There is an actual interview in this project, but names of the company and the engineer and associated products and aquisitions were changed for legal protection purposes.

1. An Analysis of Voice Systems Infrastructure for
‘Corporation A’ Product Support
With
Recommended VOIP Solutions
Term Project
TM 590 VOIP
Keller Graduate School of Management
DeVry University
Chris McCoy
Keller Graduate School of Management
DeVry University
Professor Carnes
August 14, 2007

2. Introduction
Executive Summary – ‘Corporation A’
‘Corporation A’ is an fictional Information Lifecycle Management Solutions
Provider. This means that the company sells and supports hardware and software capable
of creating, managing and storing information from its inception as data to its grave as
archived information. The company began its business as a Storage Hardware
Manufacturer, and continues to enjoy great success in this industry. However, in recent
years, ‘Corporation A’ has recognized the value of integrating software and hardware
solutions to better manage the information of businesses throughout the world. Within its
large catalog of product and solution offerings, ‘Corporation A’ has a number of software
products which require customer technical support via toll free telephone numbers. This
project will focus on three specific call centers located in Toronto Canada, Rockville
Maryland, and Nashua New Hampshire.
Each site supports a different product or set of products. Nashua, New Hampshire
supports a product called ‘Email-Archive-Xstation’, which is an e-mail archiving
solution. Burlington Ontario, Canada supports a data backup product called ‘Networker’.
Rockville, Maryland supports a storage archive product called Disk Manager X Station
and a content management Product known as ‘Application Manager Xstation’. All three
sites were former locations for Corporation B Systems, which Corporation A acquired in
2003. The Toronto site is where the upper management of the product support
organization is located. There is a business dependency between the three sites for
effective management of the overall operation of product support for these products.
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3. Problem Statement
The current customer SLA’s for tech support contracts require a 24/7 phone
presence for each of the 3 centers. The current system in place consists of separate Avaya
Definity PBX systems in each of the 3 sites, utilizing a traditional PBX over PSTN
solution. Five digit dialing is in place, due to the addition of Avaya CLAN and MEDPRO
Cards, but the DID ranges remain separate, and this creates confusion when dialing
between sites as each sites. A second issue is that the toll charges for Canada and the US
are growing with the customer base, due to the existence of separate 800 numbers for
each site and management would like to explore a more cost effective solution to
telephone support and interoffice communication between these sites.
A third problem exists with regard to personal safety and security. In order to meet the
24/7 contract support requirement, some staff members work in rotating shifts around the
clock and on weekends for their customers. This presents a physical security issue with
regards to safety in that there are only a few reps on call during the late night and
weekend shifts. An empty office building means less security for the tech support
employees, as they are the only department in the building working after regular business
hours. This presents an issue of personal safety and security for those working the
rotating shifts. When considering a solution to these issues, it is important to note that the
Engineering, HR, QA, Product Management, Sales, and other departments may be
impacted by any downtime tied to a change in the existing infrastructure. The sales
department can not afford to be without a phone system for any extended period of time,
and a learning curve required to work with different phones might cause missed revenue
3

4. deadlines.
Analysis
For several years, CORPORATION A has operated its three major product
support call centers using traditional PBX systems with 800 numbers located in the
locales nearest each center. When a call needed escalation or transfer to a different
product support group, a long distance charge was incurred by the local call center. This
costly solution was improved by the added capability of a 5-digit inter-office dialing
solution between the three sites using an AVAYA CLAN and MEDPRO Card in each
PBX to provide this connectivity across the WAN using H.323 protocol for signaling.
The CLAN card handles the call signaling and the Media Pro (Medpro) card handles the
voice channel traffic. The existing WAN infrastructure is an MPLS that serves not only
these three sites, but the entire CORPORATION A global infrastructure for its 32,000
worldwide employees. The diagram shown below is intended to show the MPLS
connectivity between the three tech support call center sites:
4

