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Passing Voice Over The Internet Is Getting Into The Mainstream

Voice communications
is without a doubt an essential part of just about every business. For
many the telephone is the most essential tool. Without it they make no
sales calls, take no orders, and provide no customer service. When a
new technology comes along, such as Voice over IP, aka VoIP, it takes a
while for the idea to catch on.
As more businesses see the potential
cost savings that VoIP systems make possible, the idea gets even more
attractive. Some of the biggest savings come when your company has
multiple locations, and you route all voice traffic between offices
over your VoIP system. Then things start to get really interesting.
Hardware & Software
One of the first questions you might
ask would be what hardware and software will I need to implement a VoIP
system? The answer depends on what you want to do. With the convergence
of data, voice, and video, the possibilities are pretty broad. The key
to making a VoIP solution work is all about QoS. If the phone call
doesn’t sound as clear as a regular analog phone line, you’ll probably
get push back from the users.
At the lowest end you can start with
a simple IP phone that has a built-in adapter. These types of phones
connect directly to your network as opposed to a regular phone outlet.
On the software side there are
basically two options. You can go with a traditional vendor and its
proprietary software, such as Cisco or Avaya, or you could go with an
open-source solution such as Asterisk (www.asterisk.org). Asterisk is a
complete PBX (private branch exchange) in software running on top of
Linux or Mac OS X. Asterisk supports most of the telephony standards
including SIP (Session Initiation Protocol) and H.323. With Asterisk
you can build it yourself or turn to a company such as Four Loop
Technologies for a turnkey solution.
Standards compliance is important if
you want to interoperate with other systems. SIP is an IETF standard
for creating, modifying, and terminating sessions with one or more
participants. It is widely used as the signaling protocol for most VoIP
systems. H.323 is a similar standard from the ITU (International
Telecommunication Union) based in Geneva, Switzerland. Many European
systems use this standard in lieu of SIP.
Communicating between different
systems requires a solution that can translate between the different
standards. Asterisk supports both SIP and H.323, so you could initiate
a call on a SIP phone and connect with an H.323 phone on the other end.
Most of the major VoIP vendors provide support for both protocols in
their products.
Avaya
Companies with distributed offices
represent one of the biggest potential savings areas for a VoIP
implementation. Avaya has several new products that directly address
this situation. The Avaya one-X Quick Edition is a SIP-based solution
targeted at small businesses and branch offices with fewer than 20
users. Embedding the functionality of an IP PBX into the individual
phone eliminates the need for a central server. All you need to set up
communications is a PSTN gateway and basic phones.
The Avaya VPNremote for IP Phones is
a solution targeted at the teleworker that combines VoIP with VPN
functionality into a single device. With the Avaya VPNremote solution
you get the same phone experience and network connectivity you’d expect
in the office from virtually anywhere. The Avaya VPNremote solution
also supports VPN environments from Cisco Systems and Juniper Networks.
Cisco
It’s just about impossible to talk
about any product that uses IP without mentioning Cisco. Cisco has been
delivering voice solutions since before the term VoIP was even
mentioned. Cisco Unified Communications represents a wide range of
products and solutions that address the needs of delivering voice,
video, and IP communications over the same network.
One of the key advantages to a Cisco
solution is the breadth of offerings that the company provides. Many
organizations already have some type of Cisco equipment in the
enterprise networking infrastructure. Adding voice capability in many
cases is nothing more than a software upgrade. Having a single vendor
providing all your voice and data hardware has both good points and bad.
Cisco has five basic technology
components that make up the unified communications system. These are:
IP Telephony (call processing, telephones, and endpoint); Unified
Communications applications (unified communications clients, messaging,
and conferencing solutions); contact center applications (a complete
line of products for call center support); communications
infrastructure (traditional routers and switches); and management tools
(unified operations manager and service monitor).
Cisco’s Unified CallManager and
Unified CallManager Express products utilize existing Cisco networking
products along with Cisco Unified IP phones to deliver enterprise-ready
voice solutions. The CallManager Express product offers a low-cost
solution suitable for the small business or a branch office.
Cisco’s high-end products provide
the hardware and software to support the largest call center.
Management tools are a key part of all Cisco products, and that
includes its Unified Communications offerings. The Unified Operations
Manager and Unified Service Monitor tools make it possible to keep
high-volume, mission-critical communication centers up and running with
a minimal amount of staffing.
Edgewater Networks
The EdgeMarc 4500 series converged
networking router provides a host of interconnection options to
implement a VoIP solution. Interface options include up to four T1 WAN
connections or one Ethernet WAN, a 4-port managed VLAN switch, optional
802.11 wireless access point, and integrated analog phone and line
ports.
Flexibility is key to the EdgeMarc
4500 with the ability to function as a low-cost WAN router all the way
up to a SIP trunking and enterprise video over IP platform. Another
part of flexibility is the ability to reroute calls locally or over the
public switched telephone network in case of a WAN link failure.
The EdgeConnect 800PoE managed power
over Ethernet switch is a perfect companion to the EdgeMarc device. It
provides plug and dial capability to just plug in an IP phone and have
it work. It also implements a variety of QoS and VLAN techniques to
ensure that voice gets first place in most every case. PoE eliminates
the need for additional power supplies and AC wiring with IP phones.
Four Loop Technologies
In most traditional business
telephony installations you will find some type of PBX or public branch
exchange. Four Loop Technologies has taken the approach of replacing
the traditional analog hardware PBX with a software digital one.
Switchvox is a complete business phone system in software supporting
unlimited extensions and voicemail.
The hardware part of the Switchvox
system is essentially a Linux server with the appropriate hardware
cards to interface to any existing telephony equipment. You can
purchase a T1/E1 interface directly from Four Loop Technologies that
will connect your Switchvox server to a traditional PRI or other T1 or
E1 service.
Telephone handsets can be either
traditional analog devices or SIP digital phones. Analog handset
connections to the Switchvox server require an ATA (analog telephone
adapter). Each analog handset can have a corresponding ATA, or you can
use a channel bank to use existing phone wiring.
Switchvox supports a number of
advanced features such as call routing to enable you to receive calls
anywhere, voicemail and email integration where your voicemails get
delivered to your email inbox, and the ability to create interactive
voice response systems using a graphical editor.
by Paul Ferrill
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