Switching to VoIP
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More and more businesses today have their receive phone service through Internet instead of local phone company lines. Many businesses are also using their internal local and wide-area network infrastructure to replace legacy enterprise telephone networks. This migration to a single network carrying voice and data is called convergence, and it's revolutionizing the world of telecommunications by slashing costs and empowering users. The technology of families driving this convergence is called VoIP, or Voice over IP. VoIP has advanced Internet-based telephony a viable solution, piquing the interest of companies small and large. The primary reason for migrating to VoIP is cost, as it equalizes the costs of long distance calls, local calls, and e-mails to fractions of a penny per use. But the real enterprise turn-on is how VoIP empowers businesses to mold and customize telecom and datacom solutions using a single, cohesive networking platform. These business drivers are so compelling that legacy telephony is going the way of the dinosaur, yielding to Voice over IP as the dominant enterprise communications paradigm. Developed from real-world experience by a senior developer, O'Reilly's "Switching to VoIP" provides solutions for the most common VoIP migration challenges. So if you're a network professional who is migrating from a traditional telephony system to a modern, feature-rich network, this book is a must-have. You'll discover the strengths and weaknesses of circuit-switched and packet-switched networks, how VoIP systems impact network infrastructure, as well as solutions for common challenges involved with IP voice migrations. Among the challenges discussed and projects presented: building a softPBX; configuring IP phones; ensuring quality of service; scalability; standards-compliance; topological consideration; coordinating a complete system? switchover? migrating applications like voicemail and directory services; retro-interfacing to traditional telephony; supporting mobile users; security and survivability; and dealing with the challenges of NAT. To help you grasp the core principles at work, "Switching to VoIP" uses a combination of strategy and hands-on "how-to" that introduce VoIP routers and media gateways, various makes of IP telephone equipment, legacy analog phones, IPTables and Linux firewalls, and the Asterisk open source PBX software by Digium. You'll learn how to build an IP-based or legacy-compatible phone system and voicemail system complete with e-mail integration while becoming familiar with VoIP protocols and devices. "Switching to VoIP" remains vendor-neutral and advocates standards, not brands. Some of the standards explored include: SIP; H.323, SCCP, and IAX; Voice codecs; 802.3af; Type of Service, IP precedence, DiffServ, and RSVP; and 802.1a/b/g WLAN. If VoIP has your attention, like so many others, then Switching to VoIP will help you build your own system, install it, and begin making calls. It's the only thing left between you and a modern telecom network.
Product Details
- Amazon Sales Rank: #211000 in Books
- Published on: 2005-06-30
- Original language: English
- Number of items: 1
- Binding: Paperback
- 502 pages
Editorial Reviews
From the Publisher
Based on real-world experience, this handy solutions manual addresses the most common VoIP migration challenges. Find out how to build your own VoIP system, install it, and begin making calls--so you can start saving today. Ideal for IT managers, network engineers, and system administrators.
About the Author
Ted Wallingford is an executive technologist and the co-founder of Best Technology Strategy LLC, a company which helps entrepreneurs and established companies alike in the adoption, integration, and successful use of communication systems and business processes. A global thought leader on the subject of VoIP and Internet Protocol communications, Ted has emerged as an expert in the emerging fields of network convergence and unified business communication. Ted has written two technology books for O'Reilly Media, and has appeared on NPR Science Friday. He also periodically writes for Macworld Magazine and maintains the Signal to Noise blog. He resides in Cleveland, OH.
Excerpted from Switching to VoIP by Theodore Wallingford. Copyright © 2005. Reprinted by permission. All rights reserved.
Chapter 14 Traditional Apps on the Converged Network
When first designed, landline phone service was intended to carry sound signals, and its uses as a carrier of data were years away from realization. It’s ironic that the technology that predated the telephone was itself a data transport technology: the telegraph. This device carried encoded messages from terminal to terminal across the 19th-century equivalent of a peer-to-peer network.
A lifetime later, in the 1960s, sound-encoding devices emerged, and, very soon, computers were able to send data, represented as sound, across the telephone network. Those devices were modems, and later fax machines—the descendants of the telegraph. Modems, fax machines, voice mail systems, emergency 911 service, and a slew of other messaging tools evolved around the international telephone network.
