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- Herausgeber: Packt Publishing
- Kategorie: Wissenschaft und neue Technologien
- Sprache: Englisch
Master the art of advanced VoIP and WebRTC communication with the most dynamic application server, FreeSWITCH
About This Book
- Forget the hassle - make FreeSWITCH work for you
- Discover how FreeSWITCH integrates with a range of tools and APIs
- From high availability to IVR development use this book to become more confident with this useful communication software
Who This Book Is For
SysAdmins, VoIP engineers – whoever you are, whatever you're trying to do, this book will help you get more from FreeSWITCH.
What You Will Learn
- Get to grips with the core concepts of FreeSWITCH
- Learn FreeSWITCH high availability
- Work with SIP profiles, gateways, ITSPs, and Codecs optimization
- Implement effective security on your projects
- Master audio manipulation and recording
- Discover how FreeSWITCH works alongside WebRTC
- Build your own complex IVR and PBX applications
- Connect directly to PSTN/TDM
- Create your own FreeSWITCH module
- Trace SIP packets with the help of best open source tools
- Implement Homer Sipcapture to troubleshoot and debug all your platform traffic
In Detail
FreeSWITCH is one of the best tools around if you're looking for a modern method of managing communication protocols through a range of different media. From real-time browser communication with the WebRTC API to implementing VoIP (voice over internet protocol), with FreeSWITCH you're in full control of your projects. This book shows you how to unlock its full potential – more than just a tutorial, it's packed with plenty of tips and tricks to make it work for you.
Written by members of the team who actually helped build FreeSWITCH, it will guide you through some of the newest features of version 1.6 including video transcoding and conferencing. Find out how FreeSWITCH interacts with other tools and APIs, learn how to tackle common (and not so common) challenges ranging from high availability to IVR development and programming advanced PBXs.
Great communication functionality begins with FreeSWITCH – find out how and get your project up and running today.
Style and approach
Find out how it works, then put your knowledge into practice - that's how this advanced FreeSWITCH guide has been designed to help you learn. You'll soon master FreeSWITCH and be confident using it in your projects.
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Seitenzahl: 339
Veröffentlichungsjahr: 2016
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Table of Contents
Mastering FreeSWITCH
Mastering FreeSWITCH
Copyright © 2016 Packt Publishing
All rights reserved. No part of this book may be reproduced, stored in a retrieval system, or transmitted in any form or by any means, without the prior written permission of the publisher, except in the case of brief quotations embedded in critical articles or reviews.
Every effort has been made in the preparation of this book to ensure the accuracy of the information presented. However, the information contained in this book is sold without warranty, either express or implied. Neither the authors, nor Packt Publishing, and its dealers and distributors will be held liable for any damages caused or alleged to be caused directly or indirectly by this book.
Packt Publishing has endeavored to provide trademark information about all of the companies and products mentioned in this book by the appropriate use of capitals. However, Packt Publishing cannot guarantee the accuracy of this information.
First published: July 2016
Production reference: 1260716
Published by Packt Publishing Ltd.
Livery Place
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Birmingham B3 2PB, UK.
ISBN 978-1-78439-888-0
www.packtpub.com
Credits
Authors
Anthony Minessale II
Giovanni Maruzzelli
Reviewers
Ayobami Adewole
Brian West
Commissioning Editor
Amarabha Banerjee
Acquisition Editors
Neha Nagwekar
Rahul Nair
Content Development Editor
Kajal Thapar
Technical Editors
Pramod Kumavat
Mohita Vyas
Copy Editors
Dipti Mankame
Safis Editing
Project Coordinator
Shweta H. Birwatkar
Proofreader
Safis Editing
Indexer
Tejal Daruwale Soni
Graphics
Disha Haria
Production Coordinator
Arvindkumar Gupta
Cover Work
Arvindkumar Gupta
About the Authors
Anthony Minessale II is the primary author and founding member of the FreeSWITCH Open Source Soft-Switch. Anthony has spent around 20 years working with open source software. In 2001, Anthony spent a great deal of time contributing code to the Asterisk PBX and has authored numerous features and fixes to that project. In 2005, Anthony started coding a new idea for an open source voice application. The FreeSWITCH project was officially open to the public on January 1 2006. In the years that followed, Anthony has been actively maintaining and leading the software development of the FreeSWITCH project. Anthony also founded the ClueCon Technology Conference in 2005, and he continues to oversee the production of this annual event.
