place enormous pressures on vendors to lower costs at the rate of 100 dollars per port range within the next years. There is a change in ports of the carriers for utmost utilization of bandwidth. Standardization adds to scaling economies, and in total, all companies of higher-technology engineering do improve, thereby lowering costs. Internet enjoys more users in the US when compared with the ones in Europe. VoIP market machinery has likeliness to split equally in between two geographical areas. Telephonic US prices are lower than in Europe, and new entrant opportunity is low in the US, and so, enthusiasm is reduced generally. But, because of new networks of IP being achieved by ITSPs, and fact that the US has more IP developed places, there is still an expectation that VoIP has higher uptakes. Worldwide marketing for gateways of VoIP reaches one billion dollars by 2000. It grew by 75 percent in 2001 and 40 percent further in 2002. The suggested decline in the rates of increases is because of the falling prices for such equipment. The number of shipped ports greatly increases over next time span, and over eight million by year 2002. The increasing rate significantly decreases after year 2000. VoIP uptakes affect WAN switching carrier markets, along with switches specifically bought for the purposes of VoIP for increasing

from lesser than 1 percent in 2001 to nearly 6 percent by 2002. This produces revenues nearly of 0.25 billion dollars.

Conclusion

This chapter introduces us to VoIP and its wonderful world. IP phone, VoIP gateway, and PSTN services are addressed with booming, that is worldwide, in internet usage, making this a conceivable substitute. It results in possible integrations that have been predicted longer of data and networks of voice.

The next chapter deals with Covid-19 and VoIP with regard to WFH users.

Points to remember

VoIP is a generalized term utilized toward all traditional telephony realms, with the applications offered truly and further over networks of public internet and IP private.

Regulators start to impose registration licensing or various needed factors of ITSP because the VoIP becomes a serious competitor toward older telephonies. Gateway development of VoIP has changed the basic necessities of early VoIP products where utilizers who at each end have an internet connection such that PC-to-phone and phone-to-phone calls are made.

The original form of VoIP is PC-to-PC telephony.

The transmission of fax is traditionally associated with PSTN, and it is not different from sending any varied file.

The functionalities of IP gateway are provided by DSP.

Multiple-choice questions

What are sub-segments of VoIP? Fax

Multimedia services

Data All of the above

Why key obstacles need to be overcome before it is being successful in a mass market?

Quality of Service (QoS) B. Infrastructure management

Government policies

Non-standardized methods What are the regulations of VoIP?

ISPs

ESPs

ITSPs

All of the above

What are VoIP methods?

PC-to-PC

FoIP

PC-to-phone None of the above

What is the equipment of VoIP? PBX phones

IP phones Doubler phones Non-PBXphones

Answers to MCQs

Q1: D Q2: A, C

Q3: D

Q4: B Q5: A

Questions

What is VoIP? How VoIP gateway, PSTN, and IP phone help in day-to-day life?

What do you mean by VoIP software and regulation?

What is ESP, ITSP, and ISP? Why PC-to-phone communication and LAN must be used?

What is purpose of PSTN-IP gateway and PBX?

CHAPTER 2 VoIP Video Conferencing and Coronavirus

Introduction

There are multiple factors for believing that the working days in the tall towers of an office are over. What 9-11 could not accomplish, 2020 pandemic succeeded. There might be a recognition that lots of jobs are present that cannot be remotely done, but many are existing. Overhead costs for rent, operating most offices, cleaning, insurance, electricity, all go out when doors are locked and let employees WFH. To lower overhead costs often is good tradeoff for a lack of controls perceived over employees. Use current situations for trying remote working with a team of employees.

Structure

In this chapter, we will cover the following topics: VoIP video/audio conferencing

Video surveillance and streaming

Software for video display Coronavirus in relation with VoIP, video conferencing

Using a platform of e-mail provider for video conferencing in coronavirus times

Using a platform of provider of VoIP in Covid-19 times

Using a third-party paid vendor

Situation in India – timeline Governance initiatives

Objective

In this chapter, we will discuss corona and its ramification for WFH practitioners with respect to VoIP and video conferencing. People are accustomed to communicating with voice, and because a huge number of messages are conversed with queue assistance that is non-verbal, to possess audio-visual talk has far more effectiveness. VoIP encompasses a collection that is technological with the capability to truly encode and deliver content of multimedia in real time across digitalized network. When there are established requirements for this type of interaction, and if the mesh networks have an ability for supporting it, then numerous reliable options are available for implementing the VoIP and video conferencing. Let us have a look.

VoIP video/audio conferencing

Today, people try to emulate communications, but nothing can be a replacement for actual discussions in an interactive manner. Individuals are actually accustomed to conversations with voice, and because much of information is communicated with assistance of queues that are non-verbal, to have audio-visual conversations is far more effective. These advantages of interaction come at a little cost. Programs of multimedia typically have a greater impact over network performances than other programs. VoIP helps in providing those services that enable both video conferencing and voice on the network. It encompasses a technological collection being capable of delivering and encoding content of real-time multimedia across a digital network. When there is an established requirement for this communication type, and if mesh networks possess capability to support it, then there are numerous reliable options to implement video conferencing and VoIP.

VoIP server

VoIP servers are capable of providing a more complete set of communication services for audio and video conferencing all in the real-time scenario.

Asterisk server

Asterisk is truly one of original software PBX server. It was designed firstly to run really on computers of Linux. But presently, it is available for OpenWRT and MacOS routers. There it is used for building large-scale systems of telephony. So, multiple features of proprietary and commercial PBX systems, which involve menus of interactive voice response (IVR), voice mail, automatic call distributions, and conference calling, are available. Asterisk is documented widely. It serves as an underlying engine of interaction for many other PBX software packages. It is a robust and extremely tried-and-true software of IP-PBX, but there is need of specific skills and knowhow for implementing it. The following figure 2.1 states the Asterisk server functionality in relation to VoIP and video conferencing:

Figure 2.1: Asterisk Server

FreePBX server

FreePBX is a graphical user interface (GUI) based on the Web to manage Asterisk. It is deployed most commonly like a part of integrated FreePBX Distro. The FreePBX Distro installs all of the operating system of Linux with FreePBX, Asterisk, and dependencies of software included.

All Asterisk’s extensive features have availability with benefits to have a Web interface of the FreePBX for facilitating management of Asterisk. This makes it much easy for those users who do not have telephonic expertise. Many mesh networking operators that deploy VoIP take FreePBX Distro’s advantage while implementing their services of PBX. The following figure 2.2 shows a FreePBX server that is a GUI component for managing Asterisk:

Figure 2.2: FreePBX Server

Endpoints of VoIP

Once VoIP PBX is provisioned on a mesh network, there shall be a need of VoIP endpoint for communicating through a server. VoIP phone specialized hardware has availability from numerous manufacturers to provide endpoints of conversation on a network. There is a possibility of using the legacy hardware of an analog phone to be connected to a network by utilizing Analog Telephone Adapters There are pure phones of software (softphones) to support many devices.

Linphone softphone

Linphone is phone of software that is supported on Android, Windows, iPhone, Raspberry Pi, Linux, and MacOS. It is used for placing video and voice direct calls and calls through VoIP PBX. Utilizers can share files or pictures, transfer calls to other numbers, merge calls to the group conference, and send messages of chatting done. The softphone has the ability for managing the contact lists with call histories having existence for future references.

Mumble

Mumble is package of VoIP possessing availability on Windows, Linux, and MacOS to support the Qt platform. There is a presence of mobile apps also, like Plumble for Android and Mumblefy for iPhone.

To host Mumble requires localized downloading of the Murmur server, which is involved as an option in the installer of Mumble. Primary Mumble users are video gamers of the internet who wish to talk with one another during playing of game. It is used as a non-gaming service of conversation of voice that does not need the presence of an IP-PBX server on a network.

Software of video conferencing

Often effective video conferencing tools depends on experience and expertise, and we can establish communication by the various tools explained as follows:

FreeSWITCH server

FreeSWITCH is recent platform of communication used for building systems of voice PBX with menus of voice response, full WebRTC support, and video conferencing, with capabilities of sharing of screen and chat messaging. The modular design creates a possibility of installing only what actually is needed for meeting interaction requirements. Currently, the package of FreeSWITCH is installed on Windows and Linux servers. But, if needed, it can have MacOS computer’s compilation. The following figure 2.3 shows a FreeSWITCH server, which is a recent communication platform:

Figure 2.3: FreeSWITCH Server

FreeSWITCH gives robust video and voice communication, IVR menus, call accounting, user directories, voice mail, hold music, chat messaging, screen sharing, call recording, and lots of other features to be implemented based on requirements. It is a communication platform with extreme flexibility. There is a need of specific skills and understanding for managing, installing, and configuring it as a service.

TeamTalk

TeamTalkis a system for audio-visual conferencing that enables potential masses to share data and communicate across a network. Often, it is classified as freeware, but the server of TeamTalk is proprietary. The source code has no public presence. During a conference, utilizers talk by means of a computer microphone, create messages instantly, see others through webcams, applications of desktop are shown, and files are shared. Software package of TeamTalk bundles a server and a client; therefore, any computer plays the role of a server or client. Video and voice conversations occur in rooms or channels, and a single server hosts multiple rooms. When there is participation in the channel, users write messages of text in the Chat tab, see applications shared in the Desktops tab, view streams of AV webcam in the Video tab, and files are downloaded from the Files tab. The owner of the server specifies a wider range of accessed permissions for all available rooms. TeamTalk is supported currently on Raspberry Pi, Windows, MacOS, and Linux computers. The following figure 2.4 shows a screenshot of the TeamTalk server:

Figure 2.4: TeamTalk

Video surveillance and streaming

Video and audio traffic are transmitted across network of AREDN for facilitating interaction. Multimedia streams have been supported on meshed networks, so use it for other executions. Video surveillance is helpful during an event or emergency, and AREDN networks are utilized for delivering this traffic type to Emergency Operations Multimedia traffic incurs a significant cost with respect to computing resources and network performance. Keep this in mind. Be sure that a mesh network is designed with appropriate bandwidth for handling traffic.

A Mobile Command Center (MCC) deploys supporting a larger event in place, namely San Juan Capistrano, California. 35,000 estimated people attend the annual gathering, and local team of RACES provides video coverage in real time of parade routes for emergency response and sheriff’s department agencies.

Greater than a dozen of IP cameras having high definition were collocated at the AREDN portable node site across the area. Individual streams of video were consolidated on many large displays in the MCC. Orange County Sheriff’s Administrator Sgt. Joseph Cope stated “This system of mesh camera provided by members of RACES was valuable technology for command staff. Parade was safest in future. When calls were taken, see activity happening in real time. There was incredibly only one fighting arrest, that happened just to occur in view of camera.”

IP video cameras

Cameras of IP video might have fixed focus and direction, or they can have remote controlled Pan, Tilt, Zoom (PTZ) models. Features and cost for video cameras vary widely. On the lower end, there is a very least costly Raspberry Pi Zero computer. It has an integrated camera that exists next to the Ubiquiti Bullet radio. On the high end, there is a commercial ruggedized PTZ camera that costs hundreds of dollars. It is shown with infrared LEDs and bubble dome. Many streaming video of IP cameras use the Real-Time Streaming Protocol (RTSP), where missing packets simply skip during video displays. It is challenging to decide the URL of the RTSP stream. There is a utility of packet capture as Wireshark and handy utility at ispyconnect, which helps. Camera supports numerous RTSP URLs all with different resolution and frequently. So, advertise whichever of these, as service on AREDN, node as needed. Currently, multiple cameras support Open Network Video Interface Forum whichis a set of standards and protocols, including the RTSP. This supports the PTZ camera controlling and camera discovery. The following figure 2.5 depicts an RTSP data flow, which is used for meta file and streaming commands:

Figure 2.5: RTSP A 1920*1080 resolution video stream at 60 frames per second consumes up till 8 megabits per second of network bandwidth. Some networks of AREDN support this load consistently, but low rates of frame reduce the needed bandwidth proportionally. 720p at 10 frames per second is adequate, more than and typically, for video surveillances. IP cameras with an Ethernet port is preferred for simplifying network connectivity, for ensuring speeds of data transfer adequately. Configure camera for obtaining the IP addresses meshed from node and reserving addresses for those cameras in Dynamic Host Configuration Protocol (DHCP) settings of the node. So, there is a consistent way for connecting to it. Camera with PoE supports truly is very useful because it simplifies site cablings.

Few cameras have easy deployment and configuration when compared with other cameras. So, ensure to study them carefully prior to making investment in costly hardware of camera. There is a topic of cameras forum on the AREDN website. Here, post experiences and questions: arednmesh.org camera forum.

Software of video display

Software helps in providing services of video surveillance on network. The list is not complete or comprehensive. Programs primarily with licenses that are open sourced were involved here in the list, although there was a successful usage of software with proprietary licensing.

iSpy

iSpy is well-known package of video management for Windows computers. It has Windows 7 and above certification, but might work on various systems supporting the .NetV4 Framework. It runs as a Windows program with a localized user interface (UI) that has accessibility on the computer with its installation. Services that are additional have availability after the subscription fee is paid. Program parts have licensing under LGPLv3, and others are proprietary portions. The following figure 2.6 shows an iSpy that is used for video management:

Figure 2.6: iSpy

Program of Windows gives workspace or surface where configuration and addition of microphones or multiple cameras is done. Then, interact and monitor them for listening to the live audios or displaying live videos from networking devices. Lots of media streams are locally recorded for utilization in future. PTZ cameras are manipulated with controlling in the UI. There is a motion detection’s configuration, which provides a way for automatic recording of multimedia snippets wherein specified events happen.

iSpy is connected to IP cameras by using sources of JPEG or MJPEG. It supports connections of camera using RTSP, MP4, or ASF. It is accomplished by means of a VLC plugin. It happens after the installation of the VideoLAN software. The VLC needs passwords and usernames to be directly present in the URL, so enter these in clear text.

MotionEye

MotionEye is a lightweight video displaying program that runs on computers of Raspberry Pi and Linux. It connects to a variety of IP cameras or a USB. It has the capability of displaying video streams truly in a grid format with accessibility by Web browsers on mesh networks. Regular user or administrator authentication displays varied menu options: options for full control of administration or regular users for utilizers of admin is viewed. The following figure 2.7 shows a lightweight video display – MotionEye.

Figure 2.7: MotionEye

Motion engine backend is built for providing robust event triggering and motion detection. It enables customized scripts for extending features. Example, printing temperature of a system and updating it every 10 seconds on displaying. Many AREDN operators have MotionEye implementation on portable lower-power Raspberry Pi computers. MotionEyeOS distro does install an operating system, with all of the dependencies on such a platform.

ZoneMinder

ZoneMinder is a video package fully featured, running on computers of Linux. Its displaying has accessibility across meshed networks by Web browsers. IP cameras are supported, which use MJPEG streams, else JPEG images interface. Connections of the camera areconfigured to record, motion detection, monitor, or to have a combination of preceding factors. The following figure 2.8 shows a Linux-compatible video package –ZoneMinder:

Figure 2.8: ZoneMinder

Regions or zones of image are defined according to ZoneMinder by administrators; each of this has different sensitivity levels of motion detection. At time of motion detection, every frame has its comparison with prior frames while checking for differences. In case the change amount is more than the percentage specified, events are triggered that can send alerts of e-mail, execute programs that are external, or capture recordings. ZoneMinder has features that are extensive for comparing and filtering video images. These features are useful to monitor the area of high traffic with a single interest point like door next entry to busy walkway. A robust set of features comes at a cost of little administrative complexity. It makes ZoneMinder a good operator candidate with experience and skills in video systems and Linux. The open designs and abilities for executing external programs does make ZoneMinder too flexible to have other system’s integration.

Shinobi

Shinobi is a video project fairly recent that implements the current streaming methods for the Web. It supports RTSP, JPEG/MJPEG, and FLV legacy streams and new WebSocket and HLS ways. A Web browser interface (UI) is responsive and clean, which renders nicely on mobile devices and tablets. This has been designed for creating navigation easiness, with pop-up and drop-down snapshot menus, configuration options, event lists, and video recording. In compliance with the ONVIF, permission is given to Shinobi for providing PTZ camera control. Motion detection is accomplished via plugins, with Web UI configuration. So, in case motion detection is not required, then conserve resources without making its addition to the system.

There are three user levels that provide authority delegation:

Superuser

Admin

Sub-account System settings are controlled, and Admin accounts are created by the Superuser. The settings of camera are controlled, and Groups

and Sub-accounts are managed by Admin. The Sub-accounts possess limited privileges with profiles of camera shared by members of the group.

Computing resources are conserved by Shinobi fairly well. Therefore, high-resolution streams or more cameras are supported on the server.

Coronavirus in relation with VoIP, video conferencing

All of us are living in an age where the risk level to confidential data is growing greater with each passing day. The need to safeguard communications that are digital places burden tremendously on business, thereby requiring constant diligence for mitigating risks that are ever-evolving. During the event of Black Swan like spreading of the coronavirus, where people are forced for making critical decisions such as sending their staff house with notice of few days only, it has been critical that they must not let their guards down. Hence, choosing the correct platform for communication is of utmost importance in this virtual world.

Choosing a platform of video conferencing

To communicate with one’s team of employees in as transparent and easy a way as possible assists in keeping his team motivated and focused, while working remotely even. There are lots of items on market providing video conferencing, group chats and texting, conducting and scheduling meeting of online teams, and sharing and collaboration of files. For keeping things in a simplified manner for those employees who just have become accustomed to remotely operate, combining such abilities on easy for using, single application eliminates lots of headaches, in turn getting team back for quick works. Some upfront fees that are minimal are there, but they have been smaller in case ability of maintaining employee productivity.

Employees have a desire to WFH and see ability of doing so anytime like an advantage. No gas, no traffic, and working in PJs is wished. They then complete their first remote working day, and soon, their lists can modify additionally. No escaping from children, no quiet offices, no company launches, contact loss with fellow employees, and delays to get job done, due to the reason that they cannot walk down hall to enquire boss with a query. Let us face this. Individuals have numerous reasons for liking to work from office, also. Employees can be in shock culturally. Some handle this while seeing it as benefit, while some do not.

Current WFH situation in India and worldwide

Many of us have to Work-from-Home (WFH), so for multi-users, this has now become the spotlight communication form only nearly.

The coronavirus is not a firstly occurring pandemic, but it is the first pandemic spread mainly through social media and media, to point that this has got nickname namely “Infodemic.” It means infamously false information spread throughout the internet, which causes distance and fear among masses.

Social distancing, unfortunately, with a sense of confusion and fear among many individuals, has presented hackers totally with the golden opportunities in cybercrime efforts continuously of theirs. More and more people are becoming familiarized with varied attack forms, and so, COVID-19 presents today a new threat entirely to all of us, and this is new method for hackers in order to maintain monetization at the back of the users.

A desire for staying informed leads multiple hackers to leave false sites for attracting innocents to view wrong websites. A wish to stay specifically to remain connected has attracted hackers to be after platforms of video conference, and “Zoom” is the one standing up and above all of these.

With 500 percent surging in daily usages, Zoom has launched video conferencing as popular “THE BEST and THE FIRST” platform for going after. Key points making Zoom so famous are being free and easy utilization. These two are reasons also for the platform’s vulnerabilities in the security realm, showing how unprotected and insecure the platform in question is.

Zoom never provides an end-user experience, but it always was more of a business-oriented platform.

Using platform of e-mail provider for video conferencing in coronavirus times Video conferencing need not be done with an e-mail provider, but, if one is in cloud (majority of people are currently in cloud), there is most likeliness that his provider of e-mail possesses the tool of video conferencing to be utilized.

Office 365

If a person is utilizing “Office 365” as ane-mail provider, then probably, notice the icon of Microsoft Teams, which keeps bugging him. Teams from Microsoft is revamped Skype for Business (Skype has been purchased by Microsoft in a short time span) and truly a tool that has no greatness. There is no easy usage, and no ideal quality, but it is a free tool with far more security than using Zoom, today.

Google Meet

It was earlier called “Google Hangouts.” If the industry utilizes Gmail as a platform of e-mail, then Google Meet has free access. By far, it is the best tool. Google Meet offers whatever Zoom offers, and features of security of the tool are built in already (there is no need to be an expert of IT for configuring it), and it is far more modern and advanced when compared with Zoom.

Different platforms of e-mail

If one is using Appriver, GoDaddy, Intermedia, or any third-party hosting of e-mail, then many such tools have little video conferencing sort. In case security is a concern, you can use any platform other than Zoom. Probably, it shall not attract the hackers, because it is not a “mainstream” solution. So, even if these tool’s security is not so much advanced as Microsoft or Google, but fact that this is more like one underground tool, does make it more secured than Zoom (that is all in spotlights).

Using platform of provider of VoIP in COVID-19 times for video conferencing If an organization is utilizing technology of VoIP for phones, then there is a provision of tool of video conferencing, cost-free mostly, and likely (all providers vary from one another). In these cases, use that tool, and this is a great option also. Famous ones are Intermedia Unite platform or RingCentral Office platform (RingCentral and AT&T has partnership with each other), so a person may possess a tool whether he is with directly RingCentral or AT&T VoIP. Various tools of VoIP exist, some use GoToMeeting or Cisco WebEx tools. Otherwise, it may cost good penny for direct subscription, but in case, they are not sure, conduct an enquiry with industry whether these tools have accessibility. These tools most likely have far more of securities while comparing it with Zoom, that for time being is truly figuring out a proper way for securing its platform.

Using third-party paid vendor for video conferencing in coronavirus times There are numerous different video/audio conference calling tools existing. Many vendors are stepping up now, and they offer upgraded features or free trials to “free tiers,” meaning tools assist while hosting virtual social gatherings or business meetings free of cost.

Few tools are as follows: Cisco WebEx: It offers free options for meeting all needs under free version.

Skype: It is an individual tool and is free and has a limitation of 50 participants. Downside is easy usage with signing in Microsoft’s platform. A few find it frustrating.

FreeConference.com: It is a free tool, but it has too many limitations to number of attendees.

Few paid tools to be explored, but no limitations are as follows: GoToMeeting: Best paid tool present probably.

CyberLink U Meeting.

Zoom Meeting (Paid License): Not to be confused with the free version. Paid version comes with security features that are improved. It is a good option in case one is ready to pay money.

Situation in India – timeline

In the current scenario, India has announced a series of measures to combat the COVID-19 situation. These measures were announced after video conferencing with the state government and in consultations with senior members of the various government departments as well as taking inputs from the masses, all the while taking help of video conferencing tools.

March 19: Prime Minister Mr. Narendra Modi announced Janata Curfew to be held on March 22, 2020 from 7 AM to 9 PM.

March 24 to April 14, 2020: Commencement of Lockdown 1.0.

April 14 to May 3, 2020: Starting of Lockdown 2.0.

May 3 to May 17, 2020: Commencement of Lockdown 3.0.

May 17 to May 31, 2020: Starting of Lockdown 4.0.

June 1 to June 30, 2020: Commencement of Unlock 1.0. July 1 to July 31, 2020: Starting of Unlock 2.0.

August 1 to till date: Commencement of Unlock 3.0.

Classification zone wise:

The Indian government has divided the nation into three zones:

Green Zone: District with no confirmed cases.

Orange Zone (Non-Hotspots): District with few cases.

Red Zone (Hotspots): District with higher number of cases that are active with higher doubling rates.

Governance initiatives – opportunity in sickness and misery by the Government of India VoIP stands in primary importance for all the governance initiatives, as India seeks to find an opportunity in sickness, and misery because all these measures were constructed with the mobile technology in mind. By taking into consideration these internet strategies with mobile boom, we can successfully take India one step ahead.

ArogyaSetu app

“ArogyaSetu” is a mobile application that is developed by the Government of India. It connects essential services of health with Indian citizens in combined fights against the coronavirus. The app aims at augmenting the Indian government’s initiatives, particularly the Department of Health, to proactively reach out to and inform users of app with regard to the relevant advisories, risks, and best practices that pertain to COVID-19 containment.

PM CARES Fund

The Prime Minister’s Citizen Assistance and Relief in Emergency Situations Fund (PMCARES Fund) was launched on March 28, 2020, following India’s COVID-19 pandemic. The fund is used to combat and relief and containment efforts against the outbreak of the corona virus, and similar situation of pandemic like in future. The Prime Minister of India is the chairman of the fund. Trustees involve the Minister of Finance, Minister of Defense, and Minister of Home Affairs in the Government of India.

Ayushman Bharat

Ayushman Bharat–Pradhan Mantri Jan Arogya Yojana (PM-JAY) – is a flagship Government of India’s scheme for giving cashless tertiary and secondary care treatments from empaneled private and public hospitals. It gives coverage to 10 crore plus vulnerable and poor families. The National Health Authority (NHA) is an apex body being responsible for Ayushman Bharat PM-JAY’s implementation.

Aatmanirbhar Bharat Abhiyan

PM Aatmanirbhar Bharat Abhiyan Yojana is a scheme created by India’s PM. The aim of this is promoting industries, local businesses, service providers, farmers, and 130 crore masses of nation with utilization of local items form a king them self-reliant. The PM has provided 20 lakh crores to Indians. He has imported weapons under this yojana. Citizens are requested to buy and promote maximum usage of services of their nation, village, state, and city. Money of the country must not go out, and business ventures possess chances of spreading throughout the globe such that government helps people in better ways.

Vocal for Local

Vocal For Local: Made In India–Swadeshi Movement or Aatma Nirbhar Bharat app gives an item list in varied categories such as FMCG products, mobile phones, clothing, fashion, mobile applications, electronics, and automobiles with lots of sections on the basis of company and production such as semi-Indian, Indian, and other brands of other nations.

Jandhan

By means of Jandhan, a person gets accessibility to numerous tools, including:

Government insurance schemes and subsidized loans: Easy access to the portals of applications and information

Storage securely of crucial digital documents Digital Ledger: Easy keeping of digital accounts

In near future, this Jandhan shall offer:

Easier access to new investment and saving items, microinsurance, and micro-loan products

Easy application of a PAN card

Conclusion

This chapter deals with VoIP, video conferencing, and the corona virus pandemic. The Free SWITCH server gives robust voice and video communications, callaccounting, sharing of screen, call recording, voice mail, IVRmenus, user directories, chat messaging, and hold music to have implementation on the basis of requirements. Team Talk is an audio-visual system of conferencing, enabling potential masses for sharing information and communicating across networks. Zone Minder possesses extensive features for filtering and comparing video images. These features have usefulness in monitoring high traffic are a with the point of single interest such as door next entries to busy walk ways. iSpy is arenowned video management package for a Windows computer. Shinobiis a video project that recently and fairly implements current Web streaming ways. Motion Eyed displays the video streams in grid formats with Web browser access on the mesh network.

The next chapter deals with VoIP benefits and challenges.

Points to remember

VoIP helps to provide the services that enables voice on both the network and video conferencing, and video displays of software server in giving video surveillance services on networks.

The Asterisk server is one among the originals of software server of PBX. The FreePBX server is deployed very common, such as part of FreePBX Distro with integration.

Today, FreeSWITCH package has been installed on Linux and Windows servers.

During a TeamTalk conference, users talk by virtue of computer microphones, creating data instantly, sharing files, seeing others by means of web cams, and showing desktop application.

E-mail provider platform, VoIP giver platform, and third-party paid vendors are used for video conferencing in COVID-19 days.

Multiple-choice questions

Whiche-mail provider platform is used for video conferencing in COVID-19 times?

Office 365

Google Meet Skype

All of the above

What are government initiatives in corona virus times?

Swachh Bharat Abhiyan

ArogyaSetu app

Vocal for Local None of the above

What is the VoIP provider platforms used in times of COVID-19?

RingCentral

AT&T

Intermedia Unite Platform

All of the above

What are the third-party paid vendors used for the video conferencing in times of Corona?

Cisco WebEx Free Conference.com

GoToMeeting Zoom

Answers to MCQs

Q1: D Q2: B,C

Q3: D

Q4: A,C

Questions

What do you mean by VoIP audio/video conferencing? What does Free PBX and Asterisk servers do?

What is the function of the Free SWITCH server and Team Talk?

What is Zone Minder, iSpy, Shinobi, and Motion Eye?

CHAPTER 3 VoIP’s Challenges and Benefits, VoIP independent Providers

Introduction

Numerous economic and technical advantages for packet switching are available because of VoIP. Some of them are as follows:

Low-cost routers replace switches.

Data overload is reduced. Swift compression updates are available.

One IP network is utilized for data and telephony with sophisticated multimedia services.

Speech over internet in un-encrypted form takes place.

VoIP challenges are delays like propagation delay, network delay, accumulation delay, processing delay, and jitters. The various protocols used are Transport Control Protocol and User Datagram Protocol for managing VoIP’s integrity. The Network Access Points congestion is largely reduced, removing delay and unreliability. The telcos, which are independent providers of VoIP, are growing by leaps and bounds, with a huge number of new entrants in the market.

Structure

In this chapter, we will cover the following topics: VoIP advantages

VoIP challenges

VoIP market’s independent providers

Objective

This chapter deals with a lot of economic and technical advantages for packet switching available because of VoIP. We will also learn the advantages and disadvantages of VoIP.

VoIP’s advantages

Today the market of ISP is largely driven by arbitrage of tariff. QoS unsatisfactorily has been compared with older telephonies. People wish to make international and long-distance calls over internet cheaply, thereby saving 20–100 percent on calls by bypassing PSTN. There are three key reasons for benefits of price:

Regulators rule that givers of VoIP have not been subjected to long distance access fees, which are levied on old providers. VoIP givers avoid settlement charges payment to operators of foreign that terminate calls. Of these, US operators approximately pay five billion dollars in one year.

Networks utilized have more efficiency because of packet switching and compression.

There are many economic and technical advantages in the realm of packet switching. It does appear that VoIPs have long-term cost benefits over traditional telephony on the basis of low equipment and efficiency costs. The technology of packet switching uses infrastructures with more efficiencies than technology of circuit switching that is utilized in old telephony. Fees of telephonic switching equipment have to sharply fall down in charges similarly as computer machinery

costs. VoIP’s port fees are higher relatively, but they fall down rapidly, and they shall undercut charges per line soon for older telephonies. Routers of low costs begin for replacing switches.

The economics unit, though debatable, seems to be like packet switching, along with the suggestion of estimates that the packet switching nearly is four times cheaper, resembling circuit switching of course at per byte levels.

A data overhead that has been associated with far lesser significant packet switching possess capacity to be spoilt in the circuit switching at time when no information has been conveyed.

Much swift compression updates could be created on internet than on PSTN hardware. On circuit switching networks, all total hardware in networks have been updated for progressing with compression advancement, whereas internet clients on standard PCs implement latest technologies with no concerns for rest of the networks.

An IP network multimedia service allows one network for being utilized for data and telephony, with more sophisticated services of multimedia. There is no practical evidence supporting an argument that management of one network has to be more economical than two network’s management.

In unencrypted forms even, speeches carried over internet are harder considerably for wire-tapping than analog speeches brought over wire pair of copper. Where encryptions are used, decryptions

by unauthorized parties are time-consuming immensely that takes lots of months or weeks for super-computers to decrypt phone conversations of two minute.

VoIP till now has not been a convenient telephonic mechanism. It needs awkward dialing, and it is cumbersome. Users need to dial the access number first, wait till this call has been routed to IP telephony access switch, and then only destination numbers are dialed. In early 1999, providers have started an introduction of services of VoIP, which no longer need that users must dial whichever differently than users dial it while making regular calls. Korea Telecom is one among the first givers who have integrated IP networks and PSTN in such ways. It is core development for ensuring VoIP’s speedy evolution. Nortel Networks and Lucent Technologies have made development packages supporting deliveries of VoIP’s enhanced services that involve single stage, above dialing. The following figure 3.1 shows Nortel networks and Lucent Technologies that enhance the supporting functionalities of VoIP:

Figure 3.1: Nortel Networks and Lucent Technologies With existence of feature parities, VoIP’s future uptakes largely depend on the introductions of applications that are enhanced. Assumption is that by such stages, price differentials have a much less significance due to lowered PSTN regulation or/and prices.

Applications available already include multicast, caller ID, call waiting, and conferencing of multimedia. Simple old features are much richer on packet switching networks. Advantages of these enhancements are too obvious. The requirement of separated lines for information and voice disappears, and functionality and flexibility are increased.

A possibility of these IP network applications extends certainly beyond the ones available on PSTN and for applications yet not

envisaged. VoIP does drive public networks toward client server, open topology. Legacy networks and systems of proprietary switching have retarded integrations of data and voice for long time span, but competitions in areas of telephonies have initiated movement toward networks that are open. It is a trigger bringing about true convergences of message and voice networks, but it does not happen till cost parities exist between circuit switched voice and packet switched voice technologies. Equipment of VoIP rapidly is falling in price, and this is considerably becoming more costly than older machinery. It will convert, but it shall take longer duration of time for conversion and modification.

In an enterprises IP network, ubiquity ensures larger addressable markets for equipment of VoIP and facilitating VoIP machinery integration onto the networks that exist. Enterprises are starting to put VoIPs on LANs and WANs, thereby cutting charges, as voice has been treated like information. On these private networks, data control is a lot easier than on public internet. In this, the QoS exceeds that of VoIP on the internet.

More and more IP machineries are attached to the internet, so benefits are derived from becoming part of VoIP networks. It increases further fostering demands. There are huge opportunities in VoIP provision. The main attraction presently is lower tariffing given by bypassing varied access fees that must be paid by traditional and old givers. VoIP no longer is technology to be merely employed by hobbyists. Users of new generation have emerged, and convenience of richness and use of applications are

becoming attractive and satisfactory. VoIP is becoming popular increasingly among general users of telephony.

VoIP challenges

Lower quality of unreliability and service of the internet is main marketing restraint for VoIP. Numerous issues of QoS experienced by networks of packet switching not affecting networks of circuit switching are there. The internet is a network providing best effort in which dropped packets and variable latencies do occur. Voice services need real-time transmissions, so VoIP results often in QoS that is heavily degraded. Four types of delays in networks of IP are present. Overall delays are aggregate of such delays:

Propagation Delay: It is caused by signals for traveling distances. Therefore, it is a distance function. Its overcoming cannot be conducted, and it is to be governed by physics laws.

Processing Delay: It is created by a collection and actual encoding of samples that are encoded into packets for transmission. It applies to the receiving end’s decoding too. It is a function of used coding algorithms and processing timing. The following figure 3.2 shows the delays – propagation and processing:

Figure 3.2: Propagation Delay and Processing Delay

Network Delay: It is function of pipe capacity in a network, and processing of packets with its transits in network is done. Delays associated with buffers of jitter are regarded as network delay’s part.

Accumulation Delay: It depends on used voice coder types. Finite amounts of timing (with variation from single sampling time to lots of milliseconds) are required for collecting frames prior to beginning of processing. The following figure 3.3 shows the delays – network and accumulation:

Figure 3.3: Network Delay and Accumulation Delay

Jitter is variable timing of inter-packeting caused by packeted traversals of network. This is considered as the delay’s standard deviation. It is not only impossible to control or predict how many hopping of packets from call of VoIP traverses, packets through similar calls have been assigned varied routes, with variations in numbers of different volumes of traffic, and hops along way. Thus, packets from same sources experience varied delays on destination ways.

Jitter is to be removed by fast packet buffering such that the slowest packets have correct timing arrivals for accurate sequences. It creates additional delays, as buffers account for all

packets. This delays them all such that the slowest is delayed. Conflict to minimize delays and removing jitter has yielded different methodology of dynamic adaptation of buffer sizes for matching timing variations. It minimizes delays that are associated with jitters while avoiding buffer’s under flowing.