5. Figure 1.1 Existing WAN Infrastructure
This is an abbreviated drawing to show the relationship of Burlington, Rockville and
Nashua in terms of their existing WAN MPLS connection. The full MPLS cloud for
CORPORATION A serves hundreds of locations spread throughout the globe. This
infrastructure provides great scalability for a worldwide VOIP implementation in the
scenario where CORPORATION A goes to a 100% VOIP telephone system for the entire
company.
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6. Figure 1.2: Existing Telephone System Infrastructure for 3 CORPORATION A Product Support
Locations.
Note, each location is using a different phone type, but all are using Avaya Definity Series PBX.
The three sites leverage 5-digit interoffice dialing from office to office using traditional
Avaya (non-IP) telephones. In a limited capacity the company is leveraging IP to reduce
toll charges incurred by dialing between sites. However, a VOIP solution will leverage
even greater cost-benefit by streamlining product support operations in these locations.
Current call transferring can be done from PBX to PBX using the IP network via
5-digit dialing, but with a separate DID range for each call center, very often the call
center administrators must look up the site’s 5-digit dialing range and number for the site
and support product group (DID) before transferring the calls. This can increase wait
times for customers requiring urgent assistance. This technology can be more effectively
6

7. leveraged through the implementation of IP telephones for the support department.
Within the current system, if a tech support rep wants to work remotely from home, the
PBX must be programmed to forward the tech support number to his/her cell phone. This
forwarding must be programmed by a telecom engineer on the PBX so that support can
be completed from remote/home offices. A VoIP based solution would alleviate the need
to constantly administer call forwarding for each user’s cell phone as is the case with the
current system. To address this issue, the CLAN and MEDPRO cards may provide a
good basis for leveraging this capability and streamlining a remote support model for
after hours support. In order to fully understand the existing voice infrastructure for the
three tech support sites, I was fortunate to secure an interview (both over the telephone,
and also via e-mail for additional questions) with Earl H., a senior voice
telecommunications engineer at CORPORATION A. Earl provided some valuable
feedback in terms of the existing environment and options that could be leveraged using
the existing equipment. Notes from the interview begin with the next section.
The SIP Protocol
The Session Initiation Protocol has become a popular protocol for signaling in VoIP
systems. One of the reasons for its success is the tight alignment with IP enabled
applications. According to SIP Demystified, one example of the benefit of SIP
implementation can be described in the following manner:
Assume that the public address for Company A’s support department is
SIP:support@company.com. Because the company has to give support around the clock,
several people are always at work…The redirect server at company.com is able to return
different addresses depending on the time of day so that if it receives a call for
7

8. SIP:support@company.com at noon, it will return (the address of the technician on duty
at noon) (Camarillo, 2002). SIP will be considered for this initiative as a protocol to
provide scalability to the tech support environment. An article in Business
Communications review described the important benefit of SIP in VoIP architecture:
A SIP-enabled contact center can integrate with an existing PBX infrastructure and
applications, yet can also introduce new capabilities such as support of SIP trunks and
phone sets. You can take your migration a step farther by maximizing the full range of
multimedia features supported by SIP inside an intelligent application, to deliver
advanced capabilities, including multi-modal communications, to the contact center. The
SIP-enabled contact center is unique because it simultaneously allows the retention of
existing investments in infrastructure and provides a roadmap to an innovative feature set
for the future (Murashige 2007).
Interview with Senior Telecommunications Engineer at CORPORATION A
Mr. Earl H., Senior Telecom Engineer with CORPORATION A, based in
Pleasanton, California was kind enough to agree to an interview to highlight the
capabilities of the PBX systems in place for Rockville, Burlington, and Nashua. We
discussed the current capabilities of each system and the ability for these systems to
support any VOIP capability. In the course of our discussion, I learned that these systems
were capable of supporting the H.323 protocol on the existing PBX systems with the
installed CLAN and MEDPRO cards, and VOIP from PBX to PBX. We discussed the
potential of this equipment to also support the SIP protocol, but additional equipment
purchases and upgrades on a higher budget level would be required to do an SIP
implementation, and this initiative would most likely take a budgetary back seat to other
on going integration efforts for several CORPORATION A acquired companies. At the
time of this interview, the company had no current plans to do a complete voice system
8