Today voice and data networks converge and VoIP begins to replace Bell’s brainchild. IP telephony has the same fundamental goal as legacy telephony: facilitate human interaction at a distance. But, since IP telephony goes about this goal differently, not all of the specialized devices that evolved around the old system work with the new one. Fax machines, modems, and voice mail systems aren’t necessarily compatible with VoIP, because they grew into a mold that was shaped by the old network.
In this chapter, we’ll cover some of the great legacy technologies we’ve come to rely on and discuss ways of migrating their functionality to the converged network.
Fax and Modems
Fax machines and modems encode digital data into analog sound signals for transport across the PSTN. At the local CO, these analog sound signals are digitized into PCM digital signals at 64 kbps. After being transported over the network, the CO at the opposite end of the link decodes the PCM signal and plays it back for the analog receiver, whose job it is to reassemble it into a facsimile of the original digital data. Since the maximum rate of transmission on a POTS line is 64 kbps, faxes and modems can’t transmit data very quickly (one twenty-fourth the speed of a T1). Even
with compression techniques, which introduce signaling overhead, most modems
will never transmit data faster than about 54 kbps. They are limited by the digital
resolution of the sound pathway they use.
Sending a fax through an encoded audio channel such as those provided by VoIP and the PSTN is called in-band faxing.
The challenge posed by VoIP is this: codecs like G.729A distort the sound signal by compressing it using lossy vocoder algorithms such as CELP. When a modem or fax’s analog sound signal is encoded and decoded using vocoders, it becomes distorted, such that the device on the receiving end of the transmission is receiving a different analog signal than the one that was sent. The side effect of compressed VoIP codecs is that faxes and modems simply don’t work. So devices that rely on modems, such as some burglary alarm systems, TiVo consoles, and maybe that old-fashioned Amiga 1000 computer, have a hard time getting along in a VoIP network. (Series 2
TiVo devices can work using the Internet instead of a modem—check out tivo.com/adapters.)
There are a few ways to tackle this problem. The first and most obvious is to avoid lossy codecs and use only non-lossy codecs like G.711 and G.722. It won’t distort the analog modem signals as long as jitter is under control. But avoiding lossy codecs may not be possible, especially on bandwidth-starved WANs.
Fortunately, the ITU has two recommendations to aid in migration of fax to VoIP.Both are discussed in this chapter.
There is no real answer to the matter of modems, however. The ITU’s V.150.1 standard describes how to relay modem signals over a converged network, but no open reference implementations exist yet. One could argue that it’s a bit arcane to use a modem in a converged network, since any data sent by the modem could be sent hundreds of times faster using Ethernet. In other words, if you have an Ethernet connection, use that instead of the modem.
Thanks to the insistence of the U.S. Department of Defense, Version 4.1 of CallManager, Cisco’s venerable softPBX, does indeed support modem pass-through using V.150.1.
T.30, T.37, and T.38
The ITU’s T.30 recommendation describes how fax devices should work. Just about all legacy fax machines (and fax software) are T.30-compliant.
The T.38 protocol is the ITU’s recommendation for sending faxes over data networks. T.38 software runs on a server that can perform T.30 fax signaling in order to send and receive fax transmissions over analog lines to fax machines. The server encodes the fax signals into packets and sends them to a T.38 server that can decode the fax signals on behalf of a receiving fax machine. This circumvents reliance on inband faxing and increases the efficiency of fax transmission over the network. T.38 has a few nicknames, including Group 3 faxing and FoIP (Fax over IP).
The T.37 protocol is another ITU recommendation. It also allows faxes to be sent over the IP network, but T.37 doesn’t package fax signals into its own packets. Instead, it creates SMTP mail messages using MIME-encoded TIFF (Tagged Image File Format) files. These messages can be sent directly to an SMTP mail server for delivery to email users, or they can be sent to another T.37 server for delivery to a fax machine. T.37 has a few advantages over T.38, among them its ease of integration with standards-based email. T.37 is a store-and-forward protocol, meaning it can hold onto messages if they aren’t deliverable at the moment. T.38 has no such capability.
Customer Reviews
A MUST OWN
This book provides an excellent resource for anyone looking to expand their knowledge of the practical "workings" of VoIP. The information is very easy to read as the text is not filled with un-needed jargon, whilist still maintaining a high standard of detail. A must for anyone looking to start a VoIP project or just to find out a little more about the protcols behind VoIP.