If delay of one way exceeds nearly 200 milliseconds, two speakers adopt modes whereby one person speaks and another person listens. There is pause for making sure that after speaker finishes, listener talks. These pauses are timed ill-wise. Speakers do end up for stepping on each other speeches. Networks of IP are networks giving best efforts with no provision of service of guarantee. When routers become overloaded, they intentionally drop packets for relieving congestion. With older data trafficking, there is an error checking way built onto that protocol, which is called the TCP for addressing such situations and maintaining informational integrities. The protocol needs few overheads that are not conducive to traffic of real time and has no implementation for voice transports. The UDP has been used like the transport levels protocol. It needs lesser capacity, not guaranteeing delivery of information to destination, nor packet deliveries in right orders, nor avoiding the duplication of packets. Data integrity is, however, guaranteed. The following figure 3.4 shows the various protocols available for VoIP like the transport protocols (TCP and UDP) and session protocols (TLS):

Figure 3.4: TCP, UDP, TLS Uncompressed voice communications, fortunately, tend to have high tolerance for dropped packets. Human ears cannot detect packet loss in audio sequences if adjacent packets are played in place. A noticeable quality degradation occurs when only numerous packets in a row have been dropped. But, many mechanisms of voice compressions result in transmission of modifications (predicted or actual) between signals that are adjacent, instead of whole signals itself. So, corruption or loss of packets yields in errors that are propagated, degrading hugely video or voice quality beyond recognition.

UDP usage has further challenges. Many networks that are protected by the firewalls do not provide permission for traffic of the UDP. For information to pass via numerous firewalls, few registration sorts occur between the receiver and sender. It gives

requirement for open ports on clients that result in blocking itself by firewalls. When two wire telephone cables do connect to four wire exchanging interfaces, a circuit known as hybrid is utilized for converting between four wires and two wires. Echoes are caused by reflection of signals in hybrid circuits. Though circuits have good efficiency in conversion ability, small energy percentages are not changed and are reflected back toward callers. It is called

If speakers are closer to telephone switches or PBX, echoes are sent back in great speed such that they become hardly noticeable. With an increase in the reflection time of the echo, there has to be cancelation out of echo. DSPs listen to echoed signals subtracting prediction of these from the audio signal of the speaker. It is important particularly with VoIP because echoed signals have delays longer enough for being noticeable, which annoys even the speaker. Though cancelation of echoes is used, there is insufficiency because only the prediction of actual echoes is removed often.

Problems of quality are the most key barrier to VoIP development. Till management mechanism of QoS is available, it remains corerestraint. Unreliability is concerned cause in realms like public safety. PSTNs usually are very reliable and well-engineered. General services rarely are unavailable, and services of emergency are truly very reliable. Internet is unavailable or slow for varied reasons. Information gets corrupted or lost, making it unsuitable, as networks have extreme dependability. Slow traffic transfers and inability for connecting are results of network’s bottlenecks. There

are no one constraint on internet, rather, all internet junctions create performance bottleneck.

The first difficulty (for dialed up subscribers) is the connection to the telephone’s switch. The average call of phone lasts for three minutes, while the average internet users are online for 20 minutes. PSTN has not been designed for this length calling and abnormal longer holding time yields in absorption of lot of switch capacity, causing failure of many attempts of call. A phone line has a significant inferior capacity for the rest of internet. The overall connection is fast as the slowest pipe, so the last mile (from service provider or telephone switch to premises) slows down the transfer of traffic dramatically. A second complexity is the connection between the internet backbone and service provider. An approach of connection does overload, so a few users are denied accessibility, while others do experience troubles of response timing. The third and fourth bottleneck is in the backbone itself. The former is the bandwidth of backbone. Although it has high capacity to extreme extents, a dramatic demand increase has rendered it insufficient. The latter of two backbone constraints is router forwarding and receiving packets of data. The router speed has grown rapidly since the starting of the explosion of the internet. But, this growth cannot cope up with internet traffic growth, thus resulting in delays.

Finally, the original interconnection point of internet backbone is becoming overloaded severely. In the near past, the main

exchanging points of the backbone of the internet, namely Network Access Points (NAPs), have been congested. Many providers choose not to utilize it anymore. They have not got funding from the National Science Foundation ever since 1998. There is a movement for private-operated exchanging points. Meaning is that, because many givers enter an agreement of private interconnection, much trafficking data has no longer got availability. Providers and operators included have no obligations for disclosing such information. The following figure 3.5 shows the which combined with the ISP forms the internet backbone.

Figure 3.5: NAP All preceding factors contribute toward delay and unreliability.

Legacy network existence is restraint on the growth of the network of IP, within enterprises. Cases of compelling business are needed for replacing voice networks, which perfectly work, thereby offering better qualities when compared with IP networks.

The problems of varieties of interoperability have been presented to vendors of VoIP, and market pushes hard till these have been resolved. Present interoperability exists at only the level of client software. It enables clients of H.323 for communicating with different telephones by means of an H.323 gateway. No standard exists that allows gateways from various vendors for interoperating, however. This slows IP adoption by telcos. Vendors of networking companies should supply gatekeepers and gateways (software that gives addresses translations and call controls), which is interoperable. In the long run, providers of service shall not deploy proprietary solutions in larger scales. Gateway interoperability has not been addressed fully by the H.323 protocol. But, it is expected to become accomplished by extensions of protocol like Media Gateway Control Protocol (MGCP). The following figure 3.6 shows the protocols that form the core networks:

Figure 3.6: MGCP, SDH

MGCP is utilized to control gateways of VoIP from call controlling external elements known as controllers of media gateway or call agents. Call control intelligences are outside the gateway of VoIP and are to be handled by call agents. In the model of MGCP, gateways focus on the audio signal function of translation, while call agents handle call processing and signaling works. Call agents implement the H.323 standard’s signaling layers presenting it like gatekeeper of H.323. Lots of ventures address interoperability at such levels, with Lucent alliance for developing gatekeepers that enable interoperability’s between gateways of varied vendors.

The trouble to integrate gateways of VoIP with existing routers, PBXs, and switches often results in complex configurations. It is due to the lack of a closed architecture and experienced technicians of most networking elements of PSTN.

The community of VoIP has not agreed upon the Call Detail Record (CDR) that is common. This creates a possibility for one provider of network to settle and access with another owner or giver of network. Without common CDRs, the area where ITSPs operate are limited to place where there is their own gateway of IP. The following figure 3.7 shows the architecture of the

Figure 3.7: CDR Standardized Operations, Administration, Maintenance, and Provisioning (OAM&P) and billing applications are not present. This forces users to depend for complete solutions on a single vendor. This is a greater restraint, if especially proprietary applications are not integrated with operations supporting systems that exist. The following figure 3.8 shows the data flow of OAM&P:

Figure 3.8: OAM&P Scalability is complexity here due to numerous requirements in this stream. Port densities of all gateways are increased easily, as can the overall user number due to internet’s distributed nature; however, sufficient element, business, network, and service management processes and tools have no availability, making general operation’s scalabilities too troublesome. When technical difficulties are overcome, acceptable voice qualities are available, probably exceedingly that of PSTN. There is VoIP’s widespread availability with no subscription toward the basic package of internet. Truly interconnected and scalable services are available. Too much of maturing and development needs to happen prior to this becoming reality.

Independent providers of VoIP market

Telcos entry and numerous newer entrants to VoIP industries result in expectation that there is some amount of squeezing of market with maturity of the organization. The ability to survive here in the market is, in larger part, determined through a competitive provider strategy, with its ability for rapidly changing in dynamic place of market. The following figure 3.9 shows telcos and their relationship with the VoIP gateways:

Figure 3.9: Telcos

As in whichever market, ones growing larger enough for generating scaling economy is in position for surviving, than ones finding niche markets for serving. To the benefit of independent givers, time for marketing is too short for services and products of internet. Innovation in old

telephonies is not a priority because most providers have a protected monopoly with newer services not been deployed promptly and generally. Givers of service of information do realize how integral it is for bringing services to markets quickly, because of competitive advantages. The ability for doing this is seen as greater strength over in such a market. Dynamism with respect to a business process and technological upgradation is similarly becoming advantageous.

Such companies, though young, have a packet networking business and accumulated experience in these areas. Telcos find it critical for transferring their understanding to packet switching. ITSPs that are independent have willingness to overtake risks because they have small things to lose. Facility-based providers like next-generation telcos and telcos in turn have invested significant sunken fees in infrastructure, and so are more conservative while making activity choice. Once investing little and fewer, more than few venture capitals take great risks. There are a huge numbers of newer entrants in market. There is a great potentiality for joint venture among such organizations for increasing technical knowhow, geographic scope, business understanding, and customer base. There is an opportunity with suppliers to partners, bringing high reliability and billing experience. Older suppliers have a relationship with telcos. But, vendors of information incorporating capability of voice into items want partnership with industries focused on giving services of VoIP.

There is opportunity for causing disruptions to traditional industry of telephony with negative effects on incumbent telcos. The telcos not yet reacting to VoIP leave themselves vulnerable. This is the case in the long term.

There are several disadvantages in becoming independent, small provider, in spite of the aforementioned. Youth have led for strengths like willingness and dynamism for taking risks, thereby yielding in overall experience lacking. An experience of business needs accumulation over the span of time, and in turn to bill systems with need of development of sustainable models of business. Most ITSPs, which are independent, have no or little experience of voice and much of knowledge of data. Though few packets switching understanding shall transfer to voice from information, there is no assumption that such knowhow shall suffice when one deals with whole arrays of newer services. Dominant carriers have rapport with established organizations and have a feeling to conduct business with these. Companies wish to operate business along with different industries of similar sizes, and this is a disadvantage for independent givers in areas of bundling and inter-connection of value-added services. Small providers cannot work without the cooperation of larger carriers. To some degrees, they should accept market instead of shaping it. Regulation is regarded both as an aid and a threat to whichever service givers of internet. Lack has allowed flourishing of the internet, suggesting that whichever regulatory intervention sort stifles growth to few degrees, while putting small providers at

disadvantage. General regulations of telecoms have provided advantages for smaller givers and threats. Incumbent providers pose threat to ITSPs that are independent. Most do realize IP potentials and have accordingly acted. They invest in equipment of VoIP, and few have started already for offering services of VoIP. VoIP is becoming huge and is rapidly growing. There are many possibilities for start-ups and companies that are established. All these markets offer potentiality of short term and market squeezing of long term, sympathizing with ones that have been successful in strategic relation and positioning, both business-and customer-wise.

Conclusion

This chapter explains the various features related to VoIP like packet switching and circuit switching. UDP is a Transport Layer Protocol. TCP needs lesser overheads, which have not been conducive to real-time traffic and possesses no voice transport implementation. Networking company vendors must supply interoperable gateways and gatekeepers. In the long run, service providers must not deploy solutions of proprietary in large scales. Gateway interoperability has not been fully addressed by the H.323 protocol. It is expected for becoming accomplished by extension protocol such as the MGCP. Billing applications and OAM&P standardized do not have existence because they force the utilizers to depend on single vendor for complete solution. This is restraint in case proprietary applications especially have not been integrated with the supporting system’s existing operations.

The next chapter deals with voice encoding and voice detection.

Points to remember

Accumulation delay, propagation delay, network delay, and processing delay are different types of delays in IP networks.

Packet switching technology uses infrastructure with greater efficiency than circuit switching technology that is used in older telephonies. In forms that are unencrypted, speeches being carried over the internet have been truly and considerably harder for wire-tapping, when compared with analog speeches that are brought over the copper wire pair.

VoIP is not a convenient mechanism of telephony. It requires awkward dialing with cumbersome factors.

Telcos entry in industries of VoIP result in expectation that some market’s squeezing has to take place with organization’s maturity.

Multiple-choice questions

What are the disadvantages of NAPs? NAPs are congested

NAPs do not have funding

Private exchanging points are more None of the above

Why scalability of OAM&P is complex?

Standardization is not available

Billing apps are not available

No proper integration with supporting systems

All of the above How are independent givers of market of VoIP benefitted?

No benefits

Ability to survive is determined by competition

New entrants provide benefits

None of the above

Why there is a greater joint venture potentiality among these industries?

Basic internet package is available

There is no competition Scalability is available

None of the above What are disadvantages in becoming small independent provider?

Insufficient knowledge Cannot take risks Sustainable business models cannot be built

All of the above

Answers to MCQs

Q1: A Q2: D

Q3: B, C

Q4: A, C Q5: D

Questions

What does UDP and TCP need? What is the purpose of MGCP?

Why CDR is important?

Why OAM&P and applications of billing do not exist?

CHAPTER 4 Overview of Systems Level

Introduction

Asystem-level VoIP overview consists of POTS networking by Digital Subscriber Line ( DSL ) modems or cables. VoIP processing supports successful transmission of dual-tone multiple frequencies, which have presence in bands. Vocoders with compressions distort such tones, giving rise to an echo. Voice encoding is important for converting analog signals to voice packets. Voice Activity Detection ( VAD ) is incorporated in Codecs. To packetize the voice and fax samples, playout is essential, using a Reduced Instruction Set Computer ( RISC ) Processor. Real-Time Operating Systems ( RTOS ) are the supplementary devices for visual message waiting, call waiting, call transfer, and call forwarding. The combination of analog signal with PSTN voltages gives rise to echo-less transmissions. The uses of Foreign eXchange Office ( FXO ) ports are availability of power, call redirection, and remote calling.

Structure

In this chapter, we will cover the following topics: Plain Old Telephone System (POTS) mimicking

Echo

Voice encoding Voice detection

Playout

Packet network signaling

Supplementary devices

Voice gateway data functions

Objective

Explains the system-level VoIP overview consists of POTS networking by Digital Subscriber Line (DSL) modems or cables.

VoIP processing supports the successful transmission of dual-tone multiple frequencies, which have presence in bands.

This chapter explains about the voice gateways, voice encoding, and voice detection methodologies.

Overview

While the products of VoIP are deployed in market for more than seven years now, service providers’ recent announcements like Sprint, Vonage, and AT&T have created activity flurries by those equipment manufacturers of consumers who race for rolling out solutions of gateway of residential VoIP. Such lower-costing devices usually are standalone boxes, which provides functionality of VoIP for the POTS via broadband modems DSL or cable usually. It serves as a bridge between the analog/TMD POTS world and as the Internet Protocol (IP)-centric world based on an internet packet. The following figure 4.1 shows the basic structure of POTS networking:

Figure 4.1: POTS

The following figure 4.2 shows the echo dealings via a digital subscriber line :

Figure 4.2: DSL The following figure 4.3 displays the encoding of voice via a Time Division Multiplexing IP gateway:

Figure 4.3: TDM IP Gateway

As with many customer commodities, their designers usually meet aggressive product cost targeting, together with tightened schedules of development. A shopping list of solution often features those factors that involve not only the specified basic functionality of VoIP gateway but also different ancillary functions. It includes routing and bridging of data, as revealed in a common router product that is residential, voice that emerges, and features of signaling securities such as IP Security Protocol (IPSec) and voice encryption, and particularly of QoS points that is necessary for maintaining and troubleshooting services of residential VoIPs. The article is fourth in the series that is intended for assisting engineers to provide the detailed designing considerations for all key VoIP solution of residential gateways. The part serves as an overview that is functional of the core components required often in the present day’s VoIP gateways that are residential. There are details about the important elements that necessitate to quality voice call over IP and highlight some of factors of designing of circuitry of telephones. It is explored in detail in subsequent articles. An overview of QoS, security, and data routing elements of monitoring is provided also. It does serve as an introduction to a detailed analysis of designing followed in the articles that are subsequently done.

POTS mimicking

Because the majority of the current day’s installed handset base is still a type unit of POTS, most VoIP calls do originate from the Customers Premise Equipment (CPE) that comes from POTS phones being connected to the gateway of voice. The unit at the remote locations can be another device of VoIP CPE or POTS phone simply connected with a PSTN. The flow of VoIP sample calls by means of a network has been depicted. The following figure 4.4 depicts voice detection by the Customers Premise Equipment which is the conjunction of POTS with voice gateways:

Figure 4.4: CPE

For any system of VoIP to become successful, it should first show equivalent experience that the end users have been accustomed with current systems of POTS. For replicating the PSTN user’s traditional interfaces, the functions of detection and generation of tone are necessary. The digits that are dialed should be correctly replayed and collected at the receiving end to have call executions successfully conducted. Tone detection or generation is not only the core at call beginning, but a feature that is driven by the tone such as functionality of calling cards, and voice mails must have tone generation or detection as handled as mid-calls also. So, processing of VoIP supports the ability for successful transmission of the Dual-Tone Multiple-Frequency (DTMF) tones having presence in band. Vocoders with compressions distort such tones. For avoiding potential distortions, the designers require to bend toward advanced techniques like the Internet Engineering Task Force (IETF) requests to comment RFC2833 when tones are passed in conjunction with the usage of vocoders of a lower bit rate. Performance of generation of tone is important to give call progressing tones and depressed playback of tones. The abilities for detecting tones with proper processing of switch for data modem and fax signals is a requirement too. The reason is that, all telephony types available currently on the PSTN should be supported. The following figure 4.5 shows the DTMF which is a type of a tone:

Figure 4.5: DTMF

Echo dealings

To combat difficulties with an echo is an integral VoIP adoption element in the traditional world of telco. With a replacement of the system of PSTN by VoIP, robust techniques of cancelation of echo are taken into consideration for meeting the packet network’s demands.

An echo has its availability in the POTS conventional networks, and the PSTN employs cancelers of echo throughout the system. Line echoes are caused when the connections involve conversions in between four lines to two lines telephony hybrids. An echo has been generated toward the packet networks from the telephonic networks.

The specifications of the PSTN dictate that working of cancelation of an echo is a key factor when the delay exceeds 50 milliseconds. The IP networking portion of a solution of a VoIP always adds to be more than 50 milliseconds of round-trip delays. So, the cancelation of line echoes is important when solutions of VoIP interface to the PSTN. For doing this, echo cancelers compare the voice messages received from the packet networks with the voice information being transmitted to the packet networks. The echo from the telephone networks is removed by the digital filters on the transmitting paths into the packet networks.

The tail length of echo cancelation, that is, length of an echo needed to be canceled by a processor, does vary among the varied applications of the VoIP. The requirement of tail length is determined by the distance between the four-to-two-line hybrids and the equipment of the gateway (residential gateway). It typically ranges from 8 milliseconds size of tail for a Small Office/Home Office (SOHO)/residential applications to 32 milliseconds. It is the case when compared with 128 milliseconds sizes of tail for applications of a carrier.

Line echo cancelation is needed because most calls of phones that are established through the VoIP residential gateways shall, at some juncture, terminate to the equipment of the PSTN. For the gateways based on the residential VoIP complexes, the typical length of 8 milliseconds to 16 milliseconds cancelation of echoes and capability of length of tail is sufficient usually. As a benchmark of minimum quality, the functionality of cancelation of an echo is compliant to standards of G.168 and ITU

Encoding of voice

Voice encoding is very important for converting the analog signals to voice packets. It includes compression for reducing the 64 kbit/s stream that is produced by encoding the stream of G.711PCM (utilized by most PSTN traditional trunk lines) to lower bit rates for transportation having more efficiency across both the subscriber’s and the network’s links. Vocoders used in systems of VoIP presently involve G.723.1 and Vocoder of G.729ab does offer rates of data as much lesser than 8 kbit/s and G.723.1 at 6.3 kbit/s and 5.3 kbit/s.

The tradeoff between such lessened bit rate vocoder and G.711 has got reduced utilization of bandwidth versus slight high quality of voice. G.729a is an optimized implementation of too common algorithm of compression of voice of A key notable factor is that the base algorithm is G.729, and G.729 has interoperability with

Voice detection

Voice Activity Detection (VAD) in relation with silent suppression, as incorporated in the codecs or as an external operation of software, must be helped as configurable (disable/enable) features in designs of VoIP. Monitors of VAD receive signals for the activity of voice. When no activities are detected for a specific time period, the software prevents packetization that is unnecessary with the subsequence of silent transmission. The work measures the characteristics of idle noises of the telephony interfaces. The measurements of noise are relayed subsequently to the receiving gateways. Comfort Noise Generation (CNG), which is the playout of the background lower-leveled noise to receiver, has been recommended for confidence of a user in a call connection. In the case the call is too quiet, then the users anticipate that the call is disconnected. The following figure 4.6 shows the Voice Activity Detection mechanisms.

Figure 4.6: VAD

The residential gateways assist the usage of techniques of fax relay. The fax relay does offer bandwidth reductions and a more reliable, robust means to connect the fax over calls of IP. It is a well-known feature for the equipment of the Server Message Block (SMB) and SOHO. The use of a fax relay does include demodulation of the facsimile scanning data, IP packet encapsulation, and fax IP packet subsequent demodulation at the receiving gateways. It needs support of the fax protocol of T.30 that is implemented between the VoIP gateways and the fax machines and T.38 fax encapsulation of IP packet for transmission of IP. The following figure 4.7 shows the feature implementation of the Server Message Block

Figure 4.7: SMB

Interfaces of POTS

The VoIP residential gateways interface with the traditional equipment of telephony via the signaling of an FXS to interface of the Pulse Code Modulation Such an interface receives the samples of PCM from the interface of the analog codec and then forwarding them to the apt functions. Conversely, the interface forwards the PCM-processed received samples from the DSP to the digital interface. The PCM interface does perform the output sample’s continuous re-sampling for avoiding slips of sample.

Playout

When the fax and voice samples are processed, they should be packetized. The systems of the VoIP do employ Real-Time Packets (RTPs). On the receiving side, the playout unit of voice becomes integral for buffering the received voice packets and then forwards this to the vocoders for playout to the users. Such playout units serve as jitter manager/buffer for queuing lots of packets, and then to avoid packet over-run or under-run. The following figure 4.8 displays RTPs, which are used in playouts:

Figure 4.8: RTP, IP WAN

Feature implementation

Features are implemented typically in software, on Digital Signal Processing usually. They are implemented in the RISC processor. It is advisable when the additional requirement of processing for the typical functions of the RISC is minimal. For architectures of the VoIP with RISC only, the availability of cycles of CPU (MIPS) is managed carefully between the telephony/network signal processing and voice processing. The working shows the tasks of the telephonic signal processing that is needed to help the media streams of the VoIP. The gateway of the VoIP assists the protocols of the signaling, both for the packet side and the telephony side of the gateway. The following figure 4.9 shows an which is a processor:

Figure 4.9: RISC processor

Telephonic and packet network signaling

The translation of a packet’s telephony signals in only one part of the gateway solution of the VoIP exists. The gateway must support controlling the signals of telephones like off-hook and on-hook functions and protocols or network control signals like Session Initiated Protocol (SIP). Both of these place different demands that are on the host processor with its software. There are comprehensive sets of operation of processing, which execute typically on the processor of the RISC that translates the telephone’s protocols/signals to the protocols of packet and viceversa.

PSTN interfaces and analog phones

In numerous applications, state machine based on software serves as a controller of calls which then handles such functions. Call controlling executes the key functions by means of stage of all calls in a gateway. Other critical signaling process elements are network stack of the protocol itself. The SIP is a very well-known protocol for the market of the residential gateways. But, there are a few deployments of H.323 and MGCP. The following figure 4.10 shows an analog phone with the PSTN interfaces:

Figure 4.10: Analog phones, PSTN interfaces

Device provisioning and supplementary services

The gateway of VoIP that is utilized in the residential applications needs the assistance of the functionality available typically in today’s phone services. It involves features like indicator of visual message waiting, call waiting, call transfer, and call forwarding. A software is needed for interpreting such commands from the network, and then execute the functions by virtue of a gateway to the telephone. The remote devices in a network that provide service residential gateway should possess the ability to become configured either remotely or on premises, but not requiring separated monitors or various equipment. The configuration needs software in the gateway for processing and accepting provisioning. It is desirable that the user interfaces must be easy and simple to use. The provisioning software is not to be insignificant. A preference in the market that devices of the residential VoIP gateways have the capacity for program load (dual image) storage, the programming updates are downloaded with no deletion of current images prevails. This possesses an impact on the overall software design of program and requirement of SDRAM and FLASH. A comprehensive view of all the functional blocks that are required in the complete system of the gateway for the VoIP-based residential complexes is there. Additionally, Ethernet interfaces, IP stacks, device drivers, and RTOS must not be disregarded. The following figure 4.11 depicts an RTOS:

Figure 4.11: RTOS

For connecting the reliable old analog phones to the voice gateway, a Foreign Exchange Station (FXS) is required. In some applications, for the PSTNs utilized outside callings, a connection of the Foreign Exchange Office (FXO) is needed. The FXS consists of 2 components, SLIC and codec. Codecs are comprised of DAC and ADC. ADC is utilized for converting the analog signals from the analog phones into the digital signal such that this is transmitted onto the network of the VoIP. The DAC converts the digital signals to the levels of the analog for driving the analog phones. The sampling rates for the DAC and ADC usually are in the range of 8 kHz for achieving audio bandwidths of 4 kHz. A Subscriber Line IC (SLIC) device does emulate the voltage levels of network of the PSTN. There are necessities to detect off-hook, on-hook for generating the ringing voltage that ranges to 120 V. The main conduct is combination of the analog signals with the voltages of PSTN. The following figure 4.12 shows the FXS and FXO:

Figure 4.12: FXS, FXO The following figure 4.13 tells us about a SLIC:

Figure 4.13: SLIC When the CPE voice gateway device requires connecting with the local company of the phone, there is a need to interface the FXO. The FXO does consist of Data Access Arrangement (DAA) and

codec. A codec has similar task as in the FXS, while the DAA conducts emulation of the POTS phones. Its core operation is removing the higher-voltage DC bias with only passing of the analog AC signals when the system of the PSTN comes, by application of loop closures to the PSTN. The uses of ports of the FXO are as follows:

Lifeline for failure of When used, there is no availability of power to the voice gateways. This prevents the calls to connect to the packet network. In such cases, the analog phones (by means of FXS) directly connect to the port of the FXO through relays. Redirection of It is used in case when the subscribers dial numbers that are unreachable via the packet networks. For completing the calls, the voice gateway’s redirect the calls through the FXO port. For the gateway that is user-friendly, the CPE device does dial the digits to the port of the FXO. It prevents people to have redialing of the numbers. VoIP remote When customers do not dwell at home, they still can make VoIP calls by calling the home numbers by the virtue of the network of the PSTN. Voice gateways pick-up the calls via the FXO port and then forward this to the network of the VoIP.

Data functions of the voice gateway

In applications of residential CPE, the voice gateways are connected normally on the LAN side of the broadband modems. If a household has one plus PC, the voice gateway becomes a standalone device that terminates the connection of IP by connecting to the hub or the router. If the home has a single PC, then an introduction of the voice gateways includes the purchasing of the hub or router and creation of the home networks. For making easy packet voice service adoption, the most appropriate voice gateway configuration is the inclusion of the applicability of the data routing to connect with different PCs. With help of this way, the PC connects to the side of LAN, and the modem connects to the broadband modem’s WAN side. In this configuration type, the voice gateways do involve the functionality of routing of information. While deciding on the performance and functionalities of the informational functions, there is a great usefulness to know the configuration (arrangement) and application of the connection of broadband. With the VDSL exception, most broadband residential modems possess the capacity below 50 Mbit/s. So, while one designs the voice gateways, there is a necessity to be aware of the end user applications for determining the appropriate performance or price goals. Few data workings that must be involved are as follows:

TFTP

Routing

Static and dynamic NAPT and NAT

PPPoE

DHCP server/client

Firewall

When people include these useful and popular functional transitions to the VoIP, then the VoIP does become easy for the consumers to connect this with their broadband service. Further, there are incorporations of the voice gateways into the hubs or routers’ lower ownership costs, while there is no requirement for the individuals to buy separate boxes. When the households purchase other PCs, switches are bought to the network voice gateway and PCs together. The following figure shows the comparison of the TFTP with FTP:

Figure 4.14: TFTP The following figure 4.15 shows the structure of the PPPoE client:

Figure 4.15: PPPoE

The following figure 4.16 shows a DHCP, which connects through the Internet through a call center gateway:

Figure 4.16: DHCP

Elements of security of VoIP

The secured communications of the voice receive a great amount of attention by the providers of a service to deploy the VoIP residential services. The VoIP implementation that has security leverages many security elements to already establish it for communication of data. One of the key operations of the current internet infrastructure of security is to monitor the transmitted data’s integrity. The element covers the recipient authentication and assurance that the messages between the two entities are not tampered with. The same element is for supporting nonrepudiations, that is, rejection of the message that is digitally signed by security keys. The confidentiality levels of security of the internet do ensure that the message’s transmitter and recipient are only ones who view content of the messages. The authorization functions of the suite of the VoIP’s security element assure the network user accessibility to particular networking services upon verification of the identities only.

Different levels of features of security are needed when one depends on the security levels that are concerned with the providers of service or the end users. One feature is the voice payload encryption itself. Another security level requires the signaling message encryption, which establishes calls of phones.

Putting everything together

There is huge pressure for putting together Gateway at lowest cost that is possible, but it is imperative that the selected components achieve performance and optimal quality. Telephony and network signaling, Ethernet interface, voice processing, and POTS interface are the least functions needed for developing such gateways. There is an essentiality for understanding the supplementary service types and provisioning function to the extents needed for ensuring that the product has been finished. The requirement for programming and regional considerations, are in place thereby ultimately dictating cost of interface of the POTS.

Additionally, the residential gateways needing advanced features, like the functionality of routing of data or authentication or voice encryption, need additional processing powers. When these factors are included, the designers should take care that there is an assurance that the proper amounts of processing powers with sufficiency of flexible architecture are present for supporting these above-mentioned requirements.

Conclusion

This chapter deals with the voice detections and modeling. Recent announcements by the provider of service have created flurries of activity by a consumer’s equipment manufacturers who in turn race to roll out the residential VoIP’s gateway solutions. These low-costing devices are boxes that stand alone with the provision of the VoIP functionality via the DSL for the POTS. It serves as bridges between the TDM IP gateway on the basis of the internet packet. The VAD monitors to get a signal for voice activity. When no activity is detected for a particular time period, then the software avoids unnecessary packetization with silent transmission subsequently. The CNG is background’s playout in the lowerleveled noises to the receivers and is recommended for user confidence in call connections. The device of SLIC emulates the PSTN network’s voltage level. It detects on-hook and off-hook to generate the ringing voltage, POTS phone’s emulation. Its DC biasing with the passing the PSTN system comes, by

ranging to 120 V. DAA conducts the key work is to remove high-voltage of an AC analog signal only when loop closure application to the PSTN.

The next chapter deals with telcos and their SWOT analysis.

Points to remember

For replicating the traditional interfaces of users of the PSTN, tone generation and detection function are important.

VoIP processing supports the ability to have a successful DTMF tone’s transmission that possesses the band presence.

Line echoes have been caused when the connections do involve the conversions in between four-line to two-line telephonic hybrids.

VoIP systems employ RTPs because when the voice and fax samples are truly processed, they must be packetized.

Gateway supports the controlling of telephone signal such as onhook and off-hook functions, and network or protocol control signal such as SIP.

Multiple-choice questions

What is the purpose of voice encoding? Convert digital to analog

Convert analog to digital

Convert analog to voice Convert voice to analog

What is use of a fax relay?

Bandwidth redirections

Reliable method to convert the fax over IP

Demodulation of the scanning data

All of the above What is the work of an RISC processor?

Integrate the telephone service into IP networks

Single silicon device is used

Cost reduction is carried out

All of the above

What do supplementary services and device provisioning do?

Message waiting Call waiting

Call forwards None of the above

What does the TFTP use? Uses simple control commands Does not support user authentication

Requires less memory All of the above

Answers to MCQs

Q1: B Q2: D

Q3: D

Q4: B, C Q5: D

Questions

Why the POTS, TDM IP gateway, and DSL used? What do you mean by the CPE and POTS mimicking?

What does the CNG and VAD do?

How are the FXO and FXS required? How SLIC and DAA work for us in real life?

CHAPTER 5 Interfaces of VoIP Telephony

Introduction

Telcos take the IP telephony challenge very seriously. It is a threat and an opportunity. The SWOT analysis is as follows: For strengths, we have, telecommunications, which become largely custom-centric. Even the start-up companies have large venture capitals and technical knowledge. Over-expectations is one of the weaknesses of telcos. The rates are also very marginal due to cross-subsidization. The cannibalization of ideas is also another weakness. So, unprepared telcos have a less profit margin. The opportunities are the network backbone, local loop, and leased lines. An online one-stop shopping destination, online banking, phone and internet access providers are the other opportunities. Huge competition is a major threat among basically the VoIP start-ups. The PSTN interfaces consist of two parts: a codec and a Subscriber Line Interface Circuit ( SLIC ). The codec consists of the DAC and ADC. The FXS is mirrored by the FXO. Various standards are GR57 , TA909 , GR303 , and GR1089 .

Structure

In this chapter, we will cover the following topics: Mobile service provider and the VoIP

SWOT analysis of telcos

Analog phones and PSTN interfaces Standards

Objective

Telcos take the IP telephony challenge very seriously. It is a threat and an opportunity. The SWOT analysis is as follows:

For strengths, we have, telecommunications, which become largely custom-centric. Even the start-up companies have large venture capitals and technical knowledge. This chapter deals with SWOT, standards, and mobile service providers of the VoIP.

VoIP and mobile service provider

Telcos is beginning to take the IP telephony challenge too seriously. It is seen as a threat, but it is a huge opportunity too. The incumbents have a natural tendency for working on the improvement of operations and technology that is existing within those business models that exists instead of embracing newer industrial and technology modifications not desiring to cannibalize that business that is lucrative. Telcos own the infrastructure of local loop and tend toward owning of the backbone networks. So, there are those opportunities for these to leverage revenues from all parts of the networks such as either the provider of capacity or the giver of retail services.

IP gives new organizations one chance for entering the markets to offer services posing no major troubles for the existing old operators in telecommunications. Technologies have progressed and matured, so a new range of applications that are available have rapid threatening of the older core business operators. The most notable is the VoIP of course.

The VoIP has been originally regarded too unreliable for the development of mass market, giving the new entrants cost-efficient and easy ways for competing with operators, which is incumbent

by means of the current pricing arbitrages that are associated along with communications of internet. With such pricing benefits, VoIPs offer platforms for data and voice integration, and rich applications with increased functionalities. Opportunities presented by the internet have resulted in numerous newer entrants, with no traditional investments of business for protecting and wishing the exploitation of the VoIP commercial possibility. Short-term economic advantages are present for allowing the entrants that are new for building up consumer bases with knowhow (business and technical both) in turn to leave these for gaining potentials for long term. Telephonic operators, which are incumbent, require consideration very seriously with respect to the IP and related services. VoIP experimentation has begun already while to tackle problems for under-cutting of own costs prior to the conducts of different ITSPs. The incumbents who think for future are quicker for responding to such challenges. The incumbent telco of Germany, namely Deutsche Telekom, has been the largest ISP of Europe. The high numbers of newer entrants’ plan to utilize the VoIP has bought VocalTec stake. They have started offering service of the VoIP, namely T-NetCall. Industries have expanded Germany’s service outside installing gateways in the US, the UK, and Japan. They are taking benefits of full liberalizations in the market of Europe. In the US, the MCI has been most notable in terms of traditional operators that enter the race of VoIP. It has launched already the technologies enabling information and integrated voice transmission to call centers. Lots of services are planned for future because the MCI continues expansion on its telephonic revenues.

Large operators in the US seem excited about IP possibility. They are yet reluctant for offering services in their own domain. AT&T has offered phone-to-phone commercial services in place of AsiaPacific (in Japan particularly) for nearly one year prior to starting of commercial trials in the US. A Japanese venture of AT&T does include 27 Japanese organizations offering the residential and business service of VoIP throughout the nation that has callings being offered at nearly 80% cheaper rate than the standardized services. The international and long-distance tariffs in the area of AsiaPacific are high generally, so the volume of calls is expanding rapidly, and many new entrants have been trialing the VoIP in this. It results in response by homely incumbents that offer these services. Companies of Japan have started offering calling cards that are prepaid toward phone-to-phone IP and FoIP services. Australia Telstra is under pressure for offering services of the VoIP between London and Sydney.