9. replacement. However, Earl recommended looking into SIP due to the potential business
value it might add in its scalability and close alignment with the OSI 7 layer model. On
the West Coast where Earl is based, the Pleasanton, California office had used Avaya IP
telephones with some of its support reps in a limited capacity, and Earl recommended
that these be implemented as the desktop telephone equipment for VoIP, particularly if
the solution involved retaining the existing Avaya PBX. We discussed the requirement
for Power over Ethernet to provide power to the IP telephones. Because none of the three
sites had any PoE (power over Ethernet) capability in the existing data network switching
equipment, we agreed that the phones would need to utilize external Avaya power
supplies containing the onboard PoE capability (The alternative would be a more
expensive replacement of modular Ethernet blades in the core switches). From our
interview, I learned of three requirements to implement a VOIP solution that would
leverage the existing PBX infrastructure at a significantly lesser cost to CORPORATION
A than a full VOIP implementation for the entire site at each location. The existing
technology responsible for the H.323 capability consists of 2 cards currently installed in
each of the three PBX’s. First, a CLAN (This card is responsible for H.323 call signaling
including setup and teardown of the calls, and remote system administration.) and a
second card, called a ‘Media Processor Card’ or ‘Medpro’ card handles the actual voice
channel traffic of the calls. To implement VoIP telephones on the existing network,
additional licenses must be purchased from Avaya for the number of users the VoIP
system will support. In addition to our discussion of the possibilities for PBX to PBX
VoIP implementation, and VoIP telephone implementation at the desktop, we also
discussed the possibilities for a remote solution to allow techs to work from home while
9

10. utilizing a VoIP telephone. We arrived at a workable solution consisting of a newer
model Avaya IP Telephone that is capable of providing a VPN connection over the home
broadband internet connection. This component is critical to implementation because the
existing remote VPN model is two-part, consisting of the Cisco VPN software client and
the RSA SecurID fob, which are required to be used in tandem for authentication. I also
asked about QoS for the phones. Because CORPORATION A runs an MPLS WAN, and
MPLS technology is well suited for QoS for VoIP, this capability would be leveraged in
the implementation.
Earl and I also discussed 800/Toll-Free numbers. Currently there are at least three
separate numbers for Tech Support at CORPORATION A. One serves the Canadian call
center in Burlington Ontario Canada. Another serves Rockville Maryland, and yet
another is listed as “All calls outside the US and Canada”. My hope had been to
consolidate the global 800 enterprise into a single number that would leverage the power
of VoIP to do call routing globally and cut monthly toll-free charges. However, I learned
that this is not entirely possible, not only for technical reasons, but also a business barrier
I had not considered: spoken language. The US and Canada share English as a common
spoken language. They also share membership in the North American Numbering Plan.
(NANP, 2007) Therefore, a consolidation of the Canadian and US toll free numbers into
a single number that can be centrally routed to VoIP telephones is worthy of
consideration for an implementation to reduce monthly costs. However, removing the
International toll free number is not recommended. I learned that there are separate call
centers in foreign countries where CORPORATION A’s employees can more effectively
communicate with tech support customers speaking their native language. From a
10

11. business view, this is more productive and less time consuming to provide such overseas
services to customers in similar language geographies. Earl recommended a quality
improvement measure in the network setup for VoIP: Implementing the IEEE 802.1q and
IEEE 802.1p protocols in the vlan in which VoIP will be located. Though this would be a
difficult decision, weighing cost versus scalability, Earl’s final recommendation was to
strongly consider an SIP based solution to accommodate future growth of the company as
well as greater application flexibility and scalability with the VoIP solution (Helsley,
2007).
Analysis of possible solutions prior to final recommendation:
An article in InfoWorld makes a valid point regarding the trend in VoIP integration:
Most agree that a major transition to VoIP in the enterprise is inevitable, but in
most companies it will probably be a gradual process of greenfield branch office rollouts,
deploying IP where it brings the most benefit, replacing obsolete legacy equipment, and
gradually upgrading the data network infrastructure. (Erlanger, 2004).
For the Tech Support VoIP implementation, three potential solutions have been identified
for consideration. The first would involve a complete overhaul of the Voice Systems in
all three sites, replacing all PBX equipment with either AVAYA or Cisco ‘SIP capable’
VoIP equipment, including in the solution, SIP compatible IP telephones, SIP Gateways
and SIP Proxy Servers. The second possibility is to implement an incremental solution
that involves some SIP VoIP combined with the existing PBX solution (this will provide
a potential migration path to a complete SIP solution in the future). The third possibility
is to leverage the existing PBX equipment utilizing H.323 compatible equipment and
deploying IP telephones utilizing all existing equipment. All three of these solutions can
11