Anthony has been the author of several FreeSWITCH books, including FreeSWITCH 1.0.6, FreeSWITCH 1.2, FreeSWITCH Cookbook, and FreeSWITCH 1.6 Cookbook.
I'd like to thank my wife Jill and my kids, Eric and Abbi, who were in grade school when this project started and are now grown up. I'd also like to thank everyone who took the time to try FreeSWITCH and submit feedback. I finally thank my coauthor Giovanni Maruzzelli for working on this book.
Giovanni Maruzzelli (<[email protected]>) is heavily engaged with FreeSWITCH. In it, he wrote a couple of endpoint modules, and he is specialized in industrial grade deployments and solutions. He's the curator and coauthor of FreeSWITCH 1.6 Cookbook (Packt Publishing, 2015).
He's a consultant in the telecommunications sector, developing software and conducting training courses for FreeSWITCH, SIP, WebRTC, Kamailio, and OpenSIPS.
As an Internet technology pioneer, he was the cofounder of Italia Online in 1996, which was the most popular Italian portal and consumer ISP. Also, he was the architect of its Internet technologies (www.italiaonline.it). Back then, Giovanni was the supervisor of Internet operations and the architect of the first engine for paid access to www.ilsole24ore.com, the most-read financial newspaper in Italy, and its databases (migrated from the mainframe). After that, he was the CEO of the venture capital-funded company Matrice, developing telemail unified messaging and multiple-language phone access to e-mail (text to speech). He was also the CTO of the incubator-funded company Open4, an open source managed applications provider. For 2 years, Giovanni worked in Serbia as an Internet and telecommunications investment expert for IFC, an arm of the World Bank.
Since 2005, he has been based in Italy, and he serves ICT and telecommunication companies worldwide.
I'd like to thank all people who made writing this book a challenging journey for me, all who helped, all who supported, all who gave me obstacles to overcome. This book has been brought to you by the knowledge that was socially cumulated by humans through the centuries, let's praise them. I finally want to thank my coauthor Anthony Minessale II for being so patient and "Always See Everything."
About the Reviewers
Ayobami Adewole is a software engineer and technical consultant with experience spanning over 5 years. Ayobami has worked on mission critical systems; these include solutions for customer relationship management, land administration and geographical information systems, enterprise-level application integrations, and unified communication and software applications for the education and business sectors.
Ayobami is very passionate about VoIP technologies, and he continues to work on cutting-edge PBX solutions built on FreeSWITCH. In his spare time, he enjoys experimenting with new technologies. His blog is at http://ayobamiadewole.com.
My unending gratitude goes to my parents for instilling in me the culture of discipline and hard work.
Brian West is a founding member of the FreeSWITCH team. He has been involved in open source telephony since 2003. Brian was heavily involved in the Asterisk open source PBX Project as a Bug Marshal and developer. In 2005, Brian joined the initiative that eventually lead to the FreeSWITCH Open Source Soft-Switch. Today, Brian serves as the general manager of the FreeSWITCH project and keeps the software moving forward. Brian has countless skills as a developer, tester, manager, and technologist, and he fills a vital role in the FreeSWITCH Community.
Contributors
Moises Silva wrote the entire 6th chapter, PSTN and TDM.
The following people contributed substantially to this book:
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Preface
Real Time Communication (RTC) is a huge sector, in perennial growth. It spans from VoIP to FAXes, from VideoConferencing to CallCenters, from PBXes to WebRTC, using many interworking technologies to connect the past with the future, legacy applications to new users and markets, creating and developing new ways for saving time and money, fostering collaboration, and enjoying leisure.
FreeSWITCH covers it all; it is the most reliable, scalable, and flexible open source foundation, and is used to build services and products worldwide.
