| Updating Your PBX For The
IP Telephony Market
BY HENRY DEWING
Internet telephony is revolutionizing telecommunications, delivering
benefits like low-cost long-distance calling and fax communications, and
offering new and flexible services to improve productivity. At the same
time, most mid-sized and large companies already have installed PBXs they
aren't prepared to scrap. Fortunately, there's no need to start from
scratch to enjoy the benefits of IP telephony. Today, there are several
ways in which enterprises can take advantage of Internet telephony without
sacrificing their existing PBX investment.
Reaping the benefits of VoIP takes two essential ingredients: products
that create value for the enterprise, and people who can correctly
implement them. Both ingredients are equally important. You presumably
already understand the benefits of IP telephony, but it might be useful to
discuss some of the ways in which a PBX can gracefully be brought into the
IP network, delivering lower-cost and more flexible services to a broader
cross-section of the enterprise's employees.
As the volume of calls between corporate locations grows, enterprises
look for ways to lower their incremental cost. One way to do this is by
using IP gateways to route traffic over their intranet. By using IP
transport to the desktop -- on the opposite side of the PBX from the IP
gateway -- enterprises can:
- Install and manage only one network on their premises.
- Provide full-featured PBX service to remote users.
- Easily integrate third-party services.
Once the PBX is surrounded by VoIP connections, the enterprise is in a
position to easily migrate to a fully converged architecture.
DEPLOYING VoIP IN THE ENTERPRISE
Telecom managers tasked with converging the enterprise's network have
their work cut out for them. Evaluating all the VoIP vendors -- making
certain the one you choose will still be around after burning through its
IPO money -- is only the beginning of the challenge. It's also essential
to ensure the solutions you deploy adhere to accepted industry standards.
This is the only way to be certain that today's investment will not be
left stranded by tomorrow's interoperability failures.
Unfortunately, making sure your solution adheres to industry standards
is not an easy task in the VoIP market. Many standards are still only in
the proposal stage, and all of them are being refined at some level. If
the PC model holds in the VoIP market, consumers of IP telephony systems
will eventually be able to easily combine solution components from
multiple vendors. But today, ensuring that components from different
vendors are interoperable is a monumental task. VoIP standards include
both H.323 and H.225, as well as SIP and MGCP, which ensure that both ends
of the call know when to ring a phone and open a channel and how to create
a conference call or perform other call control functions. Digitized voice
can be compressed in a variety of different ways, and must be decompressed
in the same way at the opposite end of the call. Compression algorithms
include G.723.1, G.729, and others. There are other standards for silence
suppression, fax transport, and a variety of other parts of the market. In
other words, buyers beware in this market: not all solutions adhere to
accepted standards. Those that do adhere to standards don't necessarily
comply with the same standards.
Enterprise telecommunications managers -- being rightfully cautious
about risking failure of the corporate voice network -- often trial these
IP gateways before making a deployment decision. Frequently during these
trials there is a temporarily rapid rise in the demand for packet-switched
data connectivity and a slower-than-expected reduction in demand for
circuit-switched voice network capacity. Because the test is a trial,
employees often make one IP call and one voice call simultaneously to hear
for themselves the voice quality during increasing peak calling traffic
volumes. In some enterprises, this activity can continue for a
surprisingly long time. Remember when e-mail was new and people would send
volumes of e-mail as soon as they got a new address -- just to see if it
worked?
IP GATEWAY BETWEEN ENTERPRISE LOCATIONS
One of the most obvious first steps to implementing VoIP in an
enterprise is by installing gateways to interconnect geographically
separate locations. This can be accomplished by using IP gateways and
packet transport on the intranet (or VPN over the Internet) to replace
existing PBX tie lines. The requirement to maintain voice connectivity and
data connectivity capacity to meet peak usage means that both networks are
equipped to handle nearly the total telecommunications load, even after
the end users quit their tests. There will be smaller incremental
requirements for packet capacity when the last voice circuits are turned
off than when the first ones are turned off. The same traffic engineering
rules -- which favorably affect cost per user as the scale of an
installation increases -- hold true whether the transport is via packet or
circuit.
A crucial concern when implementing these VoIP gateways is quality of
service (QoS). Capacity planning is essential, so network managers must
understand and manage their network traffic profiles for both data and
voice. Although users become irritated while they wait a few seconds, or
even a minute for a large file to download, they go ballistic when a phone
call breaks up or is dropped. Since many datacom and telecom departments
have been established over the past years, there are many more
organizations today that can handle both voice and data than there were
five years ago.
