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VoIP Readiness

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Chapter 8

CHAPTER 8

VoIP Readiness

Let’s face it: VoIP isn’t exactly new, but IP telephony’s readiness for enterprise consumption is a fairly recent development. When it first appeared on the Internet scene, VoIP offered the ability for people to make free long-distance calls over the

Internet. In fact, products like Internet Phone came with substantial buzz about how they let in-laws with Microsoft Windows have half-duplex speakerphone conversations through their PCs over the Net.

Lack of interoperability, poor quality of service, and a drop in traditional long-distance calling rates ultimately killed the first generation of consumer VoIP software.

The short-lived voice-over-Internet craze of the late 1990s died. VoIP is still what historians might call a disruptive technology—it is changing the status quo—but as it becomes more standardized, quality-driven, and accepted, it also becomes a more sustaining technology, just as the PSTN has been for decades. In this regard, VoIP has proven much more valuable in the enterprise than in the home.

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Security and Monitoring

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Security and Monitoring

Like the Web, email, and other Internet communications tools, IP telephony can be secured. This fact is one of its biggest appeals over old-school telephone equipment.

Security means enforcing system policy, recording instances of abuse for forensic and litigation purposes, encrypting or otherwise hiding sensitive information in transit, bolstering call-management systems’ resilience to exploitive attacks and computer viruses, and securing the access perimeter of the VoIP network.

Security tools and enforcement practices for VoIP applications are the same, essentially, as those for other IP-based apps, because they run on the same network. The security objective of VoIP systems is largely the same as those of other IP-based systems: in a nutshell, preserve the operational status of the system.

There are many threats to this objective and many countermeasures to the threats.

Policy enforcements points, like firewalls, protect lower layers of the network, while authentication systems like RADIUS and application proxies provide higher-layer security. This chapter describes how to secure and harden a VoIP server, the basics of DMZs, how to enable logging of VoIP traffic with iptables, how to tweak the logging configuration of Asterisk, and how to log and monitor VoIP network traffic.

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Replacing Call Signaling with VoIP

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Replacing Call Signaling with VoIP

In Chapter 6, the process of transmitting voice sounds in virtual media channels, via codecs, was presented. In order for those media channels to be set up, monitored, and destroyed when needed, a PBX calls on signaling. Different methods are used for different kinds of endpoints and trunks. On the PSTN, the SS7 network handles signaling. On a POTS voice channel, the signaling is accomplished using analog FXS signaling.

SS7, FXS, and the dozens of other signaling technologies in use on the PSTN, though all signaling protocols, are outside the realm of VoIP. They could all be considered legacy technology, since just about all of their signaling functions have been replicated using several new, modern, open TCP/IP-centric standards. Even though SS7 is a packet-based protocol and there are attempts underway to make it compatible with

VoIP softPBX systems (Asterisk included), its roots are in the PSTN, not the Internet.

This chapter describes the standards for call signaling in a softPBX-based VoIP network; it also describes the ways these standards compete with and complement one another.

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Enterprise Telephony Applications

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Enterprise Telephony Applications

In the previous chapter, the technologies of legacy voice network systems were discussed. Some might find that subject fascinating enough to have spent more than a chapter on it. In fact, there are volumes on the subject, and the ITU web site (http:// www.itu.int) is filled with papers that describe it all in painfully unsparing verbosity.

But it’s telephony, the application functionality within the voice network, that is the fun part. Telephony accommodates and assists human interaction in a very real, personal way, which is why it’s such an engaging subject. Unlike written forms of communication, such as email or instant messaging, telephony’s distinguishing traits are its use of sound and its immediate, real-time nature. It’s a much more fundamental mode of interaction than the written form—because when we use telephony, we talk, the same thing we do when we’re together.

Telephony can use live, immediate speech or speech that’s recorded, stored, and played back later, depending upon the needs of the application—and it can be largely automated using well-defined standards. In fact, computer-integrated telephony applications have even been programmed to recognize and respond to human voice commands.

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SIP Methods and Responses

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Appendix A

APPENDIX A

SIP Methods and Responses

Table A-1 shows SIP methods.

Table A-1. Methods

INVITE

A SIP device is being invited to participate in a call.

ACK

Confirms that the client has received a final response to an INVITE request.

BYE

Terminates a SIP call. Can be sent by any party involved.

CANCEL

Cancels any pending call but does not terminate a call that has already been connected.

OPTIONS

Queries the capabilities of servers without requesting to establish a call.

REGISTER

Registers an IP with a SIP registrar.

PRACK

Insures reliability of provisional 1xx responses if a UAS offers them.

UPDATE

Updates a previously made offer for a not-yet-established session.

REFER

Initiates a call transfer by telling the recipient (specified by URI) to contact a third party using the contact information provided in the request.

SUBSCRIBE

Subscribes to be notified of an event occurrence; for example a user presence update.

NOTIFY

Used to notify that an event has occurred.

MESSAGE

A method signifying the payload is an instant message.

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