This book adopts a professional approach and attitude, making available a wealth of cumulated actual industry experience in each aspect of FreeSWITCH implementation.
Written for professionals, each chapter contains the knowledge needed to frame and understand its domain, and a thorough explanation of FreeSWITCH wheels and knobs, best practices, and real-world solutions.
What this book covers
Chapter 1, Typical Voice Uses for FreeSWITCH, gives an overview and analyzes each sector where FreeSWITCH is in production.
Chapter 2, Deploying FreeSWITCH, shows best practices in FreeSWITCH installation and management.
Chapter 3, ITSP and Voice Codecs Optimization, suggests what to look for when choosing an Internet Telephony Service Provider, and how to get the best from DIDs, terminations, T38, and voice traffic.
Chapter 4, VoIP Security, exposes specific measures and tools used to keep FreeSWITCH protected from unwanted attention and hostile behavior.
Chapter 5, Audio File and Streaming Formats, Music on Hold, Recording Calls, covers all that is related to audio manipulation with FreeSWITCH, from prompts optimization to call center barge in, from playing live streams to HD codecs.
Chapter 6, PSTN and TDM, happens to be the first published, thorough explanation of all possible interactions between FreeSWITCH and Sangoma, Digium, and other compatible hardware for interfacing traditional and legacy telephony networks.
Chapter 7, WebRTC and Mod_Verto, provides a detailed overview of what WebRTC is and what techniques it entails, and then follows the development of a complete FreeSWITCH implementation.
Chapter 8, Audio and Video Conferencing, delves into the intricacies of setting and managing FreeSWITCH multiuser conferences both via SIP and WebRTC, with chatting, screen sharing, moderation, and advanced techniques for videocomposing the screen.
Chapter 9, Faxing and T38, explores all facsimile transmission aspects, and how to reliably fax via VoIP, send office documents, and integrate with mail.
Chapter 10, Advanced IVR with Lua, proves that it is not your average code snippet or more of the same example. Starting from the thoroughly described script techniques, it will be possible to build your industry-grade applications.
Chapter 11, Write Your FreeSWITCH Module in C, describes exactly what is needed to add or modify FreeSWITCH functionalities at the most fundamental level: interfacing your custom hardware, or your legacy OSS, or whatever.
Chapter 12, Tracing and Debugging VoIP, shows the art of SIP packet tracing, using the latest open source tools.
Chapter 13, Homer, Monitoring and Troubleshooting Your Communication Platform, walks through the operation of the most advanced VoIP/WebRTC monitoring and data warehousing solution: Homer. Once implemented, your support staff will reach Nirvana!
What you need for this book
For implementing the same solutions described in this book, you will need a (virtual) machine with Debian 8 (Jessie) 64 bit, and some Linux admin and networking knowledge.
Who this book is for
This book is for skilled professionals who want to jump right into the depths of FreeSWITCH, such as system administrators, programmers, and telephony technicians who want to augment their ability to create real-world VoIP and WebRTC products and services.
Conventions
In this book, you will find a number of styles of text that distinguish between different kinds of information. Here are some examples of these styles, and an explanation of their meaning.
Code words in text, database table names, folder names, filenames, file extensions, pathnames, dummy URLs, user input, and Twitter handles are shown as follows: "Several built-in modules exist to assist in this, such as mod_lcr and mod_nibblebill, but the real beauty of FreeSWITCH's handling of calls in a wholesale scenario is due to four core building blocks."
A block of code is set as follows:
When we wish to draw your attention to a particular part of a code block, the relevant lines or items are set in bold:
New terms and important words are shown in bold.
Note
Warnings or important notes appear in a box like this.
Tip
Tips and tricks appear like this.
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Chapter 1. Typical Voice Uses for FreeSWITCH
FreeSWITCH (FS) is one of the world's most robust Real Time Communication (RTC) switching tools. It packs a rich feature set, and its modular approach allows it to stay ahead of the curve as new technologies emerge in the marketplace.
With this strong foundation, FreeSWITCH has matured into a product which is in use in a multitude of environments. However, FreeSWITCH can also be complex and overwhelming because of its rich feature set.
This book unravels some of the ways FreeSWITCH can be utilized.