12. be deployed to reduce at least one of the toll free numbers Canada’s number may be
eliminated, and Canada’s customer base would then be instructed to use the US based toll
free number from which calls can be routed back to Canada using VoIP.
Solution Alternatives
Based on the research pertaining to potential VoIP solutions, there appear to be three
possible solutions for CORPORATION A to take in regard to the tech support phone
system issues.
Each of these should be explored to assess the business impact of the final decision.
The matrix shown below identifies some of the issues and benefits for each possible
solution:
SOLUTION A SOLUTION B SOLUTION C
Complete SIP based VoIP
solution, Complete
Incremental SIP replacement of existing PBX
H.323 based VoIP solution Implementation with some equipment with 100% VoIP
Technology over existing Avaya PBX additional equipment solution
More than Solution A, Less
Cost Least Expensive than Solution C Most Expensive
Some Interuption to Tech Planned Downtime required
Business Impact No Interruption to Business Support for Entire Site
An Upgrade Path is Possible,
leading to eventual PBX Full SIP Compatability
Scalability None replacement provides greater scalability
None - Staff Voice Training on additional Cisco
Engineers have required Some Training on SIP and new products, some cross training
Training knowledge Cisco products for Network Engineers
12

13. (Continued on next Page)
Solution A – IP Telephone implementation with existing PBX Equipment:
Figure 1.3: Simple H.323 solution using Avaya VOIP telephones.
The first solution option is an implementation of VoIP for the tech support call centers in
Canada, Maryland and New Hampshire using the existing Avaya PBX Medpro and
CLAN cards. These are capable of supporting H.323, but not SIP signaling. Despite this
limitation, the current systems will support desktop IP telephones for Tech Support
13

14. including a single DID range, and support to route calls between sites as needed. This
solution is a low cost alternative that allows VoIP desktop connectivity with only cost
involved in the purchase of the telephones and licensing fees for Avaya. The growth of
CORPORATION A including continued integration of acquired software companies
means that voice system integration will increase over the next few years. While the
immediate need to satisfy tech support with a better voice solution would imply a simple
leveraging of the existing PBX infrastructure, the bigger picture for CORPORATION A
would indicate that scalability of any implemented VoIP solution should be considered so
that the entire organization would better benefit from such investment.
14

15. Solution B Implement Cisco SIP equipment with PBX H.323 equipment :
Phased SIP approach including SIP proxy support
15

16. Cisco offers a solution that may provide a ‘best of both worlds’ scenario. This means that
CORPORATION A will leverage benefit from the existing H.323 equipment integrated
with a new SIP architecture. The benefit to this scenario is that it provides a scalable
architecture solution that will allow tech support to leverage the benefits of SIP VoIP
without any interruption to the existing H.323 5-digit dialing system in place on the
current PBX to PBX infrastructure. The information from Cisco pertaining to this type of
integration is found in a Whitepaper titled, “H.323 and SIP Integration”. (Cisco, 2002)
According to Cisco:
Cisco gateways provide support for the coexistence of SIP and H.323 calls
beginning with Cisco IOS Software Release 12.2(2)XB (p.2)
16

17. This same paper includes an important diagram showing the relationship of the
SIP/H.323 gateways as they route traffic to and from the PSTN with Cisco SIP proxy
servers providing SIP call routing. CORPORATION A’s call centers would be adapted
with these gateways and proxy servers to maintain the H.323 environment while
providing enhanced SIP services to the new VoIP infrastructure. (Cisco, 2002)
Solution C: Replace all PBX equipment with a Complete VoIP (SIP-based) Solution
In a full replacement scenario where the organization decides to do away with the Avaya
PBX infrastructure, the end solution would most likely involve a Cisco IP telephone
product as well.
17