Telcos in VoIP

The market has evidence that the incumbent telcos move into market of the VoIP and has significant disadvantages and advantages to enter the market neighboring itself. SWOT analysis is done: The following figure 5.1 depicts the SWOT analysis of telcos:

Figure 5.1: SWOT analysis

Strengths

Telcos own local loops owning large internet backbone parts. This has a huge benefit because of the elimination of higher cost percentage. Generally, telecommunications are becoming customercentric to great extents, meaning that the key parts of the value chain are the nearest to people. While one owns local loops, telcos manage to be nearer to individuals, whether they give services or lease infrastructure to others. The incumbent telcos is a telecommunication service provider for a long span of time now, and the result is that there is an accumulated understanding of network managements while dealing with larger bases of customer and outlook of a multinational industry. A general practice of business has no abundance in the start-up companies, and many of these have little greater than few venture capitals and technical knowledge.

Telcos have larger customer bases already, and because of convenience and loyalty, lots of people possess happiness to accept the services of the VoIP from such companies, instead of further looking afield. Known names are there. Huge capital resources, both potential and existing, are excellent for undercutting numerous different organizations on price.

Weaknesses

Weaknesses experienced by telcos while entering the market of the VoIP are due to those expectations that are provided by communications and that are similar across boards. They fail to know that the new technologies and networks that deal with no whole analogy have a familiarity with it. It leads to the usage of business with different processes, which are outdated or unsuitable. Telcos-associated business models have no basis on costs that are marginal, with a provision of the VoIP. The old telephony relying on the accessibility charges and crosssubsidization has universal service provisions. VoIP models are based on a fee that is marginal and that minimally is compared with fixed fees. The abilities of the great incumbent industries do adapt to arguable new schemes because they always have difficulties with rapid conversions. In case telcos manage movements apart from the traditional costing models with low prices to costs that are marginal, attraction by virtue of fee arbitrages is minimized.

Ideas cannibalizing business that exists retard few large operators. The ones enjoying the status of monopoly in the past suffer at the hands of the new entrants, with a version of shortening reduced short-term margins further, if even it results in a downfall that is long termed. The regulation (or lack thereof) of the VoIP has to be questionable. Numerous telcos have attitude that regulators deal with the ITSP threat. This results in unprepared

telcos for coming markets, which are left to play catch-ups when they do realize the VoIP uptakes extents.

Opportunities

Telcos that enter the market of the VoIP possess expertise of many opportunities of vendors, and ISPs like acquisitions and mergers of medium-and small-sized companies. Alliances and deals are formed for generating economies both of scope and scale. Lots of such takeovers and joint ventures have evidence already because telcos visualize it as easier way to enter the market of the VoIP and simultaneously acquires the technical expertise and consumer base. Larger telcos that own much of network have opportunities for leveraging revenues at all of the network points: backbone, local loop, leased lines, and to avoid payment to others for carrying traffic. This appears that it is the greatest telcos benefit in the market. With breaking down of the NAPs, an agreement of interconnection is commonly placed. Standing for gaining most from the operators and owners of networks of this type is there.

As they have the telecommunication industry’s experience, and because they offer services that are bundled (shopping one-stop like online banking, phone, and internet access), they have an ability for squeezing much of competitions on service and price offerings. Small providers, who are in primary position for niche service markets modifying rapidly, are not simply in position to offer bundled services with no large operator cooperation.

Threats

Telcos enter the market of the VoIP at high levels. Therefore, there is competition not only among themselves, but also from players that are established. These players have an understanding about the customers and market. It is unlike telcos that make mistake to think that their knowhow shall transfer to such markets. The accumulated knowledge of larger ITSPs and ISPs needs to become learned by telcos. While the opportunity for telcos expansion, with respect to both expertise and size, exists in the form of mergers and acquisition, similar activities have to occur among different ITSPs that make surviving organizations more dangerous.

The emergence of telcos of next generation has shown necessities for such strategically focusing. Telcos that give those services that are bundled lack these dedicated heeds. The result is that there is no ability of giving services to customers adequately. The regulation in such a scenario is debatable highly. So, it is regarded as a threat. The three basic strategies adopted by telcos in the VoIP are as follows:

First, it does involve to stay, hoping that the phenomenon of the VoIP shall never cease to exist. It was until recently the general factor among telcos, expect that the voice quality has no sufficiency on such geographically distributed networks that are packet switched like the internet. Expectation was also that the differentials of price shall dissolve eventually. This may prove to be true, but arbitrage of the pricing fuels adoption of the VoIP

currently is there. The availabilities of enhanced functionalities take over because the costs of the PSTN fall.

Internet voice quality eventually exceeds that of the PSTN. Various current advantages of the PSTN, like intelligence integration into network, have already entered the internet. Nortel Networks have announced SS7/IP next-generation application of a signaling gateway, which delivers largely scaled, carrier class integration of the PSTN/IP.

The preceding approaches maximize revenues in short terms, but as costs are forced down because of competitions from VoIP providers and new entrants, revenues similarly slip. At a similar time, innovation givers gather the establishment of credibility and base of customer. Operators desiring to enter IP arenas, at later stage, find themselves to be in the race for catching up with the tarnished image of the brand.

Operators who seek to be early movers need to make difficult decisions for offering services of low prices with the existing business in the competitions that are direct. Strategies take benefit of names that are known with the available capitals for obscuring the competition. Telcos is in a position for offering differentiated services on the internet or the PSTN or different IP networks, depending on the sensitivity and requirement of consumers. They move into new markets with new entrants. AT&T has done it successfully in Japan while knowing about such new services and markets.

The approach does seek revenue maximization in long terms by compromising few short-termed margins. But, market segmentation and differentiated service offerings do not importantly mean that the revenues significantly drop in short terms.

The final strategies involve that operators should look into climbing the value chains by exploring the VoIP services value added and to offer them to corporate people. It includes the movement into any primary realm of value-added provision associated generally with ISPs—web hosting, content, and system integration. These services are too far from the industrial older focusing strategically and core competency. Many fail to become successful in such an area.

Functionalities of the FXO and FXS

When a VoIP system is developed, one key consideration area is the interface to the analog telephone. The designer should have knowledge about the requirements of the telephony existing in the PSTN that are supported in systems of the VoIP. Two very common interfaces to standardized POTS phone are as follows: FXO and FXS. The following figure 5.2 shows the architectural overview of foreign central office:

Figure 5.2: Foreign Central Office

FXO and FXS are common words in the analog telephony world. What are the differences between the two, and why they are core in applications of VoIP to be known? In the traditional telephonic connection over the PSTN, the switch of a telephone-centralized office feeds the battery, and then provides ringing for calls in the phone. The phone itself finishes ring/tip circuit for requesting to answer or service calls from the PSTN. For the internet-placed calls, the circuit of the FXS does emulate the central office switches of the telephone. The residential gateway pretends to become a switch for giving both ringing and battery to phone, and then detecting looping currents. In the FXO circuit, there is an emulation of the phone to provide the closure of a loop with detection of the incoming ringing. The FXO and FXS terminology is derived from the wish to enlarge localized calling areas. Prior to the availability of 800 toll free calling, customers of businesses seeking alternatives to costlier long-distance charges are offered with foreign dialer tone service. The analog and digital carrier systems are created for assisting such service by extending the dialer tones from the foreign central office (FXO) to multiple localized central office sites (FXS). Such application is one among the earlier usages for the interface of the FXO and has the responsibility for till-date existing terminology.

PSTN interfaces and analog phones

The circuit of FXS consists of two parts, a Subscriber Line Interface Circuit and a codec. The codec comprises of the DAC and ADC. In the ADC, there is a conversion of the analog signals that comes from the analog phones into the digital signals for the VoIP network transmission. In the DAC, there is a conversion of the digital signals to the analog levels for driving the analog phones. For achieving a 4 kHz audio bandwidth, the sampling rate for the DAC and ADC is around 8 kHz usually. The device of SLIC emulates the voltage levels of PSTN. It should detect if the phone is off-hook or on-hook and then generate 120 V ringing voltages.

The FXO circuitry consists of the codec and DAA. The codec has a similar functionality, as is present in the FXS. It is the conversion of the analog speech to the digitalized signals and vice-versa. The DAA emulates the POTS phone, and its important function is removal of high-voltage DC biasing, passing only the analog AC signals from the PSTN by further application of loop closures to the PSTN.

FXS is mirrored by the FXO

In the residential VoIP gateways, the circuit of the FXS is the primary interface to establish outgoing calls and to receive calls that are incoming over the packet networks. In application of the central office, two wire SLIC interface on the line card of the POTS serves as an FXS interface. In the CPE applications, the circuitry of the FXS exists in the gateway to give battery currents, dialer tones, and ring voltages, and then detecting loop closures from the phones. A direct connection to the PSTN is not important because this functionality of the switch resides at the level of the CPE. But, there are cases when the PSTN connection is useful for working out the FXO interface. A presentation of similar interface type to the central office like the POTS ordinary telephone with a few improvements is done.

A few crucial usages of the FXO ports include the following:

Lifeline for failure of power

When there are no powers to voice gateways, the gateway does not possess the ability for connecting to the packet networks to receive or place calls. In such a case, relays are utilized to connect the analog phones to the PSTN directly. When such situation rises, then theFXO circuit becomes intelligent, detecting calls that are in progress. It avoids disconnection of calls once powers are restored.

Re-direction of calls

When the packet networks have unavailability because of networking congestion, the circuit of the FXO remembers the numbers dialed by subscribers, then routing calls to the PSTN by means of FXO circuits, for completing calls. This procedure prevents consumers from possessing redialing of phone numbers when the packet networks are down.

VoIP remote calling

When VoIP people are not present at home, they make VoIP calls by calling the home numbers through the network of the PSTN. The voice gateways receive calls by the virtue of the FXO ports and then forwarding it to the VoIP networks.

The FXO and FXS interfaces provide a few well-known functions that are referred like function of The terminology of BORSCHT has been oriented to the functionality of the FXS, while the FXO tends to be the mirror image of little of such functions.

B: Battery feed functioning is found in the interface of the FXS linefeed. The complementary functions in the FXO interfaces are battery sink. The connection is established between the central office ring and the tip that leads by the off-hook FXO relays, with total current limiting given by the FXO.

O: Over-voltage protections are provided by the FXO because of the exposure to conditions of power cross and lightning. SLIC’s inputs of and in the FXS circuits are designed to provide additional protection of over-voltage. R: Ringing is given by the central office, but the FXO has an ability for detecting ringing, and to forward this data. The FXS circuit provides ringing to phones. Low-voltage ring signals

generated either by SLIC or codec are amplified by the SLIC and then placed on local loops for ringing phones.

S: Signaling refers to the ability of the FXO for receiving the ON/OFF hook data and then present the off-hook on-commands to the central office. This must detect ringing with different conditions and transmit such information. The FXS has the ability for detecting the OFF/ON hook states, generating and detecting tones of DTMF, and generating signals for caller IDs. The following figure 5.3 shows the signaling of VoIP camps:

Figure 5.3: Signaling

C: Coding is the functioning of devices of the codec that is a part of the FXO and FXS interfaces both. This refers to the D-to-A and the A-to-D coding of voice signals.

H: Hybrid functionalities are key for good quality of voice and stability and are equally core in both the FXO and FXS interfaces. The following figure 5.4 shows the hybrid functionalities of the head office and branch offices:

Figure 5.4: Hybrid Functionalities T: Test is normally not a function of the FXO because testing and automated maintenances have been given by the central offices. The required tests and diagnostic functionality are involved in SLIC/codec due to bypassing of the PSTN by the circuit of FXS.

Echo

The importance of excellent voice quality and stability is crucial whether the call is made over the packet network or the PSTN. An echo’s potential impact is difficult for functions of both the FXO and FXS interfaces. A special hybrid functionality is needed in two cases within the CPE devices to handle various line impedance in world. POTS ordinary telephones relatively have uncontrolled impedances in between 200 ohms and 400 ohms. As the current to the subscriber from the office is two wires with no added gains, the variations of impedance encountered typically does not affect performances. The issues of a line echo’s and stability can arise when the carrier system utilizes two-to four-wire hybrids of voice frequency (VF) on all ends, with possibility of gaining in four-wire paths, as shown in Figure

Figure 5.5: VF

The result of the line echo is made either from the delayed of transmitted signals of voice into receiving paths at hybrids (two-

wire to four-wire conversion points) or from the reflection in the local loops because of the mismatches in impedances. Line echoes are present always in the PSTN and are necessarily not the problem. The fact is that, a few telephonic transmitted signals are coupled into the receiving paths for generating sidetones. The sidetone allows speakers to hear their own voices in the receivers. Without the sidetones, the speakers do not have surety to hear themselves on the other ends, thereby making awkward conversations.

Excessive line echoes when uncontrolled affect the experience of the caller in the two following ways: Louder the echo, more disruptive it becomes at the time of voice calling. Many times, the lower levels of echoes have presence on lines, although they do not possess detection by users.

Delay length of echoes affects the quality of voices greatly. Such delays represent the time elapsing between when the user speaks and at the instance when she or he hears their own echo. An echo delay of round trip that is more than 25 milliseconds begins to affect the voice quality. An important function of hybrid circuits completing two-to fourwire conversions with factors that are vice versa is limiting the amounts of the outgoing transmitted signals bleeding into incoming receiving paths. As a result of the trans-hybrid imbalances (mismatches of impedance and hybrid component’s imperfection), some amounts of Tx signal’s get always into path of Rx. The following figure 5.6 deals with two-to-four-wire conversions:

Figure 5.6: Two-to-Four-Wire Conversion Line echoes are caused by the mismatches of hybrid termination impedances. An echo is caused by the imbalance of trans-hybrids. If a line has no correct termination with characteristic impedance, echoes are generated. Improper CPE-terminated equipment like modems or phones might generate echoes in local loops also. For getting better knowledge about line echo sources, little background data on PSTN is needed. Telephonic network comprises of two sections basically:

Transport and switching core: This section has the responsibility to route and transport the billing of calls and services of calls. Within such a part, all data and voice signals have to be digitally transmitted by using separated paths for Rx and Tx signals. With the help of this transmission of signals of longer distances, allowing the usage of towers of microwave transmission and

repeaters becomes easy. The following figure 5.7 depicts digital transmission in a classical approach:

Figure 5.7: Digital Transmission

Local loop: It consists of the last mile of copper connecting the centralized office to subscribers. The following figure 5.8 shows the local loop from the head office to sub-offices in an echo:

Figure 5.8: Local Loop

For eliminating unwanted echoes, providers of telephone services install a canceler of an echo in the PSTN typically. To help eliminate echo on the connection of Phone A, an echo canceler needs to have installation just prior to or in #2 Class 5 switch. Echo cancelers are inserted into the four-wire network section. Its work is modeling characteristics of a local loop’s echo section containing EGP 4 and 3. With understanding of the Tx Phone A signal, the cancelation of echoes generated at EGP 4 and 3 before transmission of the Tx Phone B signal back toward Phone A is to be done. In systems of the VoIP, echoes are completely handled in gateway of VoIP for those calls that terminate and originate at the gateways of VoIP CPE’s. For calls that are routed to the PSTN, there is a continuation of handling of operation of echo cancelation by the PSTN.

Standards

Although the placement of calls of the VoIP over packet networks does not need connection directly with the PSTN, there is no preclusion of system designers to meet the PSTN requirements. Designer of the VoIP residential gateway should know telephonic requirement existing presently with application of these to the interfaces of the FXO and FXS of system of theirs. Four primary standards that have essentiality while knowing PSTN needing factors and behavior are as follows: GR57 specifies Digital Loop Carrier’s behavior system from the ring/tip pair at the central office to the ring/tip pair at system’s subscriber end.

7.2. TA909 describes Fiber-to-the-Curb behavior’s system from the digital interfaces at the central office (for North America, it is T1) to the interface of the tip/ring at the subscriber end of system. This system of Fiber-to-the-Curb is a 16 to 24-channel system typically. The following figure 5.9 shows TA909 Fiber-to-the-Curb:

Figure 5.9: TA909: FTTC 7.3. GR303 discusses digital loop carrier behavior systems. Behaviors have been specified from digital interfaces at (T1) central office to the ring/tip interfaces at the system’s subscribing

end. This type of system tends to be larger with lots of tip/ring pairs at the end of the subscriber. The following figure 5.10 shows the integrated digital loop carrier behavior system:

Figure 5.10: GR303 7.4. GR1089 does explain the standard that is environmental for North America’s infrastructure of telephone. It includes specification of the immunity to the EMC, lightning, and power cross. A limit to RF unintended radiation (EMI) is stated too.

With an increasing interest in the service of the VoIP, phones have been designed specifically for calls of voice over the packet networks (that is, IP phones), replacing eventually the standard

POTS analog phones. However, till this time, the VoIP gateway designers consider how their system interfaces to such standardized analog phones. The circuits of the FXO and FXS provide means to these ends.

Conclusion

This chapter deals with the telcos and their functionalities alongside analog phones and standards. GR57 tells about the digital loop carrier’s behavior system from the tip/ring pair at the central office to tip/ring pair at the subscriber end of system. TA909 says about the fiber-to-the-curb behavior system at the central office from the interfaces that are digital to the interface of ring/tip the system’s subscriber end. GR303 explains the integrated system of behavior of digital loop carrier. GR1089 discusses about the environmental standard for telephonic infrastructure of North America.

The next chapter deals with VoIP networking.

Points to remember

Line echo result is either made from “Bleed-Though” delayed of voice signal that are transmitted into paths that have been receiving at hybrids or from reflection of local loop due to impedance’s mismatches.

With the help of digital transmission, the long-distance transmission signal allows the usage of microwave transmission’s towers with repeaters becoming easy. Echo cancelers have been inserted into the network section of four wire with operation and the work of modeling echo section’s local loop characteristics that contains EGP 4 and 3.

Multiple-choice questions

What are internal strengths in VoIP telecoms? Established processes

Intelligent people

No proper staff No understanding of company network

What are the external opportunities in telcos of VoIP?

Cloud computing

Managed services

Mobile device management

All of the above What are the internal weaknesses in VoIP’s telcos?

Lack of strategic planning

Lack of scalability

Finding a model for IT

All of the above

What are the external threats in telecoms of the VoIP?

Hostile security environment Funding levels and priorities

Increases devices to support All of the above

What does the VF do? Removes echo Removes stability

Adds echo Adds stability

What is the function of hybrid circuits?

Complete two to four-wire conversions

Complete four to two-wire conversions Complete 10 to 15-wire conversions

Complete 12 to 15-wire conversions What does a local loop consist of? No function

Remove unwanted echo Connect the centralized office to subscribers

Connect the decentralized office to subscribers

Answers to MCQs

Q1: A, B Q2: D

Q3: A, B

Q4: D Q5: A, D

Q6: A

Q7: B, C

Questions

How BORSCHT works? What do GR57 and TA909 do?

What is the function of GR303 and GR1089?

CHAPTER 6 Assurance of Voice Quality for Voice Over Internet Protocol Networks

Introduction

VoIP technology has widespread deployment, the main reason being assurance of voice quality. Voice quality is voice perception of end user depending on Mean Opinion Scores ( MOS ). Wireless Local Area Networks ( WLANs ) are basically used for better quality of voice. VoIP networking has several issues like management of traffic, failure of network switches, routers, virus attacks, Denial of Service ( DoS ), failure of VoIP gateways and call servers. Delay caused by network endpoint, loss of voice packets due to concealment process, VAD, due to signal dropouts and loss and background noise are the various issues at the time of DoS. A few examples of deployment are described in this chapter which states that the call quality ranges from bad to excellent (1 to 5) respectively with 4 or 4 plus as the benchmark. Active tests based on algorithms like PSQM, PAMS, and PESQ are described.

Structure

In this chapter, we will cover the following topics: Issues of VoIP Networking

Measurement of Quality of Voice Tools

Objective

VoIP technology has widespread deployment, the main reason being assurance of voice quality. Voice quality is voice perception of end user depending on Mean Opinion Scores WLANs are basically used for better quality of voice. VoIP networking has several issues like management of traffic, failure of network switches, routers, virus attacks, Denial of Service failure of VoIP gateways and call servers. This chapter deals with VoIP networking and quality of voice tools.

Voice quality assurance

Primary reason that technology of VoIP has wider deployment, apart from decreasing cost-per-channels of enabling technology like DSP, is ability of system of VoIP to match overall quality of service that is offered by old and traditional voice circuit switched networks. Lots of local incumbent exchange and longer distance providers of service utilize VoIP technology in backhauling portions of its network with no end user’s being aware that VoIP is involved. Such traditional providers of service use techniques for managing service qualities that are developed over last hundred plus years for networks having circuit switching. It is namely customer tracking, trouble network reports and network careful designs. Service providers utilize rules that are well understood for characterizing service levels with respect to quality of voice (on basis of echo, loss and delay) with difficulties in call establishment. Networks have been pre-engineered for offering some levels of service while considering such factors. Then, service provider’s key tool for assessing service quality when network is operated has been based on problematic user reports and general network failure notification of equipment by means of system of

In reality voice quality is quality perception of end user. A true characteristic of network performance does impact quality of voice. Mean Opinion Scores (MOS) metrics measures voice quality’s subjective perception and tools of analysis are utilized for deriving such metrics. But technology of VoIP gets pushed close to edge of network along with phones of IP (wireless and wired) and voice residential gateways. Providers of VoIP service has much more complex timing while assuring voice quality for itself and subscribers for two key reasons. The following figure 6.1 shows the MOS according to bandwidth and complexity.

Figure 6.1: MOS

Lack of controls over underlying transportation Example is that, while proving service of voice from gateway for voice based residential complexes that when attached with other provider’s DSL service or Cable Modem of residential broadband is there. Usage of technology of transport which varies in Instance is, utilization of media of WLAN for transporting VoIP when especially subscribers are moving.

Fortunately, increased processing power in such edge devices that enables these to support high quality service of VoIP in first place also enables them for direct measurement and issue of

troubleshooting with consumer service qualities exists. Service provider’s make in-service calculation of end user experienced quality of voice that uses such information for separating problems with equipment of VoIP from those underlying transports which helps them with more effective addressing issues irrespective of their becoming owner of total network or not.

Issues of VoIP networking

Factors impacting voice qualities in VoIP networks are well understood fairly. While most of such are mitigated with good quality tools of assurance, careful network designs in both VoIP machinery of endpoint and network itself allows these issues for becoming addressed with best balance of cost and effectiveness.

Controlling levels over such factors varies from one network to next network. It has been highlighted by difference between unmanaged networks like internet versus well managed enterprising network. Operational issues of network impact network performances to create conditions affecting quality of voice. Such issues do include:

Management of traffic at time of Denial of Service (DoS)/ virus attacks and peak periods.

Failure/ outage of bridges, network switches and routers.

Outage/ failure of elements of VoIP-gateways and call servers.

The following figure 6.2 show DoS for Virus Attacks and Peak Periods:

Figure 6.2: DoS To scale to too large networks does increase exposure then placing more consideration and importance on implementation and planning that is effective. Following are numerous good points which must be regarded while deploying, planning and designing networks of VoIP.

Delay

It is caused by endpoint equipment (in network too) processing, collection of voice sample set for implementing voice compressions and voice collection (uncompressed or compressed) into networking packets. 400 millisecond or more one-way delays does impact ability for carrying on normal conversation. Delays are mitigated along with efficient networking designs and VoIP gateways. Example is, to prioritize voice packets for minimizing routing and switching delays. But there is selection of appropriate packet lengths to lower delay of packetization.

Jitter

This has been created by variations in delay characteristics of network of packet transports. It is nicely mitigated by jitter buffer adaptive management in packet receiving path for effective removal of jitter before samples of voice have been played out to listeners.

Loss of voice packets

It is caused by processors or packet buffers that are overloaded in networks or receiving VoIP endpoints or by errors of packet bits. It is best mitigated by utilizing concealment process of packet loss as parcel and part of voice compression algorithms for replaying previous voice received or/and comfortable samples of noise till new information is received.

Echoes

It is created by off circuit voice energy at PSTN analog interface (that is in line to telephone). Echoes that are attenuated sufficiently or/and that are delayed by lesser than 15 milliseconds are not noticed. Echoes between 15 milliseconds to 35 milliseconds give speech hollow sounds, while echoes delayed by more than 50 milliseconds are heard distinctly and must be cancelled. These echoes are exacerbated by additional delays being caused by VoIP typically having range of 50 milliseconds to 100 milliseconds. Mitigation needs cancellation solution of robust echoes in gateway between PSTN and VoIP.

Vocoder

Voice qualities are affected partially by used voice vocoders. While PSTN utilizes pulse code modulation - systems of VoIP use widely lower bit-rate vocoders like Most commonly used vocoders have MOS that are acceptable. Vocoders of wide band like G.722 support actually voice qualities on all IP voice networks that are greater than when compared with traditional circuit switching voice networks.

Voice activity detection

Voice activity detection (VAD) is one renowned extension to schemes of coding of voice which further reduces band-widths by elimination of packets containing silence. This affects sometimes quality of call by clipping start of talk burst. Such effect is mitigated by tuning of voice detection algorithms carefully. Different varied factors affecting quality of voice includes: Background Noise, Level Clipping, Signal Dropouts and Loss, Physical Interface (example digital versus analog T1/E1) and Gain Changes and Signal Attenuation.

Issues of VoIP quality in relation with example deployment

Newer networking models of deployment and technologies do cause additional challenges impacting ability of service provider of VoIP for guaranteeing high levels of service qualities in deployments of theirs. Two such instances are where VoIP service providers have no control on underlying transport packet network and usage of networks of packet with high potential delays and losses like 802.11 (WLAN) technologies. Example

Numbers of VoIP service providers (independent) enter market offering customer residential service of voice extremely at low prices. Such providers provide home gateways designed for having connection with internet broadband connection (that is service of cable modem or DSL) and operating gateway infrastructure equipment for connecting subscribers with one another and with PSTN.

Such VoIP’s service providers have complete typical independent broadband provider access such that gateways have no interoperability and interworking with transport networks for allowing assistance for end-to-end Quality of Service Since indeed transport networks include Internet, there are no ways for guaranteeing whichever levels of packet delay, jitter or loss. More aggressive points are taken in infrastructure gateways, and homes for

mitigating possible degradation because such effect takes place. Additionally, it is very crucial that such devices provide robust troubleshooting, and measurement tools for allowing quality issues to be hopefully localized and be known about by providers of service.

Example

Significant progresses are made in IEEE 802.11 and Wi-Fi Alliance to work in groups for adding aware features of QoS to WLAN MAC. It is like accessibility category for handling streaming video and voice application requirements of QoS and admission control policy for ensuring that there is no over-subscription of WLAN channels. But fact remains that media of WLAN has relatively higher delay and loss when compared with wired Ethernet. It is due to:

Changes in path of RF: It is because of moving objects that reflect energy of RF or motions from end stations itself. It means user is driving or walking. Interference of RF: There are interferences from different devices by using frequency bands of WLAN (2.4 GHz for 802.11b and 802.11g) like microwaves and cordless phones. Again, it is too crucial to have a diagnostic solution that is robust in VoWLAN handsets and in overall networks for identifying voice troubles per-call for enabling network operators and givers of

service for recognizing and addressing problems as arising with most effectiveness. The following figure 6.3 shows a VoWLAN Architecture and Data Flow.

Figure 6.3: VoWLAN

Measurement tools of quality of voice

Ability for reporting and capturing events is truly critical for management of performance of network. Such tools are extended for managing voice qualities thereby allowing operations for identifying and correcting problems of network impacting quality of voice. In few cases, problem cause is not determined in real-time in turn needing off line analysis. Information that is captured is reviewed for determining root causes. Too reliable and oldest tool of voice quality is to listen opinion tests in which human listeners rate calling quality in controlled settings. Overall result is compiled for producing a MOS that is based on listener panels that rank quality of calling sample series on scale of to (1 to 5, respectively). Aggregate scores of 4 and 4 plus is regarded toll (that is PSTN) quality. While such a test has disadvantage to be time-consuming, expensive, and subjective for producing, this is recognized like most consistent measurement of available voice quality.

Bulk of subsequent activities to calculate voice qualities are to produce tools and algorithms that measure quality of voice objectively. It is by virtue of direct mathematical calculations on samples of sound, instead of listening tests. These tests are classified roughly as intrusive (or active) and non-intrusive (or passive). Generally, active tests do perform measurement on simulated or test calls with intruding on normal networking usages (or as conducted in lab environment). Passive tests

conduct active call calculation in live networks with no service interruption. Following explores test relevancy in further categories:

Intrusive/ Active tests

Wide range of studies into objected automated testing of voice quality leads to development of lots of algorithms on basis of perceptual modelling. Most widely used are following:

Perceptual Speech Quality Measurement (PSQM): It is automated system of scoring that has been designed for circuit switching networks. Perceptual Analysis and Measurement System (PAMS): There are intrusive assessment tools of speech quality with analysis of end to end degradation of signals that are injected.

Perceptual Evaluation of Speech Quality (PESQ): This is international standard to measure end-to-end quality of voice as per human perception models. It is recent standard to assess voice quality in turn leveraging best of PAMS and PSQM algorithms to support channel errors, voice encoding, time clipping, and jitter and packet loss.

Lot of analysis is done on relative merits of different and these techniques. We can suffice this, and say that while such algorithms are evolved over span of time for better modelling of more situations arising in networks based on packets (example variable delay and packet loss), PESQ is designed for combining

best aspects of ones that are previous for recognition of provided high correlation degree for MOS subjective testing.

Equipment from many manufacturers implementing such algorithms has been used widely for testing VoIP implementation quality at system level, and at component level. But it costs to utilize this equipment for measuring active network performance since traffic producing revenue is interrupted for using it. Additionally, while such algorithms quantify deficiency in quality of speech, no data is produced for helping in identification or localization of root situation causing creating deficiencies.

Passive tests

Differently speaking, passive tests run in networks that live with no interruption of active calls and using statistics gathered often on calls that are active. Therefore, they are embedded actually into machinery in VoIP network of service provider, and at use site. Passive tests, as such can so be utilized at low costs because it eliminates interruption to dedicated equipment of test additional wise, and revenue producing trafficking both. Various tools used at this place are based on E-models. These Emodels are planning tools of transmission meant for accounting numbers of factors of real world to predict network performance. E-Model calculates transmission rating factors, namely R, being calculated as:

R=Ro–Is–Id–Ie+A

Where Ro is Signal Noise Ratio including room and circuit noise.

Is has to be impairment combination occurring simultaneously with speech that includes too lower receiving or sending loudness, quantization distortion and non-optimal side tones. Id has been impairment combinational with delays that involves absolute delays, talker echo and listener echo.

Ie is machinery’s impairment factor because of lower bit-rate vocoders.

A is an advantageous factor accounting for convenience of varied accessibility types to be added nicely. For instance, mobile telephony’s has high factor of advantages than wired telephonies. R factor is related with MOS as described with ITU specification.

Numerous vendors possess those aspects of E-models for supporting real time measurement of quality of calls on basis of information about calls (instance, used vocoder, jitter, and loss of packet) adaptively. Such calculations take those resources that are minimally processed having combination into overall software load of VoIP DSP. With help of this way, they perform measurement on active calls by means of gateways on per-channel basis. These calculations also are single ended because they do not depend on collection of end-to-end data regarding all calls. The following figure 6.4 shows a VoIP DSP which combines the overall software load.

Figure 6.4: VoIP DSP

Solution aspects of practical VQM

To have passive measurement embedded in equipment of VoIP there is representation of just first part of overall solutions for monitoring comprehensive voice qualities. Once calculations are done in VoIP gateway, such data is reported to system of network management in which this is utilized for problem isolation and detection. Different information got from VoIP gateway and networks are used for off line analysis to diagnose root causes of troubles. They are addressed to gateway or network configuration in tweaks. Following shall outline current workings in such realms:

To RTCP XR

Real Time Protocol (RTP) is IETF standard to transport real time data including video and voice in packet networks (involving IP). It includes control part and information part both.

Latter is Real Time Control Protocol (RTCP). The RTCP does offer quality message general feedback in context of multicasting grouping and information for allowing synchronization of streams of multimedia. Extended report extension to RTCP (RTCP XR) defines format to transport data that is gathered in gateways of VoIP in interoperable and standardized format.

RTCP baseline involves receiver and sender reports including little basic messages (like packets lost, jitter, and total packet counts)

about all calls. RTCP XR defines formats for sending reporting blocks utilized for a lot of more detailed data regarding sessions of RTP. RFC3611 does define seven of these blocks of reporting. Two are defined for carrying out summary metrics which is useful quality measurement of VoIP. The following figure 6.5 shows the blocks of reporting:

Figure 6.5: RFC3611, RTCP-XR, R-factor, G.107, Adapter Jitter Buffer, P.862-PESQ

Reporting block of statistic summary

Maximum, minimum, standard, and mean deviation of jitter calculations, duplicate packets and lost packets.

Values of Hop limit or packet TTL by timing intervals (defined by stopping and starting of packet sequence numbers).

Block of VoIP metrics

It includes following five categories of information: Parameters of configuration: Use of adaptive jitter buffers, used threshold and usage of concealment of packet loss.

Packet discard/loss statistics: Burst metrics (gap duration, burst density, duration of burst, and gap density), loss rate and discard rate.

Metrics of call Conversational quality MOS estimates, R factor and listening estimate of quality MOS.

Delay: End system and round trip.

Signal metrics: Noise level, signal level and residual echo returning loss.

Specification additionally defines that there is framework with the help of which various implementation specified reports are defined. To have this data available per endpoint of VoIP has greater benefit in identification of potential problematic areas for summation of networks or even users that are individual clearly.

Comprehensive QoS management and monitoring is needed for services of VoIP. Network that is implemented poorly adversely affects end user experiences thereby impacting broader VoIP acceptance as viable alternatives of traditional telephonies. Standards and techniques existing today for monitoring and measuring quality of voice are there within networking elements of VoIP. Such tools are involved as networks of VoIP are deployed.

Conclusion

This chapter deals with VoIP Networking. Delay is nicely caused by equipment processing of endpoint, set of voice sample collection to implement voice compressions into the networking packets. Jitter is caused by characteristics of delay network variations of transports of network. Qualities of voice are partially affected by utilized voice vocoders. New technological and deployment networking models cause challenges that impact VoIP’s service provider ability to guarantee higher service quality levels in their deployment.

The next chapter deals with VoIP Security and Services.

Points to remember

Metrics of MOS measures subjective perception of voice quality and analyzing tools are used to derive these metrics.

Providers of service use rules that have been nicely understood to characterize levels of service in comparison to voice quality with call establishment difficulty. It is very core to have robust diagnostic solution in handsets of VoWLAN to identify voice troubles in one call to enable operators of network and service givers to address and recognize problems that arise with lots of more effectiveness.

Multiple-choice questions

What does WLAN 802.11, WLAN 802.11g and WLAN 802.11b do? Compatibility with Ethernet

Non-compatibility with Ethernet

Relatively higher delay Relatively lower delay

What are most widely used active/ intrusive tests?

PSQM

PAMS

PESQ

All of the above What are passive tests?

No interruptions of active calls

With interruptions of active calls

Statistics are gathered

Statistics are not gathered

How to report statistics summary block?

Maximum of jitter Minimum of jitter

Standard and mean deviation of jitter All of the above

Answers to MCQs

Q1: A, D Q2: D

Q3: B, D

Q4: C

Questions

What are issues of networking of VoIP? What is cause of jitter, vocoder, and delay?

What are conditions of Quality of VoIP related with deployment example? What is function of VoIP DSP?

What are five informational categories in VoIP metrics block?

What is work of RTCP-XR and RTCP?