In this chapter, we will cover "traditional" Voice over IP usage. See other chapters for video, conferences, RTC, and so on. We will also cover the following:
Understanding routing calls in FreeSWITCH
Routing calls is the very essence of FreeSWITCH. Moving calls around can assume very different meanings and use very different techniques, depending on the scenario and with which aims it is done. You don't use the same tools and interaction level for an enterprise PBX, a telemarketing dialer, and a provider-to-providers minutes exchange.
FreeSWITCH's remote console at startup
Wholesale (provider to providers)
FreeSWITCH supports a multitude of useful features for call routing services. When we describe call routing, we are referring to connecting Party A with Party B. These routing scenarios are generally heavy on logic regarding cost analysis, interconnections with other carriers, and user permissions. These routing scenarios also typically exclude features the user directly interacts with (such as voicemail or auto attendants).
FreeSWITCH can be utilized as a powerful wholesale routing engine. Several built-in modules exist to assist in this, such as mod_lcr or mod_nibblebill, but the real beauty of FreeSWITCH's handling of calls in a wholesale scenario is due to four core building blocks:
FreeSWITCH's flexible design aids in providing a tremendous amount of customization and capabilities as well. Examples include the ability to add transcoding support for codecs at any moment during the call in a way that will automatically and inherently work with any other codecs which are installed, and the ability to add custom handling for failures in a way that suits your environment.
Wholesale services typically represent high-volume customers who want to:
FreeSWITCH is capable, out of the box, of providing all of these services with simple dial plan configuration. Additionally, FreeSWITCH can be attached to a web, mobile, or legacy user interface to allow for users to manage their account, services, and monetary assets.
Residential uses of FreeSWITCH
FreeSWITCH stands as one of the best open source class 4 (and class 5) switch options, and is often the undisputed champ in many different roles because of the number of features offered by the many ready-made modules. It is definitely an excellent choice for the Internet Telephony Service Provider (ITSP), but let's not forget one of its simplest use cases: Residential service.
Some residential options include things like Network Address Translation (NAT) when configuring end-user devices like Analog Telephone Adapters (ATA). This can be challenging when working with disparate networks and client devices residing on LANs behind residential gateways and firewalls.
FreeSWITCH has configurable options for its Session Initiation Protocol (SIP) stack (called Sofia) especially designed to overcome these hurdles and provide a viable solution for residential service.
Some reasons why FreeSWITCH makes a good choice for residential service are:
Routing with federated VoIP
Federated VoIP is a distributed Voice over IP network composed of autonomous systems.
Federated VoIP is to telephony what internet e-mail is to messaging. Particularly, it allows for the free flow of traffic without depending on a central exchange (or exchanges), just like e-mail does not depend on a central post office. It works by exchanging mail messages directly between organizations' (or even personal) Mail Servers that have the authority and capability of managing their own traffic.
Let's continue with the example of e-mail (of note, SIP was based on SMTP and HTTP protocols, that is, the protocols that orchestrate mail and the Web). So, here's the trick: no central authority is involved, it's all peer-to-peer exchange of messages in a worldwide network that works with extreme overall reliability day in and day out for billions of people and trillions of communication exchanges.
Exactly the same criteria can be applied to Voice over IP (SIP) and Instant Messaging (SIMPLE or XMPP), basing all communication exchanges around the concept of a personal address like an e-mail address, which is used both by SIP and IM, and often exactly the same for both clients. The example address [email protected] could be used for all unified communications with Joe.
Initially, VoIP had been popularized as a better and cheaper way to manage traditional telephony traffic and to connect to traditional voice carriers. Then it was adopted by the carriers themselves because of its better suitability to modern digital networks and compatibility between hardware providers. So, today's approach to VoIP often brings an unnecessary prejudice about dependency from carriers.
Federated VoIP gets rid of this, having autonomous servers exchanging their communications, finding each other via DNS (queries about destination address) without the need for central authority and/or carriers, just like the e-mail system. Around this core concept has grown an ecosystem of encryption, mapping, and resolving traditional telephone numbers via DNS (ENUM) and other additional services. It should be noted that there is no technical requirement for encryption in Federated VoIP.