18. Full Implementation of VoIP with elimination of the Avaya PBX’s
With a full replacement of the Avaya PBX systems, management will likely require a
Cisco based IP telephone rather than an Avaya telephone. Although several
CORPORATION A sites have Avaya PBX equipment, Avaya is not the corporate voice
standard. Because Cisco is the standard for IP data switching and routing systems, and
due to a pre-existing purchasing relationship with Cisco, CORPORATION A will best
benefit from the purchase of Cisco IP telephones if the replacement solution is approved.
The implementation of a complete SIP solution would involve a total replacement of the
existing PBX infrastructure with SIP based VoIP equipment. Were CORPORATION A
to consider this alternative, there are a number of options available for a complete VoIP
on premise solution. First, the company could utilize a Cisco Integrated Communications
18

19. Server (shown in the above diagram). AT&T is an example of a vendor that offers VoIP
for Enterprise, with the flexibility of premise or non-premise based systems. One major
advantage to this approach is that it can utilize the existing MPLS architecture. From the
AT&T website, here is a list of the features that would be leveraged, should
CORPORATION A choose a full VoIP solution:
C In-bound and out-bound calling
C On-net calling between VoIP-enabled U.S. sites
C Flexible calling and dialing plans
C Dynamic bandwidth sharing
C Prioritization of voice across access and network
C Virtual Telephone Numbers (VTNs)
C Voice-quality SLAs
C Full Local service
(AT&T, 2007)
(Continued on next Page)
Selected Solution for CORPORATION A
With the valuable input from the Interview with Earl, and the additional information from
the TM590 course readings, lecture notes, discussions, and research from the ProQuest
19

20. library, there is now sufficient information to construct a solution for CORPORATION
A’s product support group. Although the most cost efficient short term solution would be
Option A, as a simple deployment of IP telephones on existing H.323 equipment, there
are many reasons why Option B, an integrated SIP solution would be a wise decision for
CORPORATION A. First, although CORPORATION A uses AVAYA equipment in
some sites, there are others that are not AVAYA based. Prior to the stream of
acquisitions, the corporate standard for voice systems was Nortel. The data routing and
switching standard is Cisco. With this information, one might justify the replacement of
AVAYA with Cisco’s SIP solution as a giant step toward corporate compliance. A 100%
VoIP solution as described in Option C, would require a major interruption to business.
CORPORATION A simply can not afford a major interruption of this type in any of the
three sites due to ongoing Software Engineering, Quality Assurance and Sales operations.
An incremental solution involving a ‘phased’ upgrade to SIP will provide a better fit by
allowing some SIP equipment to be implemented in parallel with the existing PBX
infrastructure. Therefore, Option B is selected for implementation as described in the
implementation plan in the following section.
Implementation Plan
The sample schedule shown below is intended to provide a working implementation
based on required tasks.
20

21. Baseline Testing
A proper pre-assessment of all three sites is required of the Network Operations team in
order to look at existing IP traffic in terms of bandwidth loads and peaks as well as bursty
applications. An assessment of the existing Voice system (PBX) calling times, peak busy
hours, and call durations should also be taken to assess the ability of the existing WAN
circuits to support the additional VoIP traffic load. CORPORATION A’s existing 5-digit
intersite calling should also be assessed as this too will be impacted by additional VoIP
traffic. The network team uses a LAN analysis tool called “Finisar Surveyor”. This tool
21

22. provides traffic analysis in each of the three call center sites, and can filter packets to
analyze a specific type of traffic. The Surveyor software is installed on three monitoring
stations, each accessible via VNC. Additionally, the network engineering team utilizes a
WAN analysis tool called ‘Netflow’ to analyze traffic across the WAN in all three sites.
Netflow provides a granular analysis level to which a network engineer can identify
traffic peaks at any point in the provided analysis graph to quickly pinpoint the sender
and receiver of the data via IP address and system name. To measure voice data across
the PBX, an analysis tool from TelSoft solutions called “MegaCall” offers a Traffic
Statistics and Analysis tool. Once this data has been gathered, a proper assessment of the
LAN and WAN environments should be considered before moving forward with VoIP
deployment. All three sites are running 100Mb LAN connection speeds, so there is at
least one consideration for upgrading to Gigabit that may be considered as a possible
performance improvement. If the traffic proves too heavy at the WAN level, a WAN
optimizer might be considered to help improve performance.
According to an article in the International Journal of Electronics and Communication:
The measurements have shown that quality of speech in VoIP is very much
dependent on the basic network load. Small packets in the environment have a larger
influence on VoIP than larger packets. From this behavior it can be concluded that it is
not the net load as such that causes poor quality of speech but rather the number of
packets sent within a time span that must be considered as the overriding factor. If more
than a certain number of packets are sent within a certain time span, jitter will increase
exponentially bringing with it all the negative consequences for transmission quality and
delay (Uhl, 2004).
22