Telecommunications managers can ensure reliable voice quality over
their packet networks in a couple of different ways. On the intranet,
network managers can always increase capacity to preserve voice quality.
Telecom managers can also use VoIP gateways with PSTN fall-back
capabilities. The best of these gateways sense the delay in communications
and seamlessly switch a call between the packet network and the PSTN while
a call is in progress.
Many industry participants are working on QoS standards. Implicit in
QoS is that the IT manager can assess in advance the relative priority of
the CEO's video conference with Wall Street analysts, the SAP data dump to
the factory for the quarter's final production run, and a phone call
between sales offices to decide where to go for happy hour. The IP Layer 2
class of service bit and Layer 3 type of service bit allow all voice and
data to be differentiated across the LAN and WAN, ensuring that neither a
fax nor SAP will slow down a voice call on the intranet. Other proposed
standards, like RSVP, try to assign priorities to applications and
particular communications. How these standards are implemented can have a
wide variety of effects on voice quality.
IP GATEWAY WITHIN ENTERPRISE LOCATIONS
There are many options to consider in migrating that last mile to a
converged network. Putting voice communications over packet networks
between the PBX and the end user requires more equipment to be distributed
around the enterprise, making traffic analysis and QoS assurance more
complicated. Putting the voice communications on the enterprise LAN allows
easy interconnection of the phone system with other client server
applications like unified messaging or CTI-enabled CRM applications.
Telephone calls can go from the PBX to the LAN in two different ways.
Most large PBX vendors offer IP-enabled line cards. These line cards
connect directly to the enterprise LAN, delivering the full functionality
of the PBX to the LAN. These cards completely replace existing PBX line
cards that connect to the twisted-pair enterprise telephony network. An
alternative way to get the telephone calls from the PBX to the LAN is to
deploy PBX extenders that connect on one side to the existing ports on the
PBX and on the other side to the LAN. Either of these options puts the
telephone calls onto the LAN, allowing access to remote users via RAS
connections, to make and receive telephone calls as if they were at their
desk in the office.
Across the LAN, at the desktop, there are at least three deployment
options: softphones, IP telephones, and gateways that put calls back onto
twisted-pair connections.
Softphones are telephone emulation programs that run on a desktop
computer, delivering a full-featured telephone set emulation to end users.
If end users are comfortable depending on their PC to make telephone
calls, this requires the least incremental investment.
Deploying IP telephones is costly, since the price of these sets is
still generally high. IP phones connect directly to the LAN. Some have a
two-port hub, enabling users to plug their PC's network interface card
into their phone for connection to the LAN. These sets are becoming more
sophisticated, with some offering Web browser capabilities and access to
other corporate data in a more traditional client/server fashion. Most of
these phones require one connection to the LAN and another to external AC
power. There is a proposed IEEE standard to allow the delivery of power to
these sets from routers on the LAN via CAT 5 cable.
Using gateways to return telephone calls to twisted pairs allows
enterprises to maintain their investment in existing digital or analog
telephones, while requiring the IT department to maintain twisted-pair
connections from the wire closet to the desktop. Some more sophisticated
gateway pairs pass all the features of digital PBXs to their digital
telephone sets just as if they were directly connected to the PBX.
BRINGING IT ALL TOGETHER
Ultimately, telecom managers need a compelling business case to
undertake the risk associated with deploying a new architecture for their
telephone systems. While cost savings may get the accountants' attention,
the executive staff will be most interested in how this creates
competitive advantage for the enterprise.
An open implementation of VoIP allows great flexibility, enabling rapid
deployment of new services that will allow employees, customers, and
suppliers to interact more effectively using applications that can now
easily interact with voice communications systems. This means there will
be little tolerance for failures in interconnecting voice and data
systems, since these are crucial corporate systems. This deployment will
require converged skills on the part of IT staffs and VARs who will deploy
and maintain these solutions. Not only will the staff need to understand
why and how to configure an ISDN line and a 100 MB switched LAN, they will
also have to evaluate the relative priority of digitized voice versus
traditional data traffic to implement QoS that serves the interest of the
enterprise. This is no small task, but it's one that all of us in the
industry depend on for our own success.
Henry Dewing is senior enterprise marketing manager for Dialogic
Corporation (an Intel company).
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