CHAPTER 7 Implementation of VoIP Security

Introduction

Security in VoIP environment has authentication, integrity, secrecy, non-repudiation and DoS protection. IETF protocols are SIP, Megaco and MGCP, deploying many methodologies for finding out the fatal flaws. Various services of network are backbone network services and access network services. Retail services are call waiting functions, call direct forwards to voice e-mails and busy tones. Virtual Private Networks ( VPNs ) are the private international networks which uses internet to maintain privacy by means of tunneling protocols. VPN gives private organizations an edge over public infrastructure.

The VoIP security areas are CPE gateway, consisting of two key components, Digital Signal Processing and Micro. Four realms of VoIP security are data streams, call controls, voice streams and configuration. Encryption protocols like Triple Data Encryption Advanced Encryption and Rivest Cipher are most crucial. Various methods of key exchanging are Symmetric Public Hybrid Key and Diffie-Helman key

Structure

In this chapter, we will cover the following topics: VoIP services

VoIP security

Encryption protocols Key exchanging methods

Objective

This chapter explains how to provide security in VoIP environment has authentication, integrity, secrecy, non-repudiation and DoS protection. IETF protocols are SIP, Megaco and MGCP, deploying many methodologies for finding out the fatal flaws.

This chapter deals with VoIP Security, Services, Encryption protocols, and Key exchanging methods.

Security

Security in environment of VoIP is complicated significantly by possibilities that networks utilized are in part, or, all, one global internet. These different attacks are subjected to these factors that are aware off to people painfully. Most protocols of VoIP include sophisticated mechanisms of security on basis primarily of cryptographic algorithms. Thorough outs and ins of those methodologies of security that have been based on cryptography are not stated. A person can see methods that give the following in general:

Authentication: It assures that identity of sender is received.

Integrity: It does assure that messages are not dropped or not changed by third party or additional data are not inserted by receiver.

Secrecy: It pays consideration to the matter that information is not read by those entities that are unauthorized.

Nonrepudiation: It pays attention to fact that senders do not later falsely deny that message was sent.

Denial of Service It pays regards assuring that desired data is received in face of attacks seeking for denying its ability for doing this.

VoIP employs those ways that typically and significantly does not depend on those facilities that are separated or by by These have proven that there is difficulty in delivering and maintaining effective security.

It is crucial to recognize that while modern protocols like Session Initiation Protocol (SIP) and Megaco have core way of security for mandatory implementations written into the specifications, there are still uncommon factors for seeing these implementations and too most common for seeing them enabled. Nevertheless, there are expectations that fully deploy more methodologies of securities in systems of VoIP that shall be present in near future.

Protocols of IETF (SIP, Megaco, and MGCP) deploy generally more or one of developed methods of IETF securities like S/MIME TLS and IPSec To use predefined ways does increase security portability because such mechanisms are deployed widely and are extensively studied. To invent new algorithms and methods is to be fraught with complexities. Lots of security ways that are improved and new are found with fatfl flaws. The following figure 7.1 shows Megaco which is a protocol of IETF.

Figure 7.1: Megaco

The following figure 7.2 shows S/MIME which is a security method of IETF:

Figure 7.2: S/MIME (RFC2632/2633) Important recognition is that mechanisms of securities are deployed by or to In former, methodologies are to be deployed between entities that are adjacent. Instance is that, message sent from SIP phones, through two proxy servers toward other SIP phones use TLS (that replaces SSL) securities between first proxies and originating SIP phones. IPSec between two proxies, and again TLS between destination SIP phones and 2nd proxies is there.

In latter, data originators employ security methods passed unchanged (and unverified usually) by entities that are intermediary till message reaches ultimate recipients. The cryptographic ways for assuring integrity, non-repudiation,

authentication, protection of Denial of Service, and secrecy are performed by recipients. Few specifics are as follows: Megaco (currently RFC3015, earlier H.248) does specify that IPSec are integral for implementations. MGCP (RFC3435) have no mechanisms that are mandatory. But it does recommend IPSec. SIP (RFC3261) does specify that TLS is core, and key for implementing hop-by-hop. It uses S/MIME for that security that is end-to-end. SIP utilizes also HTTP authentication methodology which is deployed widely and fairly. Methods for preventing attacks of DoS are built into such protocols. It can be overwhelmed by elements that are rogue thereby generating valid data. Implementations that are specific give throttling ways for mitigating such kinds of problems and difficulties.

Services of VoIP

Industry of VoIP is modifying from ones catering essentially for hobbyists into ones where services have been offered to general public. Services are lower in quality. Tariffs are lower substantially than for telephony of PSTN. Next transition phase shall be to markets in which qualities are comparable with PSTN. Functionality is more advanced significantly.

Providers of service offer reduced prices for VoIP services due to packet switching efficiency over circuit switching and because of regulation lacking, and so accessibility charges. Telcos that are traditional lessen prices of PSTN, develop own strategy of VoIP or face loss of share of market which is substantial.

Lots of operators based on facility have primary heed on market of wholesale capacity with almost all offering consumer services that is of end users. Additional factor is that there are varied ITSPs and ISPs that offer services of VoIP with multiple new entrants that make their presence known in markets.

Key focus on VoIP retail providers are international and longdistance markets. Avoidance of accessibility to long-distance, and international fees result in significant cheap tariffs for such markets. The markets that are international generally are difficult for entering, but providers that are small today use internet for

giving international services. Apart from difficulty which obtains local network accessibility, case is also that in local loops, trunk link lengths have been significantly reduced (and few times even completely eliminated) and because VoIP’s tend to utilize PSTN for last and first leg connections, so to use internet for shorter trunking lengths lessens advantages of costs. In long terms, local accessibility become a lot more core and integral as technology of last mile becomes sophisticated in increased manners.

Fax is necessary and shall move to IP’s prior to voice telephony because of latency tolerance. To send fax over network of IP has no difference to send whichever file. Fax traffic does use internet as trunk connection of itself. Estimated 30 percent to 40 percent of international and long-distance traffic has to be fax (estimation on basis of lots of various sources). It is too easily switched to internet. With increase in numbers of givers who have desire to provide service of longer distance, ones that own networks are in primary positions for taking benefit of this.

Network services

Network of communication consists of two sections: Backbone network

Access network

Providers of backbone network supply bandwidth of wholesale to providers of service which are close to end users. They lease network or own it from different givers of backbone. Giver of backbone tends to be larger traditional operator or incumbent telcos. But new generations have started to emerge with industries like Qwest and Level 3. Building of new branded IP based networks must be done. Large cable operators and ISPs are involved to offer capacity of backbone.

Technologies like frame relay and Asynchronous Transfer Mode (ATM) allows backbone network to carry various traffic types and for offering guarantee of quality of service for services in real time. In conditions where latency and quality are issues, there is likeliness that IP over ATM is dominant technology. But where efficiency is core and key, IP directly run over Synchronous Digital Hierarchy (SDH) or Synchronous Optical Network (SONET). Whatever is link/physical layer technology, backbone of future appears to become IP. It is network protocol being accepted universally, and has independence of low layer technology. Issues

of QoS have responsibility of operators of backbone as are Virtual Private Networks obligations being offered like value added services. The following figure 7.3 shows the VoIP Network Architecture:

Figure 7.3: ATM, IP PBX, IP Centrex

The following figure 7.4 describes a Synchronous Optical network which is the backbone of IP:

Figure 7.4: SONET

Access networks belong essentially to telcos that is incumbent and, to few degrees, operators of cable. Complexity to get accessibility to local loops and construction of totally new access network is impractical economically. Regulation in few nations is to push for unbundled local loops. Its owners have market strength still enormously. Speed of local loop greatly increases with mass deployments of higher speed technology like radio, cable, extended DSL (xDSL) and satellite. Improvement of quality of VoIP when compared with that of PSTN has impact on local loops with its owners. The following figure 7.5 deals with the extended DSL Architecture:

Figure 7.5: xDSL Unlike telcos, lots of small providers do not evolve billing systems that are sophisticated. It is troublesome for those who have systems development over short time periods. It has pushed ideas of calling cards which are prepaid by smaller givers. It offers many advantages to the customers and providers. Consumer authentication and identification is simple. People dial in account number and data is captured. If calling cards are prepaid, there is no accumulation of bad debts. No billings essentially are involved. Cards are recharged and obtained directly by means of website of organization with credit card’s automatic billing.

Managers having employees with frequent changes to calling requirements can do it. Method gives excellent short-term ways to promote services. In long terms users need convenience in good amounts.

Services of retails

Enhancements to old telephonic services is main attraction of market once arbitrage of tariffs has somewhat dissolved, as this appears that it will. Caller ID of current PSTN allows party that receives to see numbers of calling parties with no other factors. Environment of IP helps more information to who caller is, where is he, then requesting for multimedia connections or whichever data he desires to be send. Call waiting functions having availability with VoIP has been enhanced greatly to version of PSTN. When call arrives, new windows are displayed on screen of user which shows number and name of caller (with other data he sends). Then users choose how to take care of calls. Call is then answered on phones by dis-connecting connection of internet and taking this on PSTN or to receive it by VoIP over internet connection. The calls are forwarded to various numbers like as mobile. Call is directly forwarded to voice mails or busy tones are given to callers. This applies if users do not wish to get disturbed. Messages are played for callers to hear, in turn to instruct him what next is to be done.

Clicking for Individuals establishing or initiating voice connection must click on web pages. Such services have applications for call centers and catalogs. Among others there is enabling of customers for speaking to representatives while looking at website of company. People click on items for initiating calls to catalogs, and

speaking to sales representatives by means of PC with no taking down of connection of internet.

Service integration like e-mail, voice mail and fax become integral as individuals have more mobile factors. Number of industries of mobile telecommunication has begun already to offer few sorts of integrated services that allow masses to check e-mails over mobile phones. Such services allow sending of message from internet to mobile phone directly. Deutsche Telekom does offer extension to it, by ways of novice systems, utilizing software of voice synthesis for converting e-mail information to speeches and people have data read over phone to it. Various organizations give services whereby data like quotes of stock market or football results get on terminals of mobile through Short Message Service (SMS). Administration in real time (such as provisioning and billing) is becoming service in its own kind. Bill unification for all services with abilities to view online bills (up-to-date and real time) is seen as huge needed factors of convenience. Service provisioning is available over internet. There is a growth of penetration of internet, and technical literacies of average users’ decreases. More comprehensive and robust service of consumer is important for serving lesser savvy users.

Virtual private networks

It is private informational network utilizing internet, maintaining privacies by means of usage of tunneling protocols with procedures of security. This is contrasted with line system leased from telcos used only by one industry with incredible costs. Virtual Private Network gives the company similar capability at much low cost by using public infrastructures instead of private one. Market demands is not high, so services of VPN internet are not pushed like key element of lot of offering of provider. Supposition is that, market is demanding leads, and so multiple givers have technical abilities for offering services of this kind and these have not visualized necessities for promoting it. They feel that VPN modify market of service provision in a dramatic way for offering greater opportunity to provider. Small business and people’s VoIP VPN services over internet is likely to become offered by carriers based on non-facilities, but whereas carriers by virtue of facility carriers does target larger people of business. The following figure 7.6 shows a virtual private network service.

Figure 7.6: VoIP VPN Services

Individuality of all networks with services priced accordingly to customer’s requirements that is specific is there. Particular market is not targeted, with providers that offer services to the company that is individual. Few segments of market present great opportunity for those early adoptions including IT firms, sectors with larger numbers of smaller sites like estate agents, travel agencies, manufacturing firms of higher technology and insurance brokers. To target organizations is determined by bandwidth with large industries being primary target for firms that are medium sized and contracts which are termed long to show potential of short terms.

Fax over Internet Protocol (FoIP)

In spite of those predictions that are made in face of e-mails, there is growth in market of fax. Despite e-mail popularity for file transfer and general communications, few functions like to send signed hand written documents that is not effortlessly accomplished. E-mail reliability and speed is unsatisfactory too. The following figure 7.7 deals with Fax over Internet Protocol

Figure 7.7: FoIP

A fax machine which is traditional is directly connected to telephonic network, resembling old telephone. Phone numbers lets numerous long distance and local carriers for initiating connections and delivering faxes. Fax server of internet collect

traffic from standalone fax machines and routing this to internet for deliveries. User dials phone numbers of destination, but this is internet fax servers interpreting dialed digits that are not PSTN and fax have been transmitted through internet to fax server remotely.

Global fax market is flourishing due to adoption of fair consistent standards used worldwide. One among criticality to use fax over internet is standardization lacking. This likely seems that first standard based FoIP is mail type standard to transport files of fax image, but real time and session-based fax must closely follow with more growing necessity of confirmation delivery and immediate delivery of document.

Extensive numbers of services of fax are present over internet, by mainly larger operators of backbone (Sprint, AT&T, and MCI) or organizations paying heed on FoIP provision (such as FAXiNET and FaxSav).

Services of mailbox are provided. This comprises of unified or fax messaging mailbox which forwards and stores fax of users. It is configured for forwarding information automatically to other locations, and can allow users for checking e-mail, voice and fax simultaneously.

Fax multicasting is given. It allows sending of single faxes to lots of destinations, be it PCs or fax machines.

Overflow services takes into consideration that fax of receiver is not answering or it is busy. Fax is stored by service providers, and then forwarded automatically to destination machines at later time. Interactive fax-on-demand/fax services allows customers for storing documents on server of service giver, such that those clients who do not have internet accessibility can retrieve these documents by requesting and calling interactive voice system of response. Translation of data converts fax information to e-mail, e-mails to voice or varied conversion for message delivery. This is significant feature with respect to unified mailboxes.

Primary markets targeted for FoIP with unified messaging is mobile professionals and SOHO market. Former receives all information e-mail, fax, and voice mail all at once. Latter has the increased functionalities, while saving costs. Market for FoIP is mainly initially international one, because of charge differentials between utilization of PSTN and internet. Till 1997, all total global international market of fax virtually was based on PSTN. As per Probe by year 2001, FoIP accounted for nearly 10 percent of international fax that originates. By 2006, it was almost 30 percent.

Multinational industries have ability for saving substantial average part of Dollar 15 million in one year on transmission fees related with fax. It is on basis of those costs which has been by virtue of fax usage of Fortune 500. Revenues that were lost by operators

of PSTN are in regions of Dollar 1 billion by year 2002, and then reaching Dollar 2 billion by year 2006.

VoIP security

VoIP Security is of prime importance for removing the vulnerability of sensitive data and communication.

Demand for security of VoIP

CPE - Media gateways and IP phones with capability of VoIP is vulnerable for many attacks of internet like packet floods or malformed frames. Both of these lead to attacks of DoS. The DoS does consume significant CPU equipment processing cycles, resulting in impaired voice quality in real time scenario of call processing. VoIP CPE is open for alteration, intrusion and monitoring of packet contents, addresses of destination and identifying frauds in unmanaged environments. So, VoIP securities are elements having criticality of mission for VoIP product deployment. Voice CPE gateways address security implementation in such voice residential gateways.

VoIP security areas

Architecture of VoIP CPE gateways consist of two key components: DSP and Micro (voice applications). These are inside IP phones or in separate boxes such as media gateways. Voice streams are packetized utilizing IETF RFC1889 Real Time Protocol and is processed by DSP using voice encryption protocols and key exchanging methods. Encryptions are done either by Micro or DSP. Key exchanging ways for encryption of voice are between two micros that are relayed by means of Call Manager/Server using IETF RFC2327 Session Description Protocol (SDP)as shown in figure

Figure 7.8: SDP

Signals of call processing are communicated between Call Manager/Server and Micro. In few situations, after some messages are there between Server/Call Manager and Micro, data of processing calls might not go anywhere through Call Manager/Server, but directly among two Micros. Common protocols of call processing are MGCP, SIP, and H.323.

In architecture, security of VoIP is divided into four realms: Configuration, data streams, voice streams and call controls. Configuration is done at start-up stage of machinery with server of configuration. After completion of configuration, equipment starts trafficking of streams of data. Data streaming is independent toward voice streams or call controls. When machinery detects incoming information or off-hook signals, it begins controlling of call process with Server/Call Manager. Once calls are established, voice streams are transmitted between two gateways of CPE.

Components of security of VoIP

Though four areas have varied mechanisms of security, but basic components of security are similar. Core goals of security are nonrepudiation, authorization, privacy, authentication and integrity. For achieving these goals, security mechanisms consist of encryption, exchange of key, configuration and authentication. Configuration is initial stage for authorizing network device. Authentication occurs at later point of time or during configuration. Encryption is the way to get privacy and integrity. It needs security keys which could be obtained dynamically, assigned statically by means of key exchanges. Non-repudiation is got by signature from the receiver and sender reports like using reports from senders and receivers nicely with IETF RFC1889 Real Time Control Protocol (RTCP).

Measurement of performance of VoIP security

Major calculation of performance of VoIP securities consist of levels of processing powers, security, messaging delays, and encryption delays. If usually key size is small, processing power, security, and encryption delay will be less. Security key sizes lesser than 56 bits are broken in three hours with those computers that are sophisticated. 128 bits are desirable key size for security. Security keys of size 192 bits does consume too much of computation powers. Though it provides higher security levels, there are no desires for real time processing of call. Complexities of security algorithms impacts level of power of processing, security and delay of encryption too. Message delays happen during call controlling process, authentication and key exchanges. In real time application of call processing, delay cause significant degradation of voice and interference with establishment of call. So, delays must be minimized. Security mechanisms that introduce more than one second delay are not correct for VoIP applications of real time.

Protocols for encryption

Following does summarize well-known encryption protocols utilized in applications of CPE, with their respective trade-off:

(Triple) Data encryption standard (3DES/DES)

Pioneers of encryption of voice used IPSec with Advanced Encryption Standard Data Encryption Standard and Triple Data Encryption Standard AES, 3DES, and DES have all been endorsed by US National Institute of Science and Technologies DES uses keys of 56 bits for blocking encryption of 64 bits plain texts. Key lengths are not so long enough for providing security. 3DES uses keys of 192 bits. 3DES gives more securities, but time of computation is too long such that there is not suitability for real time processing of voice as shown in figure

Figure 7.9: DES

Advanced encryption standard

AES utilizes 128 bits keys. AES does provide a much high level of security than DES, while computational powers are three to ten times lesser than 3DES. AES has been ideal encryption protocol for signaling and voice systems shown in figure

Figure 7.10: AES

Rivest cipher (RC4)

RC4 had been invented at Rivest, Shamir, and Adelman by Ronald RC4 is one key stream shared cipher algorithm. This algorithm is identically utilized for decryption and encryption because data streaming has been merged with generated key sequences. Algorithm is serial since it needs successive state entry exchanges that are based on key sequences. So, implementations are too intensive computationally. Still RC4 is most renowned method of encryption to encrypt configuration files. The following figure 7.11 shows RC4 which is used for encryption.

Figure 7.11: RC4

Protocol for Voice Encryptions – Secure RTP (SRTP)

SRTP is IETF RFC3711. SRTP gives encryption framework and authentication of message of RTCP and RTP streams. SRTP has added two parts to RTP headers: and Authentication has to be optional for SRTP, while it is needed for SRTCP. Encryption is needed in case of SRTP. AES scheme of encryption is supported in SRTP only. The following figure 7.12 shows a secure RTP:

Figure 7.12: SRTP

Key exchanging methods

Common key exchanging ways are as follows:

Symmetric key

Scheme utilizes one key only for decryption and encryption. Both ends of phone calls utilize similar keys. Keys are generated by one periphery and distributed to another periphery. Or, it is assigned by servers to all parties in one domain. This way has no scalability. But, this is a very simple method.

Public key

Method uses two keys. Public key of remote end is utilized for encrypting outgoing messages. Private keys are used for decrypting receiving messages. Method has scalability, but requires hundred to thousand times more power of computation.

Hybrid key

This way utilizes public keys for encrypting symmetric keys. Once symmetric keys are got, it is utilized for decrypting messages. This actually is most important method to provide good efficiency with usage in numerous applications like Netscape Communicator, secured storage of data and MS Outlook shown in figure

Figure 7.13: Netscape Communicator

Diffie-Helman keys

Endpoints having two interactions should agree on passwords for calling to pass through. It is named as Diffie-Helman scheme. One among two CPE devices picks random numbers of base 2 and different devices must match this number. There are five groups or algorithms of DH. If group is high then algorithms shall be more complex. This leads to high security levels with more intensified computations. DH ways are lesser when used in applications of voice because computation powers are required. IETF RFC2401 IPSec provides security frameworks for key exchanging. But this refers to International Security Association (ISA) IETF RFC2409 Internet Key Exchange (IKE) protocols for key exchanges. IKE utilizes DH key exchanging methodology. IPSec is used a lot in pioneering voice applications. Alternative to IPSec is using Multimedia Internet Keying (MIKEY) for SRTP’s key exchanges. MIKEY currently is IETF draft but it is in process to become RFC. MIKEY needs support of symmetric key and public key ways both, while DH keys is optional. Key exchanging ways are carried in SIP SDP attributed fields. Field is used for any exchanging way of keys to stream media. MIKEY possesses limited implementations. But it is getting attention from industries. The following figure 7.14 shows an IKE.

Figure 7.14: IKE The following figure 7.15 shows Multimedia Internet keying for SRTP’s key exchanges.

Figure 7.15: MIKEY

Security association

It is virtual connection in between two or two plus devices for security purposes. During association establishment stages devices conduct authentications, and exchanging of certificates or tokens used for creating encryption keys. Once SA is established few mechanisms of security really perform key exchanges. Presently there is at least one SA among one Call Manager/Server and all CPE. In case there are separated configuration servers then there exists SA between Configuration Server and all CPE. There are SA among all CPE pairs.

Establishment of SA often consumes time, due to mainly exchanging of data. Establishment of the SA’s has so been recommended at configurational stages between server and CPE. In case SA does expire and needs renewal, this is done when devices are actually not in stage of call processing.

Additionally, with SA among servers and CPE, there is requirement to have establishment of SA between two or two plus CPE. Preestablishment of SA among all CPEs has not only got unlikeliness, but it creates meshed up connections with difficulty for managing by using processing power of CPU and memory. There are recommendations to establish SA between CPE on basis as needed. Voice connections are short often, so SA should be terminated prior to its expiring. There are possibilities for reusing previously established SA among 2 CPE, in case there is one. This reduces few steps in establishment stage of SA.

Configuration security of VoIP

In beginning, CPE does provide secured pre-installed ID to network server of configuration. Configuration servers respond with authentication keys. CPEs utilize keys of authentication for starting process of authentication. Once CPE gateways are authenticated, configuration servers give encryption keys. From this point on, encryption keys are used for encrypting all information between server of configuration and CPE. Most renowned protocols utilized in such a process are Secure Hyper Text Transfer Protocol (SHTTP), Rivest Cipher Transport Layer Security (TLS), and Session Security Layer (SSL). The following figure 7.16 shows a SSL.

Figure 7.16: SSL Establishment of SA is part of process of configuration. Configuration has no uniqueness to applications of voice. While information networks may not need configuration totally, but configuration for application of voice is truly a must. RC4 is symmetric/shared key streaming algorithm of cipher. Key size ranges from 54 bits to 128 bits. Algorithm is serial because it

needs state entry’s successive exchanges based on sequence of keys. So, implementations are too intensive computationally.

VoIP security for call controlling process

VoIP signaling or call control system uses encryption/authentication keys when generated at stage of configuration or utilizes key exchanging ways for getting encryption keys.

Internet security (IPSec)

Cable industries use IPSec utilizing Kerberos key exchanging methods for controlling data and messages in MGCP. IPSec is implemented under IP stacks and above networking driver namely bump-in-the-stack (BITS). Alternative is for implementing IPSec away from hosts in firewalls or gateways or routers. This operation is namely bump-in-the-wire (BITW). If IPSec is implemented in IP stacks, it is used for all device applications and these applications might not even consider that device is placed correctly. This is employed often on PC for setting up VPN to Corporate Local Area Network (COLAN). If IPSec is applied in gateways, routers or firewalls, then lots of devices need to share security association of IPSec. It is practiced often between two branches of office.

Transport layer security (TLS)

Derivation of TLS 1.0 is from Session Security Layer v3.0 but there is no backward compatibility with SSL. SIP communities are recommended first by using IPSec but it is then converted to TLS. SIP old spec had been by virtue of UDP that requires IPSec to give more reliability while latest specification of SIP has been on basis of TCP. TCP gives reliability that is sufficient, and so TLS over TCP has no concerns of reliability. TLS equips function of exchanging of keys. Since TLS is greater than TCP, it provides SA on two devices among two applications.

IPSec is utilized for reliable and long connection, while TLS has been used more for those applications that are based on web with bursty and short traffic. Even after termination of SA, with TLS, application reuses prior SA information with re-establishment of connection that shortens time of SA establishment. IPSec reutilizes previous SA data for establishing new connection of SA.

Voice processing security

Voice encryption pioneer use IPSec with AES, DES, and 3DES. Latest standard of encryption of voice is IETF RFC3711 Secure Real-Time Transport Protocol with AES. SRTP has no definition of what protocol of key exchanging is used. Industrial trend is utilization of MIKEY for SRTP’s exchanging of keys.

Denial of Services (DoS)

Attacks of DoS are common in internet. Methodology for handling attacks has no uniqueness with VoIP. Few instances of DoS actions and attacks are highlighted. There are websites that is public like CERT advisory boards offering solution to attack of DoS and commercial item that addresses such problems.

Open issues

Although industries have provided numerous VoIP security solutions, there still are resolving issues. Most of challenges come from security key management. No consensus on how keys are stored distributed and updated and not stolen are still present. FCC has given issuing of requirements for compliance of VoIP with Communications Assistance for Law Enforcement Act Meaning is that service providers of VoIP should give ways for agents of law enforcement to tap into lines of VoIP or risking to face bigger fines. Despite few of outlined challenges, security of VoIP has been achievable currently. With securities placed accurately, utilization of VoIP is expected to become proliferated in forthcoming years.

Conclusion

This chapter is a brief analysis of VoIP security and services. Mechanisms of security are deployed to or by In former, originators of information employ methods of security that have been passed unverified and unchanged by intermediary entities till the message reaches the ultimate recipients. In latter, methodologies have been deployed in between adjacent entities. Lots of industries give services where information like football results or quotes of the stock market get on the mobile terminals by means of SMS. Small business and services of VoIP VPN of people over internet have likeliness to be offered by the carriers on the basis of non-facilities. Extensive fax service numbers have been present over the internet, by backbone’s larger operators or corporations that pays heed on provision of FoIP, such as FAXiNET and FaxSav is present.

The next chapter deals with router performances and functionalities.

Points to remember

IETF protocols and deploys one or more of developed IETF security methodology like TLS IPSec and S/MIME

Call control system or VoIP signaling utilizes authentication/encryption keys when it is generated at configuration stage or uses key exchanging methods for getting keys of encryption. Most well-known protocols used in configuration of VoIP security are SSL, SHTTP, TLS, and RC4.

IPSec has been implemented above networking drivers and under IP stacks namely BITS.

BITW is alternative way to implement IPSec away from hosts in routers or firewalls or gateways.

Multiple choice questions

What are VoIP services? Non-network services

Network services

Retail services None of the above

What are VoIP security methods?

Repudiation

Non-Repudiation

Denial of Service

Non-secrecy What are key exchanging methods?

Non-public key

Public key

Non-hybrid key

All of the above

What is SA?

Security Association Security Access

Secure Access None of the above

Answers to MCQs

Q1: B, C Q2: A, C

Q3: B

Q4: B

Questions

How are security mechanisms deployed? What is ATM, SONET and xDSL?

What are functions of SMS, VoIP VPN, and provision of FoIP?

How DES, AES, RC4 and SRTP is used? What is utilization of MIKEY and IKE?

CHAPTER 8 Functionality of Data Router

Introduction

Network Address Translation ( NAT ) is a standard of IP on LAN where address key translation is kept separate from public view. Customer, provider and internet are segregated by a security layer. The various factors of router performance are QoS and number and type of information supported features. Routing architecture use VoIP integrated chip set. Microprocessor without Interlocked Pipeline Stages ( MIPS ) must be adequately sized for proper router functions. Fortunately, VoIP markets have evolved and matured over last some years along with tightly integrated data silicon and voice solutions for meeting requirements of designers. Customers need robust voice features, and qualities for best router methodologies requiring best voice architecture.

Structure

In this chapter, we will cover the following topics: Information about router

Performance of router

Designing process

Objective

Here we will talk about Network Address Translation is a standard of IP on LAN where address key translation is kept separate from public view. Customer, provider and internet are segregated by a security layer. This chapter will help understand router performances.

Data routing

Provide consideration of designing for core portions of residential VoIP gateway items. The design plethora has emerged like offering of movement beyond market of early Japanese and into mainstream opportunities of US.

Growth in market of residential gateways is attributed to continued broadband adoption in homes with nearly 100 million worldwide users. VoIP functionality and quality continues toward technological advanced improvement in software and hardware both. Economics to migrate to VoIP continues to give significant saving of cost to residential voice consumers, and old telephonic providers of service both. It creates new voiced service giver paths like Vonage for emerging.

With continuation of maturity and growth of VoIP residential market variety of configuration of end products have emerged. It includes solutions which not only integrates functionality of voice but combines home routers also. Early adoptions of oriented items of terminal adapter giving only basic to conversions for service of VoIP, has been replaced by these much more featuring voice gateway of rich solution. Technical architectural issues together with designing consideration and trade-off, important for equipment configuration optimally ended has been addressed.

Basics of information router

It is essential to know what functions of end product information routers have been needed as they drive requirement of performance, specification of software/hardware and affect item architecture. Routing of data directs information from external WAN toward computer IP addresses addressed properly on internal LAN. Other functions involved in residential routers are firewalls protecting LAN from sabotage or corruption by means of WAN. Features that should be included in residential informational router are as follows:

Network address translation (NAT)

NAT is standard of internet enabling address of IP on internal private networks (LANs) to be hidden or separate from public IP corresponding address. Function of NAT gives key address translations such that data passes to and fro, from LAN to WAN and vice-versa. Protectively shield internal address of IP from view that is public. NAT’s most well-known concept used extensively in broad-band routing is extension called Port Address Translation (PAT) or/and Network Address Port Translation (NAPT). Broadband router LAN to WAN context is where NAT implies usual extension of NAPT/PAT that introduce high desirable feature of one IP public address mapped to lots of IP private addresses through inclusion of port identification numbers additionally with LAN IP addresses. NAT (with implication of PAT/NAPT) does allow lots of additional addresses of IP to become internally used to LAN while appearing as single public address of IP. The following figure 8.1 depicts Network Address Translation which is an IP Standard.

Figure 8.1: NAT

The following figure 8.2 depicts a Port Address Translation which is another IP Standard.

Figure 8.2: PAT

The following figure 8.3 shows NAPT which is a type of Transversal.

Figure 8.3: NAPT

IPv4 is current standard for addressing of IP. Limited number of address of IP is present in it. IPv4 uses 32-bit address that results in nearly slightly 4 billion unique internet addresses. But lots of such addresses have been reserved for extremely special purposes. This leaves far less availability for consumption by public. Givers of internet services allocate one address for one entity or resident usually. This single address is adequate at time when there is typically single computer in one household only. With proliferation of PCs throughout the households, networking at

home has significantly increased in form of LANs and routers. NAT is enabler. It allows multiple devices or PCs on home networks for appearing as single address of IP to network of public, so only consuming single address. Security layer by isolation of internal address from public view and access is provided. There is allowance of addressing schemes that is internal and managements that has no conflicts to IP public models of addressing. The following figure 8.4 deals with IPv4 which is a current IP addressing standard.

Figure 8.4: IPv4

When packet from LAN is delivered to WAN, NAT:

Records private internal source IP LAN number, and number of source port is its table of translation.

Replaces private source IP packet address with own external public address of IP. Assigns specific port number for outgoing packets, entering that into translation table, and replacing number of source port with it.

When packet from WAN enters LAN, NAT: Checks port number of destinations of packet. Finds out which machine that is inside has been assigned such port number, in case it matches source number of ports assigned previously, then discard packet if matching port numbers are not found. Found matches are rewritten. IP address and port number of destinations with original address of machine and port numbers used for packets on inside initiates connections.

Transmits such packets to inside of intentional private networks. Maintains entry of translation table till opening of connection.

Firewall

Firewall is hardware piece or program of software that enforces security between two networks determining which traffic should be allowed accessibility, and which one to block. Firewalls are configured for protection against logins that are unauthenticated from outside world while keeping internal network segmentation to become secured. The following figure 8.5 shows us a firewall which acts as a security mechanism between the source and the target.

Figure 8.5: Firewall

Features that are supported by firewall which includes protection against are as follows:

Login that is remote: Without approval, connection to computer is there.

Hijacking of Simple Mail Transfer Protocol (SMTP) session: Accessibility to lists of e-mail address on computers, for sending spam that is unsolicited is allowed. The following figure 8.6 shows how an SMTP session works in real-time.

Figure 8.6: SMTP Bugs of operating system: Remote accessibility because of insufficient security bugs/ controls is done. DoS: Hackers inundating servers with requests of unanswerable session causing crash servers is present.

Bombs of e-mails: Sending of similar e-mail numerous times till crashing of system of e-mail is there. Macros: Hacker creates Macros depending on applications destroying information or crashing computer is present. It quickly spreads among different systems. It ranges from messages that are harmless to erasing of all information of computer. Spam: It links to websites accepting cookies giving backdoor to computer.

Redirecting bombs: One of the ways of setting up of attacks of DoS is there. Routing of source: Data appearing from trusted sources or even from inside network is present. Depending on vulnerability of network, the designers enable protection against attacks that is external in all forms. Such maximum protection takes up extra cycles of CPU while reducing performance of informational routers.

Factors of performance of router

For application that is residential there is designing of voice gateway with router functionality which have been impacted by lots of performance factors. It includes following:

Type and number of informational supported features.

QoS levels assisted for service of voice. Features of router often impacts packets throughput performances. For instance, in case NAT is needed, additional packet processing itself is integral to keep such NAT packets filtered at rate of wire. This type of item combines residential data traffic with voiced traffics, so it is key to maintain QoS and throughputs of real time for voice trafficking. If packets of voice do not allow maintenance of relative constant cadences, with minimal delays, because of larger packet size or higher traffic of data between ports of WAN and LAN, the qualities of voice degrade, yielding in unacceptable architecture of commodity which does not muster service provider or market acceptances.

Designing process

Designing processes target at performance, and other criterion when it comes to data routing processes.

Targeting performance

Designer of voice gateways should possess excellent ideas of those requirements that affect performance, prior to commencing of product designs. Whether a person is taking existing item of voice gateway and adding capability of data routing to commodity, or initiating voice/ information router product from inception, there is factor of essentiality for determining how much power of processing is required for meeting requirements. Other crucial factors for implementing optimized router/ voice products are costs. Residential gateways of voice are currently available in higher volumes, mass marketed channels of retail and facing Bill of Material (BOM) pricing that is aggressive. Time-tomarket finally is the consideration critically in segment of dynamic market. Single one-month delay in rollout of commodity costs dollars in millions in lost opportunities with promotion in retail that hinge on availability of product for specified events like holiday and back-to-school promotions.

Voice gateway has various characteristics when compared with application of information. Voice traffic has to be data of real time relying on timely packet reception, that is, in case voice message arrives too much late, then it gets lost with distortion of what listener hears. In voice gateway, the voice traffic receives preferential treatments having priority prior to data trafficking. Such priorities come in QoS forms. With implementation of QoS, router causes processing of voice or RTP packets before varied

packets of message by examining type of packet and placing packets of voice ahead of different packets in transmitting queues exists.