23. The ultimate goal of CORPORATION A’s baseline data network testing is to look for
spikes in the data flow graphs. LAN and WAN utilization should both be considered in
this testing. These spikes in the data graphs (found in the LAN/WAN tools mentioned in
the previous sections)measure peak data transfers where the network bandwidth has been
completely utilized to its capacity and traffic may have even surpassed the limit into what
is described as a ‘burst’. Often this scenario will occur when QA engineers in the lab
send large file transfers over the wire. CORPORATION A builds large scale data
products, so this type of testing happens from time to time, and must be considered as a
part of the business activity. If the graphs in Netflow indicate an increase in the interval
with which high level transfers occur, then the network team might consider upgrading
the existing partial DS-3 circuits in each site (currently running at 10Mbps) to a full DS-3
capable of approximately 45Mbps.
Provisioning Service QOS with the WAN provider
CORPORATION A’s MPLS WAN is well suited to take on VoIP traffic, but the majority
of its current utilization is IP data traffic. Although the company isn’t currently involved
in a total replacement of the PSTN, the MPLS vendor should be engaged to provide QOS
provisioning for the increase in tech support VoIP traffic that will follow. A consultation
with the vendor to discuss future growth of CORPORATION A’s VoIP potential would
also be a prudent business decision to explore additional options and services that the SIP
protocol might provide in terms of applications for the call centers.
Tech Support Telephone Training
23

24. Training sessions will be conducted by the vendor on site. The senior voice engineer will
travel to each site with the vendor to answer any company specific questions regarding
the phone system or the new equipment.
VoIP live testing with a Few selected reps in each site
Once the new VoIP equipment is received in each location, the internal systems will be
installed in the telecom room and connected to the network for testing. A Change
Management Request will be submitted prior to installation of the equipment to provide a
paper trail in the event of any unusual occurrence on the network at time of installation.
Two tech support reps will be selected from each of the three locations to do live testing
with the new VoIP telephones. These phones will co-exist with the older phones until
testing has been completed and all required functions are verified over a period of two to
three weeks.
Go-Live Approval Date from Tech Support Management
Once testing has been completed and management approval obtained for a go-live date,
the IT group will schedule a “Change Management Request” with the CMR approval
board. This approval is required before making any changes to the network environment.
Formal Communication to Support Reps for Go-Live
After the CMR (Change Management Review) group has approved the changes; IT will
send an e-mail to the tech support group indicating the date of the change.
Internal Office Rollout
In the evening prior to rollout, IT will install all new desktop IP telephones and test
connectivity between sites, voicemail accessibility, and call queue functions to ensure
24

25. that the integration will be transparent. Once this is confirmed, all existing legacy
telephone equipment will be removed and placed in storage.
Retiring the Toll Free 800 number in Canada
Once all internal go-live initiatives have been completed, the Toll Free number in Canada
will be retired. All Canadian support customers will be directed to use the US Toll free
800 number for all tech support calls. Communication will be done via e-mail, printed
mail to customers, and a series of calls from tech support management to customer sites
to verify that they have received the new information. Call center administrators will be
instructed to provide the new 800 number to all incoming calls as a courtesy. The
Corporate website will for product support will also be updated to reflect this
information.
Remote Office User Rollout
The after hours support techs will be assigned an IP telephone to use in their home office.
The Avaya IP telephones provide an additional onboard data port, which will allow the
user’s to connect a PC through the phone’s additional data port to their broadband router
without the requirement of an additional mini-switch to provide extra ports. Remote users
will receive additional training on how to establish a VPN session using the softclient
feature on the IP telephone.
High Speed Internet Access
In order to leverage the full power of a remote access VOIP solution across a VPN
tunnel, the home office should be equipped with High Speed Internet. Recent competitive
initiatives from ISP’s such as Verizon and Comcast have made High Speed Internet
25

