An Analysis of Voice Systems Infrastructure for‘Corporation A’ Product Support

With Recommended VOIP Solutions

Term Project

TM 590 VOIP

Keller Graduate School of ManagementDeVry University

Chris McCoyKeller Graduate School of Management

DeVry UniversityProfessor Carnes

August 14, 2007

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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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

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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:

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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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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

leveraged through the implementation of IP telephones for the support department.

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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

SIP:support@company.com at noon, it will return (the address of the technician on duty

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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

replacement. However, Earl recommended looking into SIP due to the potential business

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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

utilizing a VoIP telephone. We arrived at a workable solution consisting of a newer

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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

business view, this is more productive and less time consuming to provide such overseas

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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

be deployed to reduce at least one of the toll free numbers Canada’s number may be

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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:

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(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

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

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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.

Solution B Implement Cisco SIP equipment with PBX H.323 equipment :

Phased SIP approach including SIP proxy support

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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)

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)

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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.

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

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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

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:

In-bound and out-bound calling On-net calling between VoIP-enabled U.S. sites Flexible calling and dialing plans Dynamic bandwidth sharing Prioritization of voice across access and network Virtual Telephone Numbers (VTNs) Voice-quality SLAs Full Local service

(AT&T, 2007)

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(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

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

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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.

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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

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.

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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).

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

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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

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

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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

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

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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

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:

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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

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

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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.

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

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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

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

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A continue its competitive advantage in the Information Lifecycle Management industry

by keeping its information and communication flow one step ahead of its competitors.

References

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Camarillo, Gonzalo (2002) SIP Demystified, USA: McGraw-Hill.

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