Combination of data and voice in gateway solution has various requirements with regards to throughputs and packet sizes. Voice involves smaller size of packet (less than typically 200 byte/packet, with dependency on codecs) and needs small bandwidth portion because of low packet throughput requirement. For 10 millisecond voice payloads, packet throughputs have to be 200 packet/s in two directions. A data transfer involves usually much large packets (1Kbyte/packet to 1.5 Kbyte/packet) with high throughputs and to consume more bandwidth. Example of information packet is File Transfer Protocol (FTP) downloads of MP3 files. The following figure 8.7 shows FTP Downloads behind a firewall.

Figure 8.7: FTP Downloads The following figure 8.8 shows the downloads on MP3 files:

Figure 8.8: MP3 Files, AMR-WB A rate of broadband in residences varies greatly depending for provided services. In Japan, there exists large presence of Fiber-ToThe-Home (FTTH); here bandwidth approaches 100 Mbit/s. In US, typical cable and DSL are available, here rates as fewer than 24 Mbit/s is maximum (with bandwidth practically between 500 Kbit/s and 5 Mbit/s). Varying broadband rates of access are core points of data for determining item architecture and performance as shown in the figure 8.9 which depicts a FTTH.

Figure 8.9: FTTH

Throughput of packet has been characterized in packets/second egressing or ingressing from port of LAN of router/gateway to port of WAN. Term “wire speed” or routing does imply that device of router achieves maximum throughputs effectively or bit/s by means of port of Ethernet. For instance, for Ethernet interface of 100 Mbit/s, wire speed throughputs are 100 Mbit/s. It is crucial to know what packet size of information intended gateway has to be expected for operating at, and which associated throughput of packet there is expectation for being achieved. For example, if information routers are connected to DSL modems in US, where rates of downstream are lesser significantly than 10 Mbit/s, designing data routers of more than 100 Mbit/s for intended deployments is there.

With performance of desired routing at given size of packet, along with requirement of voice that includes budget for growth of feature and performance, design engineers presently are considering numerous architectures for voice residential gateway with functionality of router.

Options that are architectural for routing

Architecture of first product is one among discrete components that performs collectively voice gateway solutions with functionality of data routing. Architecture does allow DSP to become sized for voice performances, where typically telephonic media processing has to be performed. RISC is sized at high frequency relatively for assuring apt megahertz (MHz) to be available for performance of routing. Approach does not afford typically software/hardware solution; designers find routing software, devices, and software of voice processing from varieties of sources. This leaves significant work amount to develop gluing software and to perform integration to designing team. This affects time for marketing with robustness of overall solutions. Discrete solutions, representative of earlier implementation of VoIP, bears typically high overall costs because of lack of integration of solution. Prevailing architectures in market of residential gateways is VoIP integrated chipset. The following figure 8.10 shows data routing mechanisms.

Figure 8.10: Data Routing

Such devices include DSP functions for telephony/voice processing typically. Processor that belongs to RISC domain for telephony and network protocol processing together with general management of device, all in integrated, single solution is involved too. RISC processors are utilized for performing functionality of routing. Integrated devices have availability in variety of performance, and speed levels. Some have been designed for functionality of basic voice gateways while others have capacity for performing routing also. These architectures have been optimized for applications of VoIP. It has processor of RISC and DSP running at varied frequencies that offer optimized performances for routing and robust voices often. Yet architectures for VoIP have been emerged in few residential items. It comprises of high-speed RISC processor executing not only router functions, telephony and network protocols, but also

operations of voice processing running typically on DSP. It of course, needs higher speed and, so, more costly processors. For running voice functions on RISC, it typically takes two times amount of RISC MHz than it does for DSP MHz. So, the designers are careful for determining adequate sizing of Microprocessor without Interlocked Pipeline Stages (MIPS) for overall application of VoIP and router functions and for planning functional growths. The following figure 8.11 shows a Microprocessor without Interlocked Pipeline Stages-MIPS:

Figure 8.11: MIPS

Designing factors that is additional

Broad levelled designing/costs complexity are affected by chosen architectures. High speed processors possess high density memory buses and high-speed memory access, requiring subsequently more challenging efforts of layouts. They are more subjected to issues of EMI. It needs shielding (additional) or possibly, even, high numbers of layers of Printed Circuit Board (PCB) (usual goals are 4 layers). These designing difficulties are paid heed when comparison of time to markets, and overall cost of commodity is done.

Other designing attention should be paid to numbers of ports of LAN on products. Presently most routers of home have been shipped with LAN switches of four ports. In lots of houses there are not more than 2 PCs. With WLAN acceptances, there are few additional reasons for LAN ports to exist. There are few times requirement for at least one configuration LAN port, in case port of WLAN requires to be set up.

Execution size and software image plays role in overall billing of material cost, while create and add development and maintenance fees. Static/Synchronous Dynamic Random-Access Memory (SDRAM) and FLASH components on gateways for residential complexes are core significant portions of overall BOM. Solutions based on only RISC, for instance, typically has large consumption of SDRAM and FLASH. So, it is taken into consideration at overall item’s levelling charges. The following figure 8.12 shows an

SDRAM of a Digital Still Camera which is used for video conferencing.

Figure 8.12: SDRAM The following figure 8.13 shows a FLASH Memory, and its uses.

Figure 8.13: FLASH

Conclusion

With interest and growth in VoIP of consumers, designers of residential gateways of voice have to face more challenges today, to come up with complete integrated solutions of data/ voice. Customers do desire robust voiced features and qualities, coupled with all functionality of information together with NAT, routing and firewall services. Overall development of these solutions is enhanced greatly by knowing relationship between such pressures and designing requirements for delivering low cost marketing solutions. Market of VoIP fortunately has evolved and matured over last some years with too much more tightly integrated data silicon and voice solutions. They are available for meeting requirements of demanding designers, these days.

The next chapter deals with technical description of VoIP with respect to the various transmission layers.

Points to remember

IPv4 is today’s IP addressing standard. Hijacking of SMTP sessions means access to e-mail address lists on the computers to send unsolicited spam that is allowed.

Voice and data combination in solutions of gateway have varied needed factors with respect to packet sizes and throughputs, example of informational packet of MP3 files is FTP downloads.

In Japan, large FTTH presence is there with bandwidth approaching 100Mbit/s.

Multiple choice questions

What are router performance factors? Type of Supported Features

Number of Supported Features

QoS All of the above

How to target performance?

High volume of voice products

Low volume of voice products

Optimized costs

Non-optimized costs Why BOM pricing is aggressive?

Voice gateway is cheap

Voice gateway is high cost

Voice gateway is moderate

All of the above

Why are FLASH and SDRAM components core for residential complexes gateways?

RISC consume large SDRAM and FLASH

RISC consume low SDRAM and FLASH RISC do not consume SDRAM and FLASH

None of the above

Answers to MCQs

Q1: D Q2: A, C

Q3: D

Q4: C

Questions

What is PAT, NAT and NAPT? What is a firewall?

What is router architecture of data?

What does MIPS do?

CHAPTER 9 Technical Description Related with VoIP

Introduction

TCP/IP which is an industrial protocol suite is used for communicating and accessing over transmission medium. Various layers are Application Layer, Transport Layer, Internet Layer, and Physical Layer . Decentralized implementation makes it easy for physical media to be transferred through fiber optics. Types of media are voice, video and data. SIP Proxy Server or IP Communication Server are connected with VoIP end points for residential gateways. Architecture of internet telephony consist of trunk replacement, hybrid (hop on/hop off) methodology and endto-end IP connection. Advanced usage methods of VoIP are PBX replacements, PBX extension, and IP Centrex (which uses third party tools for internet telephony). Least Cost Routing ( LCR ) uses gateway devices for on-net calls. Programmatic or API interfaces facilitates control management, creation, and deployment of services which includes provisioning, accounting and billing.

Structure

In this chapter, we will cover the following topics: TCP/IP

Decentralized implementations

Types of media Call control separations

Mobility and virtual locations

Internet telephony architecture

Usage methods

API/ programmatic interfaces

Objective

This chapter presents TCP/IP which is an industrial protocol suite is used for communicating and accessing over transmission medium. Various layers are application layer, transport layer, internet layer, and physical layer. In this chapter we will understand the decentralized implementations with regards to interfaces and layers.

TCP/IP Primer

TCP/IP is defined as industrial standardized protocol suite that is used by devices to access, find and communicate with one another over transmission mediums. Protocol is set of rules and standards which should be followed. In networking case, protocol is set of rules as well as standards that must be followed by software and hardware of device such that it is nicely understood and recognized by different devices. Suite of protocol is implemented through software very commonly called TCP/IP stack as shown in the figure

Figure 9.1: TCP/IP Model, OSI Model

TCP/IP is made of the following:

Under hood, architecture of TCP/IP consists of numerous layers that perform some functions. All layers contain protocols. Four layers of TCP/IP Primer are as follows:

Application

Data sent begins at top of stack of TCP/IP in application layer. It contains applications and services of network that user interfaces work with for communicating with network. There is service utility like name resolution and print and file services. Good example is NetBIOS that is application programming interface (API) supporting environment of desktop operation. All such utilities working with TCP/IP live in this layer. Instances of few familiar protocols with primary usage of theirs includes figure 9.2 which are NetBIOS, ICMP, RARP.

Figure 9.2: NetBIOS, ICMP, RARP The following figure 9.3 depicts an API:

Figure 9.3: API Dns: Domain Name Service ftp: File transfer http: Transporting web traffic

pop3: Retrieving e-mail rtp: Real-time transmission of multimedia data smtp: Transporting e-mail Remote terminal access

Transport layer

Once application layer is through with information, there is passing down of stack to the transport layer. Two key components of this layer are UDP and TCP. Layer in question is interface used by applications for connectivity of network. The TCP/IP designers wish to be sure that sent data does get received by correct application and accurate device running on those devices in correct sequences. The layer gives functionality of this kind. There are mechanisms in turn for flow control, verification and error checking ensuring completeness and integrity of data being received and sent by this transport layer.

Though UDP and TCP have same functionalities, there is a key difference between both. TCP is regarded to be protocol oriented with connection. UDP is a protocol that is connectionless. Connection oriented norm establishes connection with other machine, then maintaining such a connection for entire data transmission duration. Functional varieties are built into TCP that rechecks and checks information while two devices are connected. So, TCP is transmission, reliable, albeit slower protocol. UDP has no connection at all with target machines. UDP is stated by Application Layer to those devices where there is a supposition for transmission to, with no queries asked. UDP is a much fast norm when query of transmission of data rises. But UDP has issues related with reliability, rudimentary checking of error and flow control. Factor to note is that ordered reliable TCP delivery service is not exploited by communication of real time because there is

no timing for re-transmissions. For such a reason, RTP, that is used for applications of VoIP, does ride over UDP.

Internet layer

This layer is beneath Transport Layer. Three integral protocols reside in Internet Layer: Internet Control Message Protocol (ICMP), Internet Protocol and Address Resolution Protocol (ARP). All of these serve specific purposes. Two lesser used protocols are namely Internet Group Management Protocol (IGMP), and Reverse Address Resolution Protocol (RARP). Address resolution and IP addressing occurs within this layer. IP addressing is one scheme which does standardize how devices have been differentiated and identified from each other. Scheme does allow whichever TCP/IP running device for communicating with varied devices that run TCP/IP in any part of world. Irrespective of whichever type of network topology, machine or operating system lived on by devices, till the time both devices use TCP/IP, they are expected to speak similar language. Job of ARP is resolution of logical address of IP into equivalent physical address of itself. ICMP is used mostly by routers for sending data back to source computer regarding transmissions which devices try to make. The following figure 9.4 shows the stacks of address resolution protocol.

Figure 9.4: ARP The following figure 9.5 shows Internet Group Management Protocol stacks:

Figure 9.5: IGMP

Physical Layer

It is final layer on this TCP/IP stack. It is at base of stack. It is last section’s packet that needs to go through prior to it, sending out across medium of transmission. It contains collection of specifications and services that manage and provide accessibility to network hardware.

Crucial responsibilities involve: Error checking in information’s incoming packets.

Interface with network hardware of device.

Tag out-going packets with information of checking errors.

Acknowledge receipt of packet.

Re-send packet in case no acknowledgement is returned by recipient.

Above layer is totally invisible to regular users. Given its complexity, this is not that much a bad ideology. It is an almost factor.

Implementation that seems to be decentralized

When revolution of PC took place two decades ago, movement was there away from monolithic architecture of computing such that decentralized environment of computing was provided by mainframes. Companies leveraged mobile and more distributed workforce while constantly staying in touch through connected organizers, cell phones, laptops, and pagers, because of internet. VPNs makes this far less costly (when compared with industry owned traditional WAN) with more security for connecting remote office premises to headquarters. Lack of capabilities of networking in existing telephonic systems yielded that decentralization efforts have focused largely on applications only with data. With the emergence of IP telephony technology, business today decentralizes information and voice applications over common networks with remote accessibility.

Internet telephony does allow industries to distribute equipment of processing of IP calls to most apt locations on basis of specified organizational requirements. Unlike older circuit-switched technology that primarily is monolithic architecture with switch/PBX controlling and central device, service of IP telephony is distributed across entire enterprise. Signaling server, application server and IP gateway is situated at remote places in varied buildings even on internet or private network that is external.

Physical media

In independent packet network, message is routed through those networks that communicate with sets of end points configured properly. These data deliveries occur over lots of different physical connections like cabling of fiber optics or broadcasting through airs or twisted pairs. Delivery of information for packet networks has not been constrained by underlying method of delivery. Data between two end points takes various routes across networks. Decision of routing is made on individual packets irrespective of how previous packets are routed. This sort of independence from such physical media makes networks of IP adaptable particularly to distinct environments. It provides too high availability levels by opening lots of dissimilar options for providing connectivity.

Types of media

Latency, packet loss and jitter are regarded detrimental to network performance. Behaviors of network have miscellaneous impacts on varied types of media. Audio or video are media types needing regular and timely IP packet delivery. Differently speaking, data is a factor that can be forgiven. Time sensitive and non-time sensitive applications are stated. The effects of latency and jitter are minimized by virtue of usage of QoS and jitter buffer ways.

Data

To browse content on networks does not necessarily require connection with lower jitters or packet loss. Applications of information re-order packets for handling delays with users that have less to no network impairment perception. Transmissions of data that is viewed by one while using internet browsers are not too sensitive to jitter or delay. So, networks that are fine to surf internet are not acceptable for video or voice.

Voice

Voice traverses’ networks of IP and has PSQM score equivalents or better, with which is seen over switches of circuit. Quality of voice has been impacted with packet loss, delay or jitter introduction. Delay or long latency is nuisance on two-way voice calls. When a person speaks there are delays while audios are transmitted to distant connection ends. Such delay is again seen when other responds. Higher jitter on connection results in garbling of audio sounding if audio packets are discarded totally or processed out-of-order. Packet loss that is minimal is not problem because human ear has no precision for perceiving enough periods of audios that are missing. So, packet loss that is excessive seriously impacts quality of voice.

Video

Videos of high fidelity have IP transmissions with no quality degradation. Lots of television signals of HDTV are transported across IP. But like voice, quality of video is impacted with packet loss, delay, and jitter introduction. Regular latency is not much of issue because videos tend to become one way. In case of video conferencing of two ways, similar awkward delays are present, which one experiences in higher latency connection of voice. Jitters on the video connections result quickly in video image’s perceivable degradation. Fact is that, packet loss and jitter both negatively impact video connection’s perceived quality.

Transport of media and call control separation

IP telephony finishes evolution from signaling that is in-band, and seen in analog telephony for completing separation of flows of media and signaling.

Mobility and virtual locations

For most part, old circuit switched telephonies are based on end point linkage (that is phones) to switch/PBX by using hard wired, dedicated connections. With the help of usage of wiring closets, patch panels and distribution centers, cabling is needed providing services of media transports and call controls from and to phones. While some capabilities exist that allow the wired digital and analog phones to move dynamically from one place to another with no re-configuration at PBX/switch levels, for most parts, business telephony and current residential infrastructures need that particular phones remaining connected to particular cables unless these configuration modifications have been performed.

This no longer is a restriction with VoIP. For service of residential and business IP telephony, there is no possibility for easy movement of end points of IP to and fro places. This is one among marketed advantages. With IP telephony, end points need to possess connectivity of IP with SIP Proxy Server or IP Communication Server and end point of IP with whom they shall communicate. Example is that, business users with IP phones that have been connected to IP Communication Server simply unplugs phone from networks, walking down hallways, or even, to different buildings for plugging phones back into networks, and they shall have ability to instantaneously use those phones. The following figure 9.6 shows SIP trunking system:

Figure 9.6: SIP Proxy Server

This instance assumes that new locations where IP phones have been plugged into provides accessibility back to communication server. For users who reside in a locality, this mobility virtually extends to any place where internet is present.

This mobility offers multiple benefits. It also creates real complexity while trying to respond to people making emergency calls through one of such phones. End points and IP phones are very mobile. So, ability of utilization of infrastructure and technology of current E-911 for correctly locating callers and routing emergency calls to appropriate Public Safety Access Point (PSAP) is even impossible or is challenging. The following figure 9.7 shows the infrastructure of E-911:

Figure 9.7: E-911

The following figure 9.8 depicts Public Safety Access Points:

Figure 9.8: PSAP

Architecture options of internet telephony

In lots of cases challenges encountered with IP telephony is its integration into existing older telephonic infrastructure which exists presently. Based on these specific goals, all companies choose from three dissimilar approaches for beginning with assimilation of internet telephony into its current infrastructure of telephony. In various cases, organizations use more than one among these methods for meeting all requirements of IP telephony within its enterprise.

Trunk replacement

In such approach, both person originating call (that is caller) and person receiving call (that is callee) continue usage of telephone services that are circuit-switched from end points (that is phone) to equipment of call processing (that is switch, PBX). Call travels from endpoint of caller to switch/PBX that then connects through public internet, IP based private network, or few combinations, to PBX/switch which is closer to callee. Methodology needs no change in phone and dialing behavior with only minimal modifications at switch/PBX. In lots of cases this way is utilized with no callee or caller having awareness about it. Numerous PBX/switch vendors offer interfaces of IP trunk which simply replaces T-1 trunk with connections that are packet switched.

Hybrid (hop-off/hop-on)

This is another way. It is called hop-off or hop-on depending on directions. In hop-on, calls are directed from traditional digital or analog phone to phone based on IP. Vice versa is true too. In two cases, IP phones are addressed by regular telephone numbers, although phones do not necessarily be located in typical geographic areas associated with area code in question. Many industries have begun to offer phones of IP for the small businesses and residential subscribers who follow this pattern. Architecture that is closely related is known as IP PBX or IP Communication Server, where phones inside enterprise have connection with gateways providing PSTN connectivity’s.

If IP communication server is shared among lots of companies with operations by service providers namely IP Centrex, then economic models have similarity somewhat to service of Centrex that is offered by old local exchanging carriers.

End-to-end, direct IP connection

This is final method. It dispenses with gateways using direct communications that is based IP wise, and it is end-to-end between callee and caller. With this methodology, end points of IP communicate directly with other end point of IP, till the time they establish IP links over one or one plus private networks or the internet.

Usage methods

Businesses have numerous options that are chosen while internet telephony is implemented. Such options depend on expertise levels, specific goals of business and current infrastructure of telephony. For users living in residences, option presently is usage of broadband internet connection for telephonic services. Four business options are as follows:

PBX replacement

IP Communication Server is implemented as software which runs on more than one computer server set that runs on Linux, Windows NT/2000 or other comparable operating systems. One methodology of IP telephony usage is replacement of current machinery of PBX/switch with IP communication server software and equipment. This method gives fullest set of benefits and features of internet telephony because it allows industries for implementing three options of architecture entirely. Method drawback is that it needs forklift up gradation of existing machinery of telephony and all applications well placed usually. While vendors of IP telephony pay heed on performance, features and reliability of IP communication servers in near past, in some cases, they might not meet expectations and standards set by current circuit switches of available equipment of switch/PBX. For such reasons, IP PBX’s are marketed commonly toward installations of the wherein no systems of legacy are placed already.

PBX extension

For industries having large investment in ancient machinery that is circuit switched, addition of internet telephonic capabilities to existing infrastructures give lots of functionalities and features to be delivered by IP telephony. Implementation of gateways of IP alongside PBXs, results in industry employing easy and costeffective ways for extending services of voice to users or remote locations. Features integrate with existing equipment, so, companies continue into management from central places. IP gateways provide bridge connecting network circuit switched with network packet switched by forwarding and multiplexing voice packets from and to faraway regions over private network of IP or/and internet.

Few benefits of such methodology are as follows:

It supports converged applications and networks.

There are potential savings of cost by means of toll bypass.

Flexibility for using multiple transporting media (example frame delay, DSL and cable) for connecting to sites that are distant is done.

It preserves existing investment and infrastructure.

There is elimination of need for buying PBX that is separate for far-off areas.

Economical and effective SOHO environment solutions are created.

IP Centrex

To use third party for giving services of internet telephony. Variation of replacement of PBX approach is augmenting or replacing existing telephony that is based on circuit with service of IP Centrex. Like older Centrex, there is decrease in initial enterprise’s capital investment making system maintenances to be responsibility of service providers through IP Centex’s service. Unlike old Centrex, in which all phones possess own accessibility circuit, the IP Centrex needs only those organizations having private IP or single internet connection of network to providers which is more cost-efficient. In environment of IP centrex, end points of IP reside within industries while IP communication server resides usually at provider of service.

Residential service

In near past, residential users starting to use internet as one among ways for communication of voice have grown in number. This was accomplished initially by software like Net2Phone and MS Net Meeting. Software did allow residential users to utilize internet connected home computers to place and receive calls from/to different users with similar software. Users use those speakers who are attached to headsets or computers for listening from the microphones connected to computers for talking. The following figure 9.9 depicts a Net2Phone software:

Figure 9.9: Net2Phone Software

After this, lots of companies began to offer those services to residential users which allowed them to receive/place calls with various others who had traditional phones connected with PSTN. It was given by means of usage of gateways of IP, routing calls from internet to PSTN with vice-versa being true. With this methodology, users use speakers that are attached to the headsets or computers for hearing still. Microphones are connected for purposes of speaking to their computers.

Recently, numerous organizations have begun to offer services working too much like those telephones which people possess presently in their homes. Residential customers pick up the phones, dial numbers, and then connect with those individuals whom they are speaking with. In technical terminology, such service does use phone adapters called Multimedia Terminal Adapter (MTA) or Analog Telephone Adapter (ATA) for converting analog signals to digital signals. Digital signals are sent over internet connection of high speed (that is DSL, cable). Residential people can have new number of phone or they can keep existing numbers (that is called number portability). On-net calls (that is calls made between places which is connected to internet) are free of cost. Off-net calls are delivered to network of PSTN by using gateway devices of IP with functionality of Least Cost Routing (LCR). The following figure 9.10 throws light on a multimedia terminal adapter.

Figure 9.10: MTA, NGN The following figure 9.11 shows an analog terminal adapter:

Figure 9.11: ATA The following figure 9.12 shows least cost routing:

Figure 9.12: LCR

When a person makes call to residential person with such service, they dial standard number of telephones. This number resembles e-mail address behind scenes. Number instructs call to go over internet, and then through network to ATA. Phone is connected to the rings of ATA. From such a point, rest of calls are identical to phone calls made by utilizing analog phones connected directly to PSTN via Local Service Provider (LSP).

API / Programmatic interfaces

In VoIP, lots of API are developed for facilitating control of service’s management, creation and deployment. Application servers give component of support systems that are operational. This includes provisioning, accounting and billing.

Session initiated protocol – Common gateway interface (SIP-CGI)

Like HTTP-CGI, script of SIP-CGI resides in servers, and passes parameters of messages by means of environmental variables to separated processes. Key point is that new processes are created on servers that add to overhead costs. When request arrive at SIP server, for initiation of new transactions, servers shall set environmental variables in good numbers and call CGI scripts. To body of request script pass through standard input (stdin) and return instructions to server by virtue of descriptor of output file that is standard. SIP-CGI is similar to HTTP-CGI. It is suitable particularly for those services which contain components of substantial web. The script of CGI is written in Java, Perl, C++, Tcl or C making this accessible to large developer community. The following figure 9.13 shows the Test-Bed Architecture:

Figure 9.13: SIP-CGI, CPL The following figure 9.14 shows the Common Gateway Interface based on HTTP:

Figure 9.14: HTTP-CGI

Call processing language

Topic in service creation of VoIP is how vendors give providers of service ability for offering basic services of call, and allowing end users the customization of service comfort. This does bring about varied numerous call service combinations needing few standardization levels.

Internet Engineering Task Force’s IP Telephony Work Group (IETF IPTL) have proposed the language based on XML, that is, Call Processing Language for handling vast different combinations of service and various call scenarios.

CPL is that language which is utilized for controlling and describing telephony services on basis of IP. CPL by nature is flexible. It is implemented on Agent User Server or Network Server, like SIP server. The language is editable by means of Graphical User Interface (GUI). It has independency of signaling protocol or operating system.

CPL features are as follows:

Easily parsed and written. Verified across all components easily.

Addition of more extensible XML tags.

Easy implementation.

CPL open issues are as follows:

Currently CPL is in draft phase. Most work is enhancements. But framework is nicely and accurately placed.

H.323 and SIP APIs have not been standardized because of requirement of ability for customization.

JAIN

Java API for Integrated Networks (JAIN) is ready for being applied to philosophy of Java namely once, run to company of telecommunication. Objective of JAIN is bringing all of proprietary networks of telecommunication into one network of coherence. Specification of JAIN SIP is transaction of general purpose with basis of interface of JAVA to protocol of SIP. This is rich while defining it to SIP protocol and semantically both. The following figure 9.15 shows how JAIN SIP Transactions take place:

Figure 9.15: JAIN SIP

Motivations behind JAIN SIP is development of standard interfaces to protocol of SIP used independently or with high levelled programming environments and entities. JAIN SIP is to be utilized in lots of methods. Standardized interface used by communication developers as minimum for supporting SIP in its applications is a provision of JAIN SIP. The reference implementation of JAIN SIP

gives fully functional implementation of SIP that is utilized by developers for talking SIP from environment of Java.

SIP Servlets

Servlet is small program running on server. SIP Servlets is same as concept of CGI, but in place of using separate process, data is passed to class running within Java Virtual Machine (JVM) inside server. SIP Servlets is too similar to They simply enhance interfaces for supporting functions of SIP. Since these are written in JAVA, servlets possess portability between operating systems and servers. SIP Servlets are restricted to Java. But they suffer few overheads than SIP-CGI. The following figure 9.16 shows the JVM:

Figure 9.16: JVM

The following figure 9.17 shows SIP Servlets and HTTP Servlets:

Figure 9.17: SIP Servlets, HTTP Servlets

Utilization of JVM to execute servlets means that concept of JAVA is applied to protect server from script. Like SIP-CGI, SIP Servlets mirror HTTP Servlets operation very closely. They simply enhance interfaces for supporting wide functional arrays which can be executed by proxy, when compared with origin server of HTTP.

Conclusion

This chapter deals with the fact - how VoIP interfaces with the user. IP, ARP, and IGMP are three core protocols that reside in the Internet Layer. RARP and IGMP are the two lesser used internet layer protocols. With IP telephonies, endpoints require to possess IP connectivity with SIP Proxy Server or IP Communication Server and IP endpoints with whom communication needs to be conducted. MTA or ATA convert analog signal to digital signal. Off-net calls have been delivered to the PSTN network by utilizing IP’s gateway devices with LCR functionality. Phones have been connected to ATA rings. From this point, the rest of calls have been identical to phone calls done by using analog phones directly connected to PSTN by means of LSP.

The next chapter deals with hardware and software components of VoIP.

Points to remember

TCP/IP Primer is defined as standardized industrial suite of protocol used by devices for communicating, accessing, and finding with each other over transmission medium.

NetBIOS is API supporting desktop operation’s environment, this is in application layer of TCP/IP stack. TCP and UDP are two key components in Transport Layer of TCP/IP Primer.

JAIN SIP is richer while defining it to both the semantically and SIP protocols, in SIP Servlets data has been passed to those classes that runs within JVM inside servers and HTTP Servlets enhance simply the interfaces to support SIP functions.

Multiple choice questions

What is TCP/IP model made of? Application

Transport

Internet All of the above

What is Physical Layer on TCP/IP stack?

Media

Signal

Binary Transmission

None of the above What is meaning of decentralized implementation?

Mobile

Distributed workplace

Internet boom

All of the above

What are architectural options of internet telephony?

Non hybrid Hybrid

Non-trunk Replacement Trunk Replacement

What are usage methods in VoIP telephony? PBX Replacement PBX Extension

All of the above None of the above

Answers to MCQs

Q1: B, C Q1: D

Q2: A

Q3: D Q4: B, D

Q5: C

Questions

What are three crucial protocols residing in Internet Layer? 2 lesser utilized protocols are what?

What is IP Communication Server or SIP Proxy Server?

What is PSAP and E-911? What is ATA, MTA, LCR, and LSP?

What is SIP-CGI, HTTP-CGI and CPL?

CHAPTER 10 VoIP Hardware and Software Components

Introduction

IP end-points, that is, PCs and phones are required VoIP hardware. Currently, smart phones are very much in vogue. Personal Digital Assistants ( PDAs ) are utilized in combination with service providers. Access gateways enable service givers to connect with switches using VoIP deployment support over DSL and cable infrastructures. Integrated Access Devices ( IAD ), are meant for supporting multiple types of media. The various software components are signaling conversion components, application servers, media server, signaling server gateways, and policy server. For media encoding we use analog to digital convertor and digital to analog convertor. This process is called Codec . By codec the information rates become faster. Various header types are Physical Transport Header, PPP Header, IP Header, UTP Header, and RTP Header. VoIP phones encode media streams, packetize, and create headers for all sent out data streams.

Structure

In this chapter, we will cover the following topics: Hardware components

Software components

Media encoding Protocols

VoIP reliability

Mapping of telephone number

Objective

We will deals about IP end-points, that is, PCs and phones are required VoIP hardware in this chapter. Currently, smart phones are very much in vogue. Personal Digital Assistants are utilized in combination with service providers. Thus, this chapter helps with all the hardware and software components, and protocols with relation to VoIP.

Components of hardware

The various components of hardware are as follows:

End-points of IP (that is PCs and phones)

Call of VoIP originates typically on one among three device types: Analog phones connected with internet through MTA, fully fledged IP phones and softphones. All of such devices shall support few forms of VoIP signaling protocols, and norms for transmitting packetized voices. Choices typically are H.323 and SIP for RTP and signaling for path of voice.

Softphones are least costly. But it is limited option most of the times. Softphone is a program that emulates IP telephones on PC with no usage of DSP for transformation of voice. This is cheapest way for allowing calling via internet that initially works only with different IP end-points. Providers of service begin to assist softphones by permitting users for communicating with those devices that are connected to PSTN through gateways of givers of service.

Softphone advantages are their flexibility to user interfaces, their attractiveness toward users of computers and their inexpensiveness. Quality of voice depends on audio system quality in host computer. It is inferior usually with two different end-point types under same conditions of network. Softphone installations are made possible on laptops and Personal Digital Assistants (PDAs) combinational with service providers supporting to turn softphones into true portable devices of communication due to recent advancement. Experience of users to use softphones is

assorted radically from utilizing analog phones in environment of PSTN.

MTA is device which connects to internet, and where one or one plus analog telephones are plugged in. MTA does perform all key voice conversions while running of one of signaling protocols of VoIP. Once installation is finished, mimicking of experience of user is very close toward expertise of PSTN. It includes full usage of telephones that exist.

Service providers typically supply customers with device of MTA thereby establishing one’s account that includes assignment of regular telephone numbers from pool of giver. Voice quality does depend on codec implementation quality. MTAs utilize DSPs supporting typically more codecs when compared with softphones. MTA is more costly than softphone. But in residential services it will be subsidized typically by providers of service. At this time, it has emerged like preferred option of device.

IP phones are VoIP devices of higher end with prevalent usage in environment of enterprise at such a time. Such phones have richer featuring sets particularly of phones of office when connected with PBXs in turn replacing legacy phones belonging to work stations, as older PBXs have been replaced with systems of IP PBX. Such phones have shortcut keys, lots of control buttons, multiple lines to be supported and LCD screens larger enough for displaying short messages. Devices of today give excellent voice qualities under favorable conditions of network. Such devices still are too costly for residential markets.

Access gateways

Access gateways enable service givers of VoIP for connecting with switches of Class 5 like intermediate solution prior to availability of VoIP’s Class 5 soft Interface protocol to switches of class 5 in North America particularly is In lots of varied nations, it is to be V.5.2. Access gateways give important VoIP deployment support over DSL and cable infrastructures by doing integral conversions of media between PSTN and IP by converting voices that are digitized from IP endpoints into PSTN. The following figure 10.1 shows Class 5 Soft Switches:

Figure 10.1: Class 5 Soft Switches

Integrated Access Devices (IAD)

It is general name for devices of premise such as MTA. IAD is meant for supporting multiple types of media. Typical IAD supports data sessions and VoIP simultaneously. More IADs that are sophisticated, perform coordination and resource allocation functions for the excellent supporting of sessions that are simultaneous. The following figure 10.2 depicts IAD.

Figure 10.2: IAD

Components of software

The various components of software are as follows:

Signaling Conversion

When calls originate on VoIP terminal and terminate on PSTN phones, signaling conversion is core for setting up that call. Messages of VoIP signaling protocols need to be changed either to Q.931 data and to subsequently SS-7 information or directly to SS-7 data. In the networks of carrier, separate perform conversion functions. In simple implementations, integrated device of gateway supports media and signaling conversions.

Application server (that is billing and accounting)

Application servers frequently is known as Feature They are designed for further separating service logics from network infrastructures. Servers contain all software required for performing specific applications like billing platforms or pre-paid calling card solutions. They use service of policy servers (soft-switches) and media gateways for completing calls. It results in high distributed architectural models allowing and permitting application designers for concentrating on application logics.

Media server

Media server is programmable device of hardware designed for processing media events that is associated with calls. Media server of carrier class does handle media events for ports in thousands of numbers. Events provide text-to-speech (TTS) and automatic speech recognition (ASR) capabilities, play customized announcements and support multi-user conferences. The simpler tasks of such nature are given by media (trunking, access) gateways but media server scales to high calling volumes implementing fast additional functionality with similar heed on separation of calling controls to be processed from supplementary functions is executed. The carrier class media server employs larger DSP volumes for achieving needed scalabilities. The following figure 10.3 shows TTS systems:

Figure 10.3: TTS System

The following figure 10.4 shows Automatic Speech Recognition Systems:

Figure 10.4: ASR

Signaling server gateways

Signaling gateways do signaling conversions from PSTN signaling typically) to the signaling of VoIP (SIP, H.323, or Megaco/MGCP) for those calls that originate on the PSTN, and terminate on end point of VoIP and from signaling protocols of VoIP to SS-7 for those calls that originate on VoIP and terminate on PSTN. It is demarcation point between two signaling networks. Signaling gateways reside on higher availability or computer platform with fault tolerance and interfacing with media gateways and policy servers on side of VoIP and STP on side of PSTN.

Policy server

Policy servers is known as Call Agents and Soft-switches loosely somewhat. This is database server containing scripts of call processing for numerous call types that is offered by giver of service. Policy servers receive information of signaling from signaling gateways on the calls that come from PSTN. They either select instructions of pass routing or terminating gateways to application servers. They are key elements for separating signaling from calls and media controls. New service types are implemented easily by adding new scripts, or, in some cases, new application servers to databases. They reside on fault tolerant or high availability general purpose platforms of computer, and use database items of off-the-shelf. Converse with those media gateways that use Megaco/MGCP either, or, protocols that are proprietary and with the application servers that either use SIP or H.323.

Media encoding

World was analog when telephony began. Button of carbon microphone in handsets did vary voltage or currents applied across buttons while talking with users. Analog signals had been connected by series in case of switches, wires, plugs and relaying eventually to the earpieces where diaphragms connected to coils of wires near magnets did reproduce sounds of speakers for listeners. Ever since fifties there is digitization of voice while sending it like lots of bits. For doing this, nearly always encode data by utilizing some forms of compression, transmitting bits in huge numbers, and then decoding bits before converting it to analog. The following figure 10.5 shows Media Encoding procedures.