FreeSWITCH has all the features needed by Federated VoIP:
FreeSWITCH is able to work as a complete, self-sufficient autonomous system or as a part of a bigger composite system with one or more SIP proxies, like Kamailio or OpenSIPS, taking care of routing, proxying, load balancing, and so on.
Dialers/telemarketing
The subject of dialers and telemarketing often makes system administrators and telephony switch operators queasy with anxiety when they are considering the limitations of their networks, hardware capability, and other system resource implications with the onslaught of marketing campaigns directed to their customers. This certainly does not stop FreeSWITCH from being a great choice when writing dialer and telemarketing applications, and not all dialer and telemarketing systems should have negative connotations.
FreeSWITCH is a natural front-runner when choosing a softswitch for writing dialer and telemarketing applications because of the small learning curve needed to develop applications in a variety of programming languages, and the excellent community support.
A developer can create a custom dialer application in FreeSWITCH utilizing a core switch database data in real-time to drive the logic. They can utilize modules like mod_event_socket to connect to the switch and perform API functions like initiating calls and managing IVRs for things like credit card payment, billing and collection, or opt-in and opt-out campaign functionality.
Not all telemarketing and dialer applications are used for marketing. Some ways FreeSWITCH is currently being utilized for dialers and telemarketing are:
Some rows from FreeSWITCH's remote console help
FreeSWITCH Products and Services
Another way to see what FreeSWITCH can do is to think in terms of what services it will give its users. Here, too, different technologies and techniques are deployed to cater to different kinds of users, looking for a different set of features.
Business PBX services (hosted and on-premises)
FreeSWITCH's scalability and feature set lends itself naturally to being used as the basis of an extremely powerful business PBX phone system. Successfully deployed in both on-premises environments for small SOHO businesses while scalable to hundreds of users, or utilized as the foundation for hosted PBX services hosting hundreds of thousands of users, the system lends itself naturally to powering these types of solutions.
Out of the box, FreeSWITCH includes basic PBX modules which provide powerful functionalities. These modules include features such as:
We could go on further, but this is a good general idea of the building blocks that are provided. Most of these modules can be activated by adding four to six lines of XML to a dial plan configuration file. The power of dial plan combined with modules should not be underestimated - this is powerful stuff with very little work to get it going!
Additional components exist for expanding into:
Customer demands will sometimes lead to more complex requirements that might not be handled by default modules. However, ready-made building blocks combined with the ability to run your own custom scripts within FreeSWITCH allows for providing quick time to market services even for the most demanding customer base.
Call centers
Any company doing substantial business in any market segment will attest that support is a cornerstone of a business's success. A robust and comprehensive telephony platform is crucial, and FreeSWITCH allows for a configurable, scalable and maintainable solution suitable for call centers of any size.
There is no shortage of flexibility with FreeSWITCH. If your solution requires a custom application, FreeSWITCH provides a host of development options for your call control and routing. Although you are free to use any supported language to "roll your own" solution, FreeSWITCH comes complete with robust call center modules that are being utilized in production environments in literally thousands of deployments all over the world.
Mod call center includes features like:
With IVR trees easily integrated into your call center solution and full access to databases for CRM and Knowledge Basis, your ability to create call center applications is almost limitless.
If your requirements do not dictate the granularity of complex configuration options, then there are other options available with an alternative FreeSWITCH module called Mod FIFO. As the name implies, it serves as a first in, first out call-queuing mechanism, with many features and strategies, music on hold, and announcements, that's easy to integrate in custom or third-party applications.
Value added services and games, prizes, and polls
Value Added Services (VAS) are services that offer something on top of pure voice transport.
Some examples include:
Interactive entertainment and polling is a business that fits perfectly with the ease of programming and integrated messaging capability of FreeSWITCH.
Here are some examples of what has been realized in this field:
"Class 4" vs "Class 5" operations (and SBCs)
FreeSWITCH is a softswitch. That is, it is a software that handles and interconnects calls, like the manual switchboards where operators answered and distributed calls by moving jacks and cables in old black and white movies.