26. Access more affordable. In addition to the two popular mediums, DSL and Cable, Fiber
Optic Internet Service is being offered in some areas.
Typical Remote/Home Office Connectivity to Corporate:
Figure 1.3: Remote User and Call Center IP Phone Connectivity .
According to an article from Network World:
Vendors continue to streamline VoIP gear for telecommuters, Riggs says.
Recently Cisco introduced VDIP phones that indude VPN software, eliminating the need
for a separate VPN router to tunnel calls over the Internet, Riggs says.lhis can reduce the
initial cost of setup (Green, 2006).
CORPORATION A’s home office user will leverage the power of high speed internet
service through her/his own home broadband ISP provider, and utilize this new phone
26

27. technology to eliminate the need for a hardware based VPN solution to connect the IP
telephone to the call center. There are two features on the AVAYA model 4622SW IP
telephone that will allow the home office employee to leverage the greatest benefit of the
technology. First is the phone’s capability to utilize AVAYA VPN software to initiate a
VPN session with CORPORATION A. This will tie in nicely with CORPORATION A’s
existing combination Cisco and RSA VPN solution. The remote office solution should be
considered scalable as it will integrate with any of the three proposed internal office VoIP
solutions.
27

28. Conclusion
The product support groups studied in this project will see a three-fold business
enhancement from a migration to VOIP. First, the 800 number charges can be reduced by
retiring two of the three numbers and going to a single 800 number for worldwide
service. Second, the security of after-hours employees will be greatly enhanced by
allowing them the capability to work from a home office environment. Third,
productivity and efficiency will be increased by creating a unified dialing environment
where the three call centers are a virtual single global call center. These improvements
will help CORPORATION A’s competitive edge by reducing customer wait times and
increasing customer satisfaction. This satisfaction will be returned to the company
through repeat business for software support contracts, and potentially expanded product
purchases and upgrades. A satisfied customer is likely to tell partners and colleagues of
the positive experience with CORPORATION A, and therefore help to expand
CORPORATION A’s customer base and partnerships in the future. Within the
CORPORATION A IT Infrastructure, there are several existing entities that make it a
good candidate for an eventual complete VoIP enterprise level solution. First,
CORPORATION A’s WAN infrastructure is a global MPLS that provides high
performance WAN connectivity. Because MPLS is a private network solution, it is also a
good candidate for VOIP. Second, CORPORATION A has a long standing relationship
with Cisco. The IP network infrastructure is based on current Cisco products and trends,
and CORPORATION A has provided positive alignment with Cisco’s initiatives in many
of its sites. One current pilot Cisco deployment is TelePresence at corporate HQ and
28

29. Santa Clara, California. This new state of the art videoconferencing system can work
with SIP, meaning that on a larger scale, each of the call center sites may justify a future
full SIP implementation for VoIP to provide a conduit into TelePresence conferences
using the SIP protocol. (Cisco, 2007) This type of expansion will help CORPORATION
A continue its competitive advantage in the Information Lifecycle Management industry
by keeping its information and communication flow one step ahead of its competitors.
29

30. References
AT&T VOIP. Retrieved August 14, 2007. Website:
http://www.business.att.com/service_portfolio.jsp?
repoid=ProductCategory&repoitem=eb_voip&serv_port=eb_voip
Camarillo, Gonzalo (2002) SIP Demystified, USA: McGraw-Hill.
Cisco SIP VOIP. Retrieved August 10. 2007. Website:
http://www.cisco.com/univercd/cc/td/doc/product/voice/sipsols/biggulp/bgsip.pdf
Cisco Telepresence FAQ. Retrieved August 13, 2007. Website:
http://www.cisco.com/application/pdf/en/us/guest/products/ps7073/c1167/cdccont_0900a
ecd80544106.pdf
Leon Erlanger (2004, June). THE 411 ON VoIP. InfoWorld, 26(23), 42-46,48,50,52.
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