Analog to Digital - Encode - Transmission and Switching - Decode - Digital to Analog

Figure 10.5: Media Encoding

Mechanism of decoding and encoding when taken together is abbreviated as These codecs exist in telephonic systems ever since conversion to digital. Compression have existed just for this long because human voices have quite range of pitch and volume. But it has little ability for discriminating very small volume changes. Due to such characteristic of the human ear, engineers of Bell Labs compressed voice signals by using method of logarithmic coding. They used specific methodology known as and utilized 4bit mantissa and 3 bits characteristic. Slightly different algorithms of compression are used in all places of world by name of ITU did standardize such a codec like This G.711 has 8 kHz sample rate (it is rate where conversion of analog to digital runs); resulting data rate is 64 kHz. Lots of references to G.711 claim that this is uncompressed. But fact is that, it employs such way of logarithmic for increasing dynamic signal ranges.

Stating simply, scientists have designed those codecs which deliver lower or similar speech reproduction quality with low bit rates. Among common codecs present today that possess low rates of data are G.723.1 and Mobile systems utilizing codecs with more aggressiveness are AMR. These codecs compress speeches to lesser than 3 kHz, but they do not have similar quality like Codecs delivering better quality of voice are also there. These codecs are called They possess a lot more frequency, range and precision responses in comparison with Wideband codecs include The variations namely G.722.2 and G.722.1 are called AMR-WB.

Video is encoded by utilizing varied codec classes. Common codecs of video include H.261 (existing in current technological systems of video conferencing) and MPEG-2. The MPEG-4 is latest video codec. Subset of this is known as H.264. The following figure 10.6 shows an MPEG-2 Codec:

Figure 10.6: MPEG-2

The following figure 10.7 shows MPEG-4 Codec:

Figure 10.7: MPEG-4

Past the codecs, there is bit stream in few informational rates. In VoIP world, do packaging of data into those blocks that are called One packet is between 64 bytes and 1000 bytes. Packet has header that has content and addressing data which is used for getting packets from senders to correct recipients. of packets contains speech information. VoIP is then deployed typically in layered environment of transmission. All layers have own headers. In case one looks at bits on wires, they shall see probably these

below-mentioned headers. The following figure 10.8 shows the headers types:

Figure 10.8: Header Types

(Layer 1) Physical transport header

This header has dependency on physical ways that join two parties. For example, on Ethernet, a person first finds Ethernet headers. Then header is modified from hop-to-hop because there is general finding that lots of transports have been involved in paths from originating phones to terminating phones.

PPP Header

The header is utilized on few transports where numerous hops are present between two parties having contractual relationships. Packets are encapsulated by PPP sending over varieties of transports, over lots of hops, until this gets to entities unwrapping encapsulation prior to its routing onwards. PPP has availability but not always. It is found between ISP and residence, over transports like modems or DSL. PPP is used for routing all packets to common points from single subscribers in network before being dispatched to ultimate destination.

IP Header

It is base norm for internet. IP has addresses of destination and source of packet. In NAT’s absence IP Header is created at source, and remains unchanged till it arrives at destination.

UDP Header

UDP is utilized for voice, very unlike most files or web data that use TCP. UDP is simple. It has lesser overheads for processing when compared with TCP. But it behaves poorer when it has congestion. UDP does implement those that allow single addresses for handling multiple information streams. UDP Header is unchanged always to destination from source.

RTP Header

RTP provides error monitoring, timing and codec selection functions for streams of media. RTP header stays with no conversion from source to destination.

VoIP phones encode media streams, packetize streams and create all headers for all sent out media streams. PPP and physical headers are recreated and removed because packets traverse networks. At destination, IP headers get packets to correct devices, UDP header gets it to the right and RTP header gets it to accurate media streams in case there are one plus. Pay-loads are played and extracted to users.

Protocols

The various protocols for VoIP are as follows:

H.323

It was conceived originally by ITU-T. It has IP equivalence of H.320 for video-conferencing. H.323 is deployed widely for these uses today. H.323 is used as protocol of VoIP, in basically applications of enterprise telephony. It is peer-to-peer protocol. The following figure 10.9 shows peer-to-peer protocols:

Figure 10.9: H.320, H.323

Terminal: It is like phone or video-conferencing system. End-points are intelligent moderately. Basic calls are finished between two endpoints with no other entity.

Gateway: It is interface between systems of H.323 and varied systems like PSTN. Gateways appear to become set of terminals of H.323 to different H.323 terminals. Gatekeeper: It is service of routing and centralized registration. Terminals register with local gatekeepers and gatekeepers route all calls. Systems of H.323 have been divided into “Zones”. Each zone is governed by gatekeeper. Calls have been routed by source gatekeepers to destination gatekeepers and then by destination gatekeepers to destination gateway/terminal. Once calls are established, media does flow between end-points and not through gatekeeper. The possibility of hierarchical gatekeeper arrangement is there. Multipoint Conference Unit (MCU): Calls of two plus parties are bridged by MCU. Terminal maintains basic calls to ports on MCU and MCU mixes streams of media for delivering audio (and data/video possibly) to all endpoints. H.323 comprises of key version 5 of H.323 document and many subsidiary protocols:

H.225 is protocol for call signaling. Three parts are there. Not all of this has usage for all calls: and basic call signaling. Protocol of signaling of call is same as ISDN Q.931. The following figure 10.10 describes audio standards:

Figure 10.10: G.723, G.728, H.225, H.245, RAS, G.729, G.711, G.722

The following figure 10.11 shows the signaling protocol stack:

Figure 10.11: H.235, H.450, Q.931 H.245 is control This is concerned mostly with negotiation of parameters of calls. It is that codecs are utilized for streams of media. H.235 is H.323’s security specification.

H.350 defines directory mechanisms to find identifiers of endpoint. The following figure 10.12 shows the mechanisms for endpoint identifiers:

Figure 10.12: H.350 H.450 does define like call transfer and call forward.

SIP

It is standard of IETF for multimedia session P2P control. For these discussions’ sessions are equivalent to calls. SIP is primarily deployed in applications of VoIP, but it is heart of instant messaging and presence (SIMPLE) and standards of 3GPP2 wireless signaling. The following figure 10.13 include the SIP Architecture:

Figure 10.13: SIP User Agent: These are endpoints of calls. Call originator is User Agent Client (UAC) and call terminator is User Agent Server

(UAS). SIP phones have UAC and UAS both such that it possibly can receive and place calls. User agents are too intelligent. Calls that are advanced and basic are placed between two user agents with no different other entities. The following figure 10.14 shows user agent client and user agent server:

Figure 10.14: UAC, UAS

Proxy server: It is intermediary in signaling path, but it is not the case in media path. These servers give varied services of calls to SIP calls like routing, registration, and redirection. Signaling

messages traverse one plus or one proxy servers in paths to UAS from UAC. Once calls are established, media flows directly between UAC and UAS.

Media Servers: It gives services of media to endpoints like conference, playback/ record bridging or announcements.

SIP specifications do define basic protocols. There are number of extensions that are common. The following figure 10.15 shows the basic protocols:

Figure 10.15: SIP Specifications (RFC 3261)

DHCP option for Session Initiation Protocol servers SIP UPDATE method SIP refer method SIP refer specific event notification

Private Extensions toward SIP for Asserted Identities inside trusted networks (RFC3325)

Megaco/ H.248

The protocol is joint collaboration of IETF and ITU. IETF names it Megaco. ITU does call it Current releases are Updated version is being finished. Version 2 is found as Megaco is protocol of Master Slave. The following figure 10.16 shows the Entities of Megaco.

Figure 10.16: Megaco Entities

Media Gateway (MG): It is client or slave portion. MG has terminations of media and signaling connections to Media Gateway Controller (MGC). MG makes connections among its terminations under MGC’s direction.

MGC: It is server or master portion. MGC controls one plus or one MG. MGC have connections with various MGCs. But these do not use Megaco for such connections of P2P. Instead, they utilize protocols like SIP or ISUP (in case of Q.1901, and Bearer Independent Call Control (BICC)). The following figure 10.17 shows bearer independent call control:

Figure 10.17: BICC

Megaco is deployed in architecture of soft switch where softswitch is MGC and phone is MG. It is close to PSTN that exists with phones being non-intelligent relatively and having relationship of slave/master with controlling MGC. Typically, Media Gateway does not provide maintenance for call states. It maintains state of media bearer. Megaco is deployed at large gateways while connecting to PSTN wherein costs of implementing more complex and difficult protocols like SIP at scales of gateways with tens of 1000s of ports is not apt, often.

Media Gateway Control Protocol (MGCP)

It is Megaco’s direct predecessor. It enjoys deployment that has significance. Few vendors have a feeling that Megaco’s deviated from original MGCP to great extents, is more critical than required. MGCP is defined with Informational RFC and not with Standards Track, as Megaco and SIP standards are. Current version of the MGCP is RFC3435. Entities of MGCP are as follows. The following figure 10.18 tells us about the MGCP entities:

Figure 10.18: MGCP Entities

Media Gateway (MG): It is portion of client or slave. MG has signaling connection and media endpoints to Call Agent (CA). MG makes connections with its endpoints at CA’s direction.

CA: It is portion of server or master. CA controls one plus or one MG. CA often has connections with different other CA’s. But they do not utilize Megaco for these P2P connections. In its place, they use protocol like SIP.

MGCP is deployed often between trunk, gateways or residential and calling controllers (soft-switches). As in Megaco, Media Gateways of MGCP have no maintenance of call states. CA does have it. So MGCP Media Gateways are non-intelligent relatively.

Others

BICC extends signaling protocols of SS7 (ISUP) for specifying Voice over ATM or VoIP bearer. BICC idea is retaining all methods of SS7 signaling, but use it for controlling devices of VoIP. Call signaling traverses those networks of SS7 that exists. Bearers’ traverses ATM or IP networks. Q.1901 defines BICC like minor enhancement series to Q.761-Q.764. BICC yet needs significant deployment. Skinny-Cisco Systems: It is protocol for proprietary phones. Numerous installations of Cisco Call Manager use proprietary protocols between Call Manager and phones. It is called Skinny Client Control Protocol (SCCP). The following figure 10.19 explains the working of Skinny Control Protocol:

Figure 10.19: SCCP

Telephony Routing over IP (TRIP): RFC3219 describes protocols used for identifying as to which gateways shall be used for sending calls. By using variations of advertising BGP route protocols being used in routers of IP, TRIP is utilized where there exists lots of gateways in networks. More than one of this reaches desired phone numbers. TRIP gives information for selecting gateway. The following figure 10.20 tells us about TRIP.

Figure 10.20: TRIP The following figure 10.21 explains about the BGP route protocols:

Figure 10.21: BGP Route

Reliability

Reliability is TDM switched service’s traditional sense that is judged often by principle of availability of 99.999 percent that is principle. With service of VoIP, operating models deployed for some environments influences reliability. Variations of standard VoIP service is set below with attention to reliabilities for all.

Reliability of trunk replacement

Reliability is sustained at higher levels in model of trunk replacement because of nature of transport of IP. Trunks of TDM are replaced with Trunks of IP over managed network of IP. In this environment end users have unawareness to place calls over networks of IP. With IP networks, most models of trunk replacement have overflow of TDM and in turn they fail over for redundancies.

Reliability for hybrid architecture

Providers of service offer VoIP on basis of one way. It offers only termination or origination points for traffics of VoIP for hopping off or on the PSTN.

Model is utilized in applications of Greenfield where no switch of legacy exists, or Class 5 switches are present. They are augmented by voice application servers or soft-switches. MG is used. But this model is used in native environment of IP where no media conversions are important. Reliability hinges on redundancies of Proxy Servers and eliminate failure in terminating/originating managed IP networks and be hand from MGC and PSTN.

VoIP reliability that is end-to-end

Network of IP is resilient and distributed inherently. True unified voice system is distributed across lots of sites by using architecture that is P2P which has no failure points at all.

Switches of IP voice specifically designed for voices have one-toone incorporation of all call processors. All switches are peers with fuller complements of routing data held safely in local memories, and can commit operations like standalone switches in case its site has been cut-off from backbone of IP. It makes calls of best effort by itself, then using fail-over PSTN trunks, if required. When switches are restored or added to networks, existing switches, and they at all sites discover one another and then begin to work together.

If switch giving PSTN accessibility to a site fails, the peer switches in IP network elsewhere provides alternate accessibility to PSTN to users at site in question. Till the time backbone of data stays up, distributed voice networks of this type cannot have outage up till all switches simultaneously go down. Reliability comes built-in with such approach. Availability of voice requirement is met. This is increased to 10 nines with installation of redundant switches with PSTN accessibility at all sites.

Mapping of telephone number

ENUM is core factor of VoIP though it is not protocol. Methodology in internet that is used for converting to server’s IP address that supports name of domain is namely Name or DNS. The following figure 10.22 shows ENUM which is a core factor of VoIP:

Figure 10.22: ENUM

The following figure 10.23 shows server’s domain name:

Figure 10.23: DNS

DNS is distributed database that is quite large. ENUM extends DNS to provide E.164 phone numbers to Uniform Resource Identifier (URI). Protocols like H.323 and SIP utilize directly URI for routing calls. URI is used for determining carrier which serves that number of phones. This is the way with help of which ENUM has analogy with query of LNP TCAP, but with a lot more utility.

Given E.164 in the ENUM, a person can determine if numbers are endpoints of VoIP and it is reachable from VoIP end-point through gateways and to which carriers serves presently these numbers. ENUM is actually global. ENUM is defined in

Conclusion

The chapter is a detailed explanation of all the components and protocols of VoIP. Media events provide ASR and TTS capability, playing announcements that are customizable and supporting conferences of multiple users. SIP is IETF standard for P2P multimedia session controls. Entities are User Agent (UAS and UAC), Media Servers and Proxy Servers. ENUM is VoIP’s core factor, though not a protocol. Internet methodology utilized to convert to IP address of server supporting domain naming is DNS. ENUM extends the DNS for giving E.164 numbers of phones to URI.

The next chapter deals with a use case scenario in most excruciating detail.

Points to remember

VoIP calls typically originate on one of three types of device: Softphones, analog phones having internet connection by means of MTA and IP phones that are fully fledged.

Common video codecs are MPEG-2 and H.261. Latest codecs of video are MPEG-4 and H.264 is subset of this. RFC3261 is basic SIP protocol. Extensions has been like RFC3361, RFC3325, RFC3311, RFC3265 and RFC3515.

BICC idea retains all SS7 signaling ways, but it is used for controlling VoIP devices and SCCP is proprietary phones protocol.

Multiple choice questions

What are hardware components? PC

Phones

PDAs All of the above

What are software components?

Non Signaling

Signaling

Billing

Accounting What are header types?

Layer 0

Layer 1

PPP

None

Where TRIP and BGP route is used?

Gateways Protocols

All of the above None of the above

What are factors of reliability? Trunk Non-Replacement Trunk Replacement

Hybrid architecture Non-Hybrid Architecture

Answers to MCQs

Q1: D Q2: B, C, D

Q3: C

Q4: C Q5: B, C

Questions

What is ASR and TTS ability? What is media encoding?

What is H.323 protocol? How is it related with H.320?

What is key version 5 of the H.323 protocol with subsidiary protocols?

What is SIP? What are its entities?

What are Megaco and MGCP entities?

What is DNS, ENUM and URI?

CHAPTER 11 Business Model and Market Model in Relation with Internet Telephony

Introduction

SYSteam Nat AB is a premier re-seller product of Cisco, which provides cabling installation and wireless communication with internet accessibility and fiber optic measures. Market model for internet telephony is marked for student housing facilities, and societies for tenants in general. The business models of VoIP are Build-it-Yourself, Resale, Outsourcing and Recommendation. There is a great variation in the client’s needs. The people require faster marketing time, shorter deployment and development time, scalable, and stable core and support of sub-supplier in pre-sales and post-sales. Skype is a global peer to peer ( P2P ) telephonic means for superior and free calling worldwide. It is the most downloaded software till date. The prices of residential VoIP’s require a normal service activation fees for initial set-up and varies from industry to industry.

Structure

In this chapter, we will cover the following topics: SYSteam Nat AB current situation

Requirements of customers

ISP goals Competitors of broadband

Customer proposition of internet telephony

Residential VoIP prices

Cost of internet telephony

Objective

In this chapter, we will talk about the in depth about SYSteam Nat AB is a premier re-seller product of Cisco, which provides cabling installation and wireless communication with internet accessibility and fiber optic measures. This chapter is all about a use case scenario.

Use-case scenario

SYSteam Nat AB operates as ISP local broadband in Uppsala. It is called formerly as It possesses 25 years of experience in documentation, installation, and service of the systems that have cable connections. Currently there are 15 employees in this. Industry is subsidiary of SYSteam AB being wholly owned. It offers sophisticated solutions that are based on IT to customers of Europe. SYSteam AB group has 1000 employees with turnover in year 2003 of MSEK numbered 1046. In Uppsala 100 employees are present.

SYSteam Nat is integrator with authorized system for generalized systems of cabling from Avaya and AMP. It is premier reseller for products of communication from Cisco.

SYSteam Nat AB does offer:

Internet accessibility.

Installations of fiber optics.

Wireless communication. Installations of broadband in buildings where people reside.

General cabling system installation.

Operation of networks of broadband in those places that is residential with lots of people dwelling in such areas.

SYSteam Nat AB does operate as local ISP broadband in

Current situation of SYSteam Nat in model of market and business Present day customers mainly are residential individuals.

Market model

In this section we will learn various marketing models related to SYSteam Nat.

Students housing facilities

Major parts of candidates staying in Uppsala hostels are connected to SYSteam Nat through varied networks. There are nearly 9900 flats or rooms in Uppsala’s student housing which is owned by nearly 20 foundations and industries. Studentstaden i Uppsala AB is largest owner with 6200 circa flats and rooms. While including on-going constructions, estimation of total flats and rooms in hostels for increasing to nearly 10300 by year 2005 end is done.

Societies owned by tenants

Much low penetration is in hand to tenant residences. Exact numbers have not been revealed because of reasons of confidentiality. As per statistics for 2003 year, Uppsala has 28693 flats in tenant owner societies that consist of semi-detached homes numbered 2613 and buildings of apartments countable 26080. Nearly flats numbered 995 in apartment buildings had been under construction till 2004. It does mean that at end of year 2005 flat numbers in Uppsala in societies owned by tenants shall be nearly 30000. These human inhabitants have 20 flats at least subscribing to have contracts with SYSteam Nat. Companies does offer accessibility of internet to those industries that are situated in buildings possessing connection with SYSteam Nat dark fiber nets. Offered bandwidths vary from 1 Mb/s to 100 Mb/s over the Ethernet.

SYSteam Nat offers broadband to customers who are residential as access of internet with up till 10 Mb/s full duplexes over Ethernet. It is with IP address public dynamic for 250 SEK per month. Central firewall services costs freely and is optional. Connecting fees are 495 SEK. Contract is to roll 3 months contract with 1-month notice period. Web homepages and e-mails have not been offered. Internal securities are given by means of Virtual Local Area Network (VLAN) that is private. External security is provided through authentications while utilizing internet by virtue of central service of firewall. SYSteam Nat tries to pay priority to offering of IP telephonies to candidate housing and

tenant owner residences in Uppsala. The following figure 11.1 shows the Market Model related to VoIP:

Figure 11.1: Market Model

Models of business

ISPs of broadband, such as SYSteam Nat AB, has interesting base of consumer for delivering services of VoIP. Such industries act quickly for avoiding cherry pick competition people to and fro of existing base of customers itself. ISPs of local broadband try for creation of more values by offering QoS that is better than competitors and by services of bundling. They have benefits in initial sales to individuals of subscription of broadband. Reason is that, loyalty of brand, advertising, requirements of single bill and preference of geography is always there.

There are four models to offer services of VoIP toward end consumers:

Build-it-Yourself

It is technically challenging and serious undertaking in case skilled staffs are not present. This alternative is too unattractive for lots of organizations because of platform of Operations Support System (OSS) and billing. This is well-known to whichever veteran of telephony. The following figure 11.2 shows Operations Support System architecture:

Figure 11.2: OSS

Resale

It is easiest methodology. It is very dangerous from long term viewpoint since customers have been owned by companies of third party. They pay only commissions back to ISPs. Branding is complex here. People discover easily that they do not work at all with ISP, and as consequence skip ISPs and in turn directly going to giver of VoIP. Risk of consumer dissatisfaction with both reseller of VoIP and provider of facility is also there. The following figure 11.3 shows the business model related to VoIP:

Figure 11.3: Business Model

Outsourcing

It is the solution that is most balanced from perspective of investment and risk. Here platform of telephony back-office, OSS, network resources and termination have to be handled by industries having multiple different masses for serving scalable systems. Staffs that exist already have to be employed by ISP for handling customer supports like service on the site and hot-line supports (trouble-shooting and advice). ISP owns consumer for avoiding confusion of individuals over support and branding that might occur in conditions of resales. Other outsourcing local ISP broadband reason to VoIP services to providers of third party is avoiding recurring costs and higher capital expenditures for staffs having experience.

Recommendation

There is possibility to help masses in choosing freely IP telephonic givers by recommending correct and proper providers.

Attitudes and requirements of customers

There is variation in needing factors of people. It depends on requirements of their services that they buy from ISP and their applications. Few common factors needed are as follows:

QoS (minimal jitter, bandwidth and minimal delay that is guaranteed) Connectivity

Numerous services that use single connections

Throughputs

Reports on Service Level Agreement

Higher availabilities of services

Security

Surveys carried out in 2005 January in US stated that many barriers to VoIP people adoption is there. Sample of decision makers of business numbered 335 and US adults calculated 1473 did participate in this research. Few barriers relate to attitude that

is present to new services. Among ones who have VoIP awareness, but do not utilize it, results were:

A person not knowing a lot about VoIP (47 percent)

Givers of VoIP fail to exhibit people with convincing stories (36 percent)

VoIP is technology that is unproven (22 percent)

Individuals wait for VoIP becoming more main stream (34 percent) Too much of complications in VoIP (installation and equipment) (22 percent) Savings that is potential does not have hassle worthiness (25 percent)

Recommendations help in moving toward VoIP (27 percent) Varied barriers do stem from technology’s perceived drawbacks. Among people having awareness about VoIP, but cannot use it: VoIP is subjected to privacy and security issues (60 percent)

Quality of calls have worsened when compared with old service of phone (52 percent)

VoIP does not support 911 calls in specific areas (62 percent)

In situations of power failures, VoIP does not operate (58 percent) Study in US did focus on knowledge regarding the next generation values of voice broadband services to masses. 12 services were elaborately examined to determine which one possessed greatest influence on demands. Video Calling, Click-to-Dial, and Caller ID occupy top three spots with respect to their impacts on household broadband demand. Single Voice Mailbox (for home phones and wireless phones) is put on fourth spot. Call Screening and Routing hold fifth spot. Most interesting finding entirely is that broadband phone services with no added service of next generation have little values to customers of broadband. Lesser than 5 percent of individuals state that they shall pay for phone service of broadband, thereby, not including any added services of next generation. All lead researching companies believe in offering that in market places without multimedia services and services of next-generation management of communication, only sell on value propositions bundled with services, like discount packages of long distances.

Goals of ISP

ISP has numerous goals to be met for making profitable business: Faster marketing time.

Shorter deployment and development time of services based on IP.

Support of sub-supplier. Stable and scalable core (backbone) network for giving assistance to larger numbers of consumers.

Multiple services for using similar network.

Easiness of operations of network for modifying/upgrading requests of people and for purposes of maintenance.

Competitors of Broadb and Blixtvik Internet och Telefoni AB

It is broadband on basis of Uppsala and giver of IP telephonies to private persons, industries, and properties in area of Stockholm, Karlstad, and Uppsala. Internet telephony with ATA has been offered in only IP collaborations. Monthly fee is 62.50 SEK including those call charges that are ordinary. ATA costs are 1995 SEK. Initial fee is 375 SEK.

Bredbandsbolaget (B2Bredband AB)

It is second largest provider of broadband after Telia in Sweden with 335 people that show 24 percent market shares. It is largest when query of broadband through cable networks rises. IP telephonies are offered with ATA. Neither subscription of internet nor computer is important. No initial fee, but 99 SEK is monthly fee. Ordinary call costs for all calls excepting free calls between own subscribers. Internet telephony serves 80000 consumers today. In 2003, it was launched. IP-TV was launched recently by Bredbandsbolaget. So, this offers full triple plays. Norwegian organization Telenor signed agreements on 2005, March 23 with shareholders of Bredbandsbolaget for acquiring this industry. Acquisition was finished in 2005 June that was subjected to Swedish competition authority’s approval.

Com Hem AB

In 2004 April there was announcement that agreement between Com Hem AB and CygateMaldata was signed with regards to maintenance, supply and implementation of technical machinery in triple venture of plays (telephony, broadband, TV) on basis of communication via net of cable-TV. Com Hem AB offers telephony by means of ATA. No initial fees. Three subscription types are given: 125, 60 and 80 SEK/month with ordinary charges of calls. In 2005 February, there were declarations that offer of internet telephony had been introduced in Eskilstuna like third city after Lund and Linkoping. Uppsala customers got offer of telephony in May.

Tele2 AB

It supplies IP telephony through adapters irrespective of ISP of users. Starting package that includes ATA costs 699 SEK. Monthly fee either is 0 SEK or 99 SEK/month. Later offering is for free of costs Sweden calls. Different calls have call costs that is ordinary.

TeliaSoneraSverige AB

Telia are TeliaSonera Group’s part since 2002 December. TeliaSoneraSverige AB operates under brand of Telia in Denmark and Sweden. Telia announced during 2005 that it shall offer internet telephony, TV and broadband, that is, quattro play or triple play in the case mobile telephones are added to 70 percent of households of Sweden. Target groups initially for IP telephonies are families with students and teenagers who moved out from their residences to study in big universities. During 2005 beginning, Telia introduced a digital TV through its network of broadband to some addresses in Sweden’s 15 large cities. Telia step-by-step and gradually has become stronger competitor in fiber networking market to Bredbandsbolaget. This organization has succeeded in connecting with 100000 homes that made a contract of fixed accessibility of 10 Mb/s. Telia has nearly 30000 customers of LAN presently.

TeliaSoneraSverige AB made an announcement in 2005 March that it plans to launch new generation’s internet telephony-based systems of communication both for future and present mobile and fixed telephonic services. In most aggressive scenarios AXE networks for this day’s fixed telephony is replaced by converging networks for integrated fixed and mobile telephony services in 5 years (2007 to 2012). New networks have capacity of 8 to 9 million subscribers. TeliaSoneraSverige AB shall make first decision regarding 2006 migration. Pre-requisites are Next Generation

Networks (NGN), new global standards developed currently by European Telecommunications Standards Institute (ETSI), and 3GPP, for integrated fixed and mobile networks, that handle all types of services of real time like TV services, telephony services, and information services.

In 2005, May 3 TeliaSoneraSverige AB declared that they shall begin tests in place Denmark with new concept wherein mobile telephones are integrated with internet telephony at homes. aim with the tests is gradually providing integrated solution of IP telephony and mobile telephony where people only need single wireless phones for all of their telephonic needs. Phone serves like internet phone inside 4 walls of residence. When individuals leave home, phones switch over automatically to network of mobile. TeliaSoneraSverige AB evaluates and monitors alternative technological factors for integrated solutions of IP telephony and mobile telephony. Possible launches in various markets depends on terminal and general marketing condition According to Kenneth Radne, broadband telephony of Telia has characteristics that follow: Accessibility that is independent and internet based at greater than or equal to 128 kb/s

Hotsip platform On basis of SIP

Based on software clients

Integration of service with WLAN, PSTN and mobile networks

Part of wide concept-integrated communication Charges are monthly fees, initial fees, and ordinary costs of call and not flat rates.

Uppsala Stadsnat AB (Uppsala Metropolitan Area Network)

It is daughter industry to TDC Song. It is local organization with Metropolitan Area Network (MAN) accessibility for 90 percent of all companies and 75 percent of all of the households in Uppsala. Its network has linkage with lots of international, regional and national networks. SWOP named theory of broadband is there, where subscribers choose one out of five providers (Ownit, Blixtvik, Bahnhof/One.se, Ready, and Tele2) for approachability to internet. Uppsala Stadsnat AB offers internet telephony from four givers (Ready, Blixtvik, Tele2, and AllTele). It shall soon offer from Canal Digital, a broadband TV. So, this shall within short time span have ability for providing triple plays to consumers.

Skype

Skype is Global P2P Telephonic Industry that changes world of telecommunication by giving people superior quality and free calling world-wide. Skype Group Technologies has its headquarters in Luxembourg. A Dane namely Janus Friis and a Swede named Niklas Zennstrom created Skype. KaZaA was also founded by these individuals. KaZaA is program that utilizes P2P such that computer files are allowed to be shared. Software, music, games and video are few files of computer. It is claimed to be most downloaded software of internet till date.

Skype is software of P2P IP telephony. In this call are directly routed between computers of users. Calls to varied global users of Skype are cost free. Income does come from SkypeOut. It allows people to call from PDAs or PCs to cell phones or fixed line conventional phones. Software of Skype allows transfer of files across platforms. Instant messaging to varied users of Skype is there. Enterprise clients are currently not targeted by Skype. Significant benefits of Skype when compared with various telephony givers are better quality of sound, low costs, security is improved and low and easy cost of installation. Critics have pointed that the Skype technology still is proprietary closed environment with accessibility and availability only for its own masses. For survival reasons, Skype has to peer with dissimilar networks. For risk of losing people, Skype peers with networks that are more open. People numbered 1.2 million and plus use SkypeOut from 2005, April 15 onwards.

On 2005 January 6, good report with name of Skype on Telecom Service was published. Conclusion of report was: of VoIP that do not use technology of P2P in retail segments cannot survive on current business models. P2P technologies possess cost advantages that are inherent with retaining of quality benefits too. Competitiveness of Skype in enterprise segments is seen when Skype introduces solutions of business. Skype shows key disruptive forces for all providers of telecom services. It results in swifter and speedier data convergences and voice traffics. Skype accelerates those changes that already underway in business models of telecoms, to ones where basic services of voice are given fee free. They are subsidized by generation of revenue from service that are value

Registration of Skype is more than for 114 million cost free downloads with call minutes of 9 million served as of on 2005, May 25. Skype has reached between 140-250 million users of retail by 2008 end with respect to market. On 2005, April 15 Skype announced public beta launches of Skype Voice-mail and SkypeIn. Customers of SkypeIn receive inbounded calls to clients of Skype from ordinary mobile phones or fixed telephones. Consumers of SkypeIn choose area code and country. They are assigned regular telephone numbers. Users can buy nearly three numbers from home countries in US, UK, Denmark, Sweden, Hong Kong, France and Finland during beta period. A person can purchase phone numbers from Skype in those countries where they do not live. It means that in case your crucial contacts are present in US and you stay in Sweden, then buy 12-month US area code and phone number subscription, and then contacts of US can possibly call

you costing much less amounts. Price is thirty euros for subscription of 12 months that includes free voice mails. Skype works together with partners on router-based items for reaching out to large groups that do not desire to have that their computers shall run continuously. Skype integrates its software into either base stations or cordless handsets that users then can directly connect to routers of theirs. Skype does claim that it has far superior call quality in comparison with those traditional calls that use narrow-band technology of PSTN. Skype does use codecs of wideband, with much broadened range of frequency. NiklasZennstrom does predict that telephonies shall become software application to be run on device simply. It changes rules completely. Example is that, one cannot charge for calls, just as he cannot charge for different tasks of software. Skype makes money by offering free services but there is expense for services that are added value wise like SkypeOut and for Skype phones later. Today Skype is affiliating to sell voice mail services and varied premium commodities by taking increasingly pragmatic approaches to business. Nearly organizations numbered 1800 have agreed to selling of products of Skype with returning of 2 to 10 percent of generated revenues. Registered sites have linkage with both Skype Store and home page of Skype, wherein people can buy premium services of Skype from affiliated websites. Referral period’s does credit publishers for all buying made from visits of returns to Skype Stores up till thirty days after original clicking.

Microsoft

Intelligence of global market and IDC advisory firm has reported that Microsoft is poised for integral push of VoIP: After building strategy of VoIP quietly over long span of time on lots of fronts, Microsoft currently is stepping its efforts of VoIP to its core initiatives both in the carrier and enterprise space. Centerpiece of such increased activities is SIP based applications of collaborations that are developed for Live Communications Server (LCS) of 2005 item of Microsoft In past, Microsoft kept lower profiles around marketing and development of VoIP strategy, however this is modified. The following figure 11.4 depicts the Live Communications Server:

Figure 11.4: LCS

Industry has started partnership with important vendors of IP-PBX, like Alcatel and Siemens, for helping in jump development for moving into enterprise’s IP telephony. Collaborations of this type show key steps toward strengthening of position of Microsoft in space of VoIP. But, on flipside, multiple IP-PBX market vendors worry regarding Microsoft like competitor, in especially realm of high-end partnership applications. Collaborative operation of real time like web conferencing and instant messaging increasingly has features of VoIP. Association dramatically enhances stature of VoIP.

Customer proposition of ISP internet telephony

As instance of end user value propositions, followings are quoted from Hotsips:

To get more values out of connection of broadband.

Quality of voice that is most worthy than ones received in PSTN are worthier when compared with cellular networks. There are options to make cheap phone calls (example, free or low-cost calls from PC to PC).

Better and convenient handling of call like incoming phone call control and click-to-call.

High quality video telephony that is inexpensive.

Second telephonic line (with own number of telephone) that is less costly for owning and installing.

Get own number that is personal and private for bringing with one when moving, for instance.

Easy for using service with self-provisioning by virtue of web that is available for 24 hours in one day.

So, there are lots of convincing reasons why end users can probably ask ISPs or others for the solutions of IP telephonies.

To price VoIP that is residential

There are normally service activation fees or initial setup fees or service connection fees. Few organizations as promotion offer first service month for free, though they still charge fees for initial setups. Most industries charge a fee for monthly services, but some offer fee for yearly services with corresponding discounts that is additional for payment in advance. Fees for the traffic of telephonic calls have been charged (for terminated calls particularly in PSTN) and for services of video conferencing and video telephony. There is finally list of features added value wise like games, personal controlling of handling of calls, transfer of PC to PC files and phone numbers provided additionally. Use routing based on presence of subscribers seeming to have willingness to pay in extra amounts.

Offers from operators of internet telephony that use SIP can be:

Cost free

Downloading software of client to PC. Call to varied operator subscribers and different users of SIP on “online”.

Voice mailboxes with e-mail notifications or e-mail transfers of voice mails. Representation of number like “Caller ID”.

Call transfer.

Video telephonies.

Possible charges that are extra

One or one plus numbers of telephones. For receiving calls from mobile net or PSTN.

Costs that are extra

Calls through mobile net or PSTN. Adapters to be used with normal phones.

IP phones that involves setups.

Requirement is great accessibility link or 128 kb/s.

Cost of internet telephonies for consumers

In reports PTS has released that they calculate average costs for IP telephony from various providers. It was done for six groups of customers. Two extremes were parents of teenagers and single people who are stressed for jobs. Monthly charges for first category vary between 250 SEK (Com Hem Large) and 80 SEK (Tele2 broadband). Corresponding images for second category were 385 SEK (Wx3) and 155 SEK (Rix broadband). Telephony of Rix broadband and telephony of Tele2 broadband is cheapest alternative. Conclusion of report is that a home with lower telephony consumptions relatively saves money with IP telephonic changeovers if supplied especially via cable-TV networks or LANs. If communication is instead by means of Asymmetric Digital Subscriber Line (ADSL), wherein people in lots of cases need to make a payment of subscription fees to TeliaSonera, similarly saving of costs have no availabilities. Subscribers with lower gain of consumption from fixed low monthly charges are there while volume individuals have this with small parts of total fees in fixed costs. Further fact to tell is that international call pricings is of core importance. The following figure 11.5 shows the Asymmetric Digital Subscriber Line:

Figure 11.5: ADSL

Market Cohesion research shows that services of VoIP rarely give users saving of costs when compared with services of conventional telephones. between users have freedom, so if too large call proportions are between the users-relations say, or, branch offices, in case of business, then such calls shall be with no costs. But the typical users having inability to make these arrangements shall not

Fact is that, for numerous users VoIP proves to be a lot more costly in comparison with choosing of appropriate packages from providers that are conventional. Based on their model of analysis of cost, Skype is more expensive significantly for typical family. So, question rises that, why should lots of utilizers use Vonage and Conclusion is that, model of cost analysis used does not provide full credits to such suppliers.

Conclusion

This chapter is all about the various functionalities of a particular use case. Build-it-Yourself is challenging technically with serious undertaking when compared with no presence of skilled staff. This alternative is very unattractive for lots of industries because of billing and OSS platforms. If interaction is through ADSL, in which numerous cases require making of subscription fees payment to TeliaSonera, then there is no cost saving. Centerpiece of Microsoft is SIP collaboration application based which have been developed for LCS of 2005 items of the Microsoft office. Connectivity, security, QoS, high service availability, throughputs, Service Level Agreement reports and lots of services using single connections are needed factors and attitudes of masses in general. Com Hem AB, Blixtvik Internet ochTelefoni AB, Tele2 AB, Bredbandsbolget (B2Bredband AB), Uppsala StadsnatAb (Uppsala Metropolitan Area Network) and TeliaSoneraSverige AB are six Broadband Competitors.

The next chapter is a detailed explanation of technology model, economics model, and In-Practice models of VoIP.

Points to remember

SYSteam Nat AB has 25 years of installation, documentation, and system’s service experience that possess cable connections, and this organization is subsidiary of the wholly owned SYSteam AB.

SYSteam Nat gives broadband to consumers who have been residential as internet access with nearly 10Mb/s over Ethernet’s full duplex and internal security is given through private VLAN. ISP internet telephonic customer proposition are as follows: Get more values from broadband connection, calls that have convenient and better handling, inexpensive higher quality video telephonies, cheap phone calls are made, voice qualities with most worthiness are selected, own private and personal number is got and second telephonic line with least costs is installed and owned.

Multiple choice questions

What is market model of SYSteam Nat? Student housing facilities

Senior housing facilities

Girls housing facilities All of the above

What are four business models?

Build-it-Yourself

Don’t build it yourself

Work it out yourself

Don’t work it out yourself What are broadband competitors?

Blitch AB

B2B BredBand AB

B2B Bloat AB

B2B Blink AB

What is pricing of residential VoIP?

Monthly fees Initial fees

Ordinary fees All of the above

Answers to MCQs

Q1: D Q2: A, C

Q3: B

Q4: D

Questions

What is OSS, ADSL and LCS? What are people’s requirements and attitudes?

What is Skype and ISP goals?

What is people’s cost of internet telephonies? What are broadband competitors?

CHAPTER 12 Technology, Economics, In Practice, to be Concerned with IP Telephony

Introduction

The best practices are open standard’s deployment, WAN links, provisioning of bandwidth, prioritizing, and voice trafficking over data. Facility upgrades, controllers of session border, port-cards, media gateways, test equipment, soft switches and systems of back office are the economics of scale. When it comes to economics, operations account for 6 percent, transport for 50 percent, and service of customer for 8 percent, when it comes to distribution of cost for VoIP telephony. If outsourced the core offering remains with whole sale providers. Quality of call is represented by R factor and MOS. Three multiple R factors are: R factor CQ, R factor LQ and G.107.

Structure

In this chapter, we will cover the following topics: Present Situation of SYSteam Nat

IP Telephony Solutions in Technology

Solutions of Internet Telephony in Economics IP Telephony Solutions In-Practice

Objective

In this chapter, we will discuss the best practices are open standard’s deployment, WAN links, provisioning of bandwidth, prioritizing and voice trafficking over data. As we know the facility upgrades, controllers of session border, port-cards, media gateways, test equipment, soft switches, and systems of back office are the economics of scale. This chapter deals with Technology model and Economics model of IP Telephony solutions.

Situation of SYSteam Nat presently

SYSteam Nat AB is ISP local broadband situated in Uppsala of Sweden. This offers broad-band to people living in residences as internet accessibility with nearly 10 Mb/s full duplexes over Ethernet with IP address that is dynamic and public. Security of internet is provided by means of VLAN that is private. Security that is external is given through service of central firewall and by means of authentications. Core network of its own kind is network of Gigabit Ethernet that connects today through darkened links of fiber, around active users numbered 1100 in housing of students (hostels) and rental societies. There is Transit Access (TA) via SYSteam office to Skanova and Uppsala stadsnät (Uppsala Metropolitan Area Network) with 200 Mb/s capacities. The office of SYSteam Nat is linked to network of Uppsala University namely Uppsala University Computer Network for Students (UpUnet-S), with 8000 pupils to be connected. Students through UpUnet-S university network have accessibility to Swedish University Computer Network (SUNET) and varied internet parts. SUNET is WAN. UpUnet-S is MAN. The ISP is university delivering link of WAN and machines of net log-on. Various owners of real estate and foundation owning buildings of student housing own the active and passive network infrastructure parts. SYSteam Nat has the responsibility to operate supplying and network supports. Uppsala University requires upgrading of WAN link. Building owners need active equipment’s up gradation and delivering appropriate QoS. General conditions for SYSteam Nat broadband

residential people contain some Service Level Agreements (SLAs) that are rather generic.

SYSteam Nat AB has no varied activity of internet telephony than some test implementation with softphones.

Solutions of IP telephony in realm of technology

IP Telephony is very important in technology realms.

General

Following are best practices that have been recommended while designing internet telephonic network solutions:

Deployment of open standard based solutions for reducing likelihood of massive re-designing down roads.

Provision of bandwidth that suffices on links of WAN. Prioritizing voice trafficking over data.

Considering SLA adoption with ISP for specifying throughput, average round trip delays, and availability of network.

Steps included to design network solutions of telephony of IP are as follows:

Perform current network assessment.

Determine additional performance, and bandwidth needed for new services of voice. Designing network topologies.

Doing analysis of capacity.

Signal network designs.

End-to-end QoS is implemented.

Network as required is upgraded.

Complete broadband end to end solution of telephony of internet consist of Business Support System (BSS), CPE, OSS, soft-switches and gateways. It is telephone or different service giver machinery that is has been situated on premises of customers (physical locations) instead of premises of provider or among it. This is owned by giver or by person.

It is used for interconnecting dissimilar applications or networks. Gateway of IP-PSTN does perform translation between two network types. Software switch (soft-switch) is general term for whichever open software for API utilized for bridging VoIP and PSTN by differentiating functions of call controls of phone calls from media gateways (transport layers). This is program in sets which assists communication service providers for monitoring, managing, analyzing and controlling

problems with networking of computer or telephones. Systems that are sophisticated are required for these activities like keeping track and ordering components of network, usage tracking, reporting and billing. It is required for billing. This includes customer relationship management, order entries, trouble ticketing, customer care, and consumer self-services. The following figure 12.1 explains the Business Support Systems

Figure 12.1: BSS

Build-it-Yourself

ISPs pursuing strategy of that is, ISP operating on its own and choosing to build infrastructure of VoIP, should establish them like deploying media gateways, soft-switches, servers, and teleoperators. Obtain space of collocation and establish agreements of interconnection with numerous service givers. Following is requirement list for ISP operating itself and opting to build VoIP infrastructure. The following figure 12.2 supports the technology model of VoIP:

Figure 12.2: Technology Model

Interconnection of PSTN

Management of soft-switch

Planning of capacity and analysis of traffic

Termination and transiting voice

Portability of local numbers

Managing bandwidths

Support of emergency numbers Variable-distance or rate sensitive billing

CPE and installation assistance

Operator directory/services aids

Integration and testing equipment costs Compliance of regulation

Outsourcing

ISP can work optionally with third party vendors for all or few of services of VoIP. There are lots of industries in markets that offer varieties of solutions enabling ISP to outsource part or all of VoIP services itself. With changes in needed factors of VoIP of ISP, for example, with increasing focus or increasing penetration to give commercial services, vendors of third party provide migration paths for adjusting mixes of specific capabilities of service which is often outsourced. Partnership with the organizations means technological risk avoidance such as IP networking technologies continuing for undergoing continuous and rapid modifications.

With SYSteam Nat, people participate in meeting possibly with the collaboration partners of Sweden. Because of confidentiality reasons, refer such company as Company cooperates with Citylink AB. From this all minutes of call is bought. Citylink AB claims to lease most modern multiservice fiber optic commercial network for data communication and telephony that is fixed and that is of Sweden. Network has 8 points of connection from North to South of Sweden, where traffics are locally switched. Citylink AB takes good care of all communication of SS7 to rest of world. The connections physically are Integrated Services Digital Network (ISDN) Primary Rate Interface (PRI multi). All links handle thirty calls that are simultaneous. Domestic trafficking is switched to different ISP indirectly or directly via transiting with direct accessibility or through public nodes in SOLIX, Netnod-IX or (D-

GIX) and NorrNod. International trafficking goes by means of powerful transits by using leading ISP. Domestic important networks have 99.99 percent availability at least with accessibility capacity up till 10 Gbit/s. MPLS is used for purposes of routing. Various levels of service are present: 99.9 percent, 99.7 percent, and 99.8 percent, availability is guaranteed with starting of troubleshooting within 1 hour and service or support on consumer’s site within four hours. Varied levels of QoS are offered: Assured forward, expedited forwarding, and the best efforts giving lots of priority levels to traffic is present. Citylink AB routes traffic from/to Industry X. This Industry X utilizes own structure of VLAN for IP residential telephony that is sold as telephony being fixed. People use own ordinary connection of analog telephone. Digital or analog conversion in some cases is done via DRG-22 Residential in the residential building basement. The following figure 12.3 shows ISDN Architecture:

Figure 12.3: ISDN

The following figure 12.4 describes the physical connections:

Figure 12.4: ISDN PRI multi

The following figure 12.5 describes the public nodes connections:

Figure 12.5: Netnod-IX or D-GIX

With such solution SYSteam Nat must notify PTS that they become teleoperator. One benefit of conducting this is that it does enable SYSteam Nat for offering same services as whichever Public Telephone Operator (PTO), and use own number of prefixes. PTS separate applications is required for reservation and allocation of numbering capacities. In such contexts there are opportunities for PTO to give elements of service for multiple ISPs. There is provisioning of higher capacity transportation service for backbone that is nationalized. Instances of these services are shown in the following figure

Figure 12.6: PTO

Maintenance of network Service of integration (clearing houses and interconnection points) Collection and billing services Gateways between PSTN for IP telephony and internet

Shared presence of facility points

Resale

Another service provider is business model of resale. It does pay commissions back to ISP and owns individuals. Here third party handles all IP telephonic related technical matters.

With SYSteam Nat, a person participates in meeting with teleoperator, who are interested in agreement of resale. Company Y offers the residential dwellers one fixed telephonic subscription through broadband IP networks not utilizing internet. Meaning is that, customers need to have connection kit and ordinary analog telephone, but necessarily not subscription of internet. To offer low expenses Company Y has no offering of connection to operators who are pre-selected. Technical responsibility of SYSteam Nat is own network that includes giving priority to voice traffic (QoS), between residential consumers and connection points with network of Company Y in Uppsala. Residential gateway (ATA) for people either is supplied by SYSteam Nat or Company Y. All calls in Uppsala terminates by Company Y.

Recommending provider of IP telephonies

In such a case ISP helps its individuals simply to make a choice of internet telephonic giver by suggesting those providers who are suitable. Giver handles all matters that are technical which has relation with IP telephony, then.

Solutions of internet telephony related with economics

Economics models are as follows:

Build-it-Yourself

As per reports in 2005 January from stratecast partners, model of build-it-yourself forces ISP of broadband to invest upfront capital significantly in form of facility upgrades, soft-switches, test equipment, trunks, controllers of session border, port cards, media gateways and systems of back office needed for managing and paying provision to services. The backhaul costs and collocations are incurred additionally. There is a requirement that ISP should expand or build human and organizational resources. Such costs are incurred with significance to advance in generation of revenue and not involving charges of deploying and developing new service capabilities. Costs are high in new/small markets, wherein scale economies are not realized.

SYSteam Nat’s Financial Business case:

With input from SYSteam Nat meeting, the e-mail correspondences with Sweden’s Cisco and literature searches have made indicative rough financial analysis for IP telephony’s build-ityourself solution for SYSteam Nat. Three alternatives with 3000, 100 and 1000 subscribers have been analyzed by using VoIP model of Leida. This model has mixed bottom up and top down approach. Model uses present fee of ISP (top-down way) as IP networking charges and this quantifies internet telephonies impact on such cost (bottom up methodology). Model uses five subscriber types that are as follows: T1 leased-line, residential dialin, 56 kb leased-line, 128 kb dial-in ISDN and business dial-in subscribers. Overall fee distribution of ISP substantially varies as per subscriber mix. People (for their analysis) use subscriber case of T1 leased line although SYSteam Natoffers Ethernet of 10 Mb/s to its dark fiber consumers.

Model considers five categories of costs of ISP with cost elements that are as follows:

Machinery of capital: BSS, gateways, OSS and soft switches. Transport: Charges for ISP to NAP and leased lines.

Customer services: Facilities and staff to assist individuals, for example, technical supports to subscribers.

Operations: Personnel of operations, maintenance of facility and billing.

Different expenses: Administrative/general and marketing/sales expenses.

Distribution of costs for T1 subscribers as per Leida is as follows:

Equipment of capital: 8 percent

Transport: 50 percent Operations: 6 percent

Service of consumers: 8 percent Varied other fees: 28 percent

Outsourcing

Outsourcing does allow ISP to derive benefits of existing facilities of partner and agreement of (co-carrier) interconnection with various networking operators. It permits also successive capacity scaling with growth of service penetrations. It helps in limiting operational costs, and reducing initial requirements of capital charges. Partnering also gives multiple advantages, like ability for accelerating market entries, reducing human resource demands, avoiding up gradation fees over span of time and to achieve fast returns on investments in its investment of network. The figure 12.7 shows the economics model of VoIP:

Figure 12.7: Economics Model

Core offering of the wholesale providers is assisting for termination and transport of minutes of wholesale. Generally wholesale givers collect cost for successful minute’s termination and transportation. Broadband ISP, like SYSteam Nat AB must choose carriers that provide best QoS, low costs and high completion rates. ISP must look for those carriers that terminate in wholesale manners and that possess broadened network

coverage. ISP wishes to have only one contact point for not having the creation of agreements with lots of terminating carriers. ISP and carriers of wholesale terminations enter into agreed price/minute contracts for some specific period of time.

Resale

In Company Y resale arrangement SYSteam Nat gets fixed percentage of invoiced values to end users. SYSteam Nat has responsibility to market telephonic services to existing broad-band consumers. Arrangements of co-marketing have possibility to new properties wherein dwellers dwell.

The customers pay 225 SEK connection fee. Calls to various subscribers within network of Company Y are cost free. Four varied packages of subscription offered with fees that is monthly are between 0 SEK to 50 SEK. Price for calling minutes inside Sweden is between 0 SEK to 0.16 SEK. It depends at which time (18-08 or 08-18) and on what day (weekends or Monday to Friday) calls shall be placed. There are connection fees of 0.40 SEK for all calls excepting calls within own network of Company Y. Calls that are international are priced as per those price lists that are separate. Calls to mobile or fixed phones in US charge 0.31 SEK/minute plus connection cost of 0.40 SEK.

Suggesting/recommending giver of telephony of IP

This activity leads to fee of finder that is paid by provider to ISP. During work, a person may not have received precise figures, but there can be estimation of an amount that is nearly 500 SEK/subscriber referred. It is equivalent to two months ISP revenue approximately.

In practice when concerned with internet telephony

Internet telephony models are as follows:

Tests done with SYSteam Nat AB

Calls of test were performed with soft phones in dark fiber network of SYSteam Nat to get hands-on practical experience of evaluating and measuring call quality and to visualize if SYSteam Nat can give IP telephonies in its accessibility networks. No tests are conducted in UpUnet-S because of projects of maintenance and up gradation in progress.

Software of test

Software that follows is downloaded from internet, and is used to do test calls:

Capture Agent of Acterna PVA-1000: It is used for troubleshooting and for having advanced analysis of VoIP. Automation of packet network capture on desktop PC of end users is done by it. Capture files is then forwarded automatically to servers for analysis and evaluation by the support personnel. Software Analysis of VoIP by Acterna PVA-1000: It captures file’s complete analysis that is made with PVA-1000 Capture Agent. Analysis is designed for identifying the problems and emulating user experiences with troublesome calls of VoIP.

Xten’s X-Lite SIP Softphone: This is Xten Network’s free version with X-PRO soft-phone that is available for purposes of evaluation.

Call

Software is installed on desktop PC and on laptop. Both of these are connected to dark fiber network of SYSteam Nat. Execute test calls between 2 X-Lite SIP Softphones by means of IP direct dialing by using G.711 mu law as CODEC. Laptop is connected at the tenant owner building societies to networks and desktop is connected at office of SYSteam Nat. Radio programs were transmitted like voice inputs. Capture agents then capture analysis software and calls that is searched for all potential RTP conversations in capture files. Present analysis report of VoIP of call qualities in detailed way is given.

Report of analysis

Analysis report of VoIP presents call statistics, and call quality.

Quality of Call

Call quality has been represented with R factor and MOS. A factor of degradation of calls that reduces R factors has also been presented. Three varied values of MOS are shown:

Conversational Quality MOS (CQ): It is score of MOS that shows quality of conversation with regards to effects of recency and delay. Meaning of recency is that in case impairment comes nearer to call end then this shall lower quality of call to greater extents than those impairments occurring earlier in call.

Listening Quality MOS (LQ): It is MOS score which represents quality of listening with no consideration of recency or delay effects. Value in question is same as implementations of ITU P.862 Listening Quality. The following figure 12.8 depicts the MOS Listening Quality Objective, the MOS Listening Quality Subjective and the MOS Conversational Quality Estimated:

Figure 12.8: MOS-LQO, MOS-LQS, MOS-CQE ITU P.862 PESQ or PQ: It is score of MOS that is normalized to raw P.862 quality scores. PQ is measurement of objective predicting result of subjective tests of listening. Resulting score of quality possesses analogy with MOS that is subjective.

Quality of voice is shown with values of MOS between 1 and 5. In this 1 is bad, 4 is good, 2 is poor, 5 is excellent and 3 is fair. Three multiple R-factors are presented:

Conversation Quality R-factor (R-Factor CQ): It is metric of voice quality, measuring quality of voice based on recency of burst loss, delay of echo transmission, and loss of burst packet.

Listening Quality R-factor (R-Factor LQ): It is measurement of quality of voice that does not consider recency and delay effects.

G.107: It is that R factor which is calculated as per specification of ITU R factor numbered 93 is accepted for level of high quality for systems of transmission of toll quality voice. Factors of call degradations that are reported are packet loss, recency, noise level of silence period, levels of echo, one-way delay, signal level of voiced segment, packet discards of jitter buffers, and selection of voice codecs. Factors are delivered in points reducing R-factors to 93. All points of degradation are summed up equally to difference between 93 and R Factor (CQ) that is actual.

Statistics of call: List following statistics for all channels of currently selected calls: Max Jitter represents high absolute jitter values, in milliseconds.

Avg. Jitter presents average jitter values, in milliseconds.

Max Gap is largest seen gap of inter frame. Max Pkt Loss shows highest observed values for packet losses that are consecutive.

Avg. Pkt Loss represents average values for packet loss that is consecutive.

Total Pkt Loss presents total packet loss values.

Automated approach of test

Methodologies for verifying implementation of internet telephony and ensuring its on-going operation are for manually testing all phones. Such ways for testing quality of voice and functionality of IP telephony’s is costly, error prone and consumes lot of time. This method may not scale for supporting larger deployments of numerous phones across lots of varied physical locations.

Network Management Software (NMS) is there for monitoring metrics like jitter and packet loss. of conventional NMS like utilization of Central Processing Unit (CPU), packet loss or network latency has no easy translations for providing experience to end users. In IP telephonies, testing is matter to ping signal to all phones for making sure that there is a network connection. The users count on accessing broadened phone function arrays and with ability for clearly communicating and hearing with whom so ever one talks with. Attempts to maintain and implement internet telephony by using NMS singularly is supplemented with manual efforts The following figure 12.9 explains the network management software intricacies:

Figure 12.9: NMS

In current times, sophisticated tools exist like Clarus Systems giving solutions of automated testing for certifying application and to assure IP telephony on-going operations. There is a usage of software for automating manual process.

Conclusion

In this chapter, we deal with the deployment of open standards in relation to VoIP. BSS is needed for purpose of billing. It involves order entries, customer self-services, trouble ticketing, consumer relationship management and purchaser care. Citylink AB takes warming care of SS7 communication to remaining part of world. Physical connections are ISDN PRI multi. Domestic trafficking has been switched to varied ISP directly or indirectly by means of direct access transition or via public nodes in the NorrNod, SOLIX, Netnod-IX or D-GIX. Software is downloaded from the internet, and it is utilized for conducting test calls. It is X-Lite SIP Softphone of Xten’s, Acterna PVA-1000’s Capture Agent and Acterna PVA-1000’s Software Analysis of VoIP. NMS is present to monitor metrics such as packet and jitter loss.

The next chapter focusses on all the models related to VoIP.

Points to remember

MPLS is utilized for routing objectives. With outsourcing of SYSteam Nat, PTS is notified to be teleoperator and one benefit of doing this has been that this enables SYSteam Nat to offer similar services like any PTO and using own prefix numbers. Call qualities are MOS-CQE, MOS-LQO, and MOS-LQS.

R-Factor CQ and R-Factor LQ are 2 renowned R-Factors.

Call statistics are Total Pkt Loss, Max Jitter, Max Gap, Avg. Pkt Loss, Avg. Jitter and Max Pkt Loss.

Multiple choice questions

What is technological model of solution of IP telephony? Non-general

General

Multi-level Non-Multi-level

What is SUNET?

Swedish university computer network

Sweden university computer network

Scotland university computer network

All of the above What is economics model of internet telephony’s solutions?

Do not build it yourself

Build-it-yourself

Build-it-with-others

Do not build it with others

Answers to MCQs

Q1: C Q2: A

Q3: B

Questions

What is BSS and ISDN PRI multi? What is SOLIX, D-GIX or Netnod-IX and NorrNod?

What is test software and why NMS exists?

CHAPTER 13 VoIP to be Concluded

Introduction

Competition and demand determine market. There are numerous factors for VoIP to gain widespread acceptance. QoS is VoIP service’s pre-requisite for business provisions of mass market. Network should have ability to prioritize traffics to minimize network delays and different service levels must be present toward varied usage classes. An average user does not pay heed to network technologies until expectations have been met, and their behavior modifications are not dramatic. Ultimate user equipment become user friendly to nice extents. PSTN approachability must go hand-in-hand with reliability. Vendor interoperability is accomplished by open architecture and industrial standards. Accessibility requires extensions beyond fax machine, telephone to PC and numerous other compatible devices.

Structure

In this chapter, we will cover the following topics: General conclusion

Conclusion of VoIP in regards with market and business model

Technological conclusion for VoIP Ending VoIP with economics

Objective

We will be concluding the book with competition and demand determine market. There are numerous factors for VoIP to gain widespread acceptance. QoS is VoIP service’s pre-requisite for business provisions of mass market. In this chapter, we conclude that VoIP is a gamechanger for WFH practitioners with regards to improving the overall look and feel of communication among peers. This chapter comprises with Economics model, Business model, Technology model, and Market model.

Conclusion in general

Although demand and competition shall determine market, lot of factors are present for VoIP to gain acceptance in widespread manner.

QoS is pre-requisite for VoIP services of mass market, particularly services of business. The criticality is that networks must have ability for prioritizing traffics for minimizing network delays and other levels of service should be available to varied classes of usage.

Market is utilized to PSTN and has reasonable satisfaction with this. Users need seamless IP networks and PSTN’s integration. They require always reliable and available services, with enhanced and high-quality features of calling that is expected out of these. Average users do not pay heed regarding involved network technologies till expectations have been met and their behavior modifications are not dramatic.

Equipment of ultimate users must become user friendly to greater extents. Reliability should have PSTN approachability. Interoperability among vendors must be accomplished by way of industrial standards and open architectures. Accessibility need extension beyond telephone to PC, different compatible devices and fax machine.

Once price benefits are dissolved, markets shall become service driven with the enterprises pushing such services. Tariffs that are lower than ones of PSTN because of toll bypass argue with costs needed for building internet networks with features and reliability of PSTN only yielding marginal advantages of charges that is relative to PSTN. Givers of services, by virtue of applications and business-oriented services achieve high margins. Those who are building dedicated IP networks are in prime position for grabbing shares of business market. They also offer QoS guarantees over privately managed networks of theirs. In internet organizations, early movers win vital marketing spaces usually. For an example, Netscape (1994 start-up) immediately became biggest provider of software of web browsing. It could do this since there was opportunity before varied houses of software and Microsoft. Vendors of machinery require enabling givers of services for offering features of business telephonies which transfer PSTN features to world of IP. It benefits themselves and latter.

Long term view (5 to 10 years in future) sees issues of quality as matter of past. Till that span of time there shall be no requirement for making economic VoIP cases. Tariffs of PSTN drop because of competition’s responses. Users have option for VoIP. Reason is that, this has more efficiency, has functionality that is improved and is better solution altogether for serving the technology savvy needs of markets.

VoIP’s conclusion with respect to business and market model

Subscriptions of broadband are commoditized. ISPs like the SYSteam Nat require raising end user values of broadband subscription of theirs. One methodology of conducting this initially is introducing internet telephony that is followed by addition of next generation services like Video Calling, Click-to-dial and Caller ID. SYSteam Nat’s broadband competitors either are in or have to process development of services of IP telephony and triple plays even. Four models to offer services of VoIP to ultimate consumers are there. This falls in order of undertakings, revenue and complexity: Build-It-Yourself, Outsource, Resale, and giving Recommendations to suitable providers of VoIP.

One serious undertaking is It is technically challenging without skilled staffs especially. is easy method. But it is dangerous from long term views because people can skip ISP and directly go to VoIP givers. ISP owns individual in arrangement of thereby avoiding confusion of customers over support and branding.

Pricing of residential VoIP includes service activation or initial setting up fees normally. To this add yearly or monthly service fees and trafficking charges for telephonic calls (for PSTN terminated calls particularly). There are price lists for features that are value added. Calls to different subscribers of ISP respectively and to various users of SIP are free of cost.

Maximum potential of marketing for SYSteam Nat presently with Uppsala chosen strategies for targeting residential masses in societies owned by tenants and candidate’s living in hostels is 40300 subscribers approximately generating revenues around MSEK in one year. Market accessible potential today is nearly 11000 subscribers (8000 pupils via UpUnet-S and 3000 dark fiber consumers) with no regards for limitations that are technical. This shows revenues of nearly 35 MSEK in time duration of one year.

Conclusion of VoIP with regards to technology

SYSteam Nat is very small participator in the rather embryonic markets for pursuing strategy of build-it-yourself. Some technical barriers to such solutions are:

Risk of buying billing system and switch which does not deliver or work required services and functionality. Lack of experience and knowledge.

Technical employees who are experienced are needed for maintaining applications and equipment.

Substantial requirement list for method of build-it-yourself underscores that SYSteam Nat must not try for reinventing wheels in terminology of both development of application and infrastructure and network operations. ISP is exposed and better served to lesser risks by devotion of own resources to realms where one has clear core competencies to leverages. In condition of SYSteam Nat, till date at least, it does not involve IP telephonies. On topmost position of all these, there are financial risks for overcoming strategies of build-it-yourself.

SYSteam Nat offers internet broadband accessibility to residential dwellers with nearly 10 Mb/s full duplexes over Ethernet and with public dynamic IP address. Its own core and key network are that networking of Gigabit Ethernet connects people by means of links of dark fiber. They have full QoS control on such network and in turn offering that QoS which is guaranteed. Nearly 8000 pupils have approachability through UpUnet-S, wherein Uppsala University is ISP, while SYSteam Nat has responsibility to operate network and to supply supports. The test results indicate that in case SYSteam Nat desires for introducing internet telephony to such candidates then Uppsala University has to upgrade equipment. It is integral factor that a person must follow best and excellent practices while designing network of internet telephony. Else severe troubles of QoS might occur.

End to end complete IP telephonic solution consists of soft switches, gateways, BSS, OSS and CPE. SIP is used like protocol. ISP choosing to operate and build own infrastructure of VoIP should deploy above machinery establishing themselves like teleoperators with quite some demanding obligations as stipulated by Swedish Electronic Communications ISP can work optionally with third party vendors for all or some of services of VoIP, that is, follow outsourcing methods. With continuous and rapid changes in technologies of VoIP, risk of technology can be avoided. With resale of business models, other tele-operators or service providers owns people paying commissions back to ISP. Third party handles all matters that are technical in relation with internet telephonies. Option is recommending IP telephonic giver managing all scenarios that includes technology.

Ending VoIP when concerned with economics

Conclusion is assisted by Skype success and new consumer offer introduction that threatens establishment of tele industries and their methodologies. CEO of Skype stated in interview in 2005 April: are nearly 34 million users who are active. Of these 1 to 2 million are the users who pay. Since there is no need of any investment in marketing or networks, there is no requirement of earning money on all individuals. This is part of business model resembling Google and Yahoo. Within 10 years there shall be no call costs or subscription charges. Tele organizations having networks shall survive probably because they earn money on internet and broadband availability instead. But those selling minutes of calls require reassessment of their idea of

Stratecast partners of cash flow and profit not only needs identification and pursuit of newer opportunity of revenues but also abilities of optimizing operations of network, controlling expenditure of capital and reducing recurring operational Such SYSteam Nat results with least complexity are got by third party partnership.

Models of build-it-yourself forces ISP broadband for investing in significant upfront capitals in form of systems of back office, equipment and facility upgrades. Costs (that is, expansion of organizational and human resources, collocation and backhaul fees) shall be incurred in advancement of generation of revenue

significantly. Expenses are higher in new and smaller markets because of scaling reason economies. Rough financial business case for the SYSteam Nat approximately indicates that 1500 subscribers of internet telephony are needed for the positive contribution done financially. Number shows that 50 percent of today’s darkened fiber accessibility people shall most likely take little time to come to existence.

Outsourcing assists in limiting functioning costs and reducing initial capital fees. This allows successive capacity’s scaling up. Partnering provides ability for accelerating market entries and putting avoidance on upgrading expenses. In reselling agreements SYSteam Nat receives fixed percentages of invoiced values to end consumers, that is 10 percent of revenues. IP telephonic gives recommendation leading to fee paid by finder. Estimation is that amount is nearly 500 SEK for one subscriber who is referred. Third party partnership shall considerably lessen operating and financial risks.

Concluding VoIP in relation with Covid-19

To conclude, we expect that corona vaccine and drug will be out in the market soon, and we will get a respite from the dreaded pandemic.

Conclusion

The concluding chapter of this book deals with the ramifications that make VoIP user friendly and robust for the masses in general. Competition and demand determine market, but there are multiple factors availing for VoIP for gaining acceptance in the widespread manners. Barriers include: Lack of knowhow and experience, risk to buy switch and billing system not working or not delivering functionality and services and experienced technical employees are required to maintain equipment and applications. VoIP Servers are capable of providing a more complete set of communication services for audio and video conferencing all in the real time scenario. Often effective video conferencing tools depends on experience and expertise and we can establish communication by the various tools explained above.

Points to remember

Broadband subscriptions are commoditized and ISPs such as SYSteam Nat needs to raise values of end user to have their subscription of broadband. This is the marketing model’s VoIP conclusion.

Economic model conclusion has been assisted by the Skype success with new buyers offering introduction threatening tele industries establishment with its methodologies.

Multiple choice questions

What is conclusion of VoIP in relation with Covid-19 virus? These are trying times

There are changing business models

All of the above None of the above

What is VoIP with respect to business models?

No call costs

No subscription charges

No investment in marketing D. None of the above

Answers to MCQs

Q1: C Q2: A, B

Questions

What is general conclusion? What are technological barriers (conclusion) on VoIP?

Key terms

Application Programming Interface Common Gateway Interface

enCOde/DECode

Digital Subscriber Line Digital Signal Processing

High-Definition Television

Hyper Text Transfer Protocol

Integrated Access Device

Internet Engineering Task Force

Internet Protocol IP Security Protocol

International Telecommunications Union

Least Cost Routing

Multimedia Terminal Adapter

Personal Digital Assistant

Point-to-Point Protocol

Perceptual Speech Quality Measurement

Quality of Service

Remote Access Server Secure Multipurpose Internet Mail Extensions

Session Initiation Protocol Small Office/Home Office

Transmission Control Protocol Transport Layer Security User Agent Client

User Agent Service

Voice over Internet Protocol

Virtual Private Network

eXtensible Markup Language

3G (third generation) Partnership Project Authentication, Authorization, and Accounting Asymmetric Digital Subscriber Line

Authentication Header Application Level Gateway

Address of Record Advanced Research Projects Agency

Analog Telephone Adapter Business Support System Class of Service Customer Premises Equipment

Central Processing Unit Differentiated Services Domain Name System Electronic Number Mapping

Enhanced Service Provider European Telecommunications Standards Institute Fully Qualified Domain Name Foreign Exchange Protocol defined in ITU-recommendation H.225

Protocol defined in ITU-recommendation H.323 Internet Gateway Device Interior Gateway Protocol Internet Key Exchange

Internet Low Bitrate Codec

Integrated Services Integrated Services Digital Network Internet Service Provider

Local Area Network

Local Exchange Metropolitan Area Network Media Gateway Control Protocol (H.248, Megaco) Multimedia Internet KEYing protocol Multipurpose Internet Mail Extensions

Multiparty Multimedia Session Control Mean Opinion Score The Moving Picture Experts Group Multi-Protocol Label Switching

Network Access Point Network Address Translation Next Generation Networks Network Management Software

Open Systems Interconnection Operations Support System Peer-to-Peer Personal Digital Assistant

Primary Rate Interface (ISDN) Public Switched Telephone Network

Post-ochTelestyrelsen (The Swedish National Post and Telecom Agency)

Request for Comments Resource Reservation Protocol

Real Time Control Protocol

Real-time Transport Protocol Real Time Streaming Protocol

Round Trip Time Session Description Protocol

SDP Next Generation

Service Level Agreement

Simple Mail Transfer Protocol Swedish Number Portability Administrative Centre

Secure Real-time Transport Protocol

Service Location Signaling System 7

Simple Traversal of UDP through NATs Transit Access

Type of Service

User Datagram Protocol Universal Plug and Play

Uppsala University Computer Network for Students Uniform Resource Identifier

Uniform Resource Locator

Virtual Local Area Network Wide Area Network

Wireless Local Area Network Federal Communications Commission

Internet Telephony Service Providers

Fax over Internet Protocol Private Branch eXchange

Call Detail Record

Operations, Administration, Maintenance and Provisioning Plain Old Telephone System

Dual-Tone Multiple Frequency Voice Activity Detection

Comfort Noise Generation

Server Message Block

Reduced Instruction Set Computer Real Time Operating Systems

Subscriber Line Interface Circuit Data Access Arrangement

Trivial File Transfer Protocol

Point to Point Protocol over Ethernet Dynamic Host Configuration Protocol

Foreign Exchange Station

Foreign Exchange Office Battery, Over-voltage, Ringing, Signaling, Coding, Hybrid, Testing

Voice Frequency Denial of Service

Radio Frequency

Perceptual Analysis and Measurement System Perceptual Evaluation of Speech Quality

RTCP Extended Report Extension to RTCP

Asynchronous Transfer Mode Synchronous Digital Hierarchy

Synchronous Optical Network Extended Digital Subscriber Line

Short Message Service

Data Encryption Standard Advanced Encryption Standard

Rivest Cipher International Security Association

Secure Hyper Text Transfer Protocol

Session Security Layer

bump-in-the-stack bump-in-the-wire

Corporate Local Area Network

Communications Assistance for Law Enforcement Act Port Address Translation

Network Address Port Translation Bill of Material

File Transfer Protocol

Fiber-To-The-Home

Microprocessor without Interlocked Pipeline Stages Printed Circuit Board Synchronous Dynamic Random-Access Memory

Network Basic Input/Output System Internet Control Message Protocol Address Resolution Protocol Internet Group Management Protocol

Reverse Address Resolution Protocol Enhanced 911 Public Safety Access Point

Local Service Provider Session Initiated Protocol-Common Gateway Interface

Call Processing Language

Graphical User Interface Java API for Integrated Networks Java Virtual Machine

text-to-speech

Automatic Speech Recognition Media Gateway Media Gateway Controller

Bearer Independent Call Control Skinny Client Control Protocol Telephony Routing over IP Border Gateway Protocol

Live Communications Server

Swedish University Computer Network Public Telephone Operator

Bibliography

Primary reading link: https://www.mygov.in/covid-19/

Resources for further reading:

European Commission Directive 90/388/EEC on Competition in the Markets for Telecommunications Services Internet Telephony - The Regulatory Issues http://204.191.126.89/home/mccarthy/web/docs/mt-inter.html

Nokia/Vienna Systems Homepage http://www.viennasys.com/

‘Siemens Introduces IP Telephone’, Zona Research, 15 April 1999

‘IP Telephony Markets in Europe and the US’, Woll, B., Datamonitor, 1998

The Swedish Electronic Communications Act (2003:389) accessed 2005-04-25 O. Bergquist and M. Sjöstedt, (June 2003) IP telephony: A Swedish Perspective M.Sc. thesis, Department of Microelectronics and Information Technology, KTH

J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, and E. Schooler, (June 2002) RFC3261 – SIP: Session Initiation Protocol accessed 2004-12-20

Jonathan Davidson, James Peters, and Brian Gracely, (2000) Voice over IP Fundamentals. Cisco Press; 1st edition, ISBN: 1-57870-168-6

Henry Sinnreich and Alan B. Johnston, (2001) Internet Communications Using SIP: Delivering VoIP and Multimedia Services with Session Initiation Protocol. John Wiley & Sons, Inc, ISBN 0-471-41399-2 G. Q. Maguire Jr. (2004) 2G1325 / 2G5564 Practical Voice over IP (VoIP): SIP and related protocols, lecture slides, KTH accessed 2004-12-06

ENUM – Slutrapport – PTS-ER-2004:39 accessed 2005-01-03 Kevin Brown, (2004) IP telephony Unveiled. Cisco Press, ISBN 158720-075-9 S. Andersen, A. Duric, H. Astrom, R. Hagen, W. Kleijn, and J. Linden, (December 2004) RFC3951 – Internet Low Bit Rate Codec (iLBC). ftp://ftp.rfc-editor.org/in-notes/rfc3951.txt, accessed 2005-02-01 The Essential Report on IP telephony, (2003) ITU-D accessed 200502-07

Ken Camp, (2003) IP telephony Demystified. McGraw-Hill Professional, ISBN 0-07-140670-0 [12] Nathan J. Muller, (2003) IP A to Z. McGraw-Hill, ISBN 0-07-141086-4

IP-telefoni – En tekniskmarknadsbeskrivning – PTS-ER-2003:41 accessed 2004-12-20 Evalueserve, Impact of Skype on Telecom Service Providers, (January 2005) accessed 2005-02-13 Skype’s homepage accessed 2005-05-25

Kennet Rådne, (2005-02-03) VOIP – a step towards the integrated world - IVA accessed 2005-02-13

Session Initiation Protocol (SIP) Working Group accessed 2005-01-17 M. Handley and V. Jacobson, (April 1998) RFC2327 – SDP: Session Description Protocol ftp://ftp.rfc-editor.org/in-notes/rfc2327.txt, accessed 2005-02-01

H. Schulzrinne, S. Casner, R. Frederick, and V. Jacobson, (July 2003) RFC3550 RTP: A Transport Protocol for Real-Time Applications ftp://ftp.rfc-editor.org/in-notes/rfc3550.txt, accessed 2005-0201

M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, (March 1999) RFC2543 – SIP: Session Initiation Protocol ftp://ftp.rfceditor.org/in-notes/rfc2543.txt, accessed 2005-02-01

International Engineering Consortium – H.323 Tutorial accessed 2005-01-17 [22] Overview of H.323 accessed 2005-01-17 Overview of H.323 accessed 2005-01-17 SIP versus H.323 accessed 2005-01-17 P. Srisuresh and M. Holdrege, (August 1999) RFC2663 – IP Network Address Translator (NAT) Terminology and Considerations ftp://ftp.rfc-editor.org/in-notes/rfc2663.txt, accessed 2005-01-02 Fredrik Thernelius, (May 2000) SIP, NAT, and Firewalls M.Sc. thesis, Department of Microelectronics and Information Technology, KTH

J. Rosenberg, J. Weinberger, C. Huitema, and R. Mahy, (March 2003) RFC3489 – STUN – Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs) ftp://ftp.rfc-editor.org/in-notes/rfc3489.txt, accessed 2005-02-01

Intertex’s homepage accessed 2005-02-01

Understanding the IPSec Protocol Suite (March 2000), Alcatel accessed 2005-02-05

Quick Introduction to SRTP and MIKEY accessed 2005-02-05 M. Baugher, D. McGrew, M. Naslund, E. Carrara, and K. Norrman, (March 2004) RFC3711 – The secure Real-time Transport Protocol (SRTP) ftp://ftp.rfc-editor.org/in-notes/rfc3711.txt, accessed 2005-02-01

Internet Telephony: Voice over Internet Protocol (IP) accessed 200502-06 J. Loughney and G. Camarillo, (February 2004) RFC3702 – Authentication, Authorization, and Accounting Requirements for the Session Initiation Protocol (SIP) ftp://ftp.rfc-editor.org/innotes/rfc3702.txt, accessed 2005-02-01 P. Faltstrom and M. Mealling, (April 2004) RFC3761 – The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM) ftp://ftp.rfceditor.org/in-notes/rfc3761.txt, accessed 2005-02-01 Henning Schulzrinne, The Session Initiation Protocol (SIP) accessed 2005-01-17

Thomas Pang (April 2003) SCTP support for SIP proxy server SIP session setup and teardown slide, M.Eng project presentation,

Communication Networks Laboratory, School of Engineering Science, Simon Fraser University accessed 2005-01-17 Blixtvik Internet ochTelefoni AB’s homepage accessed 2005-02-13

Bredbandsbolaget’s homepage accessed 2005-02-13 Com Hem AB’s homepage accessed 2005-05-06 Tele2 AB’s homepage accessed 2005-02-13

Image credits

https://www.bing.com/images/search? view=detailV2&id=4E7E08E39250CE722FFC42F252EC0A7DA457554A&t hid=OIP.G2v4zo4CthIfpmhTaBUwYgHaFj&mediaurl=https%3A%2F%2F image.slidesharecdn.com%2Farsitekturvoip-130320204127phpapp02%2F95%2Farsitektur-voip-1638.jpg%3Fcb%3D1363812129&exph=479&expw=638&q=VoIP+Ads&sel ectedindex=58&ajaxhist=0&vt=0&eim=0

https://www.bing.com/images/search? view=detailV2&id=E8C8685E30D8EFC7F202C12C19575012A7C038AE&th id=OIP._eXag65RcMcGlmCZVekm7gHaD3&mediaurl=https%3A%2F%2 Fsoconnect.co.uk%2Fwp-content%2Fuploads%2F2015%2F11%2FhowVoipworks.jpg&exph=366&expw=700&q=How+VoIP+Works&selectedindex= 1&ajaxhist=0&vt=0

https://www.bing.com/images/search? view=detailV2&ccid=ENBGSA%2B%2B&id=8BD43533005AA54473B1119 3AB465BEE86D5B09C&thid=OIP.ENBGSA-Nc6nq_L7HpXYtQAAAA&mediaurl=https%3A%2F%2Fwww.globalspec. com%2FImageRepository%2FLearnMore%2F20165%2FVoiptypical8a9304d0022641efbc5a21eb8eda97dc.gif&exph=302&expw=300& q=VoIP+Gateway+Information&simid=608013531611857634&selectedind ex=144&ajaxhist=0&vt=0

https://www.bing.com/images/search? view=detailV2&id=1F65A0835A91530DB517B660EB906921787E5CCB&th id=OIP.SXuOunZFdkOG3dRO8Ax4GwHaG3&mediaurl=http%3A%2F%2 Fwww.comparebusinessproducts.com%2Fcontent%2Fassets%2F20%2F infographics%2Fwhat-is-voipig.png&exph=1112&expw=1200&q=What+Is+VoIP&selectedindex=3&aja xhist=0&vt=0

https://www.bing.com/images/search? view=detailV2&id=686B5DCA477EA2A7CB02B91A38944FB66C0F199C &thid=OIP.yLl9HMRBfXwEB0HKlSvRIAHaFI&mediaurl=https%3A%2F %2Fwww.ndm.net%2Fuc%2Fimages%2Ftraditional-pbx-vs-cloudphone.png&exph=665&expw=960&q=VoIP+Cloud+Computing&selecte dindex=1&ajaxhist=0&vt=0 https://www.bing.com/images/search? view=detailV2&id=1715A566E613A124C73548BC00C12FF2436A3AB0&thi d=OIP.kpYSk4xrEcHn5U_6QGJzqwHaH8&mediaurl=http%3A%2F%2Fw ww.abcvoip.com%2Fimg%2Fvoip_gd.jpg&exph=583&expw=543&q=VoIP+Techn ology&selectedindex=96&ajaxhist=0&vt=0 https://www.bing.com/images/search? view=detailV2&id=180A2DFE6609412D64E4FDDEDC16494242845371& thid=OIP.Vpzc80kufLXE2mw6idZYAHaF3&mediaurl=http%3A%2F%2Fwww.advancedvoi p.com%2Fimages%2Fip.jpg&exph=643&expw=812&q=VoIP+PBX&selec tedindex=30&ajaxhist=0&vt=0

https://www.bing.com/images/search? view=detailV2&id=D7F15EB03AFAB12E40C6B9BFE00260F0D0FA94AA &thid=OIP.haGIFBnyT6LK8dJUALgXUQHaFj&mediaurl=http%3A%2F% 2Fhome.ubalt.edu%2Fabento%2F427%2FVOIPLASTMILE%2Fimg003.g if&exph=600&expw=800&q=VoIP+Over+IP&selectedindex=57&ajaxhist =0&vt=0&eim=0 https://www.bing.com/images/search? view=detailV2&id=9FC84FBAAEB3AC15126392A7A7858BEB6C0185EF&t hid=OIP.QsLEU25KgkRQJi0XdDanAQHaEe&mediaurl=https%3A%2F% 2Fen.vcenter.ir%2Fwp-content%2Fuploads%2F2017%2F07%2FVoIPvoice-overIP.jpg&exph=333&expw=550&q=VoIP+Over+IP&selectedindex=104&ajax hist=0&vt=0&eim=0 https://www.bing.com/images/search? view=detailV2&id=B84585FF062C85FB2A4CF2B6EB29AA567A8A5CC9& thid=OIP.6fZ6broEihQbklyWH4lwggHaHa&mediaurl=https%3A%2F%2 Fi.pinimg.com%2Foriginals%2F9f%2Ff5%2Fc6%2F9ff5c6033cf7e03fb3e a0974046f0f59.jpg&exph=700&expw=700&q=VoIP+Cloud+Computing &selectedindex=17&ajaxhist=0&vt=0&eim=0 https://www.bing.com/images/search? view=detailV2&id=702F2EE421C3422FE34AD340B42F57BD5205E95D&t hid=OIP.n0fiVKPKj0F4694X8JFGQHaFp&mediaurl=http%3A%2F%2Fwww.sumitdhawan.com%2Fw p-content%2Fuploads%2F2014%2F03%2Fvoip-without-serviceprovider.jpg&exph=568&expw=745&q=VoIP+Service+Providers&selecte dindex=34&ajaxhist=0&vt=0&eim=0

https://www.bing.com/images/search? view=detailV2&id=9C7C55F3D0853882DC9DAB2411A561BAC84E6E94&t hid=OIP.xt8ezp_2QQkOliDiJfdEIgHaDz&mediaurl=http%3A%2F%2Fww w.vocal.com%2Fwp-content%2Fuploads%2F2012%2F05%2Fvoip-mwsoftware.png&exph=555&expw=1082&q=VoIP+Software&selectedindex= 12&ajaxhist=0&vt=0&eim=0

https://www.bing.com/images/search? view=detailV2&id=0AA2336D71699A5D3C6931DD703C06439DD8D03F &thid=OIP.UiQMagfHFjfSvMFroLWYaQHaFP&mediaurl=https%3A%2F %2Fdclabproject.files.wordpress.com%2F2014%2F11%2F82096029.png &exph=1240&expw=1754&q=VoIP+Server&selectedindex=66&ajaxhist= 0&vt=0&eim=0 https://www.bing.com/images/search? view=detailV2&id=39BC9A606654268617D6BC3E5EE7C91DAF1A6397&t hid=OIP.ThRqVOy8J3pLpigLybqR8AHaFj&mediaurl=https%3A%2F%2Fi mage.slidesharecdn.com%2Fvoipsecurity-130101015517phpapp02%2F95%2Fvoip-security-13638.jpg%3Fcb%3D1357005454&exph=479&expw=638&q=VoIP+Security &selectedindex=11&ajaxhist=0&vt=0&eim=0

https://www.bing.com/images/search? view=detailV2&id=5C6992A87BF1E64C5C6E32892772B6FCAE01AEE3&t hid=OIP.xzKbohPx7FUMzrTaTTLnyAHaFG&mediaurl=https%3A%2F%2 Fsites.google.com%2Fsite%2Fsoftswitchngn%2FVOIPpic1.jpg&exph=38 3&expw=556&q=VoIP+Networking&selectedindex=55&ajaxhist=0&vt=0 &eim=0

https://www.bing.com/images/search? view=detailV2&id=BAAECDD2C76B74751E95AEFE8E6D66C6D9890623 &thid=OIP.gGKQJ8XxqN02SiJG4T8nagHaE5&mediaurl=http%3A%2F% 2Fwww.telnetnetworks.ca%2Fimages%2Fstories%2Fsuppliers%2Finfosi m%2FInfosim_StableNet_Managing_End_to_End.png&exph=529&exp w=800&q=VoIP+Networking&selectedindex=74&ajaxhist=0&vt=0&eim =0 https://www.bing.com/images/search? view=detailV2&id=B04E026ED2AA6AF058A359AEDCBD28E7D15A9B8E &thid=OIP.UD5yEJfWiD8cFMbMp5hvDAHaG3&mediaurl=https%3A%2 F%2Fi.pinimg.com%2Foriginals%2Fe7%2F89%2F4f%2Fe7894f4dd8315 655423c09a0999080d9.jpg&exph=1112&expw=1200&q=VoIP+Cloud+Co mputing&selectedindex=201&ajaxhist=0&vt=0&eim=0 https://www.bing.com/images/search? view=detailV2&id=6A1627219C04D7785131D4F2EA62ADB99DE64C38&t hid=OIP.unvfodYejpMDHtW54_htkwHaEV&mediaurl=https%3A%2F%2 Ffiverrres.cloudinary.com%2Fimages%2Ft_main1%2Cq_auto%2Cf_auto%2Fgi gs%2F91978101%2Foriginal%2F7da7bb17f2880a5587cbc03d3d65e7eb67 4240cd%2Ffix-or-implement-cisco-call-manager-and-voipsolutions.png&exph=398&expw=680&q=VoIP+Solutions&selectedindex =8&ajaxhist=0&vt=0&eim=0

https://www.bing.com/images/search? view=detailV2&id=A5E3F5D03088A86D44C9834271F4C27B20A65686&t hid=OIP.7IlIzOQqfvmsRpACO6r6eQHaFj&mediaurl=https%3A%2F%2F netbeez.net%2Fwp-content%2Fuploads%2F2015%2F07%2F28Distributed-network-monitoring-for-cloud-based-VoIP.002-

1024x768.jpg&exph=768&expw=1024&q=Cloud+VoIP&selectedindex=53 &ajaxhist=0&vt=0&eim=0 https://www.bing.com/images/search? view=detailV2&id=6B35C84918FD54CD8D004F5327FE9340A3069F8D& thid=OIP.ujb32Uxy15LqeaHmfcI2twHaDi&mediaurl=https%3A%2F%2Fi nsidemstech.files.wordpress.com%2F2016%2F04%2F041816_1704_sky peforbus1.png%3Fw%3D625&exph=298&expw=625&q=Microsoft+Clou d+PBX&selectedindex=5&ajaxhist=0&vt=0&eim=0

https://www.bing.com/images/search? view=detailV2&id=264A7C9FE152B9FB7A69E14CFFBD8D980436CC04 &thid=OIP.KgSeOMAfNl5twIkWN2BfPgHaFj&mediaurl=https%3A%2F %2Fimage.slidesharecdn.com%2Fhostednfv-140722082430phpapp01%2F95%2Fhosted-nfv-voip-solution-cloud-sourcing-next-genvoice-14638.jpg%3Fcb%3D1439935578&exph=479&expw=638&q=cloud+voip+ar chitecture&selectedindex=27&ajaxhist=0&vt=0&eim=0

https://www.bing.com/images/search? view=detailV2&id=D17939C5E373070371D9B6636292C244705FF420&th id=OIP.sO4LpcMB7HZhrDNV3UPLXQHaDT&mediaurl=https%3A%2F %2Fwww.alianza.com%2Fhs-fs%2Fhubfs%2FWebsite__Images%2FCVP_Architecture.png%3Ft%3D1500653697623%26width %3D1000%26name%3DCVP_Architecture.png&exph=447&expw=1000 &q=cloud+voip+architecture&selectedindex=53&ajaxhist=0&vt=0&eim= 0

Index

A access gateways 169 access networks 111 active tests about 98 Perceptual Analysis and Measurement System (PAMS) 98 Perceptual Evaluation of Speech Quality (PESQ) 98 Perceptual Speech Quality Measurement (PSQM) 98 Advanced Encryption Standard (AES) 118 AES algorithm 119 A factor 99 AMR-WB 174 analog phones with PSTN interfaces 80 Analog Telephone Adapter (ATA) 159 analysis report, of VoIP automated approach of test 225 call quality 222 call statistics 224 API / Programmatic interfaces about 160 call processing language 162 JAIN 163 SIP-CGI 161 SIP Servlets 164 application layer, TCP/IP 147

application server 171 Appriver 32

architecture options, of internet telephony end-to-end, direct IP connection 156 hybrid (hop-off/hop-on) 155 trunk replacement 155 ArogyaSetu app 34 Asterisk server 21 Asymmetric Digital Subscriber Line 207 Automatic Speech Recognition Systems 172 Ayushman Bharat 35 B

backbone network 110 Bell Labs 174 BGP route protocols 186 Bleed-Though 84 Blixtvik Internet ochTelefoni AB 200 BORSCHT bouncing 95 Bredbandsbolaget (B2Bredband AB) 200 broadband end to end solution, of IP telephony BSS 213 CPE 213 gateways 213 OSS 213 soft-switch 213 broadband telephony of Telia characteristics 202

business models about 197 build-it-yourself 196

outsourcing 198 Business Support System (BSS) 213

C Call Agents 173 call control separation, of media about 152 mobility and virtual locations 153 Call Detail Record (CDR) architecture 49 call originator 181 Call Processing Language (CPL) about 162 features 162 open issues 163 Capture Agent of Acterna PVA-1000 222 Cisco WebEx 33 Citylink AB 216 classification zone, COVID-19 situation in India Green Zone 34 Orange Zone (Non-Hotspots) 34 Red Zone (Hotspots) 34 Codec 167 Com Hem AB 200 Communications Assistance for Law Enforcement Act (CALEA) 126 compandor 174

competitors of broadband Blixtvik Internet ochTelefoni AB 200 Bredbandsbolaget (B2Bredband AB) 200 Com Hem AB 200 Microsoft 204 Skype Tele2 AB 200 TeliaSoneraSverige AB 201 Uppsala Stadsnat AB 202 Conversation Quality R-factor (R-Factor CQ) 223 coronavirus, in relation with VoIP. See video conferencing, during coronavirus cost elements, ISP 219 COVID-19 situation, in India about 34 classification zone wise 34 CPE gateway 105 customer proposition, of ISP internet telephony 205 customer requirements 199 Customers Premise Equipment (CPE) 58 CyberLink U Meeting 33

D Data Encryption Standard (DES) 118 data routing about 130 mechanisms 140 decentralized implementation 151 delay 94

Denial of Service (DoS) 126 designing factors 141 designing process about 136

performance, targeting 137 device provisioning and supplementary services 66 DHCP connecting to Diffie-Helman Diffie-Helman Digital Signal

Internet, through call center gateway 69 (DH) scheme 122 keys 122 Processing (DSP) 105

Digital Subscriber Line (DSL) 55 digital transmission in classical approach 85 distribution of costs, for T1 subscribers 219 Domain Name System (DNS) 189 Dual-Tone Multiple-Frequency (DTMF) 59 Dynamic Host Configuration Protocol (DHCP) settings 26 E E-911 154 echo 83 echo cancelation tail length 60 echo dealings about 60 with digital subscriber 57 echoes 95

economics models, IP telephony build-it-yourself 219 outsourcing 220 resale 221 SYSteam Nat’s Financial Business case 219 e-mail provider, for video conferencing in coronavirus times different platforms of e-mail 32 Google Meet 32 Office 365 32 platform, using 32 Emergency Operations Centers 25 E-models 99 encryption 117 encryption protocols about 118 advanced encryption standard 119 Data Encryption Standard (DES) 118 Rivest Cipher (RC4) 119 Secure RTP (SRTP) 120 Triple Data Encryption Standard (3DES) 118 endpoints, VoIP 22 Enhanced Service Providers (ESPs) 6 ENUM 188 ESPs data packets 7 extended DSL Architecture 112 F

fax machine 114 fax multicasting 115

Fax over Internet Protocol (FoIP) 115 Fax over IP (FoIP) 1 feature implementation in RISC processor 63 Feature Servers 171 Federal Communications Commission (FCC) 5 Fiber-To-The-Home (FTTH) 138 File Transfer Protocol (FTP) downloads 137 firewall about 134 features 135 FLASH Memory 142 Foreign Exchange Office (FXO) about 66 functionalities 80 Foreign eXchange Office (FXO) ports about 55 lifeline for failure of power 67 redirection of calls 67 VoIP remote calling 67 Foreign Exchange Station (FXS) about 66 functionalities 80 mirrored by FXO 81 FreeConference.com 33 FreePBX server 22 FreeSWITCH server 24 free tiers 33 fundamental VoIP applications 3 FXO circuitry 80 FXO ports usages

lifeline for failure of power 81 re-direction of calls 81 VoIP remote calling G

G.107 223 gateways, VoIP about 13 advantages 14 implementing 13 GoDaddy 32 Google Meet 32 GoToMeeting 33 governance initiatives, by Government of India about 34 ArogyaSetu app 34 Ayushman Bharat 35 Jandhan 36 PM Aatmanirbhar Bharat Abhiyan Yojana 35 PM CARES Fund 35 Vocal for Local 35 GR57 87 GR303 87 GR1089 88 H

H.323 about 177

gatekeeper 178 gateway 178 H.225 179 H.235 180 H.245 179 H.350 180 H.450 180 Multipoint Conference Unit (MCU) 178 terminal 178 hardware components about 168 access gateways 169 end-points of IP 169 Integrated Access Devices (IAD) 170 header types about 176 IP header 177 PHP header 176 physical transport header 176 RTP header 177 UDP header 177 HiNet LP 5100 IP telephones 12 HTTP-CGI 161 HTTP Servlets 164 hybrid key 121

I

Id factor 99 Ie factor 99

IETF protocols 107 IKE 122 independent providers, of VoIP market about 51 disadvantages 52 Infodemic 31 information router

basics 131 Integrated Access Devices (IAD) 170 interactive fax-on-demand/fax services 115 Intermedia 32 Intermedia Unite platform 33 Internet Group Management Protocol stacks 149 internet layer, TCP/IP 149 Internet Protocol (IP) 149 Internet security (IPSec) 125 Internet Service Providers (ISPs) 6 internet telephonic network solutions best practices 213 internet telephonies cost, for consumers 207 internet telephony architecture options 155 Internet telephony models about 221 software, for testing 222 test done, with SYSteam Nat AB 221 Internet Telephony Service Providers (ITSPs) 8 intrusive tests. See active tests

IP Centrex 157 IPCourier Ethernet Phone of Nokia 12 IP header 177 IP phones 11 IP Security Protocol (IPSec) 57 IPShuttle product of Nokia 12 IP telephony about 212 call 222 economics models 218 giver, recommending 221 network solutions, designing 213 provider, recommending 218 IP telephony system 4 IPv4 133 IP video cameras 26 ISDN Architecture 216 Is factor 99 ISP goals 199 ISP operating 214 iSpy 27 issues, of VoIP networking about 93 delay 94 loss of voice packets 95 issues of VoIP quality, in relation with example deployment 97 ITU P.862 PESQ 223 J

JAIN SIP 163 Jandhan 36 Java API for Integrated Networks (JAIN) 163 Jitter 94 JVM 164

K key exchanging methods about 121 Diffie-Helman keys 122 hybrid key 121 public key 121 symmetric key 121 L Least Cost Routing (LCR) 145 line echo cancelation 60 line echoes 85 Linphone 22 Listening Quality R-factor (R-Factor LQ) 223 Local Area Network (LAN) 1 local loop 86 M mapping, of telephone number 189 marketing models, SYSteam Nat AB

about 195 societies owned by tenants 195 students housing facilities 195 Mean Opinion Scores (MOS) 92 measurement tools, of quality of voice about 98 intrusive/ active tests 98 passive tests 99 media encoding 174 Media Gateway Control Protocol (MGCP) 184 media server 171 media transport 152 media types about 151 data 151 video 152 voice 152 Megaco 107 Megaco/ H.248 about 184 Media Gateway (MG) 183 MGC 183 MGCP entities CA 184 Media Gateway (MG) 184 Microprocessor without Interlocked Pipeline Stages (MIPS) 140 Microsoft 204 MIKEY 123 Mobile Command Center (MCC) 25 mobility and virtual locations 153 MOS Conversational Quality Estimated 223

MOS Listening Quality Objective 223 MOS Listening Quality Subjective 223 MOS values Conversational Quality MOS (CQ) 222 Listening Quality MOS (LQ) 223 MotionEye 28 MPEG-2 Codec 174

MPEG-4 Codec 175 mu law 174 Multimedia Terminal Adapter (MTA) 159 Mumble 23 N

NAPT 132 National Health Authority (NHA) 35 Net2Phone software 158 Network Access Points (NAPs) 47 Network Address Translation (NAT) Network Management Software (NMS) 225 network services about 110 access networks 111 backbone network 110 Nortel Networks and Lucent Technologies 42

O OAM&P data flow 50

Office 365 32 Open Network Video Interface Forum (ONVIF) 25 Operations Support System architecture 196 outsourcing 198 P packets 175 Pan, Tilt, Zoom (PTZ) models 25 passive tests 99 payload 176 PBX extension about 156 benefits 157 PBX replacement 156 PC-to-phone communication 10 peer to peer (P2P) 193 peer-to-peer (P2P) protocol 177 Perceptual Analysis and Measurement System (PAMS) 98 Perceptual Evaluation of Speech Quality (PESQ) 98 Perceptual Speech Quality Measurement (PSQM) 98 Personal Digital Assistants (PDAs) 168 PHP header 176 physical layer, TCP/IP 150 physical media 151 physical transport header 176 platform of provider of VoIP, in COVID-19 times for video conferencing Cisco WebEx tools 33 GoToMeeting 33

Intermedia Unite platform 33 RingCentral Office platform 33 playout 63 PM Aatmanirbhar Bharat Abhiyan Yojana 35 PM CARES Fund 35 policy servers 173 Port Address Translation (PAT) 131 POTS interfaces 63 POTS mimicking 58

POTS networking basic structure 56 PPPoE client structure 69 practical VQM solution aspects 100 Pradhan Mantri Jan Arogya Yojana (PM-JAY) 35 primary standards GR57 87 GR303 87 GR1089 88 TA909 87 Private Branch eXchange (PBX) 11 protocols, for VoIP H.323 177 Media Gateway Control Protocol (MGCP) 184 Megaco/ H.248 182 SIP 180 Skinny Client Control Protocol (SCCP) 185 Telephony Routing over IP (TRIP) 185 PSTN-IP Gateway 9

public key 121 public nodes connections 217 Public Switched Telephone Network (PSTN) 1 Public Telephone Operator (PTO) 217 Pulse Code Modulation (PCM) 62 Q Quality of Service (QoS) 12

R Real Time Control Protocol (RTCP) 100 Real-Time Operating Systems (RTOS) 55 Real Time Protocol (RTP) 116 Real-Time Streaming Protocol (RTSP) 25 Reduced Instruction Set Computer (RISC) Processor 55 reporting blocks about 101 of statistic summary 101 of VoIP metrics 102 residential gateway implementing 70 residential service 157 residential VoIP prices about 205 cost free 206 extra costs 206 extra possible charges 206 R factor 99

RingCentral Office platform 33 RISC processor 64 Rivest Cipher (RC4) 124 Rivest, Shamir, and Adelman (RSA) 119 router features 136 performance factors 136 RTCP XR 100 RTP header 177

S

Secure RTP (SRTP) 120 securities by obscurities 106 security association (SA) 124 security elements, VoIP 70 security, VoIP. See also VoIP Security about 106 authentication 106 Denial of Service Protection 106 integrity 106 nonrepudiation 106 secrecy 106 Server Message Block (SMB) 62 services, of retails 112 services, of VoIP 109 Session Description Protocol (SDP) 116 session initiated protocol - common gateway interface (SIP-CGI) about 161 Test-Bed Architecture 161

Session Security Layer (SSL) 125 Shinobi 30 signaling conversion 171 signaling protocol stack 179 signaling server gateways 173 Signal Noise Ratio (SNR) 99 SIP about 180 architecture 180 SIP architecture media servers 181 proxy server 181 user agent 181 SIP Servlets 164 SIP specifications (RFC3261) 182 SIP trunking system 153 Skinny Client Control Protocol (SCCP) 185 Skype SLIC 67 S/MIME 108 softphones about 168 advantages 168 Software Analysis of VoIP by Acterna PVA-1000 222 software components about 171 application server 171 media server 172 policy server 173 signaling conversion 171 signaling server gateways 173

software, for testing calls Capture Agent of Acterna PVA-1000 222 Software Analysis of VoIP by Acterna PVA-1000 222 Xten’s X-Lite SIP Softphone 222 solution aspects, of practical VQM 100 solutions, of IP telephony about 212 build-it-yourself strategy 214 general 213 outsourcing 215 resale 218 standards about 86 primary standards 86 Subscriber Line Interface Circuit (SLIC) 80 SWOT analysis, telcos about 75 opportunities 77 strengths 76 threats 79 weaknesses 77 symmetric key 121 Synchronous Digital Hierarchy (SDH) 110 Synchronous Optical Network (SONET) 111 SYSteam Nat AB about 194 current situation, in model of market and business 194 marketing models 195 present situation 212 use-case scenario 194 SYSteam Nat’s Financial Business case 219

T

TA909 87 TCP/IP about 146 architecture 147 TCP/IP Primer about 146 application layer 147 internet layer 149 layers 147 physical layer 150 transport layer 148 TeamTalk 24 technology model 214 telcos about 75 relationship, with gateways 51 SWOT analysis 75 Tele2 AB 200 telephone number mapping 188 telephonic and packet network signaling 64 telephonic network local loop 86 transport and switching core 85 Telephony Routing over IP (TRIP) 185 TeliaSoneraSverige AB 201 transport and switching core 85

Transport Control Protocol (TCP) 39 Transport layer security (TLS) 125 transport layer, TCP/IP 148 TRIP 186 Triple Data Encryption Standard (3DES) 118 two-to-four-wire conversions 84 U Udac 194

UDP header 177 Uniform Resource Identifier (URI) 189 Uppsala Stadsnat AB (Uppsala Metropolitan Area Network) 202 usage methods about 156 IP Centrex 157 PBX extension 157 PBX replacement 156 residential service 157 User Agent Client (UAC) 181 User Agent Server (UAS) 181 User Datagram Protocol (UDP) 39 user interface (UI) 27 user levels admin 30 sub-account 30 superuser 30 US National Institute of Science and Technologies (NIST) 118

V

video/audio conference calling tools free tools 33 paid tools 33 video conferencing, during coronavirus about 30 current Work-from-Home (WFH) situation in India and worldwide 31 platform for e-mail provider 32 platform of provider of VoIP, using 33 platform, selecting 31 third-party paid vendor, using 33 video conferencing tools about 23 FreeSWITCH server 24 TeamTalk 24 video display software about 26 iSpy 27 MotionEye 28 Shinobi 30 ZoneMinder 29 video surveillance and streaming about 25 IP video cameras 25 Virtual Private Networks (VPNs) 114 Vocal for Local 35 Vocoder 95 Voice Activity Detection (VAD) 95 voice detection 61 voice encoding 60

voice encryption protocol SRTP 120 voice frequency (VF) 83 voice gateway data functions 68 Voice over Internet Protocol (VoIP) about 2 advantages analysis report 222 and mobile service provider 75

basic system working 2 business and market model conclusion 230 concluding, in relation with Covid-19 232 conclusion, with regards to technology 231 ending, when concerned with economics 231 endpoints 22 gateways 13 general conclusion 229 IP phones 3 IP telephony system 4 market of equipment PC-to-phone communication 10 protocols 177 PSTN services 3 regulation 5 residential VoIP prices 205 security 106 security elements 70 services 109 telcos 75 video/audio conferencing 20 VoIP gateway 3

Voice over IP 3 voice processing security 126 voice quality assurance about 93 Mean Opinion Scores (MOS) 93 voice telephony 7 VoIP challenges about accumulation delay 44 network delay 44 processing delay 43 propagation delay 43 VoIPequipment 11 VoIP gateways 1 VoIP market independent providers VoIP methods 9 VoIP Network Architecture 110 VoIP networking issues 93 VoIP reliability about 187 end-to-end 188 for hybrid architecture 187 of trunk replacement 187 VoIP Security about 116 call controls 117 components 117 configuration 125 data streams 117

demand for 116 performance measurement 118 security areas 117 voice streams 117 VoIP Security, for call controlling process about 125 Denial of Services (DoS) 126 Internet security (IPSec) 125 open issues 126 transport layer security (TLS) 125 voice processing security 126 VoIP servers 20 VoIP software basic software 5 VoIP video/audio conferencing about 20 Asterisk server 21 FreePBX server 21 Linphone 22 Mumble 23 VoIP server 20 VoWLAN architecture and data flow 97 W wide band 174 Wireless Local Area Networks (WLANs) 91 X

Xten’s X-Lite SIP Softphone 222 Z ZoneMinder 29 Zoom 31 Zoom Meeting (Paid License) 33