Softswitches in telco jargon are often categorized as pertaining to a "class," and "Class 4" and "Class 5" are the only two classes you will hear about.
Because those are fuzzy terms, almost marketing terms, you will never find the exact demarcation between Class 4 and Class 5 features and capabilities; a lot of them overlap (anyway, it's mostly the same technology).
An arbitrary rule of thumb can be to use Class 4 when talking about large volume, wholesale switching of call minutes between different carriers, ITSPs, CLECs, with minimal meddling in the audio streams (apart from transcoding, if needed). The term "Class 5" applies to audio or text-based services where end user interaction is in focus and where sophisticated logic is required.
FreeSWITCH is widely used in both contexts.
A typical Class 4 usage would be to interconnect many providers of international voice routes and sell voice minutes based on algorithms about least cost route and/or route quality. Here, the sheer volume of signaling that can be managed per second and the availability of very efficient ways to lookup which route to connect to is of paramount importance. FreeSWITCH with "bypass media", mod_lcr, mod_easyroute, some Lua scripting or custom C code is a perfect platform, easy to use and modify on the fly, without service interruption.
Typical Class 5 usages would be an enterprise or SOHO PBX, a call center system, a fax server with mail2fax and fax2mail, an airport IVR to query flights' arrival times, and so on. Here, FreeSWITCH offers prized features like audio quality (that is, no glitches, distortions, and so on), programmability (how easy it is to implement complex services and business logic), capability of interfacing different media (PSTN to WebRTC, SIP to Skinny, TDM to Skype, SMS to XMPP, and so on) and different audio formats (alaw, ulaw, High Definition Audio, Silk, Siren, G729, Opus, mp3, wav, raw, and so on). Easy integration of Text To Speech and Automatic Speech Recognition, manipulation of audio prompt libraries, and easy ways to gather and interact with user pressed DTMFs are the highlights in FS Class 5 operation.
"SBC" is another very vague marketing buzzword. A Session Border Controller (SBC) is a softswitch that sits on the edge of your own telecommunication network and acts as a point of demarcation and interconnection with the external world. Let's say an SBC is a softswitch with an emphasis on security, NAT traversal, media proxying, network connectivity, manageability, audio transcoding, protocol gatewaying (connecting with different protocols), and protocol adaptation (being the compatibility layer between different "interpretations" of the same protocol). FreeSWITCH excels in those areas, as we have seen before in the two "Classes", while it sports specific SBC features like the most advanced NAT traversal, so smart that it can connect endpoint (that is, user phones) behind residential ADSLs and firewalls, or form a federation between the many international offices of a company, each SBC sitting on different NATed LANs. Also, as security goes, FreeSWITCH is one of the reference implementations for ZRTP media encryption, as well as TLS and SIPS.
WebRTC / web services / Internet-only services
FreeSWITCH's unique modular approach made it an easy choice for extending integration into WebRTC and other web-based services which need a bridge between different types of technologies. As an example, web-based communications are useful but are often hindered by their inability to connect to the rest of the established world, causing adoption to be slow. As an example, users will be reluctant to get rid of their desk phone when their browser-based replacement can't call phone numbers but only other browsers. Best of all, WebRTC support follows the same ease-of-installation and global compatibility standards that FreeSWITCH has become known for in the VoIP world. Users can make calls where one side of the conversation is WebRTC and the other is the PSTN, or WebRTC to SIP and so on. FreeSWITCH does all the hard work of normalizing the audio and signaling services between the two services and bridging any gaps that may exist when connecting from one type of service to another.
Mobile "over-the-top" SIP
As mobile services become more pervasive in the telecommunications industry, mobile network operators have responded by increasing data speeds. In this process, many service providers are now investigating "over-the-top" services which utilize data communication services to transmit and receive voice and video. These services often link to messaging or social applications and provide both real-time, semi real-time, and recorded communication services via data connections. In many cases, the user experience simulates phone technology even though it is not using traditional telephony services provided by the underlying communications service provider. In these cases, there is added complexity for handling such services.
Over-the-top services face a number of challenges, which include:
